=== release 1.19.1 === 2021-06-01 00:11:44 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * README: * RELEASE: * gst-plugins-good.doap: * meson.build: Release 1.19.1 2021-05-29 12:54:22 +0100 Tim-Philipp Müller * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: fix image corruption when compiled with MSVC on Windows On Windows with MSVC, jpeg_header_size would end up 2 bytes larger than it should be. This then leads to the first 2 bytes of the actual jpeg image data to be dropped, because we think those belong to the header, which results in an undecodable image when reconstructed in the depayloader. What happens is that when the compiler evaluates jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem); it actually uses the mem.offset value after it has been increased by the function call on the right hand size of the equation. From section 6.5 of the C99 spec: 3. The grouping of operators and operands is indicated by the syntax [74]. Except as specified later (for the function-call (), &&, ||, ?:, and comma operators), the order of evaluation of subexpressions and the order in which side effects take place are both unspecified. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889 Part-of: 2021-05-25 16:19:20 +0800 Hou Qi * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Set default latency if the frame duration is invalid If the duration of the v4l2object is invalid, use default 25fps instead. Part-of: 2021-05-26 00:23:56 +0900 Seungha Yang * gst/deinterlace/gstdeinterlace.c: deinterlace: Drop "field-order" field while transforming caps Like other basetransform subclasses are doing, drop field which can be converted by deinterlace. Part-of: 2021-05-25 20:10:34 +0900 Seungha Yang * gst/deinterlace/gstdeinterlace.c: deinterlace: Drop field-order field if outputting progressive Progressive with field-order doesn't make sense Part-of: 2021-05-21 14:19:29 +0200 Havard Graff * gst/rtpmanager/gstrtpssrcdemux.c: * tests/check/elements/rtpssrcdemux.c: rtpssrcdemux: fix "data flow before segment event" crash This crash could happen at any time a RTP and RTCP buffer arrived simultaneously in ssrcdemux. The problem was that sticky-event arriving while the rtp and rtcp pads were being set up could arrive just too late to be included in the initial forwarding. The fix checks if the stickies have been sent on the srcpad about to be pushed on, and if not sends them. It also blocks any stickes from being forwarded *prior* to this happening, to avoid them arriving on the srcpad multiple times. Since the test loops 1000 times, this will make running under valgrind take forever, so use the RUNNING_ON_VALGRIND variable to detect we are running under valgrind, and reduce the loop-count to 2 in that case. Part-of: 2021-05-21 18:45:17 +0200 Havard Graff * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: refactor destruction of GstRtpSsrcDemuxPads Part-of: 2021-05-21 18:30:28 +0200 Havard Graff * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: rtpssrcdemux: make naming consistent Use plural for GstRtpSsrcDemuxPads, since it contains two pads, and use the variable-name 'dpads' everywhere. Part-of: 2021-05-23 15:14:11 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: use g_strndup() for copying text data So we don't rely on NUL terminators inside the data. Part-of: 2021-05-23 13:29:07 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: clean up adtl/note/labl chunk parsing We were passing the size of the adtl chunk to the note/labl sub-chunk parsing function, which means we may memdup lots of data after the chunk string's NUL terminator that doesn't really belong to it. Part-of: 2021-05-23 13:24:21 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: guard against overflow when comparing chunk sizes Could be rewritten as lsize > (size - 8) a well, but the extra check seems clearer. Doesn't look like it was problematic, lsize wasn't actually used when parsing the sub-chunks. Part-of: 2021-05-21 13:31:12 -0300 Daniel Almeida * docs/gst_plugins_cache.json: doc: update gst_plugins_cache.json Part-of: 2021-05-05 13:20:04 +0200 Stéphane Cerveau * gst/matroska/matroska-demux.c: matroskademux: fix decoder glitches with H264 content To avoid decoder starvation causing glitches on screen, the demuxer shall clip only when the buffer is a key frame and the lace time is greater than the stop time. Fixes gst-editing-services#128 Part-of: 2021-05-11 20:41:38 +1000 Matthew Waters * ext/qt/gstqtoverlay.cc: qml: don't use buffers that have invalid contents If the GL context is not shareable, ignore it. A future change may also not output the relevant output either. Part-of: 2021-05-11 20:38:52 +1000 Matthew Waters * ext/qt/gstqsgtexture.cc: qml: also use the dummy texture when no buffer has been set Fixes corrupted texture output when changing OpenGL display/contexts. Part-of: 2021-05-11 17:20:00 -0400 Nicolas Dufresne * docs/gst_plugins_cache.json: doc: Update cache for RGBP format addition Part-of: 2021-04-23 14:37:46 -0400 Nicolas Dufresne * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: matroskademux: Advertise codec-alpha in caps This will be used to select the appropriate decoders. We also only attach the GstVideoCodecAlphaMeta if the AlphaMode element is set, this is to stay on the safe side and mimic what browsers (verified in Firefox and Chromium code) do. Part-of: 2021-03-22 16:58:26 -0400 Nicolas Dufresne * gst/matroska/matroska-demux.c: matroskademux: Store alpha stream in VideoCodecAlphaMeta This generalize the feature over using mini object quark data. If that feature was Matroska specifc, using the new CustomMeta would have been enough and arguably cleaner then QData, though it seems that similar technique is use with AV1 Image Format (AVIF). Part-of: 2016-12-03 14:27:57 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: extract VP8 alpha from BlockAdditionals And put it on buffers as qdata (which is easier in this case than a private custom meta because it can be picked up easily in other modules). Part-of: 2021-05-03 17:39:05 +1000 Matthew Waters * ext/qt/gstqtglutility.cc: * ext/qt/gstqtglutility.h: * ext/qt/gstqtoverlay.cc: * ext/qt/qtitem.cc: * ext/qt/qtwindow.cc: qt: return a different GstGLDisplay object when the first sink requests This allows the 'replace-gstreamer-opengl-context' context machinery to correctly replace the OpenGL context used by the pipeline when the first qmlglsink is added to the pipeline. Part-of: 2021-05-07 11:16:47 +0200 Jan Alexander Steffens (heftig) * gst/udp/gstudpsrc.c: udpsrc: Plug leaks of saddr in error cases Part-of: 2021-05-07 11:16:21 +0200 Jan Alexander Steffens (heftig) * gst/udp/gstudpsrc.c: udpsrc: Whitespace Part-of: 2021-05-07 00:43:44 +0200 Jan Alexander Steffens (heftig) * gst/deinterlace/gstdeinterlace.c: deinterlace: Plug a method subobject leak Changing the method would leak the previous method. Part-of: 2021-05-06 15:04:42 -0400 Nicolas Dufresne * ext/vpx/gstvp9enc.c: vp9enc: Add color range support When setting the colorspace, we now clear the range to reduced range, the default, and then we also set the range so the VP9 encoder encodes the right information in the bitstream. Part-of: 2021-05-06 14:51:31 -0400 Nicolas Dufresne * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxenc.c: vp9enc: Move colorspace configuration in VP9 enc This is not supported by VP8 and was causing a warning. Part-of: 2021-05-06 14:48:36 -0400 Nicolas Dufresne * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxenc.c: * ext/vpx/gstvpxenc.h: vpxdenc: Add a GstVideoCodecState to configure_encoder virtual This will be needed to configure the VP9 specific colorimetry, which is currently configured for VP8 casing warning. Part-of: 2021-05-05 16:48:10 +0200 Bastien Nocera * ext/gtk/gtkgstbasewidget.c: gtk: Remove coordinates double-translation Remove our own translation in the mouse event capture code, as that translation will be done through the navigation interface. Tested by resizing the window created by: gst-launch-1.0 -v videotestsrc ! navigationtest ! glupload ! glcolorconvert ! tee name=t ! gtkglsink and checking that the cursor follows the mouse as expected. Part-of: 2021-05-05 14:28:15 +0200 Bastien Nocera * ext/gtk/gstgtkbasesink.c: gtk: Translate navigation events coordinates If the application passed down some pointer coordinates, translate those from display coordinates to stream coordinates, so things work as expected even if the video is resized. Part-of: 2021-05-05 14:24:31 +0200 Bastien Nocera * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: gtk: Export _display_size_to_stream_size() Export _display_size_to_stream_size() so that GstNavigation implementors can translate from display coordinates to stream coordinates before pushing the events upstream to the DVD source, for example. Part-of: 2018-02-26 17:26:41 +0100 David Fernandez * docs/gst_plugins_cache.json: * gst/matroska/matroska-mux.c: matroska-mux: Change accepted caps width and height from [16, MAX] to [1, MAX] There are cases where the video size might be less than 16x16. This change allows the Matroska muxer to accept this cases. Part-of: 2021-04-20 22:08:23 +0200 François Laignel * gst/multifile/gstsplitmuxsink.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtsp/gstrtspsrc.c: * tests/check/elements/avimux.c: * tests/check/elements/flvmux.c: * tests/check/elements/interleave.c: * tests/check/elements/qtmux.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpcollision.c: * tests/check/elements/rtpmux.c: * tests/check/elements/splitmuxsink.c: * tests/check/elements/videomixer.c: * tests/examples/rtp/client-PCMA.c: * tests/examples/rtp/server-alsasrc-PCMA.c: Use gst_element_request_pad_simple Instead of the deprecated gst_element_get_request_pad. Part-of: 2021-04-30 08:12:47 +1000 Jan Schmidt * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: qtmux: Make sure to write 64-bit STCO table when needed. qtmux attempts to choose between writing a 32-bit stco chunk offset table when it can, but switch to a 64-bit co64 table when file offsets go over 4GB. This patch fixes a problem where the atom handling code was checking mdat-relative offsets instead of the final file offset (computed by adding the mdat position plus the mdat-relative offset) - leading to problems where files with a size between 4GB and 4GB+offset-of-the-mdat would write incorrect STCO tables with some samples having truncated 32-bit offsets. Smaller files write STCO correctly, larger files would switch to co64 and also output correctly. Part-of: 2021-04-22 15:01:32 +0800 Hou Qi * sys/v4l2/gstv4l2object.c: v4l2object: Add interlace-mode back to caps for camera skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and colorimetry when probe caps. gst_v4l2_object_set_format_full() will add colorimetry back to caps when iterating over the negotiated caps. There is one case that v4l2src is first in preview state then starts recording. v4l2src caps will change with an additional interlace-mode structure after renegotiation, then v4l2src needs to reset. But this camera driver can't orphan buffer pool, it causes require buffer failed as streaming is still in active state. To fix this, also need to add interlace-mode back to caps for camera to avoid reset. Part-of: 2021-04-02 18:41:28 +0200 Guillaume Desmottes * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: * tests/check/elements/rtp-payloading.c: rtpopuspay: set MARKER flag Set MARKER flag on first buffer after DTX. According to RFC 3551 section 4.1 the marker bit needs to be set on "the first packet after a silence period during which packets have not been transmitted contiguously". Part-of: 2021-03-31 11:18:30 +0200 Guillaume Desmottes * docs/gst_plugins_cache.json: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: * tests/check/elements/rtp-payloading.c: rtpopuspay: add DTX support If enabled, the payloader won't transmit empty frames. Can be tested using: opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true Part-of: 2021-04-24 11:15:50 -0400 Doug Nazar * ext/taglib/gstid3v2mux.cc: taglib: Update createFrame() to non-deprecated version. ID3v2::FrameFactory::createFrame() versions not taking a Header have been deprecated since v1.5 (Jan 2008). Part-of: 2021-04-25 02:16:45 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix divide-by-zero The estimated packet-duration can sometimes end up as zero, and dividing by that is never a good idea... The test reproduces the scenario, and the fix is easy. Part-of: 2020-06-02 19:38:33 +0200 Havard Graff rtpjitterbuffer: clean up and improve missing packets handling * Try to make variable and function names more clear. * Add plenty of comments describing the logic step-by-step. * Improve the logging around this, making the logs easier to read and understand when debugging these issues. * Revise the logic of packets that are actually beyond saving in doing the following: 1. Do an optimistic estimation of which packets can still arrive. 2. Based on this, find which packets (and duration) are now hopelessly lost. 3. Issue an immediate lost-event for the hopelessly lost and then add lost/rtx timers for the ones we still hope to save, meaning that if they are to arrive, they will not be discarded. * Revise the use of rtx-delay: Earlier the rtx-delay would vary, depending on the pts of the latest packet and the estimated pts of the packet it being issued a RTX for, but now that we aim to estimate the PTS of the missing packet accurately, the RTX delay should remain the same for all packets. Meaning: If the packet have a PTS of X, the delay in asked for a RTX for this packet is always a constant X + delay, not a variable one. * Finally ensure that the chaotic "check-for-stall" tests uses timestamps that starts from 0 to make them easier to debug. Part-of: 2021-04-23 12:07:52 +0200 Guillaume Desmottes * gst/level/gstlevel.c: * gst/level/gstlevel.h: level: make properties thread-safe Part-of: 2021-04-22 14:11:09 +0200 Guillaume Desmottes * gst/level/gstlevel.c: level: disable passthrough when audio-level-meta is enabled Ensure we receive a writable buffer to add the meta. Fix #878 Part-of: 2021-04-23 08:28:06 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Don't pass a non-GObject pointer to GST_DEBUG_OBJECT and similar Part-of: 2021-04-22 08:57:23 +0200 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Avoid generation of invalid timestamps When updating timestamps and timer timeouts with a new offset, make sure that the resulting value is valid (and not a negative (signed) value which ends up in a massive (unsigned) value). Fixes #571 Part-of: 2021-04-21 18:41:08 +0100 Philippe Normand * sys/v4l2/v4l2_calls.c: v4l2: Fix glib warning emitted when attribute query fails The v4l2object is not a GstObject. Logging has to go through its dbg_obj specially meant for this. Part-of: 2021-03-25 13:20:38 +0100 VaL Doroshchuk * ext/qt/gstqtoverlay.cc: * tests/examples/qt/qmloverlay/overlay.py: qmloverlay: Use first found GstGLVideoItem as widget property GstGLVideoItem is required to render input video in the overlay's qml. And currently qmlgloverlay requires to set this GstGLVideoItem to its widget property. Instead of fetching GstGLVideoItem from the overlay's root object (root-item prop), and setting it back as a widget (widget prop), proposing to use found GstGLVideoItem in the current object hierarchy (passed in qml-scene) by default. Also useful in Python, which solves the issue when casting gpointer <=> QQuickItem* is required. Part-of: 2021-04-19 16:39:03 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2.c: v4l2: fix debug category initialisation again Would spew warnings on the rpi4 when calling into gst_v4l2_object_get_codec_caps() from the probe_and_register() function since the v4l2_debug category initialisation would only be done later as part of the element/device provider registration. Also log things in the probe function to the v4l2 category instead of the default category while we're at it. Part-of: 2021-04-19 01:29:33 -0400 Doug Nazar * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix race saving seek event seqnum. We need to save the seek seqnum before the flush stop event since that will start the basesrc task which may send the segment event before we're ready. Part-of: 2021-03-31 10:52:14 +0200 Marco Felsch * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qmlglsink: allow to set force-aspect-ratio property Add the forceAspectRatio Q_PROPERTY to allow changing the aspect ratio from QML code as well. Part-of: 2021-04-19 11:14:00 +0100 Tim-Philipp Müller * sys/v4l2/v4l2_calls.c: v4l2src: fix spurious SOURCE_CHANGED error-level log messages They're harmless, and some drivers at least return EINVAL instead of ENOTTY for unsupported events (here: uvcvideo). Part-of: 2021-04-14 16:32:06 -0400 Doug Nazar * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: remove use of packed struct for payload Part-of: 2021-04-14 11:13:45 -0400 Doug Nazar * gst/dtmf/gstdtmfcommon.h: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: convert to bit accessors Part-of: 2021-04-13 09:23:12 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove some dead code stop is not used after this point, nor do we create a new segment here since 84725d62b57bc74ce34abde755f35bf8f948f94d Part-of: 2021-04-10 02:53:51 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Do not overwrite the known duration after a seek This breaks the duration query and also the seeking query. Broke in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7 Part-of: 2021-04-10 04:40:46 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Just assign the segment instead of memcpy Assignments copy by value, we don't need to memcpy... Part-of: 2021-04-13 11:30:51 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Check srcresult before waiting on the condition variable too It might've been set to FLUSHING between the last check and the waiting, and in that case we'd be waiting here forever now. Part-of: 2021-04-12 23:15:17 -0400 Doug Nazar * tests/check/elements/rtpsession.c: rtp: fix test_twcc_header_and_run to support big endian. Part-of: 2021-04-12 23:13:15 -0400 Doug Nazar * gst/rtpmanager/rtptwcc.c: rtp: fix rtptwcc to support big endian. Part-of: 2021-04-12 21:59:45 -0400 Doug Nazar * gst/rtpmanager/gstrtphdrext-rfc6464.c: rtp: fix rtphdrextrfc6464 to support big endian. Part-of: 2021-04-12 21:36:58 -0400 Doug Nazar * tests/check/elements/alpha.c: tests: Fix alpha test on big endian machines. Part-of: 2021-03-19 02:51:20 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Protect against writing absurd sample durations If the input DTS goes backward or is missing, the calculated sample duration goes negative and wraps around to a very big number. In that case, just write a sample with a duration of 0 and hope the problem is transient. Part-of: 2021-04-10 03:09:44 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: De-dup seek event seqnums to avoid multiple seeks Seek events are sent upstream on each sink, so if we receive multiple seeks with the same seqnum, we must only perform one seek, not N seeks where N = the number of sinks in the pipeline connected to rtspsrc. This is the same thing done by demuxers like qtdemux or matrsokademux. Part-of: 2021-04-10 01:55:28 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Using multicast UDP has no relation to seekability The transport has no relation to whether a media can be seeked. The range response having a duration is the correct thing to check for. Part-of: 2021-04-10 01:54:48 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Add more logging for range parsing and seekable Part-of: 2021-04-10 14:47:23 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: videocrop: Update documentation cache Part-of: 2021-04-07 21:57:11 +0200 Markus Ebner * gst/videocrop/gstvideocrop-private.h: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: Add support for GBR* video formats Part-of: 2021-04-07 18:54:49 +0200 Markus Ebner * gst/videocrop/gstvideocrop-private.h: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: Added support for planar pixel formats > 8bits - Added support for planar pixel formats with depths greater than 8bits to transform_planar implementation - Added a whole lot of new pixel formats to the support-list Part-of: 2021-04-07 17:52:34 +0200 Markus Ebner * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop-private.h: * gst/videocrop/gstvideocrop.c: videocrop: Move supported format list into private header - Moved declaration of supported pixel formats to private header, which can be shared between videocrop and aspectvideocrop Part-of: 2021-04-06 17:02:34 +0530 Nirbheek Chauhan * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: More logging when calculating rfc7273 timestamps This code can be fragile, since it is very exacting in the timestamps that it will accept. Add more logging so it's easier to debug issues and figure out whether it's a bug in the calculation or something wrong in the incoming buffers. Part-of: 2021-04-08 13:29:10 +0200 Stéphane Cerveau * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtpsv3vdepay.c: rtp: missing debug init after element splitting - h264depay - h265depay - sv3vdepay Part-of: 2020-03-30 09:29:07 +0200 Michal Dzik * gst/rtp/gstrtpsbcpay.c: rtp: rename gst_rtp_sbc_pay_flush_buffers() gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains any leftover queued data and pushes it downstream. "Flushing" in GStreamer typically means to throw away any queued data and not process/push it downstream. Signed-off-by: Michal Dzik Part-of: 2020-03-24 13:31:00 +0100 Michal Dzik * gst/rtp/gstrtpsbcpay.c: rtp: fix adapter flushing in sbc payloader GstAdapter must be flushed in some cases (flush, new segment, state change) Without it, it may, for example, push some leftover buffer from old segment in new segment. This, in general, breaks timestamps. See GstAdapter documentation for more. Signed-off-by: Michal Dzik Part-of: 2020-08-18 20:16:06 +0200 Jakub Adam * ext/vpx/gstvpxenc.c: vpxenc: add colorspace information into VP9 bitstream Part-of: 2021-03-26 16:26:22 +0800 Hou Qi * sys/v4l2/gstv4l2object.c: v4l2object: Use default colorimetry if that in caps is unknown Some streams have unknown colorimetry in caps, but v4l2object sets default values for each primaries. It will cause check colorimetry fail when do gst_v4l2_video_colorimetry_matches(). To fix this, need to keep the unknown colorimetry in caps same as the default value set by v4l2object. Part-of: 2021-03-31 16:37:56 +0300 Vivia Nikolaidou * gst/matroska/matroska-demux.c: matroskademux: Take segment stop into account when need_segment Otherwise, in the case of e.g. a deferred seek event, the segment stop would be replaced with GST_CLOCK_TIME_NONE. Part-of: 2021-03-29 16:45:26 +0200 Val Doroshchuk * ext/qt/gstqtoverlay.cc: * ext/qt/gstqtoverlay.h: gstqtoverlay: Add initialization and finalization to qml-scene prop Part-of: 2021-03-31 10:21:59 +1100 Matthew Waters * ext/qt/gstqtglutility.h: qt: fix build warning with clang and c-linkage of user defined type In file included from ../subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:17: ../subprojects/gst-plugins-good/ext/qt/gstqtglutility.h:35:16: error: 'qt_opengl_native_context_from_gst_gl_context' has C-linkage specified, but returns user-defined type 'QVariant' which is incompatible with C [-Werror,-Wreturn-type-c-linkage] QVariant qt_opengl_native_context_from_gst_gl_context (GstGLContext * context); Part-of: 2021-03-30 09:45:45 +0200 Stéphane Cerveau * ext/qt/gstqtelement.cc: * ext/qt/gstqtelements.h: * ext/qt/gstqtoverlay.cc: * ext/qt/gstqtsink.cc: * ext/qt/gstqtsrc.cc: qt: hotfix: allow per feature registration Fixes #869 Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-17 08:52:40 +0100 Stéphane Cerveau * ext/twolame/gsttwolamemp2enc.c: * ext/twolame/gsttwolamemp2enc.h: twolame: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:49:03 +0100 Stéphane Cerveau * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:38:46 +0100 Stéphane Cerveau * ext/cairo/gstcairo.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: cairo: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:34:34 +0100 Stéphane Cerveau * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: y4m: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:32:26 +0100 Stéphane Cerveau * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:29:40 +0100 Stéphane Cerveau * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:27:24 +0100 Stéphane Cerveau * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:27:12 +0100 Stéphane Cerveau * gst/monoscope/gstmonoscope.c: * gst/monoscope/gstmonoscope.h: monoscope: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:22:47 +0100 Stéphane Cerveau * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:19:52 +0100 Stéphane Cerveau * gst/id3demux/gstid3demux.c: * gst/id3demux/gstid3demux.h: id3demux: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:16:33 +0100 Stéphane Cerveau * gst/icydemux/gsticydemux.c: * gst/icydemux/gsticydemux.h: icydemux: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:14:26 +0100 Stéphane Cerveau * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom2k1: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:11:26 +0100 Stéphane Cerveau * gst/cutter/gstcutter.c: * gst/cutter/gstcutter.h: cutter: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:11:14 +0100 Stéphane Cerveau * gst/goom/gstgoom.c: * gst/goom/gstgoom.h: goom: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 17:10:33 +0100 Stéphane Cerveau * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 16:34:48 +0100 Stéphane Cerveau * sys/oss4/gstoss4audioplugin.c: * sys/oss4/meson.build: * sys/oss4/oss4-audio.c: * sys/oss4/oss4-audio.h: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-sink.h: * sys/oss4/oss4-source.c: * sys/oss4/oss4-source.h: oss4: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 16:11:36 +0100 Stéphane Cerveau * sys/oss/gstossaudio.c: * sys/oss/gstossaudioelement.c: * sys/oss/gstossaudioelements.h: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss/meson.build: oss: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 15:56:35 +0100 Stéphane Cerveau * gst/auparse/gstauparse.c: * gst/auparse/gstauparse.h: auparse: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 15:29:06 +0100 Stéphane Cerveau * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/gstv4l2element.c: * sys/v4l2/gstv4l2elements.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/meson.build: v4l2: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 15:05:43 +0100 Stéphane Cerveau * gst/videofilter/gstgamma.c: * gst/videofilter/gstgamma.h: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: * gst/videofilter/gstvideomedian.c: * gst/videofilter/gstvideomedian.h: * gst/videofilter/plugin.c: videofilter: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:58:57 +0100 Stéphane Cerveau * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocropelement.c: * gst/videocrop/gstvideocropelements.h: * gst/videocrop/gstvideocropplugin.c: * gst/videocrop/meson.build: videocrop: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:54:15 +0100 Stéphane Cerveau * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:49:56 +0100 Stéphane Cerveau * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudp.c: * gst/udp/gstudpelement.c: * gst/udp/gstudpelements.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/udp/meson.build: udp: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:43:32 +0100 Stéphane Cerveau * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: * gst/smpte/gstsmptealpha.c: * gst/smpte/gstsmptealpha.h: * gst/smpte/plugin.c: smpte: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:38:37 +0100 Stéphane Cerveau * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: shapewipe: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:35:51 +0100 Stéphane Cerveau * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtspelement.c: * gst/rtsp/gstrtspelements.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/meson.build: rtsp: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 14:24:33 +0100 Stéphane Cerveau * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: * gst/rtpmanager/gstrtpfunnel.c: * gst/rtpmanager/gstrtpfunnel.h: * gst/rtpmanager/gstrtphdrext-rfc6464.c: * gst/rtpmanager/gstrtphdrext-rfc6464.h: * gst/rtpmanager/gstrtphdrext-twcc.c: * gst/rtpmanager/gstrtphdrext-twcc.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxreceive.h: * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: * gst/rtpmanager/gstrtpst2022-1-fecdec.c: * gst/rtpmanager/gstrtpst2022-1-fecdec.h: * gst/rtpmanager/gstrtpst2022-1-fecenc.c: * gst/rtpmanager/gstrtpst2022-1-fecenc.h: rtpmanager: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 13:49:15 +0100 Stéphane Cerveau * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrglimiter.h: * gst/replaygain/gstrgvolume.c: * gst/replaygain/gstrgvolume.h: * gst/replaygain/replaygain.c: * gst/replaygain/rganalysis.h: replaygain: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 13:43:44 +0100 Stéphane Cerveau * gst/multipart/multipart.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: multipart: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 12:04:26 +0100 Stéphane Cerveau * gst/multifile/gstimagesequencesrc.c: * gst/multifile/gstimagesequencesrc.h: * gst/multifile/gstmultifile.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitfilesrc.h: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: multifile: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 11:14:17 +0100 Stéphane Cerveau * gst/matroska/gstmatroskaelement.c: * gst/matroska/gstmatroskaelements.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska.c: * gst/matroska/meson.build: * gst/matroska/webm-mux.c: matroska: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 10:59:34 +0100 Stéphane Cerveau * gst/level/gstlevel.c: * gst/level/gstlevel.h: level: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 10:57:58 +0100 Stéphane Cerveau * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: * gst/law/alaw.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.c: * gst/law/mulaw-encode.h: * gst/law/mulaw.c: law: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 10:26:40 +0100 Stéphane Cerveau * gst/isomp4/gstisomp4element.c: * gst/isomp4/gstisomp4elements.h: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/isomp4-plugin.c: * gst/isomp4/meson.build: * gst/isomp4/qtdemux.c: isomp4: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 09:57:27 +0100 Stéphane Cerveau * gst/interleave/deinterleave.c: * gst/interleave/gstinterleaveelements.h: * gst/interleave/interleave.c: * gst/interleave/plugin.c: interleave: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-16 09:51:16 +0100 Stéphane Cerveau * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: flx: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 17:37:09 +0100 Stéphane Cerveau * gst/flv/gstflvdemux.c: * gst/flv/gstflvelement.c: * gst/flv/gstflvelements.h: * gst/flv/gstflvmux.c: * gst/flv/gstflvplugin.c: * gst/flv/meson.build: flv: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 17:27:51 +0100 Stéphane Cerveau * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer.h: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/equalizer/gstiirequalizerplugin.c: * gst/equalizer/meson.build: equalizer: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 15:37:52 +0100 Stéphane Cerveau * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gsteffectv.c: * gst/effectv/gsteffectv.h: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effectv: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 15:03:10 +0100 Stéphane Cerveau * gst/dtmf/gstdtmf.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: dtmf: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 14:55:15 +0100 Stéphane Cerveau * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstdebug.c: * gst/debugutils/gstdebugutilselements.h: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavigationtest.h: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: debugutils: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 13:38:21 +0100 Stéphane Cerveau * gst/avi/gstavi.c: * gst/avi/gstavidemux.c: * gst/avi/gstavielement.c: * gst/avi/gstavielements.h: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/avi/meson.build: avi: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 13:02:59 +0100 Stéphane Cerveau * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautodetect.h: * gst/autodetect/gstautodetectelement.c: * gst/autodetect/gstautodetectelements.h: * gst/autodetect/gstautodetectplugin.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/autodetect/meson.build: autodetect: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 13:00:38 +0100 Stéphane Cerveau * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstaudioparserselements.h: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: * gst/audioparsers/plugin.c: audioparsers: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 12:44:31 +0100 Stéphane Cerveau * gst/apetag/gstapedemux.c: * gst/apetag/gstapedemux.h: apetag: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-15 11:00:46 +0100 Stéphane Cerveau * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxelement.c: * ext/vpx/gstvpxelements.h: * ext/vpx/meson.build: * ext/vpx/plugin.c: vpx: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 17:26:36 +0100 Stéphane Cerveau * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gsttaglibelement.c: * ext/taglib/gsttaglibelements.h: * ext/taglib/gsttaglibplugin.c: * ext/taglib/meson.build: taglib: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 17:09:19 +0100 Stéphane Cerveau * ext/qt/gstplugin.cc: * ext/qt/gstqtelement.cc: * ext/qt/gstqtelements.h: * ext/qt/gstqtoverlay.cc: * ext/qt/gstqtsink.cc: * ext/qt/gstqtsrc.cc: * ext/qt/meson.build: * ext/qt/qtplugin.pro: qt: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 16:09:53 +0100 Stéphane Cerveau * ext/speex/gstspeex.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexelement.c: * ext/speex/gstspeexelements.h: * ext/speex/gstspeexenc.c: * ext/speex/meson.build: speex: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 16:04:16 +0100 Stéphane Cerveau * ext/soup/gstsoup.c: * ext/soup/gstsoupelement.c: * ext/soup/gstsoupelements.h: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/soup/meson.build: soup: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 15:53:19 +0100 Stéphane Cerveau * ext/raw1394/gst1394.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gstdv1394src.h: * ext/raw1394/gsthdv1394src.c: * ext/raw1394/gsthdv1394src.h: raw1394: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 15:47:46 +0100 Stéphane Cerveau * ext/wavpack/gstwavpack.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackelement.c: * ext/wavpack/gstwavpackelements.h: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/meson.build: wavpack: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 15:35:11 +0100 Stéphane Cerveau * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 15:27:31 +0100 Stéphane Cerveau * gst/audiofx/audioamplify.c: * gst/audiofx/audioamplify.h: * gst/audiofx/audiochebband.c: * gst/audiofx/audiochebband.h: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiocheblimit.h: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiodynamic.h: * gst/audiofx/audioecho.c: * gst/audiofx/audioecho.h: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofirfilter.h: * gst/audiofx/audiofx.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioiirfilter.h: * gst/audiofx/audioinvert.c: * gst/audiofx/audioinvert.h: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiokaraoke.h: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiopanorama.h: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: * gst/audiofx/audiowsinclimit.h: * gst/audiofx/gstscaletempo.c: * gst/audiofx/gstscaletempo.h: * gst/audiofx/gststereo.c: * gst/audiofx/gststereo.h: audiofx: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 13:16:28 +0100 Stéphane Cerveau * gst/rtp/gstasteriskh263.c: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpL24depay.h: * gst/rtp/gstrtpL24pay.c: * gst/rtp/gstrtpL24pay.h: * gst/rtp/gstrtpL8depay.c: * gst/rtp/gstrtpL8depay.h: * gst/rtp/gstrtpL8pay.c: * gst/rtp/gstrtpL8pay.h: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3depay.h: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpac3pay.h: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvdepay.h: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpbvpay.h: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltdepay.h: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpceltpay.h: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvdepay.h: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpdvpay.h: * gst/rtp/gstrtpelement.c: * gst/rtp/gstrtpelements.h: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722depay.h: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg722pay.h: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723depay.h: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726depay.h: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg726pay.h: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpg729pay.h: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261depay.h: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph261pay.h: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263depay.h: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcdepay.h: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpisacdepay.c: * gst/rtp/gstrtpisacdepay.h: * gst/rtp/gstrtpisacpay.c: * gst/rtp/gstrtpisacpay.h: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegdepay.h: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpjpegpay.h: * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpklvdepay.h: * gst/rtp/gstrtpklvpay.c: * gst/rtp/gstrtpklvpay.h: * gst/rtp/gstrtpldacpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp1sdepay.h: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tdepay.h: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp2tpay.h: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4adepay.h: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4apay.h: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmparobustdepay.h: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvdepay.h: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpmpvpay.h: * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopusdepay.h: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqcelpdepay.h: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpqdmdepay.h: * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsbcdepay.h: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirendepay.h: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsirenpay.h: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpspeexpay.h: * gst/rtp/gstrtpstorage.c: * gst/rtp/gstrtpstreamdepay.c: * gst/rtp/gstrtpstreamdepay.h: * gst/rtp/gstrtpstreampay.c: * gst/rtp/gstrtpstreampay.h: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtpsv3vdepay.h: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: * gst/rtp/gstrtpvp9depay.c: * gst/rtp/gstrtpvp9depay.h: * gst/rtp/gstrtpvp9pay.c: * gst/rtp/gstrtpvp9pay.h: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: * gst/rtp/meson.build: * tests/check/meson.build: rtp: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 11:12:34 +0100 Stéphane Cerveau * ext/pulse/gstpulseelement.c: * ext/pulse/gstpulseelements.h: * ext/pulse/meson.build: * ext/pulse/plugin.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 10:41:29 +0100 Stéphane Cerveau * ext/mpg123/gstmpg123audiodec.c: * ext/mpg123/gstmpg123audiodec.h: mpeg123: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 10:33:50 +0100 Stéphane Cerveau * ext/libpng/gstpng.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: libpng: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 10:27:18 +0100 Stéphane Cerveau * ext/lame/gstlamemp3enc.c: * ext/lame/gstlamemp3enc.h: * ext/lame/plugin.c: lame: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 10:26:26 +0100 Stéphane Cerveau * ext/libcaca/gstcacaplugin.c: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacasink.h: * ext/libcaca/gstcacatv.c: * ext/libcaca/gstcacatv.h: * ext/libcaca/meson.build: libcaca: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 10:09:46 +0100 Stéphane Cerveau * ext/jpeg/gstjpeg.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegelements.h: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegplugin.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/jpeg/meson.build: jpeg: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 09:56:36 +0100 Stéphane Cerveau * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 08:57:55 +0100 Stéphane Cerveau * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gdk_pixbuf/gstgdkpixbufelement.c: * ext/gdk_pixbuf/gstgdkpixbufelements.h: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufplugin.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/meson.build: gdk_pixbuf: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-12 08:48:21 +0100 Stéphane Cerveau * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: * ext/gtk/gstplugin.c: gtk: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-11 19:53:30 +0100 Stéphane Cerveau * ext/flac/gstflac.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacelement.c: * ext/flac/gstflacelements.h: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/flac/meson.build: flac: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-02-11 18:57:03 +0100 Stéphane Cerveau * ext/dv/gstdv.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/dv/gstdvelement.c: * ext/dv/gstdvelements.h: * ext/dv/meson.build: dv: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2020-08-14 15:27:31 -0400 Julian Bouzas * ext/aalib/gstaaplugin.c: * ext/aalib/gstaasink.c: * ext/aalib/gstaasink.h: * ext/aalib/gstaatv.c: * ext/aalib/gstaatv.h: * ext/aalib/meson.build: aalib: allow per feature registration Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: 2021-03-19 17:19:43 +0100 Víctor Manuel Jáquez Leal * docs/gst_plugins_cache.json: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: * tests/check/elements/videocrop.c: videocrop: handle non raw caps features Currently, videocrop, only negotiates raw caps (system memory) because it's the type of memory it can modify. Nonetheless, it's also possible for the element to handle non-raw caps when only adding the crop meta is possible, in other words, when downstream buffer pools expose the crop API. This patch enable non-raw caps negotiation. If downstream doesn't expose crop API and negotiated caps are featured, the negotiation fails. Part-of: 2021-03-19 10:35:09 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Don't special-case G_SIGNAL_RUN_CLEANUP stage in signal accumulators All these signals don't run the class handler in the CLEANUP stage. Part-of: 2021-03-19 10:34:33 +0200 Sebastian Dröge * ext/shout2/gstshout2.c: shout2: Don't register signal without class handler with G_SIGNAL_RUN_CLEANUP There is no class handler to run during the CLEANUP stage. Part-of: 2021-03-23 16:59:28 +0800 Hou Qi * sys/v4l2/gstv4l2object.c: v4l2object: Avoid colorimetry mismatch for streams with invalid colorimetry video-info sets gst colorimetry to default value when colorimetry in caps is unparsable or invalid. Then v4l2object uses this gst colorimetry to do mapping with v4l2 colorimetry. This may cause colorimetry mismatch when check mapped gst colorimetry with that read from caps directly. To fix this, need to correct gst colorimetry as that parsed from video-info when check gst_v4l2_video_colorimetry_matches(). Part-of: 2021-03-19 10:52:26 +0800 Hou Qi * sys/v4l2/gstv4l2object.c: v4l2object: Add support for hdr10 stream playback Colorimetry of hdr10 video is bt2100-pq with transfer as GST_VIDEO_TRANSFER_SMPTE2084. So map GST_VIDEO_TRANSFER_SMPTE2084 to V4L2_XFER_FUNC_SMPTE2084 to support hdr10 stream playback. Part-of: 2021-03-20 10:41:29 -0500 Sid Sethupathi * gst/shapewipe/gstshapewipe.c: shapewipe: fix broken link in docs Part-of: 2021-03-18 17:42:02 +0000 Alba Mendez * docs/gst_plugins_cache.json: * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix more signals Behaviour change in GLib causes select-stream signal to discard the value returned by handlers. See !909 for more info. Part-of: 2021-03-18 19:52:53 +1100 Matthew Waters * ext/jack/gstjack.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/pulse/pulsesink.h: * ext/qt/gstqsgtexture.cc: * ext/qt/gstqtglutility.cc: * ext/qt/qtglrenderer.cc: * ext/qt/qtitem.cc: * ext/qt/qtwindow.cc: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxenc.c: * gst/audioparsers/gstac3parse.h: * sys/rpicamsrc/gstrpicamsrc.c: * sys/ximage/ximageutil.c: gst: don't use volatile to mean atomic volatile is not sufficient to provide atomic guarantees and real atomics should be used instead. GCC 11 has started warning about using volatile with atomic operations. https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719 Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868 Part-of: 2021-03-17 15:54:59 +0530 Nirbheek Chauhan * docs/gst_plugins_cache.json: * gst/rtsp/gstrtspsrc.c: Update docs cache and fix before-send signal doc syntax The docs for before-send were missing because of this Part-of: 2021-03-17 13:18:34 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix accumulation of before-send signal return values Since glib 2.62, the accumulated return values in RUN_CLEANUP override the accumulated return values in RUN_FIRST. Since: 1. We have a default handler that always returns TRUE, and 2. User handlers are only run in RUN_FIRST, and 3. Our accumulator just takes the latest return value We were discarding the return value from the user handler and always sending messages even if the user handler said not to. See https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details. This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it to RUN_LAST so that it's emitted exactly once and accumulated once. With this fix, this signal can now be used to intercept PAUSE when going to GST_STATE_NULL so that the server does a TEARDOWN (if necessary) and not a PAUSE, which will confuse other RTSP clients when playing shared media. Part-of: 2021-03-17 11:32:08 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: Revert unusable workaround for PAUSE being sent when going NULL Directly setting rtspsrc to the NULL state before putting the pipeline in the NULL state usually works, but it can cause a deadlock in some cases, so it's not a reliable mechanism to fix this. This reverts commit f37afdafff1fd0a339966116261f5cd0de53f5d1: "rtspsrc: Fix state changes from PAUSED to PLAYING" and commit 76d624b2df5594a82269b94dffe8766a372d059d: "rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL" Part-of: 2021-03-16 19:25:36 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix parsing of the mediaclk:direct= field Due to an off-by-one when parsing the string, the most significant digit or the clock offset was skipped when parsing the offset. Part-of: 2021-03-16 00:08:43 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix state changes from PAUSED to PLAYING This was accidentally broken in the last commit that touched this because I missed the fall-through in the case immediately above this. Part-of: 2021-03-04 13:05:19 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Fix extraction of multichannel WavPack The old code had a couple of issues that all lead to potential memory safety bugs. - Use a constant for the Wavpack4Header size instead of using sizeof. It's written out into the data and not from the struct and who knows what special alignment/padding requirements some C compilers have. - gst_buffer_set_size() does not realloc the buffer when setting a bigger size than allocated, it only allows growing up to the maximum allocated size. Instead use a GstAdapter to collect all the blocks and take out everything at once in the end. - Check that enough data is actually available in the input and otherwise handle it an error in all cases instead of silently ignoring it. Among other things this fixes out of bounds writes because the code assumed gst_buffer_set_size() can grow the buffer and simply wrote after the end of the buffer. Thanks to Natalie Silvanovich for reporting. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/859 Part-of: 2021-03-03 11:31:52 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Initialize track context out parameter to NULL before parsing Various error return paths don't set it to NULL and callers are only checking if the pointer is NULL. As it's allocated on the stack this usually contains random stack memory, and more often than not the memory of a previously parsed track. This then causes all kinds of memory corruptions further down the line. Thanks to Natalie Silvanovich for reporting. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/858 Part-of: 2021-03-15 12:57:19 +0530 Nirbheek Chauhan * gst/rtsp/gstrtspsrc.c: rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL This usually doesn't matter, but it is disruptive when streaming from a shared media since it will pause all other clients when any client exits. This new behaviour is opt-in and should be safe because you need to set the NULL state on rtspsrc directly, instead of just on the pipeline. See the updated documentation for an explanation. Part-of: 2021-01-18 15:54:43 +0100 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: handle GST_VIDEO_TRANSFER_BT601 V4L2 makes no difference between the BT.601 and BT.709 transfer functions [1], but GStreamer does since 1.18 [2]. Adapt gst_v4l2_object_get_colorspace() and gst_v4l2_object_set_format_full(). [1] https://linuxtv.org/downloads/v4l-dvb-apis-new/userspace-api/v4l/colorspaces-details.html#colorspace-smpte-170m-v4l2-colorspace-smpte170m [2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724 Part-of: 2021-03-11 22:22:15 +0100 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: fix title of a few properties docstrings GstRtspSrc -> GstRTSPSrc This would have been noticed by the since checker, but those properties were introduced prior to that. Part-of: 2021-03-07 21:25:01 +0000 Vladimir Menshakov * docs/gst_plugins_cache.json: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: wavpackdec: Add floating point format support This commit negotiate F32 audio format if MODE_FLOAT used in wavpack file. Wavpack float mode is always in 32-bit IEEE format. The following pipeline plays distorted audio if source file is encoded in float mode: gst-launch-1.0 filesrc ... ! wavpackparse ! wavpackdec ! pulsesink Part-of: 2021-03-04 16:40:06 +1100 Matthew Waters * gst/matroska/matroska-demux.c: matroska: also support push-mode from seek events sent to the element Otherwise sending seek events would fail to actually seek. Part-of: 2021-02-26 10:49:10 +0100 Marc Leeman * gst/rtsp/gstrtspsrc.c: gstrtspsrc: 551 should not result in an unhandled error Some cameras (e.g. HikVision DS-2CD2732F-IS) return "551 Option not supported" when a command is sent that is not implemented (e.g. PAUSE). Instead; it should return "501 Not Implemented". This is wrong, as previously, the camera did announce support for PAUSE in the OPTIONS. In this case, handle the 551 as if it was 501 to avoid throwing errors to application level. */ Part-of: 2021-03-01 14:32:40 +0800 Hou Qi * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Do not expose profiles/levels in vp8/vp9 template caps Vp8/vp9 supported profiles/levels are listed in decoder sink caps, but there is no parser for these two formats and the demuxers also don't have these information. It causes negotiation fail between demuxers and decoder when check caps "accept = gst_caps_is_subset (caps, template_caps);". To fix this, need to remove profiles/levels for vp8/vp9 formats in decoder sink caps. Fix #854 Part-of: 2021-03-03 18:30:39 +0900 Seungha Yang * gst/rtpmanager/gstrtphdrext-twcc.h: rtpmanager: Fix an MSVC compile warning We don't expect this object is a part of public library. gstrtphdrext-twcc.c(45): warning C4273: 'gst_rtp_header_extension_twcc_get_type': inconsistent dll linkage Part-of: 2021-02-24 13:25:43 +0100 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2videodec: fix src side frame rate negotiation Negotiating v4l2h264dec ! v4l2h264enc transcoding pipelines fails in case the encoder does not accept framerate=(fraction)0/1. The acquired caps used for downstream negotiation are determined from gst_v4l2_object_acquire_format(), which sets the GstVideoInfo::fps_n and ::fps_d fields to 0. To fix this, copy the frame rate from the sink side. Part-of: 2021-02-16 16:20:05 +0200 Jordan Petridis * sys/rpicamsrc/meson.build: rpicamsrc: depend on posix threads and vchiq_arm Could only test on rpi 3b+ Close #839 Part-of: 2021-02-11 14:48:07 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Silence traces around unsupported source change Don't be too spamy about unsupported source change flags as these will be commonly extended in the future. Part-of: 2021-02-11 14:24:29 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Move preferred resolution query before the probe As we lock the DV_TIMINGS (and standards in the future), we need to probe the caps after, otherwise, we may endup fixating to an unsupported resolution, which would lead to a not-negotiated error. Part-of: 2021-02-10 16:37:01 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2_calls.c: v4l2src: Calculate framerate from DV timings And use this framerate in our preference. Note that we also flush the probed caps as it seems that the format enumeration may change when a new source change event get triggered. Part-of: 2021-02-10 15:52:55 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2_calls.c: v4l2rc: Add DV_TIMINGS query and locking This adds support to DV_TIMINGS query and locking. The timing width and height is then used as a preference. Part-of: 2021-02-10 15:49:03 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Force renegotiation on resolution change As mandated by the specification, make sure to cycle through streamoff / streamon regardless if the caps have changed or not. Part-of: 2021-02-10 14:52:14 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: v4l2object: Remove unused streaming member Part-of: 2021-02-10 10:48:48 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Refactor to use PreferredCapsInfo structure Avoid passing around a bare structure for the preference, this removes the need to copy and free that structure and simplify the code. Also fix a type in the structure name, Prefered -> Preferred. Part-of: 2021-02-08 17:27:20 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Stub preferred resolution support This stubs the ability to use preferred resolution from digital video timings, analog TV standards or driver reported native resolution. Part-of: 2021-02-09 14:44:02 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: Subscribe source_change for the current input When we subscribe for source-change event, we need to specify for which input. Make sure we subscribe for the current input. Part-of: 2021-02-08 17:26:20 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2src: Add input signal status detection As part of the support to select a preferred size, we can also detect the signal status. This is a split patch so that feature is separated to ease review. Part-of: 2021-02-08 17:24:00 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: Add helper to query input status This is a wrapper around ENUM_INPUT renamed for readability. Part-of: 2021-02-08 17:22:37 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/v4l2_calls.c: v4l2: Fix input/output index sign This is an unsigned integer in the kernel API. Part-of: 2021-02-04 16:59:44 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Add source resolution change support This patch adds support for source resolution change detection. Resolution change is signaled by drivers when a change in the detected signal have been detected. This is notably seen on HDMI receivers. Part-of: 2021-02-04 14:13:32 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Handle resolution change event This patch adds the detection, dequeuing and reporting of the SOURCE_CHANGE event when the CH_RESOLUTION flag is set. The acquire function will now return a new custom success called GST_V4L2_FLOW_RESOLUTION_CHANGE. In order to use this new feature, elements must enable it by calling: gst_v4l2_buffer_pool_enable_resolution_change (pool); Part-of: 2021-02-04 11:01:38 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2object: Add event helpers Part-of: 2021-02-04 10:10:34 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: use FLOW_LAST_BUFFER This uses the GST_V4L2_FLOW_LAST_BUFFER alias instead of GST_FLOW_CUSTOM_SUCCESS to make the code more readable. Part-of: 2018-12-10 14:10:05 +0100 Lucas Stach * sys/v4l2/gstv4l2object.c: v4l2object: prefer NV12 over I420 Considering NV12 an 'odd' format is a historical artifact. This format is now quite common, and usually preferable to I420 due to more memory friendly access patterns. Part-of: 2021-02-18 10:34:25 +0100 Guillaume Desmottes * gst/wavparse/gstwavparse.c: * tests/check/elements/wavparse.c: wavparse: fix seeking in READY state wavparse claims to be able to support seeking in the READY state by saving the pending seek event and actually seeking later after having parsed the header. Problem was that this seek event was reset on the READY to PAUSED transition, making all this code useless. Fixing it by stop resetting on READY to PAUSED transition as we already reset on PAUSED to READY and when initiating the element. Note that DTS marker detection isn't support in such scenario as gst_type_find_helper_for_buffer() needs a buffer containing the beginning of the stream. Part-of: 2021-02-18 10:05:03 +0100 Guillaume Desmottes * tests/check/elements/wavparse.c: tests: wavparse: factor out create_pipeline() No semantic change. Part-of: 2021-02-18 00:34:02 +0100 Mathieu Duponchelle * docs/gst_plugins_cache.json: docs: update plugins cache with new h264 / vp8 depay properties Part-of: 2020-12-09 01:40:45 +0100 Mathieu Duponchelle * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: expose request-keyframe property When set, the depayloader will request new keyframes on packet loss Part-of: 2020-12-09 01:34:20 +0100 Mathieu Duponchelle * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: rtpvp8depay: expose request-keyframe property When set, the depayloader will request new keyframes on packet loss Part-of: 2020-12-09 01:24:57 +0100 Mathieu Duponchelle * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: expose wait-for-keyframe property Similar to rtpvp8depay, when packet loss occurs, the depayloader starts waiting for a keyframe. We try to only stop waiting when all the packets for the new keyframe have been received, by only resetting waiting_for_keyframe when encountering the first packet of a keyframe, this is slightly fragile because there is no bit that explicitly marks the start of an access unit, so we rely on the existing picture_start detection code. As a consequence, the property is only meaningful when outputting access units, and is ignored when outputting NALs directly. Part-of: 2021-02-18 00:36:43 +0100 Mathieu Duponchelle * docs/gst_plugins_cache.json: * gst/videomixer/videomixer2.c: videomixer: document as deprecated Part-of: 2021-02-16 22:20:17 +1100 Ashley Brighthope * gst/wavenc/gstwavenc.c: wavenc: Fixed INFO chunk corruption, caused by odd sized data not being padded. Code style was updated. Part-of: 2020-12-07 19:51:35 +0100 Jakub Adam * gst/rtp/gstrtpopuspay.c: rtpopuspay: add info regarding (non-standard) multichannel support Part-of: 2020-12-07 16:50:01 +0100 Jakub Adam * docs/gst_plugins_cache.json: docs: update plugins cache for rtpopus Part-of: 2020-12-01 20:09:58 +0100 Jakub Adam * tests/check/elements/rtpopus.c: tests: add rtpopus multichannel test cases Part-of: 2020-12-01 16:43:32 +0100 Jakub Adam * gst/rtp/gstrtpopusdepay.c: rtpopusdepay: support libwebrtc-compatible multichannel payload Part-of: 2020-11-30 21:49:48 +0100 Jakub Adam * gst/rtp/gstrtpopuspay.c: rtpopuspay: support libwebrtc-compatible multichannel payload When the audio has more than 2 channels, add optional fields to output caps from which webrtcbin can generate SDP in the syntax recognized by "multiopus" codec present in libwebrtc [1]. e.g. for 5.1 audio: a=rtpmap:96 multiopus/48000/6 a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5 [1] https://webrtc-review.googlesource.com/c/src/+/129768 Part-of: 2020-11-30 22:10:14 +0100 Jakub Adam * gst/rtp/gstrtpopuspay.c: rtpopuspay: make use of gst_rtp_base_payload_set_outcaps_structure() Part-of: 2021-02-09 19:31:28 -0500 Olivier Crête * gst/effectv/LICENSE: effectv: Remove redundant license file Part-of: 2021-02-05 00:55:12 +0000 Kevin Song * sys/v4l2/gstv4l2videoenc.c: Apply 1 suggestion(s) to 1 file(s) Part-of: 2021-02-05 00:55:04 +0000 Kevin Song * sys/v4l2/gstv4l2videoenc.c: Apply 1 suggestion(s) to 1 file(s) Part-of: 2021-02-04 13:43:17 +0800 Bing Song * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: support resolution change stream encode. Resolution change stream transcoding will drain before send new video frame buffer. Need encode video frame after process EOS. Part-of: 2021-02-04 11:44:53 +0100 Xabier Rodriguez Calvar * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: added support for cbcs encryption scheme Part-of: 2021-01-21 18:04:58 +0100 Guillaume Desmottes * docs/gst_plugins_cache.json: * gst/rtpmanager/gstrtphdrext-rfc6464.c: * gst/rtpmanager/gstrtphdrext-rfc6464.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/meson.build: * tests/check/elements/rtphdrextrfc6464.c: * tests/check/meson.build: rtp: add rtphdrextrfc6464 Header Extension for Client-to-Mixer Audio Level Indication as defined in RFC 6464. Part-of: 2020-06-16 12:01:30 +0200 Guillaume Desmottes * docs/gst_plugins_cache.json: * gst/level/gstlevel.c: * gst/level/gstlevel.h: * tests/check/elements/level.c: level: add GstRTPAudioLevelMeta on buffers This meta can be used by a RTP payloader to send the level information to the peer. Part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/446 Part-of: 2021-02-03 17:10:20 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Provide documentation for GST_DEINTERLACE_BUFFER_STATE More information available in https://gstconf.ubicast.tv/videos/interlacing-and-telecine-in-gstreamer/ Part-of: 2021-01-30 16:16:13 +0200 Vivia Nikolaidou * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: Fix telecine/onefield mixup https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/838 Part-of: 2021-01-30 15:49:23 +0200 Vivia Nikolaidou * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: Better alternate support Improve line offset halving based on whether this field is top or bottom. Also handle the buffer state the same as mixed. Part-of: 2021-01-14 01:12:06 +0800 Bing Song * sys/v4l2/gstv4l2h265codec.c: v4l2h265codec: fix HEVC profile string issue. Keep HEVC profile compatible with other module. Part-of: 2020-12-15 10:41:40 +0800 Bing Song * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Need keep same transfer as input caps. GST_VIDEO_TRANSFER_BT2020_12 and GST_VIDEO_TRANSFER_BT2020_10 will be mapped to V4L2_XFER_FUNC_709. Need check input caps when map V4L2_XFER_FUNC_709 back to GST_VIDEO_TRANSFER_BT2020_12 and GST_VIDEO_TRANSFER_BT2020_10 Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/816 Part-of: 2020-12-07 10:01:53 +0100 Tobias Ronge * gst/rtsp/gstrtspsrc.c: rtspsrc: Do not wait for response while flushing Due to the may_cancel flag in GstRTSPConnection, receiving might not get cancelled when supposed to. In this case, gst_rtsp_src_receive_response will have to wait until timeout instead but if busy receiving RTP data, this timeout will never occur. With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR if flushing is set to TRUE instead of continuing to receive. Part-of: 2021-01-14 19:13:03 +0000 Tim-Philipp Müller * ext/dv/meson.build: meson: allow libdv subproject fallback Part-of: 2020-12-21 13:55:58 +0100 Xabier Rodriguez Calvar * gst/isomp4/qtdemux.c: qtdemux: Allow streams with no specified protection system ID This is necessary in cases like CMAF where there won't be any events passing thru. Part-of: 2021-01-07 16:57:27 +0800 Hou Qi * docs/gst_plugins_cache.json: * sys/v4l2/gstv4l2object.c: v4l2object: Map correct video format for RGBA Map V4L2_PIX_FMT_RGBA32 pixel format to GST_VIDEO_FORMAT_RGBA instead of GST_VIDEO_FORMAT_RGB video format to support RGBA. Fixes #823 Part-of: 2021-01-02 13:06:16 +0530 Sanchayan Maity * gst/udp/gstudpsrc.c: udpsrc: Fix marker links These should be with a single ':'. The double '::' results in a CI with build failure message like below. ERROR: [links]: (mandatory-link-not-found): Mandatory link Link GstSocketTimestamp -> None (GstSocketTimestamp) could not be resolved ERROR: [check-missing-since-markers]: (missing-since-marker): Missing since marker for udpsrc:socket-timestamp Part-of: 2020-12-17 11:24:07 +0530 Sanchayan Maity * docs/gst_plugins_cache.json: * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: Allow use of socket control message timestamps for DTS Part-of: 2020-12-09 20:20:18 +1100 Matthew Waters * docs/gst_plugins_cache.json: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: * tests/check/elements/videoflip.c: videoflip: fix possible crash when setting the video-direction while running A classic case of not enough locking. One interesting thing with this is the interaction between the rotation value and caps negotiation. i.e. the width/height of the caps can be swapped depending on the video-direction property. We can't lock the entirety of the caps negotiation for obvious reasons so we need to do something else. This takes the approach of trying to use a single rotation value throughout the entirety of the negotiation and then subsequent output frame in a kind of latching sequence. Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792 Part-of: 2020-12-09 19:49:47 +1100 Matthew Waters * tests/check/elements/videoflip.c: * tests/check/meson.build: tests: add tests for videoflip Part-of: 2020-12-30 13:38:46 +0100 Ignacio Casal Quinteiro * gst/deinterlace/meson.build: deinterlace: force -DPREFIX on macos This is due to a bug in meson where it will not detect properly the compiler if the symbols need an undercore. https://github.com/mesonbuild/meson/issues/5482 Fixes #821 Part-of: 2020-12-15 11:36:27 +0200 Sebastian Dröge * docs/gst_plugins_cache.json: * gst/rtsp/gstrtspsrc.c: rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal Part-of: 2020-12-10 14:27:49 +0200 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Avoid deadlock when releasing a pad from a running muxer Might not drain correctly Part-of: 2020-12-11 11:24:14 +0800 Hou Qi * sys/v4l2/gstv4l2object.c: v4l2object: Use active resolution during fallback colorspace probe For legacy drivers that don't implement ENUM_FRAMESIZE, use active resolution to probe colorspace. This can improve the accuracy of the result when the colorspace depends on the resolution. This fixes a wrong colorspace issue on board with vendor bsp at resolution 2560x1440. Part-of: 2020-12-12 04:02:37 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpst2022-1-fecdec.c: rtpst2022-1-fecdec: don't xor out of bounds When reconstituting packets from a stream with variable packet sizes, don't xor larger packets past the length of the protected packet Part-of: 2020-12-12 04:00:41 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpst2022-1-fecenc.c: rtpst2022-1-fecenc: memset when reallocating xored payload When protecting packets with a variable payload length, we reallocate the xored payload when needed. It is a good idea to memset the extended memory to 0 so that we don't xor data with garbage! Part-of: 2020-12-12 03:56:11 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpst2022-1-fecdec.c: * gst/rtpmanager/gstrtpst2022-1-fecenc.c: rtpst2022-1-fec-*: protect additional RTP header fields While the standard is a bit vague about whether the padding, extension and marker bits should be protected: > The usage, by senders and receivers, of the following bits shall > be defined by the associated video/audio transport standards: It is obviously necessary and useful for some formats (eg VP8) that those indeed be protected. Part-of: 2020-12-12 03:28:56 +1100 Jan Schmidt * tests/check/elements/splitmuxsink.c: splitmuxsink: Unit test - check format/opened/closed sequence Check the sequence of format-location/fragment-opened/fragment-closed events is respected. There should be 1 format-location call for each fragment-opened message, and 1 fragment-closed for each. Part-of: 2020-12-09 00:40:52 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Fix for 'reference bytes muxed' check. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798 introduced a check in the need-new-fragment logic to avoid starting a new fragment unless there has been some data on the reference stream, but the check is done against the number of bytes that have been received on the input, not the number that were released for output into the current fragment. Fix the check to remember and test against bytes that have been sent for output. This also fixes a problem where starting a new fragment fails to request a new filename from the format-location signal. Part-of: 2020-09-15 00:27:24 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Add debug for fragment opened/closed msgs When posting fragment-opened and fragment-closed messages, put a debug statement in the logs Part-of: 2020-08-18 16:06:14 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Convert asserts into element errors. Change some g_assert into element errors so that they can be caught and the pipeline shut down. Part-of: 2020-07-10 15:36:54 +1000 Matthew Waters * docs/gst_plugins_cache.json: * gst/rtpmanager/gstrtpfunnel.c: * gst/rtpmanager/gstrtphdrext-twcc.c: * gst/rtpmanager/gstrtphdrext-twcc.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/meson.build: rtpmanager: update for rtp header extensions Provide an implementation of the transport-wide-cc header extension and use it in rtpfunnel. Part-of: 2020-11-15 11:30:07 +0000 Jose Quaresma * sys/rpicamsrc/meson.build: rpicamsrc: add vchostif library as it is required to build successful fix: undefined reference to `vc_gencmd' /usr/src/debug/gstreamer1.0-plugins-good/1.18.1-r0/build/../gst-plugins-good-1.18.1/sys/rpicamsrc/RaspiCamControl.c:1440: undefined reference to `vc_gencmd' Part-of: 2020-11-25 17:51:24 +0100 Marijn Suijten * tests/check/elements/rtp-payloading.c: tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function The function is renamed to be properly associated with AudioFormatInfo (its instance) instead of AudioFormat (an unrelated enum), see [1] for the rename itself. [1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940 2020-11-24 22:11:50 +0530 Nirbheek Chauhan * gst/deinterlace/meson.build: * meson.build: deinterlace: Enable x86 assembly with nasm on MSVC We need to remove x86inc.asm from the list of compiled assembly files because it is not supposed to be compiled separately. It is directly included by yadif.asm, and it exports no symbols. The object file was getting ignored on all platforms except on msvc where it was causing a linker hang when building with debugging enabled because the object file had no debug symbols (or similar). We've seen this before in FFmpeg too, which uses nasm: https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg/-/merge_requests/46 Part-of: 2020-11-19 17:47:21 +1100 Matthew Waters * ext/qt/gstqtoverlay.cc: * ext/qt/gstqtsink.cc: qml: add some docs on display and contexts Especially considering some dynamic pipeline scenarios. Part-of: 2020-11-18 20:09:24 +0100 Tim Schneider * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Added "src->started = FALSE;" to gst_rpi_cam_src_stop Makes the element reusable multiple times after a state change back to READY. Fixes #105 Part-of: 2020-11-12 09:32:30 +0800 Bing Song * docs/gst_plugins_cache.json: * sys/v4l2/gstv4l2object.c: v4l2: caps negotiate wrong as interlace feature gst_caps_simplify() will move interlace format before normal video format. It will cause caps negotiate prefer interlaced caps which isn't expected. Seperate normal caps and interlaced caps and then merge it will keep prefer progress video format. Add ARGB/BGRA for interlaced caps. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/802 Part-of Part-of: 2020-11-13 21:25:42 +0100 Havard Graff * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: never send on a non-internal source This will end up as a "received" packet, due to the code in source_push_rtp, which will think this is a packet being received. Instead drop the packet and hope that either: 1. Something upstream responds to the GstRTPCollision event and changes SSRC used for sending. 2. That the application responds to the "on-ssrc-collision" signal, and forces the sender (payloader) to change its SSRC. 3. That the BYE sent to the existing user of this SSRC will respond to the BYE, and that we timeout this source, so we can continue sending using the chosen SSRC. The test reproduces a scenario where we previously would have sent on a non-internal source. Part-of: 2020-11-13 12:39:53 +0100 Havard Graff * gst/rtpmanager/rtpsource.c: rtpsource: rewrite timeout-check to avoid underflow If current_time is < collision_timeout, we get an uint64 underflow, and the check will trigger prematurely. Part-of: 2020-11-13 14:58:44 +0200 Vivia Nikolaidou * gst/audioparsers/gstaacparse.c: aacparse: Fix caps change handling In baseparse we set the fixed caps flag on all src pads, therefore the source pad caps query in get_allowed_caps will return the current caps. Current caps won't necessarily intersect with the new caps (e.g. sample rate change). Replace get_allowed_caps with peer_query_caps. Part-of: 2020-11-12 23:39:21 +0000 Tim-Philipp Müller * tests/check/elements/qtdemux.c: tests: qtdemux: fix typo in caps field timesacle -> timescale Part-of: 2020-11-12 23:38:21 +0000 Tim-Philipp Müller * tests/check/elements/qtdemux.c: tests: qtdemux: fix crash on 32-bit architectures Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/803 Part-of: 2020-09-14 13:12:50 +0530 Sanchayan Maity * docs/gst_plugins_cache.json: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpldacpay.c: * gst/rtp/gstrtpldacpay.h: * gst/rtp/meson.build: rtp: ldacpay: Add LDAC RTP payloader Part-of: 2020-11-03 15:58:30 +0200 Sebastian Dröge * ext/qt/gstqsgtexture.cc: * ext/qt/gstqsgtexture.h: * ext/qt/qtitem.cc: qmlglsink: Keep old buffers around a bit longer if they were bound by QML We don't know exactly when QML will stop using them but it should be safe to unref them after at least 2 more buffers were bound. Part-of: 2020-11-10 18:18:12 +0000 ChrisDuncanAnyvision * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Ensure same group-id used for both TCP/UDP stream-start events Part-of: 2020-11-10 16:17:23 +0000 ChrisDuncanAnyvision * gst/rtsp/gstrtspsrc.c: rtspsrc: Use consistent URI hashed stream-id for UDP and TCP/Interleaved streams Part-of: 2020-11-04 18:43:04 +0530 Nirbheek Chauhan * meson.build: meson: Enable some MSVC warnings for parity with GCC/Clang This makes it easier to do development with MSVC by making it warn on common issues that GCC/Clang error out for in our CI configuration. Continuation from https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/223 Part-of: 2020-10-15 21:42:40 -0400 Olivier Crête * docs/gst_plugins_cache.json: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: rtpsource: Report for which local SSRC is a remote RB reporting on This is useful in the Bundle case because there may be multiple local and remote SSRCs in the same session. Part-of: 2020-10-29 15:58:38 +0100 Guillaume Desmottes * docs/gst_plugins_cache.json: * gst/rtp/gstrtpisacdepay.c: * gst/rtp/gstrtpisacpay.c: docs: update plugins cache Part-of: 2020-03-20 13:15:33 +0100 Guillaume Desmottes * gst/rtp/gstrtp.c: * gst/rtp/gstrtpisacdepay.c: * gst/rtp/gstrtpisacdepay.h: * gst/rtp/meson.build: rtp: add rtpisacdepay Depayload for the iSAC audio codec. Part-of: 2020-03-20 13:15:33 +0100 Guillaume Desmottes * gst/rtp/gstrtp.c: * gst/rtp/gstrtpisacpay.c: * gst/rtp/gstrtpisacpay.h: * gst/rtp/meson.build: rtp: add rtpisacpay Payload for the iSAC audio codec. Part-of: 2020-11-01 18:36:49 +0000 Dinesh Manajipet * ext/qt/qtitem.cc: qmlglsink: Set qtitem's implicit width/height This can be useful to let the layouts automatically resize qtitem and also easily query a video's width/height from QML Part-of: 2020-11-01 10:30:27 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Release pads via GstAggregator See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797 Part-of: 2020-10-26 12:40:49 +1100 Matthew Waters * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: support muxing multiple codec_data for h264/h265 Each codec_data is put into its own SampleTableEntry inside the stsd. Part-of: 2020-10-29 14:54:16 +0100 Stéphane Cerveau * docs/gst_plugins_cache.json: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstnavseek.h: navseek: add hold_eos property This property will tell the element to hold the EOS event and keep it until the next keystroke. Part-of: 2020-10-31 12:52:04 +1100 Jan Schmidt * tests/check/elements/splitmuxsrc.c: splitmuxsrc: Fix comment in a test Fix a comment in the splitmuxsrc robust muxing test so it describes the test properly. Part-of: 2020-10-31 12:49:08 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Change EOS catching logic. Add a new state for ending the overall stream, and use it to decide whether to pass the final EOS message up the bus instead of dropping it. Fixes a small race that makes the testsuite sometimes not generate the last fragment(s) sometimes because the wrong EOS gets allowed through too early. Part-of: 2020-10-31 02:19:07 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Don't use the element state lock Using the element state lock to avoid splitmuxsink shutting down while doing element manipulations can lead to a deadlock on shutdown if a fragment switch happens at exactly the wrong moment. Use a private mutex and a shutdown boolean instead. Part-of: 2020-10-30 03:38:15 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't busy loop on a non-ready pad. If a pad gets into the check_completed_gop method and then the underlying conditions change on the reference context, things could get stuck in a busy loop when the context should instead jump back out and wait for more data. Part-of: 2020-10-30 03:36:51 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Mark running=false on shutdown. Make sure that any late gst_element_call_async() callbacks know that the elements is shutting down and bail out instead of operating on the element we're trying to stop. Fixes a spurious test failure in elements_splitmuxsrc Part-of: 2020-10-29 02:36:35 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Forward EOS messages from async fragments. Re-enable forwarding EOS messages from fragments that are completing asynchronously, so that splitmuxsink itself won't go EOS until they are complete. This was disabled to work around a bug in core that is fixed in https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/683 Part-of: 2020-09-17 22:56:01 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Never start a new fragment with no reference buffers If there has been no bytes from the reference stream muxed into the current fragment, then time can't have advanced, there's no GOP... this fragment would be broken or empty, so wait for some data on the reference buffer. Part-of: 2020-10-29 02:38:16 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Chain up when releasing pad, and fix some locking. Release pads by calling up into aggregator so it can do the right things. Don't clean up the pad until after that. Add some missing locks around some accesses to shared pad state. Part-of: 2018-08-13 15:35:11 +0200 Stian Selnes * gst/rtp/gstrtpvp9depay.c: * gst/rtp/gstrtpvp9depay.h: * tests/check/elements/rtpvp9.c: rtpvp9depay: Improve SVC parsing, aggregate all layers - Fix start and end of picture to support multiple layers. Start of picture is the first packet of the base layer, while end of picture is when the marker bit is set (last packet of the enhancement layers). - All "layers" (aka "frames") of a picture are pushed downstream in a single buffer when picture is complete. - Forgive SID=0 for enhancement layers (invalid, but Chrome and Firefox sends it) Part-of: 2020-10-30 03:09:48 +0100 Stian Selnes * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: * tests/check/elements/rtpvp8.c: rtpvp8depay: Send lost events when marker bit is missing This means the previous frame was incomplete. Part-of: 2020-10-14 23:17:53 +0200 Knut Saastad * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: rtpvp9depay: detect incomplete frames and bail out If a packet with the B bit set arrives but we haven't received a packet with the marker or E bits set to end the previous frame, we know the current frame was incomplete. Part-of: 2020-10-14 23:17:53 +0200 Knut Saastad * gst/rtp/gstrtpvp9depay.c: rtpvp9depay: detect incomplete frames and bail out If a packet with the B bit set arrives but we haven't received a packet with the marker or E bits set to end the previous frame, we know the current frame was incomplete. Part-of: 2020-10-14 01:28:50 +0200 Mikhail Fludkov * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: * gst/rtp/gstrtpvp9depay.c: * gst/rtp/gstrtpvp9depay.h: * tests/check/elements/rtpvp8.c: * tests/check/elements/rtpvp9.c: rtpvp*depay: possibly forward might-have-been-fec PacketLost events This is ad adaptation of a Pexip patch for dealing with spurious GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets under that scheme are spliced in the same sequence domain as the media packets, it is not generally possible to determine whether a lost packet was a FEC packet or a media packet. When upstreaming pexip's ulpfec patches, we decided to drop all lost events at the base depayloader level, and where the original patch from pexip was making use of picture ids and marker bits to determine whether a packet should be forwarded, this patch makes use of those to determine whether they should be dropped instead (by removing their might-have-been-fec field). Spurious lost events coming out of the depayloader can cause the decoder to stop decoding until the next keyframe and / or request a new keyframe, and while this is not desirable it makes sense to forward that information when we have other means to determine whether a lost packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads when they carry a picture id. Part-of: 2020-10-20 23:22:36 +1100 Jan Schmidt * gst/rtp/gstrtph264depay.c: rtph264depay: Preserve SPS/PPS arrival order. Even if SPS/PPS haven't changed, make sure to move them to the end of the tracking array if needed, so we always know what the most recent entries are, in case we need to discard the oldest when generating codec_data. Part-of: 2020-10-17 00:05:15 +1100 Jan Schmidt * gst/rtp/gstrtph264depay.c: rtph264depay: Warn when max SPS/PPS are collected in AVC mode. The AVC codec_data has a flaw that it can only accomodate 31 SPS headers, even though H.264 can have 32, and 255 PPS, when there can be 256 in H.264. When streaming RTP some clients like to cycle through SPS/PPS ids when changing configuration and can eventually accumulate a full set. In that case, we have no choice but to discard one (oldest) entry, or else the count written into the codec_data is wrong and downstream decoding failures ensue. Part-of: 2020-10-28 00:29:05 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtptimerqueue.c: * gst/rtpmanager/rtptimerqueue.h: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/rtptimerqueue.c: rtpjitterbuffer: don't send multiple instant RTX for the same packet Due to us not properly acknowleding the time when the last RTX was sent when scheduling a new one, it can easily happen that due to the packet you are requesting have a PTS that is slightly old (but not too old when adding the latency of the jitterbuffer), both its calculated second and third (etc.) timeout could already have passed. This would lead to a burst of RTX requests, which acts completely against its purpose, potentially spending a lot more bandwidth than needed. This has been properly reproduced in the test: test_rtx_not_bursting_requests The good news is that slightly re-thinking the logic concerning re-requesting RTX, made it a lot simpler to understand, and allows us to remove two members of the RtpTimer which no longer serves any purpose due to the refactoring. If desirable the whole "delay" concept can actually be removed completely from the timers, and simply just added to the timeout by the caller of the API. But that can be a change for a another time. The only external change (other than the improved behavior around bursting RTX) is that the "delay" field now stricly represents the delay between the PTS of the RTX-requested packet and the time it is requested on, whereas before this calculation was more about the theoretical calculated delay. This is visible in three other RTX-tests where the delay had to be adjusted slightly. I am confident however that this change is correct. Part-of: 2020-10-27 23:43:49 +1100 Jan Schmidt * gst/matroska/matroska-mux.c: matroska-mux: Fix sparse stream crash https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656 introduced an invalid memory access when debug is enabled, by casting the wrong pointer to a GstCollectPad. Fixing that showed the original change was incorrect and leads to an infinite loop in the testsuite. This patch fixes both problems. Part-of: 2020-10-22 15:29:01 -0300 Thibault Saunier * ext/vpx/gstvpxenc.c: vpx: Fix the check to unfixed/unknown framerate to set bitrate 0/1 means unknown framerate not X/0 (which is illegal). Part-of: 2020-10-22 09:17:26 -0400 Arun Raghavan * gst/rtp/gstrtputils.c: rtputils: Count metas with an empty tag list for copying/keeping The GstMetaInfos registered in core do not set their tags to NULL, but instead use an empty list (non-NULL list with a single NULL value). Let's check explicitly for that so as to not miss some metas. Part-of: 2020-10-16 16:05:45 -0700 Bastien Reboulet * ext/qt/qtitem.cc: qmlglsink: fix crash when created/destroyed in quick succession The crash is caused by a race condition where the render thread calls a method on the QtGLVideoItem instance that was previously destroyed by the main thread. Also, less frequently, QtGLVideoItem::onSceneGraphInitialized is called when QQuickItem::window is null, also causing a crash. Fixes #798 Part-of: 2020-10-19 18:23:25 +0300 Sebastian Dröge * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2codec: Garbage collect old frames if they accumulate because of codec bugs Part-of: 2020-10-19 17:56:04 +0300 Sebastian Dröge * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2codec: Pass system frame number as timestamp and use it to retrieve back frames reliably System frame numbers are supposed to be unique and correct drivers are passing through timestamps without modification from the output/sink to the capture/src side. Part-of: 2020-09-24 13:13:00 -0400 Nicolas Dufresne * docs/gst_plugins_cache.json: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: Add clear-ssrc action This action signal will delegate to clear-ssrc onto the rtpssrcdemux element associated with the session. This allow rtpbin users to clear pads and elements for a specific ssrc that is known to no longer be in use. This happens when a pad is reused in rtpsrc or ristsrc. Part-of: 2017-09-08 20:02:13 +0100 John-Mark Bell * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: * tests/check/elements/rtpvp8.c: * tests/check/meson.build: rtpvp8pay: payload temporally scaled bitstreams. Co-Authored-By: Vincent Sanders Part-of: 2017-11-17 15:11:41 +0100 Stian Selnes * docs/gst_plugins_cache.json: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: rtpvp8pay: Add picture-id-offset property Add property to set the initial value for picture-id. RFC7741 says that picture-id MAY be initialized to a random value, thus it's also valid to simply set it to a fixed initial value. A fixed value is very useful for testing. Default behavior is not changed. Part-of: 2017-03-16 15:23:28 +0100 Mikhail Fludkov * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: move duplicate code to separate functions Two new functions to modify picture id: gst_rtp_vp8_pay_picture_id_reset - picks random picture id of appropriate bitsize gst_rtp_vp8_pay_picture_id_increment - increments picture id taking care of wrapping Part-of: 2017-09-08 08:13:05 +0100 John-Mark Bell * docs/gst_plugins_cache.json: * ext/vpx/gstvpxenc.c: vp8enc: expect bps for temporal-scalability-target-bitrate. Consistency with target-bitrate is less surprising and with modern libvpx additional configuration is required to make temporal scaling work. Part-of: 2017-09-08 08:19:20 +0100 John-Mark Bell vp8enc: finish support for temporally scaled encoding - introduce two new properties: * temporal-scalability-layer-flags: Provide fine-grained control of layer encoding to the outside world. The flags sequence should be a multiple of the periodicity and is indexed by a running count of encoded frames modulo the sequence length. * temporal-scalability-layer-sync-flags: Specify the pattern of inter-layer synchronisation (i.e. which of the frames generated by the layer encoding specification represent an inter-layer synchronisation). There must be one entry per entry in temporal-scalability-layer-flags. - apply temporal scalability settings and expose as buffer metadata. This allows the codec to allocate a given frame to the correct internal bitrate allocator. Additionally, all the non-bitstream metadata needed to payload a temporally scaled stream is now attached to each output buffer as a GstVideoVP8Meta. - add unit test for temporally scaled encoding. Part-of: 2020-10-15 18:21:54 +0200 Stéphane Cerveau * gst/isomp4/qtdemux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/udp/gstudp.c: * meson.build: meson: update glib minimum version to 2.56 In order to support the symbol g_enum_to_string in various project using GStreamer ( gst-validate etc.), the glib minimum version should be 2.56.0. Remove compat code as glib requirement is now > 2.56 Version used by Ubuntu 18.04 LTS Part-of: 2020-10-14 14:30:34 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtpst2022-1-fecenc.c: rtpst2022-1-fecenc: fix input seqnum check We need to cast the incremented last seqnum to guint16 for consistent checks on wraparound Part-of: 2020-09-12 09:02:30 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Correct time types - last_dts is in milliseconds, not nanoseconds as expected for GstClockTime. Make it a generic guint64. - Use GstClockTime for the fields that actually contain nanoseconds. None of them should become negative. Part-of: 2020-10-09 09:31:27 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpst2022-1-fecenc.c: rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs g_queue_clear_full() in this case. Part-of: 2020-10-08 18:54:55 +0200 Mathieu Duponchelle * gst/rtp/rtpulpfeccommon.c: rtpulpfec: fix potential alignment issue in xor function https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453 for context Part-of: 2020-10-06 03:03:13 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpst2022-1-fecenc.c: * gst/rtpmanager/gstrtpst2022-1-fecenc.h: * gst/rtpmanager/meson.build: * tests/check/elements/rtpst2022-1-fecenc.c: * tests/check/meson.build: rtpmanager: implement SMPTE 2022-1 FEC encoder + improve integration of FEC encoders in rtpbin Part-of: 2020-10-06 03:13:30 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpst2022-1-fecdec.c: * gst/rtpmanager/gstrtpst2022-1-fecdec.h: * gst/rtpmanager/meson.build: * tests/check/elements/rtpst2022-1-fecdec.c: * tests/check/meson.build: rtpmanager: implement SMPTE 2022-1 FEC decoder + improve integration of FEC decoders in rtpbin Part-of: 2020-07-08 17:28:31 -0400 Olivier Crête * gst/rtpmanager/gstrtpfunnel.c: * tests/check/elements/rtpfunnel.c: rtpfunnel: Also forward custom sticky event This is useful to track metadata about each group of packets Also include a unit test Part-of: 2020-09-29 09:44:54 -0300 Thibault Saunier * docs/gst_plugins_cache.json: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API Since 52b63de19ada283c1180c8fc00cacb1465fdf10f the qtmux GType was renamed GstQTMuxElement which breaks presets, revert that change. Part-of: 2020-09-28 18:25:21 +0300 Sebastian Dröge * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpklvpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9pay.c: rtp: Fix allocations to support source-info property Use gst_rtp_base_payload_allocate_output_buffer() instead of gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with correct number of CSRCs according to the meta. Part-of: 2015-10-23 11:08:56 +0200 Stian Selnes * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: Fix allocation to support source-info property Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate RTP buffer with correct number of CSRCs according to the meta. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314 Part-of: 2020-09-28 15:36:00 +1000 Matthew Waters * gst/isomp4/gstqtmux.c: qtmux: output the correct limits in error messages Having the current bytes being less than the limit was confusing! Part-of: 2020-07-31 16:47:37 +1000 Matthew Waters * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * tests/check/elements/qtmux.c: qtmux: properly support initial caps nego failure Scenario: - gap event causes h264parse to push made up caps that may fail checks inside qtmux (e.g missing codec_data). - the caps event has already been marked as received and is sticky on the sink pad - gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event using gst_pad_get_current_caps() and reject the correct updated caps with codec_data. - Failure! Keep track of the configured caps ourselves instead of relying on the sticky event on the pad. Part-of: 2020-07-22 15:34:44 +1000 Matthew Waters * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: support non-seekable downstream mode Write an mdat per buffer in that case. Part-of: 2020-09-23 15:25:36 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpbin.c: rtpbin: Remove the rtpjitterbuffer with the stream Since !348, the jitterbuffer was only removed with the session. This restores the original behaviour and removes the jitterbuffer when the stream is removed. This avoid accumulating jitterbuffer objects into the bin when a session is reused. Part-of: 2020-09-23 13:26:51 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpbin.c: rtpbin: Cleanup dead code The rtpjitterbuffer is now part of the session elements, we no longer need to do the ref_sink dance when signalling it. It is already owned by the bin when signalled. Also, the code that handles generic session elements already handle the ref_sink() calls since: 03dc22951bacb6fdc3868c8f801e6a52c33a745f Part-of: 2020-09-18 16:09:20 +1000 Matthew Waters * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: * tests/check/elements/rtph264.c: rtph26*depay: drop FU's without a corresponding start bit If we have not received a FU with a start bit set, any subsequent FU data is not useful at all and would result in an invalid stream. This case is constructed from multiple requirements in RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3. Following are excerpts from RFC 3984 but RFC 7798 contains similar language. The FU in a single FU case is forbidden: A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the Start bit and End bit MUST NOT both be set to one in the same FU header. and dropping is possible: If a fragmentation unit is lost, the receiver SHOULD discard all following fragmentation units in transmission order corresponding to the same fragmented NAL unit. The jump in seqnum case is supported by this from the specification instead of implementing the forbidden_zero_bit mangling: If a fragmentation unit is lost, the receiver SHOULD discard all following fragmentation units in transmission order corresponding to the same fragmented NAL unit. A receiver in an endpoint or in a MANE MAY aggregate the first n-1 fragments of a NAL unit to an (incomplete) NAL unit, even if fragment n of that NAL unit is not received. In this case, the forbidden_zero_bit of the NAL unit MUST be set to one to indicate a syntax violation. Part-of: 2020-09-20 21:06:19 +0900 Seungha Yang * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Response caps query from srcpad ... and chain up to default query handler for unhandled query types. Unhandled query shouldn't be returned with FALSE if there's no special needs. Part-of: 2020-09-16 12:15:09 +1000 Matthew Waters * docs/gst_plugins_cache.json: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux-doc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: make documentation happy introduce a base qtmux class that we can install documentation snippets on instead of duplicating across alll the isomp4 elements Part-of: 2020-05-28 19:40:24 +1000 Matthew Waters * docs/gst_plugins_cache.json: * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: * tests/check/elements/qtmux.c: isomp4/mux: add a fragment mode for initial moov with data Used by some proprietary software for their fragmented files. Adds some support for multi-stream fragmented files Flow is as follows. 1. The first 'fragment' is written as a self-contained fragmented mdat+moov complete with an edit list and durations, tags, etc. 2. Subsequent fragments are written with a mdat+moof and each stream is interleaved as data arrives (currently ignoring the interleave-* properties). data-offsets in both the traf and the trun ensure data is read from the correct place on demuxing. Data/chunk offsets are also kept for writing out the final moov. 3. On finalisation, the initial moov is invalidated to a hoov and the size of the first mdat is extended to cover the entire file contents. Then a moov is written as regularly would in moov-at-end mode (the default). This results in a file that is playable throughout while leaving a finalised file on completion for players that do not understand fragmented mp4. Part-of: 2020-06-25 16:37:56 +1000 Matthew Waters * gst/isomp4/qtdemux.c: qtdemux: increase some logging on streams and sample parsing Part-of: 2020-06-25 16:35:45 +1000 Matthew Waters * gst/isomp4/qtdemux.c: qtdemux: bail out when encountering an atom with a size of 0 A size 0 atom means the atom extends to the end of the file. No further valid atoms will ever follow. Avoids a subsequent scan for an atom from one byte earlier after encountering a size 0 atom. Part-of: 2020-06-25 16:33:04 +1000 Matthew Waters * gst/isomp4/qtdemux.c: qtdemux: fix subsequent moof parsing after moov with valid samples reset the moof_offset back to its original value like is done in the error case just before. Fixes subsequent parsing of a moof following a moov that contains valid samples in a non-streaming fragmented mp4. Part-of: 2020-06-25 16:30:28 +1000 Matthew Waters * gst/isomp4/qtdemux.c: qtdemux: extend edit list when fragmented When we are fragmented, the edit list may only refer to the portion of the media that is in the moov. Extend the edit list stop time when we if there is only one qt segment and we are reading a fragmented file. Fixes playback of some fragmented mp4 files generated by proprietary programs. Part-of: 2020-09-15 14:22:13 -0400 Nicolas Dufresne * meson_options.txt: meson: Allow overriding qt5 feature This will allow controlling that feature from gst-build Part-of: 2015-11-17 19:14:01 -0500 Olivier Crête * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Implement segment query Fixes #239 Part-of: 2020-09-14 10:15:35 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr" Various live555 based products are using the wrong "mode" string or seem to assume case-insensitive matching, which is wrong. Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2. Part-of: 2020-05-02 02:21:00 +0200 Stefan Brüns * gst/isomp4/qtdemux.c: qtdemux: Add support for AAX encrypted audio streams This is modelled after the DASH Common Encryption scheme, but is somewhat simpler as more parts are fixed, i.e. just one encryption scheme. The output caps are fixed to 'application/x-aavd'. All information required for decryption are part of the 'adrm' atom, which is passed on as a property. The property is attached to the buffer. Part-of: 2020-05-02 02:20:44 +0200 Stefan Brüns * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio The 'aavd' box is contained in the 'stsd' sample description. The 'aavd' box follows the layout of an 'mp4a' entry, i.e. it contains a single standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd' extension boxes. Part-of: 2014-06-23 08:46:37 +0200 Haakon Sporsheim * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxdec.h: vpxdec: request a sync point on decoder errors Part-of: 2020-09-13 18:31:57 +0200 Camilo Celis Guzman * gst/rtp/gstrtpvrawpay.c: rtp/vrawpay: use alloc_output_buffer from base class Part-of: 2020-09-07 23:20:58 +0800 Ricky Tang * docs/gst_plugins_cache.json: * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix push-backchannel-buffer parameter mismatch When using python, signal parameter must match with function. Part-of: 2020-09-10 11:24:32 +0200 Jérôme Laheurte * ext/jpeg/gstjpegdec.c: jpegdec: check buffer size before dereferencing. Fixes #541 Some cameras (Panacast) have buggy drivers/firmware which send invalid JPEG frames, containing no data, which makes jpegdec crash because it assumes the frame is at least 2 bytes long. Part-of: 2020-09-10 11:11:00 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal Part-of: 2020-09-09 15:12:53 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer Besides looking like the correct place to put this, it allows us to drop the entire aggregator queue. The old implementation only dropped at most one buffer for each call of aggregate. Part-of: 2020-09-08 17:35:50 +0200 Havard Graff * sys/v4l2/gstv4l2object.c: v4l2object: plug memory-leak Part-of: 2020-08-28 18:09:15 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvp9enc.h: * ext/vpx/meson.build: vp9enc: expose row-mt property With recent libvpx versions, multithreading can be enabled on a per-tile basis, instead of on a per tile-column basis. In combination with the new tile-rows property, this allows the encoder to make much better use of the available CPU power. Bump minimum libvpx version to 1.7.0 Part-of: 2020-08-28 17:45:48 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * ext/vpx/gstvpxenc.c: vpxenc: change default for deadline to good quality Having the deadline set to best quality causes the encoder to be absurdly slow, most real-life users will want the good quality tradeoff instead. Part-of: 2020-08-28 17:39:47 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvp9enc.h: vp9enc: expose tile-columns and tile-rows properties Based on patch by Stian Selnes . Part-of: 2020-08-28 17:35:26 +0200 Mathieu Duponchelle * ext/vpx/gstvpxenc.c: * ext/vpx/gstvpxenc.h: vpxenc: add configure_encoder virtual method For subclasses to expose format-specific properties Part-of: 2020-09-08 20:57:33 +0200 Mathieu Duponchelle * gst/multifile/gstsplitmuxsink.c: splitmuxsink: fix sink pad release while PLAYING - Release the split mux lock while removing the probes - Flush the sinkpad to unblock other pads - Turn check_completed_gop into a do while statement, when waking up we want to recheck whether the current GOP is ready for sending Part-of: 2017-10-31 09:40:33 +0000 John-Mark Bell * tests/check/elements/vp8enc.c: vp8enc: improve unit tests - make test_encode_simple cope with libvpx built with CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at runtime beyond trying to set lag-in-frames to >0, pushing a buffer and catching the GST_FLOW_NOT_NEGOTIATED return. - fix bitrot in test_encode_simple_when_bitrate_set_to_zero. - port test_encode_simple to GstHarness and introduce a separate test for the lag-in-frames property. Part-of: 2020-08-21 16:03:09 +0200 Jakub Adam * docs/gst_plugins_cache.json: docs: Update plugin cache Part-of: 2020-03-24 19:35:07 +0100 Jakub Adam * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxenc.c: vpx: Support GST_VIDEO_FORMAT_I422_10LE Part-of: 2020-03-24 17:16:59 +0100 Jakub Adam * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxenc.c: vpx: Support GST_VIDEO_FORMAT_I420_10LE Part-of: 2020-03-23 21:44:30 +0100 Jakub Adam * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxenc.c: vp9enc: support GST_VIDEO_FORMAT_Y444 Part-of: 2020-09-08 17:30:35 +0100 Tim-Philipp Müller * .gitlab-ci.yml: ci: include template from gst-ci master branch again 2020-09-08 16:58:37 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * meson.build: Back to development === release 1.18.0 === 2020-09-08 00:05:14 +0100 Tim-Philipp Müller * .gitlab-ci.yml: * ChangeLog: * NEWS: * README: * RELEASE: * docs/gst_plugins_cache.json: * gst-plugins-good.doap: * meson.build: Release 1.18.0 2020-09-07 22:39:02 +0100 Tim-Philipp Müller * meson.build: * scripts/dist-translations.py: * scripts/meson.build: meson: dist pot file in tarballs Part-of: 2020-09-07 12:13:18 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: gst: Update for gst_video_transfer_function_*() function renaming Part-of: 2020-08-31 15:01:32 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Avoid crash when best pad gets flushed The 'best' pad might receive a flush event between us picking it and us popping the buffer. In this case, the buffer will be missing. Part-of: 2020-08-31 13:43:42 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Correct breaks in gst_flv_mux_find_best_pad The code seems to use `continue` and `break` as if both refer to the surrounding `while` loop. But because `break` breaks out of the `switch`, they actually have the same effect. This may have caused the loop not to terminate when it should. E.g. when `skip_backwards_streams` drops a buffer we should abort the aggregation and wait for all pads to be filled again. Instead, we might have just selected a subsequent pad as our new "best". Replace `break` with `done = TRUE; break`, and `continue` with `break`. Then simplify the code a bit. Part-of: 2020-05-13 11:31:38 +0200 Dmitriy Purgin * ext/qt/README.md: * ext/qt/qtplugin.pro: gstqmlgl: build on Windows with qmake without pkgconfig; update instructions on building for Windows Part-of: 2020-08-21 12:12:48 +0200 Philipp Zabel * meson.build: meson: fix build failure if orc is enabled but none of its users are Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/778 Part-of: 2020-08-20 14:26:04 +0200 Zeid Bekli * gst/rtp/gstrtpL16depay.c: rtpL16depay: unref buffer on error gst_rtp_L16_depay_process to unref buffer on wrong payload size or reorder failure. Part-of: === release 1.17.90 === 2020-08-20 16:11:58 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * docs/gst_plugins_cache.json: * gst-plugins-good.doap: * meson.build: Release 1.17.90 2020-08-18 10:27:52 +0300 Sebastian Dröge * gst/rtp/gstrtputils.c: rtputils: Don't call NULL GstMeta transform function It's optional and if it does not exist then no transformation is possible. Part-of: 2020-08-13 15:27:25 -0400 Julian Bouzas * gst/rtp/gstrtp.c: rtp: Do not register rtpreddec and rtpredenc twice Part-of: 2020-08-12 12:21:43 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpmanager: Improve readability of "stats" docs by making the fields an actual list Otherwise they end up all in the same line one after another. Also add docs for the "avg-jitter" stats field of the jitterbuffer. Part-of: 2020-08-11 17:24:11 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2h264codec.c: v4l2h264codec: Map newly defined profile/levels Part-of: 2020-08-11 17:18:42 -0400 Nicolas Dufresne * sys/v4l2/ext/types-compat.h: * sys/v4l2/ext/v4l2-common.h: * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: v4l2: Sync headers with kernel 5.9 Part-of: 2020-08-06 13:15:10 +0200 Víctor Manuel Jáquez Leal * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/v4l2_calls.c: v4l2: use GstV4l2Error in gst_v4l2_open() gst_v4l2_open() is called by gst_v4l2_device_provider_probe_device(), where the GstV4l2Object is created without an associated GstElement. If gst_v4l2_open() fails, it raises a bus message, but without an element, a precondition check fails on gst_element_message_full_with_details() generating a crash if running with fatal-warnings debug mode. GstV4l2Error is a helper to raise error bus messages when it is appropiated. This patch changes the direct bus messages to this helper, and the elements will actually send the error message. Part-of: 2020-08-10 20:20:53 +0300 Vivia Nikolaidou * gst/flv/gstflvmux.c: flvmux: Return NEED_DATA when we drop a buffer When we are dropping a buffer in find_best_pad (e.g. waiting for a keyframe, or skipping backwards timestamp), return GST_AGGREGATOR_FLOW_NEED_DATA to make sure we have enough data at the next run. Otherwise, a stream that accidentally fell behind (e.g. relinking race, or just waiting for a keyframe) will never get the opportunity to catch up to the other one, because the other one will always keep advancing. Part-of: 2020-08-10 20:20:04 +0300 Vivia Nikolaidou * gst/flv/gstflvmux.c: flvmux: Return NEED_DATA when no best pad is found Part-of: 2020-08-10 20:17:38 +0300 Vivia Nikolaidou * gst/flv/gstflvmux.c: flvmux: Fix possible crash on GST_ITERATOR_RESYNC Wrong pointer type Part-of: 2020-08-10 15:49:55 +1000 Matthew Waters * ext/qt/qtglrenderer.cc: qmlgloverlay: fix multiple elements with Qt 5.15 With Qt 5.15 multiple qmlgloverlay elements would produce: ASSERT: "!m_gl->property(QSG_RENDERCONTEXT_PROPERTY).isValid()" in file /path/to/qt5/qtdeclarative/src/quick/scenegraph/qsgdefaultrendercontext.cpp, line 121 Workaround by setting the (seeminigly unused) property before initialization. Part-of: 2020-08-05 10:41:33 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtph264.c: * tests/check/elements/rtph265.c: rtph26[45]pay: Change default aggregate-mode to "none" for backwards compatibility We didn't aggregate at all in previous versions and there are apparently various RTP implementations that don't handle aggregation well at all. As part of this also document that for RTSP it is recommended to keep it set to "none" while for WebRTC it should be set to "zero-latency". Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749 Part-of: 2020-07-24 16:58:34 +1000 Matthew Waters * ext/gtk/meson.build: * ext/qt/meson.build: * meson.build: * tests/examples/gtk/meson.build: build: update for gl pkg-config file split Part-of: 2020-07-31 13:50:13 +0200 Jan Alexander Steffens (heftig) * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Make sure flushing doesn't block * Trying to disconnect a stream from a running splitmuxsink by flushing it results in the FLUSH_START blocking in the stream queue's gst_pad_pause_task because the flush did not unblock complete_or_wait_on_out, so add a check for ctx->flushing there. * Add a GST_SPLITMUX_BROADCAST_INPUT so check_completed_gop notices flushing changed and the incoming push is unblocked. * Pass the FLUSH_STOP along to the muxer without waiting. Part-of: 2020-08-04 15:49:43 +0300 Vivia Nikolaidou * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Wait until we have a clock Otherwise it can happen that it tries to get the clock in PAUSED state in live mode, which does not exist. Thanks to Sebastian Dröge for helping debugging. Fixes #775 Part-of: 2020-07-31 11:05:02 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: extract bit depth from codec data for ALAC The info in the sound sample description might not be accurate if it's an older version atom. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/771 Part-of: 2020-07-28 18:46:30 +0300 Jordan Petridis * gst/auparse/gstauparse.c: auparse: fix compiler warnings GCC 10 was complaining like following. It really is complaining about default cases returning with potentially unitialized *desval, but those cases in the switch should never be hit. ``` ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c: In function 'gst_au_parse_chain': ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:481:37: error: 'timestamp' may be used uninitialized in this function [-Werror=maybe-uninitialized] 481 | GST_BUFFER_TIMESTAMP (outbuf) = timestamp; ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:482:36: error: 'duration' may be used uninitialized in this function [-Werror=maybe-uninitialized] 482 | GST_BUFFER_DURATION (outbuf) = duration; ../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:480:34: error: 'offset' may be used uninitialized in this function [-Werror=maybe-uninitialized] 480 | GST_BUFFER_OFFSET (outbuf) = offset; cc1: all warnings being treated as errors ``` Part-of: 2020-07-29 14:06:55 +0300 George Kiagiadakis * gst/rtsp/gstrtspsrc.c: rtspsrc: drop stream-start message posted by the internal udp sink(s) See #1368 Part-of: 2020-07-22 16:24:15 +0900 Hosang Lee * tests/check/elements/qtdemux.c: tests: qtdemux: test correct pad names are created Test correct pad names are created in accordance to their media type in mss mode. Part-of: 2020-06-16 17:23:44 +0900 Hosang Lee * gst/isomp4/qtdemux.c: qtdemux: create correct pad names in encrypted streams Refer to "original-media-type" when setting stream's subtype for encrypted streams in mss mode. Part-of: 2020-07-22 14:31:13 -0400 Thibault Saunier * gst/matroska/matroska-mux.c: matroskamux: Do caps renegotiation when it only adds fields Matroskamux can accept caps renegotiation if the new caps is a superset of the old one, meaning upstream added new info to the caps. Same logic as a5f22f03aa25b04726f78ae619f40b3b648f7d48 in qtmux. Part-of: 2020-07-24 14:02:26 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpfunnel.c: rtpfunnel: protect internal srccaps with lock These are modified from sink pad event handlers, so could be accessed from multiple threads at the same time. Part-of: 2020-02-23 23:44:16 +0100 Havard Graff * gst/rtpmanager/gstrtpfunnel.c: rtpfunnel: copy caps before sending them in a caps-event Reason being we don't want downstream to own a ref to our internal caps. Part-of: 2020-07-27 15:41:26 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpmanager: fix various documentation issues Improper naming of properties, improper links, misc Part-of: 2020-07-24 17:13:04 +0100 Tim-Philipp Müller * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: hypothetical fix for data pointer calculation mmal buffer header docs say data is valid for length bytes from offset. In practice offset always seems to be 0 so far though. Part-of: 2020-07-24 16:35:43 +0100 Tim-Philipp Müller * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: mark buffers as header and keyframe/delta-unit Part-of: 2020-07-24 16:14:00 +0100 Tim-Philipp Müller * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: fix nal alignment of output buffers We claim output buffers are nal-aligned, but that wasn't actually true: We would push out a partial nal in case the nal doesn't fit into the max encoder-selected output buffer size, and then the next buffer would not start with a sync marker. That's not right and makes h264parse unhappy. Instead accumulate buffers until we have a full frame (we can't rely on the NAL_END flag, it's always set). Fixes #768 Part-of: 2020-07-13 23:43:48 +0100 Tim-Philipp Müller * sys/rpicamsrc/meson.build: rpicamsrc: fix "Could not find component vc.ril.camera" on recent raspios Make extra sure all the required mmal libs such as libmmal_vc_client.so actually get linked and stay linked. Otherwise the above error happens it seems. buster (10.4) with meson 0.55 and pi ref 2020-05-27 pi-gen, 825107f04027269db77426046f5085475b1ea22f, stage5 Part-of: 2020-07-13 17:01:42 +0100 Tim-Philipp Müller * po/POTFILES: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: rpicamsrc: deviceprovider: hook up i18n properly Part-of: 2020-07-13 16:55:48 +0100 Tim-Philipp Müller * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: rpicamsrc: deviceprovider: advertise (M)JPEG as well Part-of: 2020-07-13 16:50:58 +0100 Tim-Philipp Müller * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: rpicamsrc: deviceprovider: also advertise constrained-baseline profile Part-of: 2020-07-23 16:58:00 +0200 Stéphane Cerveau * meson.build: meson: add a plugin summary This summary displays a list of plugins which have been enabled. Part-of: 2020-07-22 09:46:47 +0800 Haihua Hu * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: enhance v4l2 control interface to support string type CID add string type cid support for v4l2 implementation Part-of: 2020-07-01 15:17:47 +0200 Stéphane Cerveau * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: qtdemux: add Dolby Vision fourcc This identifiers are registered in the MPEG-RA and defined to be used by the Dolby Vision AVC/HEVC streams. This is a first step to present the stream to the decoder. Additional box parsing of DOVIConfigurationBox is necessary to complete the media presentation with proper Dolby Vision enhancements. Part-of: 2020-05-17 15:51:09 +1000 Luke Yelavich * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Copy GstCapsFeatures to caps for source pad Allows using imagefreeze with buffers in GLMemory. The following pipeline works. gst-launch-1.0 filesrc location=image.jpg ! jpegdec ! glupload ! \ imagefreeze ! glcolorconvert ! glimagesinkelement Part-of: 2020-07-20 18:20:59 +0100 Tim-Philipp Müller * gst/rtpmanager/rtptwcc.h: rtpmanager: fix "redefinition of typedef RTPTWCCManager" compiler warning G_DECLARE_FINAL_TYPE includes this typedef as well. Part-of: 2020-07-17 16:39:25 -0400 Olivier Crête * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpvorbispay.c: rtp*pay: Allocate using the base class for audio codecs This is required to add RTP header extensions from the meta automatically. Part-of: 2020-07-14 13:14:09 +0200 Ognyan Tonchev * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix segfault with illegal free set_get_param_q is not a pointer so it is illegal to call g_queue_free_full(). Freeing the requests by popping them from the queue instead. Part-of: 2020-07-15 14:40:42 +0300 Raul Tambre * ext/qt/qtitem.cc: QtGLVideoItem: Use QSharedPointer::data() for better compatibility Older Qt versions didn't have QSharedPointer::get(), which is just a modern alias for QSharedPointer::data(). FAILED: ext/qt/libgstqmlgl.so.p/qtitem.cc.o c++ -Iext/qt/libgstqmlgl.so.p -Iext/qt -I../ext/qt -I. -I.. -I../gst-libs -I/usr/include/glib-2.0 -I/usr/lib/aarch64-linux-gnu/glib-2.0/include -I/usr/include/gstreamer-1.0 -I/usr/include/orc-0.4 -I/usr/lib/aarch64-linux-gnu/gstreamer-1.0/include -I/usr/include/aarch64-linux-gnu/qt5/QtCore -I/usr/include/aarch64-linux-gnu/qt5 -I/usr/include/aarch64-linux-gnu/qt5/QtGui -I/usr/include/aarch64-linux-gnu/qt5/QtQml -I/usr/include/aarch64-linux-gnu/qt5/QtNetwork -I/usr/include/aarch64-linux-gnu/qt5/QtQuick -I/usr/include/aarch64-linux-gnu/qt5/QtX11Extras -I/usr/include/libdrm -flto -fdiagnostics-color=always -pipe -D_FILE_OFFSET_BITS=64 -std=c++11 -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Winit-self -Wmissing-include-dirs -Wno-multichar -Wvla -Wpointer-arith -g -fdebug-prefix-map=/opt/good/src=. -Wformat -Werror=format-security -O3 -march=native -Wno-error -Wdate-time -fPIC -pthread -DHAVE_CONFIG_H -DHAVE_QT_X11 -DHAVE_QT_EGLFS -MD -MQ ext/qt/libgstqmlgl.so.p/qtitem.cc.o -MF ext/qt/libgstqmlgl.so.p/qtitem.cc.o.d -o ext/qt/libgstqmlgl.so.p/qtitem.cc.o -c ../ext/qt/qtitem.cc In file included from /usr/include/gstreamer-1.0/gst/gst.h:55:0, from /usr/include/gstreamer-1.0/gst/video/video.h:23, from ../ext/qt/qtitem.cc:27: ../ext/qt/qtitem.cc: In destructor ‘virtual QtGLVideoItem::~QtGLVideoItem()’: ../ext/qt/qtitem.cc:138:86: error: ‘class QSharedPointer’ has no member named ‘get’ GST_INFO ("%p Destroying QtGLVideoItem and invalidating the proxy %p", this, proxy.get()); ^ /usr/include/gstreamer-1.0/gst/gstinfo.h:682:31: note: in definition of macro ‘GST_CAT_LEVEL_LOG’ (GObject *) (object), __VA_ARGS__); \ ^~~~~~~~~~~ ../ext/qt/qtitem.cc:138:3: note: in expansion of macro ‘GST_INFO’ GST_INFO ("%p Destroying QtGLVideoItem and invalidating the proxy %p", this, proxy.get()); ^ Part-of: 2020-07-14 14:24:20 +0100 Justin Chadwell * gst/isomp4/qtdemux.c: * tests/check/elements/qtdemux.c: qtdemux: fix allocation explosion with stsd entries Previously, the user input for stsd entries is trusted completely, and so a maliciously crafted file could choose the length of the stsd entries arbitrarily and cause qtdemux to try to allocate up to 2GB of memory (half of a 32 bit max int). This patch fixes this by sanity checking the stsd input against the size of the entire stsd atom. Part-of: 2020-07-13 10:37:19 +0100 Justin Chadwell * gst/isomp4/qtdemux.c: * tests/check/elements/qtdemux.c: qtdemux: fix crashes when input stream contained no stsd entries During trak parsing, we need to check for the existence of stsd_entries, otherwise, we end up with a NULL pointer to them. It is entirely possible for the stsd to exist, but for it to have no entries, which the previous checks did not take into account. This patch adds a simply check to ensure that all files that do not contain a stsd entry are deemed corrupt, and adds a test case to prevent a regression. Part-of: 2020-07-15 12:40:17 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: docs: update for new pixel formats https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/753 https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/754 Part-of: 2020-07-10 21:43:14 +0100 Tim-Philipp Müller * sys/rpicamsrc/meson.build: rpicamsrc: fix build with older meson versions assert() used to require two arguments. Part-of: 2020-07-10 13:08:55 +0000 Tim-Philipp Müller * tests/examples/meson.build: * tests/examples/rpicamsrc/meson.build: * tests/examples/rpicamsrc/test_color_balance.c: * tests/examples/rpicamsrc/test_orientation.c: examples: hook up rpicamsrc examples webrtc one should probably go into gst-examples. Part-of: 2020-07-10 00:42:13 +0100 Tim-Philipp Müller * tests/examples/rpicamsrc/test_color_balance.c: * tests/examples/rpicamsrc/test_orientation.c: * tests/examples/rpicamsrc/webrtc-unidirectional-h264.c: examples: fix indentation of rpicamsrc examples Part-of: 2020-07-09 19:08:34 +0000 Tim-Philipp Müller * docs/gst_plugins_cache.json: * docs/meson.build: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: flesh out docs and add to plugin docs cache Part-of: 2020-07-09 18:04:10 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: enable video orientation/direction unconditionally Part-of: 2020-07-09 17:37:01 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstrpicam-enums-template.c: * sys/rpicamsrc/gstrpicam-enums-template.h: rpicamsrc: remove mkenums template files which are no longer needed They were still being used by the autotools build, but that's gone. Part-of: 2020-07-09 17:35:15 +0000 Tim-Philipp Müller * sys/rpicamsrc/RaspiCLI.c: * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiPreview.c: * sys/rpicamsrc/RaspiStill.c: * sys/rpicamsrc/RaspiStillYUV.c: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: rpicamsrc: fix indentation Not touching the Raspi* files. Part-of: 2020-07-09 17:31:49 +0000 Tim-Philipp Müller * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/meson.build: rpicamsrc: fix and silence some compiler warnings Some are in system headers, and in Raspi files we want to keep modifications to a minimum. Part-of: 2020-07-09 16:07:30 +0000 Tim-Philipp Müller * meson_options.txt: * sys/meson.build: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.h: * sys/rpicamsrc/meson.build: rpicamsrc: hook up to build Part-of: 2020-07-09 11:46:30 +0000 Tim-Philipp Müller Merge branch 'plugin-move-rpicamsrc' Move rpicamsrc from https://github.com/thaytan/gst-rpicamsrc/ It's a useful little element and works well, so might as well make sure it's widely available so people can stop piping raspivid output into fdsrc. Part-of: 2020-05-02 19:27:20 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstrpicam-enums-template.c: * sys/rpicamsrc/gstrpicam-enums-template.h: rpicamsrc: sync autotools with glib-mkenum usage in meson build 2020-05-02 18:28:10 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/meson.build: rpicamsrc: meson: use gnome.glib_mkenums_simple() and fix build as Meson subproject While at it also fix up the type defines, e.g. GST_RPI_CAM_TYPE_RPI_CAM_SRC_EXPOSURE_MODE -> GST_RPI_CAM_SRC_TYPE_EXPOSURE_MODE 2020-05-03 11:09:47 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstplugin.map: * sys/rpicamsrc/meson.build: rpicamsrc: meson: drop map file and fix plugin symbol export with newer gstreamer versions Use -fvisibility instead of a map file for symbol export, so that the right symbols get exported with newer gstreamer versions. Older GStreamer versions also still work of course. Fixes blacklisting/plugin-loading issues with GStreamer >= 1.14 Fixes https://github.com/thaytan/gst-rpicamsrc/issues/984, closes https://github.com/thaytan/gst-rpicamsrc/issues/94 and https://github.com/thaytan/gst-rpicamsrc/issues/67 2018-07-16 19:49:21 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Attempt to workaround MMAL timeout bug mmal_queue_timedwait() might spuriously return immediately if called at exactly the wrong instant due to an internal off-by-one bug. Attempt to work around that and just retry. 2018-07-16 19:30:26 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Improve timeout error Propagate timeout errors so they're not reported generically 2018-06-21 22:50:28 +1000 Jan Schmidt * tests/examples/rpicamsrc/webrtc-unidirectional-h264.c: rpicamsrc: webrtc example: Add a STUN server to the configuration To let the webrtc example work through NAT firewalls 2018-06-21 22:44:25 +1000 Jan Schmidt * tests/examples/rpicamsrc/webrtc-unidirectional-h264.c: rpicamsrc: webrtc example: Modify HTML to support other ports than 57778 2018-06-21 21:45:32 +1000 Jan Schmidt * tests/examples/rpicamsrc/webrtc-unidirectional-h264.c: rpicamsrc: webrtc example: Remove external fmtp insertion GStreamer 1.14.2 should contain the backport of gst-plugins-bad commit 5c450c5 adding FEC and RTX support, and incidentally the fmtp field in the SDP 2018-06-21 20:33:03 +1000 Jan Schmidt * tests/examples/rpicamsrc/webrtc-unidirectional-h264.c: rpicamsrc: webrtc example: Set the locale Make the date format in the overlay respect the current locale 2018-06-20 15:36:42 +0000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Don't destroy the camera component on startup error Just disable the camera component when it fails to start. The most common reason is that the camera device is already in use, and if we just disable the mmal component correct cleanup will happen later 2018-05-12 21:13:52 +0000 Jan Schmidt * tests/examples/rpicamsrc/webrtc-unidirectional-h264.c: rpicamsrc: Add webrtc streaming example Add an example for testing webrtc streaming from the rpi camera, based on the code from https://bugzilla.gnome.org/show_bug.cgi?id=795404 Requires GStreamer 1.14.1 or git master 2018-05-12 19:57:43 +0000 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Expose constrained-baseline profile constrained-baseline is a useful profile for streaming to iOS devices, and seems to work in the firmware, so let's publish it 2018-03-28 22:00:10 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: rpicamsrc: Add define and increase reported maximum FPS from 90 to 1000 2017-11-14 15:01:21 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Expand frame timeout from 100ms to 500ms rpicamsrc on a normal rpi camera doesn't start up fast enough, and always fails the new 100ms timeout. A better solution might be to have a longer timeout for the first frame, but shorter once frames are running - but this quick fix will at least make rpicamsrc work again. 2017-11-08 09:14:35 +0000 Georgii Staroselskii * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: RaspiCapture: use mmal_queue_timedwait() for buffer queueing If an external camera was disconnected, there were no feedback in an application. It seems reasonable to wait on mmal_queue no longer than 100ms. If it's stuck we just return a FLOW_ERROR and let the application decide what to do later. 2017-11-07 15:14:06 +0000 Georgii Staroselskii * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: RaspiCapture: handle MMAL_EVENT_ERROR 2017-07-01 00:51:13 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Implement use-stc property to disable STC timestamps If use-stc=false, then rpicamsrc won't apply the camera timestamping to outgoing buffers, instead relying on real-time timestamping by the GStreamer clock. It means slightly less accuracy and more jitter in timestamps, but might help on some CSI inputs with broken timestamping. 2017-05-19 20:55:35 +1000 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Fix the descriptions of text annotation colour properties The text annotation colour properties take an integer value corresponding to a VUY colour, not a text string like the copy-pasted description from raspivid suggests. Fixes https://github.com/thaytan/gst-rpicamsrc/issues/59 2017-01-27 12:58:29 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Implement dynamic bitrate update Use mmal_port_set_parameter_uint32 to update the encoder bitrate. Fixes https://github.com/thaytan/gst-rpicamsrc/issues/60 2016-10-08 11:12:09 +0000 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Set outgoing buffer durations based on negotiated framerate. make sure outgoing buffers have at least some duration set, otherwise it leads to strange situations, like qtmux writing out a file that doesn't include the final frame inside the playable segment, because no-duration = 0 duration there. 2016-10-08 11:10:30 +0000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Destroy mmal pool on shutdown always. Avoid hangs on the next run because we didn't clean up the mmal pool last time we shutdown. 2016-10-03 15:29:49 +0000 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Switch back to MJPEG codec for image/jpeg The JPEG codec hangs, not sure why yet. The MJPEG codec doesn't provide a quality setting, and sometimes freezes on shutdown, but otherwise seems more reliable 2016-10-03 14:00:54 +0000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Don't try and set H264 params with JPEG codec 2016-10-03 02:34:50 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: First attempt at implementing MJPEG and raw video support 2016-09-19 12:06:05 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstplugin.map: * sys/rpicamsrc/meson.build: rpicamsrc: Add experimental build using the Meson build system Builds in about 10 seconds vs. 77 seconds with autotools. 2016-08-30 17:00:41 +0200 Xabier Rodriguez Calvar * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Implement GstVideoDirection interface Instead of implementing a custom property, we implement that interface. 2016-07-21 02:29:57 +1000 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: MMAL gives buffers with nal alignment, not AU Fix the output caps, our buffers are not AU aligned, since the SPS / PPS are given in separate packets at the start. 2016-07-08 15:32:21 +0200 Xabier Rodriguez Calvar * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Create orientation property Its behavior and choices are analog to the ones present in [gl]videoflip for the method property. 2016-01-03 08:26:23 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: basesrc event handlers should not unref Don't unref the passed event when handling events via the GstBaseSrc src pad event handler - basesrc does the unref. That breaks handling of upstream force-key-unit events by unreffing twice. Fixes https://github.com/thaytan/gst-rpicamsrc/issues/43 2015-12-17 14:16:10 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add property getters for preview window position. Add the lines in get_property() for the preview-x/y/w/h properties so the values can be retrieved without causing critical warnings. Fixes https://github.com/thaytan/gst-rpicamsrc/issues/42 2015-12-02 01:20:10 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add preview-x/y/w/h properties Expose properties for setting the position of the preview window on the screen 2015-10-21 21:11:36 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add properties for configuring annotation text size and colour. Map the raspivid setting for annotation text size and colours to properties. 2015-10-08 10:32:32 +0200 ibauer * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Changed awb-gain-blue use the correct enum PROP_AWB_GAIN_BLUE and not PROP_AWB_GAIN_RED 2015-07-19 01:48:35 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Fix buffer PTS calculation Timestamps from MMAL are in microseconds, so make sure to convert to nanoseconds before using them to adjust the GStreamer buffer time 2015-05-11 11:16:52 +0200 Philippe Normand * sys/rpicamsrc/gstrpicamsrc.c: * tests/examples/rpicamsrc/test_orientation.c: rpicamsrc: Basic orientation interface support The (h,v)flip attributes are now supported through this interface. It should also be possible to support (h,v)center attributes using the ROI properties. 2015-05-11 21:29:58 +1000 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Describe awb-mode=off in lowercase Change the descriptions for the awb-gain-blue and awb-gain-red properties to say 'awb-mode=off' instead of 'awb-mode=OFF' See https://github.com/thaytan/gst-rpicamsrc/issues/26 2015-05-11 10:17:18 +0200 Philippe Normand * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: colorbalance: protect with config_lock mutex 2015-05-05 19:03:43 +0200 Philippe Normand * tests/examples/rpicamsrc/test_color_balance.c: rpicamsrc: add test-color-balance example This small test will display a live video preview of the rpicam with the balance controls being updated once a second. The controls to update can be disabled in the source by setting the CONTROL_* macros values to 0. 2015-04-29 16:36:18 +0200 Philippe Normand * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Implement GstColorBalance interface Fixes https://github.com/thaytan/gst-rpicamsrc/issues/24 2015-04-27 22:56:32 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Fix initial config setting. Make sure to update the captsure config before starting capture. Since the capture component now keeps a local copy of the config, it's not updated automatically. 2015-04-27 04:05:42 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Disable bitrate, quantisation and intra-refresh dynamic changes The firmware rejects dynamic changes of those encoder params. 2015-04-27 04:05:04 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.h: rpicamsrc: Send vcos_log_warn via GStreamer debug messages 2015-04-27 02:43:14 +1000 Jan Schmidt * tests/examples/rpicamsrc/dynamicprops.py: rpicamsrc: Add dynamic properties example Python example of adjusting saturation on the fly 2015-04-27 00:54:54 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Update properties dynamically where possible Update camera and encoder properties at runtime where possible Fixes https://github.com/thaytan/gst-rpicamsrc/issues/19 and https://github.com/thaytan/gst-rpicamsrc/issues/23 2015-04-27 00:40:23 +1000 Jan Schmidt * sys/rpicamsrc/RaspiPreview.c: * sys/rpicamsrc/RaspiPreview.h: rpicamsrc: split preview config and state 2015-04-21 02:45:59 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Clear intra-refresh MMAL param struct. Use memset on the stack allocated MMAL_PARAMETER_VIDEO_INTRA_REFRESH_T struct. Apparently mmal_port_parameter_get() doesn't retrieve all parameters, causing random failures when we set the intra-refresh param on the encoder. Fixes https://github.com/thaytan/gst-rpicamsrc/issues/22 for me. 2015-04-21 01:17:55 +1000 Jan Schmidt * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCamControl.h: * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Merge changes from userland repo Current to b69f807ce59189457662c2144a8e7e12dc776988 No integration of stereoscopic support as yet 2015-04-21 00:02:27 +1000 Jan Schmidt * sys/rpicamsrc/gstrpicam_types.h: rpicamsrc: Map intra-refresh cyclic-rows to the correct MMAL param. 2015-03-10 00:22:40 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Use MMAL PTS and STC to calculate GStreamer timestamps Don't apply timestamps based on output time from the encoder, but use the MMAL STC and capture PTS to generate a GStreamer timestamp that more accurately resembles the input (and would preserve reordering should the encoder ever add B-frames). Fixes https://github.com/thaytan/gst-rpicamsrc/issues/16 2015-03-07 02:11:25 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: rpicamsrc: Defer encoder creation until after caps are negotiated This ensures the encoder is created with the profile negotiated with downstream 2015-03-07 01:17:30 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Read and set H.264 profile from negotiated caps 2015-03-06 03:43:07 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicam_types.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add intra-refresh-type property, and set default keyframe spacing to -1 (auto) This plus other recent commits mostly fix bug https://github.com/thaytan/gst-rpicamsrc/issues/16 2015-03-06 03:05:24 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicam_types.h: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Add annotation-mode and annotation-text properties 2015-03-06 02:42:00 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: implement sensor-mode property 2015-03-06 01:27:44 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: rpicamsrc: More conversion to GStreamer logging 2015-03-06 01:15:48 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicam_types.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Implement drc property 2015-03-06 01:09:16 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: add awb-gain-red and awb-gain-blue properties 2015-03-06 00:52:37 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add camera-number property 2015-03-06 00:45:05 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: add inline-headers and shutter-speed properties 2015-03-06 00:21:31 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add quantisation-parameter property, support variable bitrate Allow birate=0 and implement the quantisation-parameter property Fixes https://github.com/thaytan/gst-rpicamsrc/issues/21 2015-03-05 17:01:33 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCLI.c: * sys/rpicamsrc/RaspiCLI.h: * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCamControl.h: * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/RaspiPreview.c: * sys/rpicamsrc/RaspiPreview.h: rpicamsrc: Incorporate raspivid changes from upstream Merge all changes for new features from upstream raspberrypi userland, up to commit 0de0b2 2015-01-05 02:21:16 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add keyframe-interval property to the element 2014-10-30 00:45:18 +0000 Tim-Philipp Müller * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCamControl.h: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: rpicamsrc: deviceprovider: check if camera is detected and supported 2014-10-29 00:43:51 +0000 Tim-Philipp Müller * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.c: * sys/rpicamsrc/gstrpicamsrcdeviceprovider.h: rpicamsrc: Add GstDeviceProvider for rpi camera module 2014-09-27 14:31:10 +0100 Tim-Philipp Müller * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: avoid single-element lists in template caps 2014-10-09 20:38:41 +0000 Vivia Nikolaidou * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Add force-key-unit event support 2014-03-13 00:16:18 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/RaspiPreview.c: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Move all debug output to go via GStreamer logs Fixes https://github.com/thaytan/gst-rpicamsrc/issues/9 2013-10-19 18:52:25 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Update maximum framerate to 90 fps 2013-10-14 02:39:00 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Enable image effects 2013-10-13 18:01:00 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Re-flow element source code with gst-indent 2013-10-13 17:46:07 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicam-enums-template.c: * sys/rpicamsrc/gstrpicam-enums-template.h: * sys/rpicamsrc/gstrpicam_types.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Implement a bunch of the raspivid command-line params Add properties for controlling various parts of the capture 2013-10-13 01:29:08 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: Tell basesrc to timestamp buffers for us, for now. 2013-10-13 01:20:51 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Initial caps nego and properties. Support caps negotiation for H.264 frame size and framerate. Add bitrate, saturation, brightness, contrast, sharpness properties. 2013-10-12 19:23:03 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/gstrpicamsrc.c: rpicamsrc: First version which generates buffers on the src pad Fixed to 1920x1080 h264 regardless of caps. 2013-10-12 12:42:07 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/RaspiStill.c: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Checkpoint. Version which writes directly to test.out Switch to plain basesrc for parent class 2013-10-11 19:17:05 +1100 Jan Schmidt * sys/rpicamsrc/RaspiCamControl.c: * sys/rpicamsrc/RaspiCamControl.h: * sys/rpicamsrc/RaspiCapture.c: * sys/rpicamsrc/RaspiCapture.h: * sys/rpicamsrc/RaspiPreview.c: * sys/rpicamsrc/RaspiPreview.h: * sys/rpicamsrc/RaspiStill.c: * sys/rpicamsrc/RaspiStillYUV.c: * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: checkpoint 2013-10-10 23:47:38 +1100 Jan Schmidt * sys/rpicamsrc/gstrpicamsrc.c: * sys/rpicamsrc/gstrpicamsrc.h: rpicamsrc: Initial commit Simple modified gst-template to use BaseCameraSrc Incorporate Broadcom mmal headers 2018-04-19 13:57:26 +0200 Michael Olbrich * ext/soup/gstsouphttpsrc.c: souphttpsrc: don't fail when seeking past the end of the content Range errors are already turned into EOS when the size is not known. Do the same thing if the request as outside the known content size. This can be triggered by seeking in a queue2: - Ensure that the range containing the end of the file is available. - Seek into this range from a different range. - queue2 creates a seek event with start= - this results in a "Requested Range Not Satisfiable" error Fixes #452 Part-of: 2019-11-10 21:19:09 +0100 Michael Olbrich * ext/soup/gstsouphttpsrc.c: souphttpsrc: don't update the size on error Any data corresponding length in the message is not part of the requested file. Part-of: 2020-06-18 19:12:46 +1000 Matthew Waters * ext/qt/qtglrenderer.cc: qt/gloverlay: fix using OpenGL after destroying Qml Qml somewhat unhelpfully seems to uncurrent our OpenGL context on its destruction. Work around that by uncurrenting and recurrenting again. Part-of: 2020-07-08 17:02:34 +0100 Tim-Philipp Müller * meson.build: * scripts/extract-release-date-from-doap-file.py: meson: set release date from .doap file for releases Part-of: 2020-07-07 12:36:01 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Don't lock object lock twice in prefill mode Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/762 Part-of: 2020-07-04 01:02:02 +0100 Tim-Philipp Müller * gst/audiofx/meson.build: * gst/deinterlace/meson.build: * gst/videobox/meson.build: * gst/videomixer/meson.build: * meson.build: * scripts/update-orc-dist-files.py: meson: add update-orc-dist target Add target to update backup orc -dist.[ch] files. Part-of: 2020-05-26 10:27:35 -0400 Xavier Claessens * sys/v4l2/gstv4l2videodec.c: v4l2: Do not renegotiate if only framerate changed Part-of: 2020-07-02 09:15:08 +0300 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Pass audio info from set_format() to query_total_samples() explicitly This fixes writing of the seek table header. gst_audio_encoder_get_audio_info() will still return old/unset audio info until set_format() has actually returned, which then results in query_total_samples() to always return 0. Thanks to Jacob Kauffmann for debugging this and finding the main cause. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/756 Part-of: 2020-07-03 02:03:33 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * meson.build: Back to development === release 1.17.2 === 2020-07-03 00:27:47 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * docs/gst_plugins_cache.json: * gst-plugins-good.doap: * meson.build: Release 1.17.2 2020-07-02 07:53:14 +0530 Nirbheek Chauhan * gst/deinterlace/meson.build: * meson.build: deinterlace: Disable nasm support on x32 The assembly assumes pointers are 64-bit, so just disable it. Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/757 Part-of: 2020-07-01 18:19:09 +0530 Nirbheek Chauhan * gst/deinterlace/meson.build: deinterlace: Fix build on x32 Need to pass `-f elfx32` to nasm in that case. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/757 Part-of: 2020-07-01 16:17:19 +1000 Jan Schmidt * gst/matroska/matroska-mux.c: matroska-mux: Wait for caps on sparse streams Don't set sparse streams to non-waiting at the collectpads level until after capa arrive, as we need caps on all pads before the file headers get written, or else the subtitle track will be silently absent in the final file. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/724 Part-of: 2020-07-01 16:13:27 +1000 Jan Schmidt * gst/matroska/matroska-mux.c: matroska-mux: Warn on late caps arrival As well as warning when caps change after the headers were already written, make sure to warn if the *first* caos arrive late too. Part-of: 2020-06-30 18:37:06 +0300 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Return TRUE from the LATENCY query handling We always answer it successfully no matter what. The default return value in the function is FALSE even if the code below sets it again to FALSE. Part-of: 2020-06-29 11:53:39 +0300 Sebastian Dröge * tests/check/elements/imagefreeze.c: imagefreeze: Add test for new live mode Part-of: 2020-06-29 10:10:09 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Add a live mode Previously imagefreeze would always operate as non-live element and output frames as fast as possible according to the configured segment (via SEEK events) and the negotiated framerate from start to stop or the other way around. With the new live mode (enabled via the is-live property) it would only output frames in PLAYING. Frames would be output according to the negotiated framerate unless it would be too late, in which case it would jump ahead and skip over the requirement amount of frames. This makes it possible to actually use imagefreeze in live pipelines without having to manually ensure somehow that it would start outputting at the current running time and without still risking to fall behind without recovery. Part-of: 2020-06-28 22:26:23 +0300 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Correctly answer the LATENCY query We never run as a live element, even if upstream is live, and never output any buffers with latency but immediately generate buffers as fast as we can according to the negotiated framerate. Passing the query upstream would proxy whatever mode of operation upstream has, which has nothing to do with how we produce buffers. Part-of: 2020-06-25 14:15:51 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2: Fix threading issues in orphaning mechanism The pool orphaning function was colling internal _stop() virtual function implementation. This is not thread safe, as a private lock inside the buffer pool is supposed to be held. Fix this by keeping delayed _stop() and orphaning the GstV4L2Allocator instead (REQBUFS(0)). Then, protect the orphaned boolean with the object lock for the case a buffer is being released after we have orphaned the buffer. That would otherwise cause a QBUF to happen while the queue is no longer owned by the buffer pool. This boolean is otherwise used and set from the streaming lock, or after threads have been stopped (final cleanup). Part-of: 2020-06-26 16:43:37 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpoool: Fix requeueue after seek when importing When the buffer pool is importing buffer, it will requeue num_allocated on streamon. As this value was not set for DMABuf import and USERPTR, no buffer was queued back. Part-of: 2020-06-26 16:39:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: Revert "v4l2bufferpool: request the maximum number of buffers for USERPTR" This reverts commit 6bf9f4bd77a4c6cce8786893feea7d601a6e6030. Part-of: 2020-06-26 16:37:06 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: Revert "v4l2bufferpool: request the maximum number of buffers for DMABUF" This reverts commit 94e323c10f2d7fa85bf63f357d203ca5305800c6. Fixes #754 Part-of: 2020-06-26 14:48:14 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Only resurrect the right amount of buffers On streamon, we need to resurrect (queue back) some buffers, as during flushign seek we'd endup with an empty queued. We initially started with resurrecting as many as we could without blocking, but that miss-behaved with dynamic CREATE_BUFS, causing the pool to grow dramatically. This was limited by the number of allocated buffers, but this still tried to resurrect too many buffers for the first run, as activating the pool will queued buffers. In this patch, we calculte the missing detal in the queue and only try and resurrect that amount of buffers. Part-of: 2020-06-26 13:11:04 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Only offer inactive pools and if needed Avoid offering a pool if it's not needed or if it's still active. This works around the fact the we only have one pool in V4L2. Part-of: 2020-06-24 21:58:07 +0530 Nirbheek Chauhan * ext/qt/gstqtglutility.cc: * ext/qt/meson.build: qt: Rework how we find the Qt QPA header Instead of querying the Qt include path from the dependency or from qmake, rely on the qt5qml_dep to set the include path to QtGui correctly, and look for the header inside the private includedir. Then we can use that path to include the header directly. Reported in https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/780#note_548092 Part-of: 2020-06-24 22:04:55 +0530 Nirbheek Chauhan * ext/qt/meson.build: qt: Only check for moc-qt5/moc in PATH if not cross-compiling This is an extra check that's only needed for working around Linux distribution packaging. `moc` is not required in the cross file. Part-of: 2020-06-26 13:10:00 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Don't do REQBUFS(0) on inactive allocator If the allocator is no longer active, it means the memory has already been freed, calling REQBUF(0) would make no sense. Part-of: 2020-06-26 11:05:25 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Avoid set_flushing warning The gst_buffer_pool_set_flushing() warns when that function is called on an inactive pool. Avoid the warning by checking the state, this is similar to what we do in gst_v4l2_object_unlock(). Part-of: 2020-06-26 09:53:13 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix data offset / bytesused size validation The check was too strict causing spurious warning. Now check for <= so that 0 sized buffer do not cause a warning. Part-of: 2020-06-25 16:46:23 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Fix negotiation caps leak Part-of: 2020-06-26 19:28:31 +0100 Tim-Philipp Müller * gst/multifile/gstsplitmuxsink.c: splitmuxsink: flesh out docs for format-location* signals Make explicit that the returned strings need to be g_free()-able. Fixes #753 Part-of: 2020-06-25 16:47:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Skip negotiation of profiles/level if no codec The codec structure is optional and not used for fwht test codec. This was leading to a crash dereferencing NULL pointer. Part-of: 2020-05-03 13:17:46 +0200 Havard Graff * gst/rtpmanager/rtpstats.c: rtpstats: guard against division by zero Part-of: 2020-06-17 23:23:58 +0200 Havard Graff * gst/rtpmanager/rtptwcc.c: rtptwcc: fix pruning of ack'ed twcc-packets Fixes #750 Part-of: 2020-06-24 21:15:47 +0530 Nirbheek Chauhan * tests/examples/qt/qmloverlay/meson.build: * tests/examples/qt/qmlsink-dynamically-added/meson.build: * tests/examples/qt/qmlsink/meson.build: * tests/examples/qt/qmlsrc/meson.build: meson: Build Qt5 tests with -std=c++11 We already do this for the plugin. https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/780#note_548179 Part-of: 2020-06-25 12:58:48 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Add new properties for setting muxer/sink presets Part-of: 2020-06-24 17:04:51 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * gst/autodetect/gstautodetect.c: autodetect: mark filter-caps property as DOC_SHOW_DEFAULT When generating the cache we inspect the base class through an instance of one of its subclasses. We don't want potential assignments in subclasses initialization to leak into the base class documentation Part-of: 2020-06-24 16:45:27 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * ext/vpx/gstvpxenc.c: vpxenc: mark all properties as GST_DOC_SHOW_DEFAULT When generating the cache we inspect the base class through an instance of one of its subclasses. We don't want potential assignments in subclasses initialization to leak into the base class documentation Part-of: 2020-06-23 19:04:03 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * gst/equalizer/gstiirequalizer.c: docs: mark GstIirEqualizer as plugin API 2020-06-23 12:47:44 -0400 Thibault Saunier * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vpx: Fix links to baseclass properties 2020-06-23 02:50:35 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: * sys/v4l2/tuner.c: * sys/v4l2/tunerchannel.c: docs: mark more types as plugin API 2020-06-23 00:02:34 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: plugins_cache: add base classes 2020-06-23 00:02:21 +0200 Mathieu Duponchelle * docs/meson.build: meson: mark plugins cache target as always stale 2020-06-21 01:34:43 +0200 Mathieu Duponchelle * ext/gtk/gstgtkbasesink.c: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxenc.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/autodetect/gstautodetect.c: docs: mark more types as plugin API 2020-06-19 22:54:38 -0400 Thibault Saunier * docs/gst_plugins_cache.json: doc: Stop documenting properties from parents 2020-06-21 20:11:06 +0800 He Junyan * gst/deinterlace/yadif.c: deinterlace: Add the missing ORC_RESTRICT define. ORC_RESTRICT may not be defined in yadif.c and cause build error. Part-of: 2019-06-06 09:41:13 +0200 Havard Graff * tests/check/elements/rtpsession.c: rtpsession: make tests more stable Part-of: 2020-06-20 20:42:37 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: docs: update plugin cache for new version Some default values include our version string, like user agent strings. 2020-06-20 00:28:11 +0100 Tim-Philipp Müller * meson.build: Back to development === release 1.17.1 === 2020-06-19 19:18:59 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * docs/gst_plugins_cache.json: * gst-plugins-good.doap: * meson.build: Release 1.17.1 2020-06-19 20:24:12 +0900 Seungha Yang * gst/deinterlace/meson.build: meson: deinterlace: Check host cpu type for asm build Add host cpu type check as we would enable asm only for x86_64 Part-of: 2020-06-19 19:54:08 +0900 Seungha Yang * meson.build: meson: Fix build error with MSVC caused by ARCH_X86_64 define ARCH_X86_64 define will enable GCC specific code path in dv_types.h while building dv plugin. Part-of: 2020-06-19 10:32:45 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: * ext/shout2/gstshout2.c: shout2: advertise documentation caps properly shout2send caps depend on what the libshout2 version in question supports, but the documentation caps should always be the same. Part-of: 2019-05-26 20:20:03 +1000 Jan Schmidt * gst/isomp4/meson.build: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux_tags.c: * gst/isomp4/qtdemux_tags.h: qtdemux: Split tag reading functions out Move some code out of the enormous qtdemux.c into a separate qtdemux_tags helper, and make some structs available via qtdemux.h to accommodate that. Part-of: 2019-05-26 05:05:06 +1000 Jan Schmidt * gst/isomp4/meson.build: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_tree.c: * gst/isomp4/qtdemux_tree.h: qtdemux: Move some tree parsing files out to a separate file. Reduce a tiny bit of the bulk of qtdemux.c by moving some agnostic helper functions out. Part-of: 2019-05-26 01:24:54 +1000 Jan Schmidt * gst/isomp4/atoms.c: * gst/isomp4/qtdemux.c: qtdemux: Factor out svmi parsing. Fix bounds checking. Move the SVMI stereoscopic atom parsing out to a helper function to shrink qtdemux_parse_trak a bit. Add a bounds check that the received atom is large enough before parsing it. Add a note to the atom parser that svmi comes from the MPEG-A spec 23000-11. Part-of: 2020-06-15 13:05:49 +0200 Guillaume Desmottes * ext/pulse/pulsedeviceprovider.c: pulse: fix discovery of newly added devices Fix regression introduced in 7bc5e28d85992b03e5852879b8d4d96043496caf preventing the device provider to send the device-added message for new devices. By early returning the patch was discarding add/remove events. Fix #735 Part-of: 2020-06-18 10:47:28 +0100 Tim-Philipp Müller * tests/examples/qt/qmlsink-dynamically-added/meson.build: * tests/examples/qt/qmlsink-dynamically-added/play.pro: * tests/examples/qt/qmlsink-dynamically-added/qmlsink-dyn-added.qrc: examples: qmlsink: rename qrc file to avoid naming conflicts with older meson versions Would get "Tried to create target "qt5-qmlsink_qrc", but a target of that name already exists." with older meson versions. Work around that by renaming the qrc file. Part-of: 2020-06-17 16:42:16 +0530 Nirbheek Chauhan * meson.build: meson: Check the nasm version with run_command Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/751 Part-of: 2020-06-16 19:34:01 +0900 Seungha Yang * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't return TRUE for unhandled query Expected return value for unhandled query is FALSE Part-of: 2020-06-16 11:52:38 +0300 Vivia Nikolaidou * gst/deinterlace/meson.build: * gst/deinterlace/x86/x86inc.asm: * gst/deinterlace/x86/yadif.asm: * gst/deinterlace/yadif.c: * gst/deinterlace/yadif.h: * meson.build: * meson_options.txt: deinterlace: Add yadif ASM optimisations Measured to be about 3.4x faster than C Part-of: 2020-06-12 13:21:02 +0300 Vivia Nikolaidou * gst/deinterlace/yadif.c: deinterlace: Fix invalid read in yadif Part-of: 2020-06-12 12:18:11 +1000 Matthew Waters * ext/qt/qtglrenderer.cc: qt/gloverlay: reset OpenGL state after Qt drawing Reset to the original OpenGL state as required by the GStreamer OpenGL API contract. Fixes output with a glimagesink element downstream. Part-of: 2020-06-12 12:16:49 +1000 Matthew Waters * ext/qt/qtglrenderer.cc: qt/gloverlay: reset current OpenGL context after Qt Qt may replace the drawable with its own which breaks output if Qt is not displaying the resulting video as used with e.g. glimagesink. Part-of: 2020-06-12 09:52:56 +0300 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Change a GST_ERROR_OBJECT() back to GST_DEBUG_OBJECT() It was accidentally changed in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/436 Part-of: 2020-06-11 20:39:33 +0300 Jordan Petridis * gst/isomp4/gstqtmux.c: * sys/v4l2/gstv4l2videodec.c: Use gst_element_class_set_metadata when passing dynamic strings gst_element_class_set_metadata is meant to only be used with static or inlined strings, which isn't the case for the 2 elements here resulting in use-after-free later on. https://gstreamer.freedesktop.org/documentation/gstreamer/gstelement.html?gi-language=c#gst_element_class_set_static_metadata Part-of: 2020-06-10 13:56:22 +0000 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: Revert "rtpjitterbuffer: Avoid deadlock on flush" This reverts commit 54810bf44f27d9c180730f58f16f6e172c7b0bc8 Part-of: 2020-06-09 15:12:13 -0400 Thibault Saunier * docs/gst_plugins_cache.json: docs: Update plugins cache 2020-06-09 13:09:20 -0700 U. Artie Eoff * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: g_queue_clear_full introduced in glib 2.60 Define g_queue_clear_full if glib < 2.60. Fixes #747 Part-of: 2020-06-08 11:33:16 -0400 Thibault Saunier * docs/gst_plugins_cache.json: * gst/rtpmanager/rtpsession.c: rtpsession: Make internal-ssrc as show default for doc 2020-06-08 10:56:02 -0400 Thibault Saunier * docs/gst_plugins_cache.json: docs: Update plugins cache 2020-06-09 15:21:25 +0100 Tim-Philipp Müller * tests/check/meson.build: tests: don't pull in all -bad plugin, only allow the one we need Set up our plugin include list for tests in such a way that we don't pull in *all* plugins from -bad but only the one used in the splitmuxsink unit test, i.e. the timecode plugin, so we don't accidentally use other encoders/decoders such as nvenc/dec for example. Part-of: 2020-06-08 17:41:13 -0400 Nicolas Dufresne * gst/rtpmanager/rtptimerqueue.c: rtptimerqueue: Fix leak on timer collision While the caller should make sure this does not happen, make sure timer collision are not silently ignored and leaked. Fixes #726 Part-of: 2020-03-27 15:48:32 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Keep JBUF lock while processing timers Until now, do_expected_timeout() was shortly dropping the JBUF_LOCK in order to push RTX event event without causing deadlock. As a side effect, some CPU hung would happen as the timerqueue would get filled while looping over the due timers. To mitigate this, we were processing the lost timer first and placing into a queue the remainign to be processed later. In the gap caused by an unlock, we could endup receiving one of the seqnum present in the pending timers. In that case, the timer would not be found and a new one was created. When we then update the expected timer, the seqnum would already exist and the updated timer would be lost. In this patch we remove the unlock from do_expected_timeout() and place all pending RTX event into a queue (instead of pending timer). Then, as soon as we have selected a timer to wait (or if there is no timer to wait for) we send all the upstream RTX events. As we no longer unlock, we no longer need to pop more then one timer from the queue, and we do so with the lock held, which blocks any new colliding timers from being created. Part-of: 2020-06-08 09:33:10 +0200 Guillaume Desmottes * tests/check/elements/vp9enc.c: tests: vp9enc: enforce I420 format Test was not enforcing a video format on videotestsrc. I420 was picked as it was the first format in GST_VIDEO_FORMATS_ALL which will no longer be true (gst-plugins-base!689). Part-of: 2020-05-30 08:55:19 +0200 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Avoid deadlock on flush When a GST_EVENT_FLUSH_START reaches the jitterbuffer, there is a chance that our task is currently blocking waiting for a timer. There was two problems: * That wait wasn't checking for flushing situations * The flushing handling wasn't waking up that conditional (to check whether it should abort) Part-of: 2020-06-06 00:42:25 +0200 Mathieu Duponchelle * ext/aalib/gstaasink.c: * ext/aalib/gstaatv.c: * ext/dv/gstdvdec.c: * ext/flac/gstflacenc.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/jack/gstjackaudiosink.c: * ext/jpeg/gstjpegdec.c: * ext/lame/gstlamemp3enc.c: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacatv.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexenc.c: * ext/twolame/gsttwolamemp2enc.c: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxenc.c: * ext/wavpack/gstwavpackenc.c: * gst/alpha/gstalpha.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gstop.c: * gst/effectv/gstradioac.c: * gst/effectv/gstripple.c: * gst/flv/gstflvmux.c: * gst/isomp4/gstqtmux.c: * gst/multifile/gstmultifilesink.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9pay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideomedian.c: * gst/videomixer/videomixer2.c: * sys/v4l2/gstv4l2object.c: plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-05 11:49:17 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Initialize uninitialized variable correctly `last_out` would be used uninitialized if the element has no `set-active` signal. Initialize it to -1 as that's what the "default" value is further below. CID 1455443 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/727 Part-of: 2015-11-26 17:52:29 +0100 Mikhail Fludkov * ext/vpx/gstvp9enc.c: * ext/vpx/gstvpxenc.c: * ext/vpx/gstvpxenc.h: * tests/check/elements/vp8enc.c: * tests/check/elements/vp9enc.c: vpxenc: Add new bit-per-pixel property to select a better "default" bitrate As part of this also change the default bitrate value to 0. The default value was 256000 previously. In reality, if the property was not set the bitrate value would be scaled according to the resolution which is not very intuitive behavior. It is better to use 0 for this purpose. Now together with newly introduced property "bits-per-pixel" 0 means to assign the bitrate according to resolution/framerate. The default bitrates are now - 1.2Mbps for VP8 720p@30fps - 0.8Mbps for VP9 720p@30fps and scaled accordingly for different resolutions/framerates. Previously the default bitrate was also not scaled according to the framerate but only took the resolution into account. This also fixes the side effect of setting bitrate to 0. Previously encoder would not produce any data at all. Addition from Sebastian Dröge to assume 30fps if no framerate is given in the caps instead of not calculating any bitrate at all. Part-of: 2020-06-03 18:35:58 -0400 Thibault Saunier * docs/meson.build: doc: Require hotdoc >= 0.11.0 2020-06-02 14:58:47 -0400 Thibault Saunier * gst/rtpmanager/gstrtpjitterbuffer.c: doc: Fix wrong link to GString in rtpjitterbuffer 2020-05-27 16:01:22 +0300 Sebastian Dröge * docs/gst_plugins_cache.json: docs: Update gst_plugins_cache.json 2020-05-30 01:29:03 +0200 Mathieu Duponchelle * ext/aalib/gstaasink.c: * ext/aalib/gstaatv.c: * ext/dv/gstdvdec.c: * ext/flac/gstflacenc.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/jack/gstjackaudiosink.c: * ext/jpeg/gstjpegdec.c: * ext/lame/gstlamemp3enc.c: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacatv.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexenc.c: * ext/twolame/gsttwolamemp2enc.c: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxenc.c: * ext/wavpack/gstwavpackenc.c: * gst/alpha/gstalpha.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gstop.c: * gst/effectv/gstradioac.c: * gst/effectv/gstripple.c: * gst/flv/gstflvmux.c: * gst/isomp4/gstqtmux.c: * gst/multifile/gstmultifilesink.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9pay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideomedian.c: * gst/videomixer/videomixer2.c: * sys/v4l2/gstv4l2object.c: plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2018-02-28 15:46:51 +0100 Stian Selnes * ext/vpx/gstvpxdec.c: * tests/check/elements/vp8dec.c: vpxdec: Check that output width and height != 0 For VP8 it's possible to signal width or height to be 0, but it does not make sense to do so. For VP9 it's impossible. Hence, we most likely have a corrupt stream. Trying to negotiate caps downstream with either width or height as 0 will fail with something like gst_video_decoder_negotiate_default: assertion 'GST_VIDEO_INFO_WIDTH (&state->info) != 0' failed Part-of: 2020-05-29 00:45:03 +0900 Seungha Yang * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: Fix crash on Windows caused by cross-CRT issue Use speex_header_free() to free memory which was allocated by library. Cross-CRT issue should not happen on 1.17 Cerbero build but might happen custom build or so. Part-of: 2020-05-27 22:33:31 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.h: rtspsrc: Use the correct type for storing the max-rtcp-rtp-time-diff property It's an integer property and rtpbin also expects an integer. Passing it as a GstClockTime (guint64) to g_object_set() will cause problems, and on big endian MIPS apparently causes crashes. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/737 Part-of: 2020-05-27 12:42:38 +0100 Tim-Philipp Müller * tests/check/meson.build: tests: fix meson test env setup to make sure we use the right gst-plugin-scanner If core is built as a subproject (e.g. as in gst-build), make sure to use the gst-plugin-scanner from the built subproject. Without this, gstreamer might accidentally use the gst-plugin-scanner from the install prefix if that exists, which in turn might drag in gst library versions we didn't mean to drag in. Those gst library versions might then be older than what our current build needs, and might cause our newly-built plugins to get blacklisted in the test registry because they rely on a symbol that the wrongly-pulled in gst lib doesn't have. This should fix running of unit tests in gst-build when invoking meson test or ninja test from outside the devenv for the case where there is an older or different-version gst-plugin-scanner installed in the install prefix. In case no gst-plugin-scanner is installed in the install prefix, this will fix "GStreamer-WARNING: External plugin loader failed. This most likely means that the plugin loader helper binary was not found or could not be run. You might need to set the GST_PLUGIN_SCANNER environment variable if your setup is unusual." warnings when running the unit tests. In the case where we find GStreamer core via pkg-config we use a newly-added pkg-config var "pluginscannerdir" to get the right directory. This has the benefit of working transparently for both installed and uninstalled pkg-config files/setups. Part-of: 2020-05-25 20:11:31 -0400 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtspsrc: Error out when failling to receive message response And let it rety twice. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/717 Part-of: 2020-05-21 17:12:55 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2: videodec: Fix broken template caps The profiles and levels were applied to the common caps instead of the copy. That had the side effect of setting profiles/level from one CODEC onto another. Leaving to encoder not being registered or not-negotiated errors. Part-of: 2020-05-21 17:09:39 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2codec.c: * sys/v4l2/gstv4l2codec.h: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2: codec: Fix GValue leak The levels and profiles probe function returned a dynamically allocated GValue that was leaked. Simplify this by using a stack allocated GValue and a boolean return value. Part-of: 2020-05-21 16:39:53 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2codec.c: v4l2codec: Remove uneeded factorisation There is only one user of that function and the split only increase complexicity. Part-of: 2020-05-20 17:30:59 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Ignore non-increasing sequence number With older kernel, older driver or just broken drivers, the sequence number may not be increasing. This simply ignore the sequence in this case. This would otherwise miss-leading large amount of lost frame being reported. Fixes #729 Part-of: 2020-05-18 13:17:14 +1000 Matthew Waters * ext/qt/gstqtoverlay.cc: * ext/qt/gstqtoverlay.h: * tests/examples/qt/qmloverlay/main.cpp: qtoverlay: add the root item as a property Part-of: 2020-05-20 13:17:13 +0300 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Send gap events if one of the streams falls behind the other by more than 3s Same mechanism and threshold as in other demuxers. Part-of: 2020-05-20 12:53:56 +0300 Sebastian Dröge * gst/flv/gstflvdemux.h: flvdemux: Remove unused audio_linked/video_linked booleans Part-of: 2020-05-20 10:46:45 +0200 Edward Hervey * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Answer bitrate queries from upstream If upstream (such as queue2 in urisourcebin) asks for our bitrate, check if we have stored audio/video bitrates, and use them. Part-of: 2020-05-20 10:45:16 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Handle empty metadata strings g_utf8_validate() errors out on empty string. But empty strings are valid, so only check if they're not Part-of: 2020-05-20 10:44:19 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Set ACCEPT_TEMPLATE flag on sinkpad A demuxer can accept any caps matching its sinkpad template caps Part-of: 2020-05-15 19:20:45 +0300 Raul Tambre * ext/qt/qtglrenderer.cc: qtglrenderer.cc: Fix compiling 46bfb7d247aef880c15300dad63eb2bbf6dc4928 fixed a format warning without checking if it actually compiled. toUtf8() returns QByteArray so we need to assign it to a temporary variable to be able to get the raw string data from it. Part-of: 2020-05-15 06:07:25 +0000 Raul Tambre * ext/qt/qtglrenderer.cc: qtglrenderer.cc: Fix -Wformat-security warning Part-of: 2020-05-12 04:35:37 +0530 Nirbheek Chauhan * ext/qt/meson.build: * ext/taglib/meson.build: * meson.build: * sys/osxvideo/meson.build: meson: Pass native: false to add_languages() This is needed for cross-compiling without a build machine compiler available. The option was added in 0.54, but we only need this in Cerbero and it doesn't break older versions so it should be ok. Part-of: 2020-05-12 04:33:43 +0530 Nirbheek Chauhan * ext/qt/meson.build: * ext/taglib/meson.build: * meson.build: meson: Make C++ compiler detection not be automagic It is now controlled by the qt5 and/or taglib options. We won't silently fail to build taglib now. Part-of: 2020-05-12 04:32:01 +0530 Nirbheek Chauhan * ext/gtk/meson.build: * ext/qt/meson.build: * meson.build: * tests/examples/gtk/meson.build: meson: Fix gstgl checks for qt and gtk Also rename from build_ to have_, which is more accurate. Part-of: 2020-05-12 04:30:13 +0530 Nirbheek Chauhan * ext/qt/meson.build: * tests/examples/qt/meson.build: * tests/examples/qt/qmloverlay/meson.build: * tests/examples/qt/qmlsink-dynamically-added/meson.build: * tests/examples/qt/qmlsink/meson.build: * tests/examples/qt/qmlsrc/meson.build: meson: Revamp qt5qml plugin and example build code Stricter and simpler. For example, now we properly error out when gstreamer-gl-1.0 was not found when the qt5 plugin is enabled or when a C++ compiler is not enabled. Part-of: 2020-05-09 03:09:03 +1000 Jan Schmidt * gst/deinterlace/yadif.c: deinterlace: Split out NULL checks in yadif Separate out explicit NULL checks for fields we depend on so that coverity can hopefully verify dependencies better. Part-of: 2020-05-09 03:07:33 +1000 Jan Schmidt * gst/deinterlace/tvtime/greedy.c: deinterlace: Handle NV12/NV21 for the greedyl mode. Don't fall back on the default interpolate_scanline function, which blindly tries to copy from the next field, which can be NULL in mixed progressive/interlaced streams Part-of: 2020-05-05 16:59:56 +0300 Vivia Nikolaidou * gst/deinterlace/yadif.c: deinterlace: Support packed formats for YADIF Part-of: 2020-05-06 11:04:18 +0300 Vivia Nikolaidou * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: Call the planar functions for the Y plane of nv12/nv21 In some algorithms (like yadif), the Y plane has to be handled different than the UV plane. Therefore, the planar_y functions are now called for the Y plane, and the nv12/nv21 functions are handling only the UV/VU planes respectively. Part-of: 2020-01-03 02:34:59 +1100 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/meson.build: * gst/deinterlace/yadif.c: * gst/deinterlace/yadif.h: deinterlace: Add C implementation of YADIF Import the YADIF deinterlacer from ffmpeg and modify it to match the simple deinterlace scanlines structure. Part-of: 2020-01-03 02:33:25 +1100 Jan Schmidt * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: deinterlace: Allow for 5 fields for interpolation Add an extra field to the simple deinterlace implementation, so that methods can potentially use 5 fields - the current field, and 2 before and 2 after. Part-of: 2020-05-07 01:17:25 +1000 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: deinterlace: Force renegotiation when changing mode Switching the deinterlacing mode on-the-fly from disabled to auto used to work, but was broken by commit #1f21747c some years ago. Force re-negotiation with downstream when the mode or fields properties are changed, otherwise deinterlace never switches out of the passthrough mode. Part-of: 2020-04-23 15:32:58 +0800 nian.yan * ext/jpeg/gstjpegenc.c: jpegenc: remove meta copy in jpegenc GstVideoEncoder takes care of the Meta copy, so there is no need in jpegenc Fixes http://gstreamer-devel.966125.n4.nabble.com/jpegenc-copy-GstMeta-twice-tt4693981.html Part-of: 2020-05-05 17:47:28 +0300 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Handle flushing correctly First of all get rid of the atomic seeking boolean, which was only ever set and never read. Replace it with a flushing boolean that is used in the loop function to distinguish no buffer because of flushing and no buffer because of an error as otherwise we could end up in a GST_FLOW_ERROR case during flushing. Also only reset the state of imagefreeze in flush-stop when all processing is stopped instead of doing it as part of flush-start. And last, get a reference to the imagefreeze buffer in the loop function in the very beginning and work from that as otherwise it could in theory be replaced or set to NULL in the meantime as we release and re-take the mutex a couple of times during the loop function. Part-of: 2020-05-06 06:48:24 +0200 Edward Hervey * gst/videobox/gstvideobox.c: videbox: Use MIN instead of CLAMP for uint an unsigned int is always positive. CID #206207 CID #206208 CID #206209 CID #206210 CID #206211 Part-of: 2020-05-06 06:35:27 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Avoid potential double-free stream->name was being freed (without being NULL-ed) before we were certain it would be set again. CID #1456071 Part-of: 2020-05-05 17:30:48 +0200 Edward Hervey * gst/deinterlace/gstdeinterlace.c: deinterlace: Don't leak frame in error case CID #1455494 Part-of: 2020-05-05 15:19:49 +0200 Edward Hervey * gst/multifile/gstsplitmuxsrc.c: slitmuxsrc: Properly stop the loop if not part reader is present Previously this would end up in a refcounting loop of hell. Part-of: 2020-03-31 14:32:19 +0300 Vivia Nikolaidou * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Add skip-backwards-streams property Backwards timestamps confuse librtmp, even if they're only backwards relative to the other stream. If the timestamp of a stream is going backwards related to the other stream, this property allows the muxer to skip a few buffers until it reaches the timestamp of the other stream. Part-of: 2020-03-31 14:10:35 +0300 Vivia Nikolaidou * gst/flv/gstflvmux.c: flvmux: Allow requesting streamable pads after header is written Allows us to request pads after writing header for streamable flv's. For non-streamable it doesn't make sense to request a new pad after writing the header, because the headers have been written already and we can't add the new stream. But for streamable, any clients that connect after the new pad has been added will be able to see both streams. Part-of: 2020-04-27 18:11:32 +1000 Matthew Waters * ext/qt/gstqtglutility.cc: qt/x11: also pass the window for gstgl -> qt context Removes this warning from Qt: QGLXContext: Multiple configs for FBConfig ID -1 QSGContext::initialize: depth buffer support missing, expect rendering errors Part-of: 2020-04-27 15:34:15 +1000 Matthew Waters * ext/qt/qtglrenderer.cc: * ext/qt/qtglrenderer.h: qt: perform surface creation in the main thread As is required when creating a QWindow instance set out in the Qt documentation. Part-of: 2020-04-22 15:32:31 -0400 Olivier Crête * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Add 'mp3 ' fourcc that VLC seems to produce now Part-of: 2020-04-22 14:09:37 +0300 Sebastian Dröge * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: Properly free internal packets queue in finalize() As we override the GLib item with our own structure, we cannot use any function from GList or GQueue that would try to free the RTPJitterBufferItem. In this patch, we move away from g_queue_new() which forces using g_queue_free(). This this function could use g_slice_free() if there is any items left in the queue. Passing the wrong size to GSLice may cause data corruption and crash. A better approach would be to use a proper intrusive linked list implementation but that's left as an exercise for the next person running into crashes caused by this. Be ware that this regression was introduced 6 years ago in the following commit [0], the call to flush() looked useless, as there was a g_queue_free() afterward. Signed-off-by: Nicolas Dufresne [0] https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/commit/479c7642fd953edf1291a0ed4a3d53618418019c Part-of: 2020-04-20 19:43:57 +0900 Seungha Yang * tests/check/elements/splitmuxsink.c: * tests/check/elements/splitmuxsinktimecode.c: * tests/check/meson.build: tests: splitmuxsink: Add more timecode based split test ... and split test cases to run tests in parallel 2020-04-10 23:52:45 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Enhancement for timecode based split The calculated threshold for timecode might be varying depending on "max-size-timecode" and framerate. For instance, with framerate 29.97 (30000/1001) and "max-size-timecode=00:02:00;02", every fragment will have identical number of frames 3598. However, when "max-size-timecode=00:02:00;00", calculated next keyframe via gst_video_time_code_add_interval() can be different per fragment, but this is the nature of timecode. To compensate such timecode drift, we should keep track of expected timecode of next fragment based on observed timecode. 2020-04-11 00:35:16 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Post error when requested timecode interval is invalid In case we cannot rely on max-size-timecode for split decision, post error instead of crashing 2020-04-16 16:47:50 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: don't use RTX packets in rate-calc and reset-logic The problem was this: Due to the highly irregular arrival of RTX-packet the max-misorder variable could be pushed very low. (-10). If you then at some point get a big in the sequence-numbers (62 in the test) you end up sending RTX-requests for some of those packets, and then if the sender answers those requests, you are going to get a bunch of RTX-packets arriving. (-13 and then 5 more packets in the test) Now, if max-misorder is pushed very low at this point, these RTX-packets will trigger the handle_big_gap_buffer() logic, and because they arriving so neatly in order, (as they would, since they have been requested like that), the gst_rtp_jitter_buffer_reset() will be called, and two things will happen: 1. priv->next_seqnum will be set to the first RTX packet 2. the 5 RTX-packet will be pushed into the chain() function However, at this point, these RTX-packets are no longer valid, the jitterbuffer has already pushed lost-events for these, so they will now be dropped on the floor, and never make it to the waiting loop-function. And, since we now have a priv->next_seqnum that will never arrive in the loop-function, the jitterbuffer is now stalled forever, and will not push out another buffer. The proposed fixes: 1. Don't use RTX in calculation of the packet-rate. 2. Don't use RTX in large-gap logic, as they are likely to be dropped. 2020-04-15 12:36:29 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Increase internal bitstream pool size This patch will now set the maximum of buffers to 32, allowing to grow the pool for drivers that supports that and will respect the minimum buffers reported by the driver. This was made to fix a stall with the virtio CODEC driver. Fixes #672 2020-04-15 17:50:31 +0300 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Do split-at-running-time splitting based on the time of the start of the GOP If the start of the GOP is >= the requested running time, put it into a new fragment. That is, split-at-running-time would always ensure that a split happens as early as possible after the given running time. Previously it was comparing against the current incoming timestamp, which does not tell us what we actually want to know as it has no direct relation to the GOP start/end. 2020-04-15 13:21:05 +0300 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix off-by-one in running time comparison for split-at-running-time If we get a keyframe exactly at the requested running time we would only split on the next keyframe afterwards due to wrong usage of > vs. >=. 2020-04-09 12:23:44 -0400 Thibault Saunier * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Properly set segments seqnums after seeks 2020-04-08 19:49:00 +0300 Vivia Nikolaidou * gst/flv/gstflvdemux.c: flvdemux: Don't write an empty string as a tag To stop warnings like: GStreamer-WARNING **: 19:47:48.186: Trying to set empty string on taglist field 'encoder'. Please file a bug. 2020-04-08 12:34:40 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: request the maximum number of buffers for USERPTR This is to match what we now do for DMABuf importation. 2019-11-20 15:32:29 +0100 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: request the maximum number of buffers for DMABUF There are often only two buffers queued in the kernel so no new buffers are requested. With every qbuf, the kernel receives a new DMABUF for the specified index. This most likely differs from the last DMABUF and the old cached entry is released. This results in a lot of map/unmap overhead if the kernel driver needs a mapping for the buffer. With a larger queue, it's quite likely, that both old and new DMABUFs are also mapped for another index. So the map/unmap is skipped, because the mapping is reference counted. The corresponding allocated buffers don't contain any actual memory, so allocating them is quite cheep. So the log message is updated to clarify this. 2020-04-08 09:45:17 -0400 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtspsrc: Avoid stack overflow recursing waiting for response Instead of recursing, simply implement a loop with gotos, the same way it was done before 812175288769d647ed6388755aed386378d9210c Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/710 2020-04-06 16:25:59 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Add property for enforcing the creation of chunks in single-stream files This is disabled by default as it unnecessarily creates bigger headers but it is something that is required by some applications and most notably the Apple ProRes spec. 2020-04-03 00:16:10 +1100 Jan Schmidt * gst/flv/gstflvmux.c: flvmux: Fix invalid padlist accesses. Request pads can released at any time, so make sure to hold the object lock when iterating the element sinkpads list where that's safe, or to use other safe pad iteration patterns in other places. When choosing a best pad, return a reference to the pad to make sure it stays alive for output in the aggregator srcpad task. Should fix a spurious valgrind error in the CI flvmux tests and some other potential problems if the request sink pads are released while the element is running.. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/714 2018-10-22 15:41:56 +0300 Vivia Nikolaidou * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Add option to create a timecode trak in non-mov flavors Even if timecode trak is officially unsupported in non-mov flavors, some software still supports it, e.g. Final Cut Pro X: https://developer.apple.com/library/archive/technotes/tn2174/_index.html The user might still expect to see the timecode information in the non-mov file despite it being officially unsupported , because other software e.g. QuickTime will create a timecode trak even in mp4 files. Furthermore, software that supports timecode trak in non-mov flavors will also display the file duration in "timecode units" instead of real clock time, which is not necessarily the same for 29.97 fps and friends. This might confuse users, who see a different duration for the same framerate and amount of frames depending on whether the container is mp4 or mov. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/512 2020-01-16 09:30:39 +0200 Sebastian Dröge * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpL8depay.c: rtpLXXdepay: Set the UNPOSITIONED flag on the audio-info when configuring an unpositioned layout Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/688 2020-04-01 13:19:46 +0200 Kristofer Björkström * gst/rtp/gstrtpjpegpay.c: * tests/check/elements/rtpjpeg.c: * tests/check/meson.build: rtpjpeg: Use gst_memory_map() instead of gst_buffer_map() gst_buffer_map () results in memcopying when a GstBuffer contains more than one GstMemory. This has quite an impact on performance on systems with limited amount of resources. With this patch the whole GstBuffer will not be mapped at once, instead each individual GstMemory will be iterated and mapped separately. 2020-04-01 13:17:03 +0200 Kristofer Björkström * gst/rtp/gstbuffermemory.c: * gst/rtp/gstbuffermemory.h: buffermemory: keep track of buffer size and current offset Added the possibility to get current offset and the total size of the buffer. 2020-04-03 10:29:18 +0200 Havard Graff * gst/rtp/gstrtpopuspay.c: * tests/check/elements/rtpopus.c: * tests/check/meson.build: rtpopuspay: make depay ! pay work There is a use-case for a server to re-payload opus going through it. Problem was that the payloader requires channels in the caps, but this is not something the depayloader can parse out of the stream, meaning caps-negotiation would fail. Removing the requirement of channels in the template-caps fixes this. 2020-04-03 16:49:25 +0900 Seungha Yang * tests/check/elements/splitmuxsink.c: * tests/check/elements/splitmuxsrc.c: * tests/check/meson.build: tests: Split splitmux test case Since we are adding more and more tests into splitmux, we need to split it to avoid CI timeout. 2020-04-03 13:45:56 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmuxsink: Don't send too many force key unit event splitmuxsink should requst keyframe depending on configured threshold and previously requested time in order to avoid too many keyframe request. 2020-03-20 21:32:36 +1100 Jan Schmidt * gst/matroska/matroska-demux.c: matroska: Check the return value of gst_segment_do_seek() gst_segment_do_seek() can fail. 2018-06-08 13:12:01 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Send instant-rate-change event if requested in the SEEK event Handle an instant rate change seek immediately by reflecting it downstream as an instant-rate-change event, and do no further seek handling. 2018-05-15 18:26:16 +0300 Sebastian Dröge * gst/matroska/matroska-demux.c: matroska-demux: Send instant-rate-change event if requested in the SEEK event Short-circuit instant rate change events by generating a downstream instant-rate-change event and doing no further seek processing. 2020-03-10 23:16:00 +0900 Seungha Yang * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: Update for video-hdr struct change See the change of -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/594 2020-03-31 15:51:27 -0400 Aaron Boxer * gst/rtpmanager/gstrtpbin.c: rtpbin: make warning messages more meaningful 2020-03-27 19:24:03 +0100 Nicolas Pernas Maradei * gst/rtpmanager/gstrtpsession.c: rtpsession: rename RTCP thread RTP session starts a new thread for RTCP and names it "rtpsession-rtcp-thread" which happens to be longer than the maximum 16B allowed by pthread_setname_np and causes the naming to fail. See docs for more details. This commit simply shortens the thread's name so it can actually be set. 2020-03-30 22:26:33 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: create specific API for appending buffers, events etc To avoid specifying a bunch of mystic variables. 2020-02-10 17:33:54 +0100 Havard Graff * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: various test-improvements Mainly generalize all the latest tests that have found various stalls in the jitterbuffer, so that they only consist of a series of packets with various seqnum/rtptime/rtx combinations, arriving at a specific time. This means future tests can be more easily written to prove certain behavior does not cause stalls. Also fix the warning on windows: warning C4244: 'initializing': conversion from 'double' to 'gint', possible loss of data 2020-03-27 14:07:04 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix waiting timer/queue code Changing the types from boolean to guint due to the ++ operand used on them, and only call JBUF_SIGNAL_QUEUE after settling down, or else you end up signaling the waiting code in chain() for every buffer pushed out. 2020-03-23 19:55:37 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Error out instead of crashing if reserved-max-duration is 0 or no samples could be created in prefill mode 2020-03-12 15:16:44 +0200 Sebastian Dröge * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: Use GST_VIDEO_OVERLAY_COMPOSITION_BLEND_FORMATS for the supported formats We don't do any blending by ourselves since a while now. Note that this is a regression in "supported" formats: previously ARGB64 was supported, for example, but in practice it caused blending to not take place at all. 2020-03-24 00:23:24 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: splitmuxsrc: Fix some deadlock conditions and a crash When switching the splitmuxsrc state back to NULL quickly, it can encounter deadlocks shutting down the part readers that are still starting up, or encounter a crash if the splitmuxsrc cleaned up the parts before the async callback could run. Taking the state lock to post async-start / async-done messages can deadlock if the state change function is trying to shut down the element, so use some finer grained locks for that. 2020-03-24 00:18:54 +1100 Jan Schmidt * tests/check/elements/splitmux.c: splitmux: Make the unit test faster The playback test is considerably faster if it runs with the appsink set to sync=false 2020-03-25 22:14:36 +0900 Seungha Yang * tests/check/elements/splitmux.c: * tests/check/meson.build: tests: splitmux: Add test for timecode based split 2020-03-25 21:20:07 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Split fragment only if queued time is larger than threshold The queued time includes the duration of the last queued frame (i.e., new keyframe) so the condition check should not be inclusive. Note that the new fragment will be cut excluding the last frame and therefore if the condition is inclusive way, the fragment might have one frame shorter duration for all keyframe stream such as jpeg or all-inter video streams. 2020-03-25 21:01:00 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't need to trace next timecode for split decision Since the commit 94bb76b6b9c48981d3ad42a8c4370b9658db4229, splitmuxsink will split fragments based on queued time and the threshold of that. So don't need to store the next timecode for split decision. 2018-08-08 09:27:19 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: add alternate interlace mode When using this mode each frame is split in two fields, each one being transferred using its own buffer. This is implemented with the V4L2_FIELD_ALTERNATE field format in v4l2. This mode is enabled using a caps filter such as "v4l2src ! video/x-raw\(format:Interlaced\)" Here are the main changes related to this feature: - use the INTERLACED caps feature with this mode. - in this mode both fields of a given frame have the same sequence/offset so adjust the algorithm checking for lost field/frame accordingly. - double pool's min number of buffers as each frame requires 2 buffers. Fix #504 Co-authored-by: Zeeshan Ali 2020-02-05 13:03:51 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: display field when setting or trying format Ease debugging interlacing pipelines. 2020-01-30 12:35:02 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videoenc.c: v4l2: pass v4l2object to GST_V4L2_MIN_BUFFERS() Will be used to double the number of buffers in alternate interlace mode. 2020-01-30 12:09:12 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: use GST_VIDEO_INFO_FIELD_HEIGHT() Use GST_VIDEO_INFO_FIELD_HEIGHT() instead of GST_VIDEO_INFO_HEIGHT() when we actually want the field height rather than the frame height. So far both are equals but that won't longer be the case when implementing alternate interlace mode. 2020-03-24 22:08:27 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Mark some split decision related properties as MUTABLE_READY The change of various criteria for split decision while muxing is on progress wouldn't work well as expected. 2020-03-24 13:45:00 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Take account queued time and max-size-timecode for split decision Not only the requested keyframe time, the queued size should be a criterion for the split decision of timecode based mode (same as max-size-time based split case). 2020-03-24 12:55:27 +1100 Matthew Waters * ext/qt/gstqtoverlay.cc: qmlgloverlay: fix usage without an qmlglsink in the pipeline Without a qmlglsink, we need to retrieve the window system display ourselves rather than relying solely on qmlglsink to have priority on the choice of display. 2020-03-23 21:32:04 -0400 Xavier Claessens * gst/rtpmanager/rtptwcc.c: * gst/videocrop/gstvideocrop.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpsession.c: Fix usage of C99 It's 2020, way too early for that, let's stick to C89 for now. 2020-03-23 16:34:46 +0900 Seungha Yang * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.h: v4l2bufferpool: Use unique name for v4l2bufferpool object Assign unique sequence number to an object name for better debugging 2020-03-23 14:02:22 +1100 Matthew Waters * ext/qt/qtglrenderer.cc: qmlgloverlay: don't leak resources freed on a different GL thread deleting a QOpenGLFrameBufferObject needs to occur on the same thread it was created on in order to actually free the relevant resources immediately. Otherwise, they will be queued for deletion and not freed until the associated QOpenGLContext is destroyed. 2020-03-20 09:14:01 +1100 Matthew Waters * ext/qt/gstqtglutility.cc: qt: reorganize include defines 2020-03-19 23:17:21 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtptimerqueue.c: * gst/rtpmanager/rtptimerqueue.h: * tests/check/elements/rtptimerqueue.c: rtptimerqueue: remove ->num from the timer This concept was only used by the "multi"-lost timer, and since that one is not around any longer, the "num" concept is superfluous. 2020-03-19 23:37:26 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: remove the concept of "already-lost" This is a concept that only applies when a buffer arrives in the chain function, and it has already been scheduled as part of a "multi"-lost timer. However, "multi"-lost timers are now a thing of the past, making this whole concept superflous, and this buffer is now simply counted as "late", having already been pushed out (albeit as a lost-event). 2020-03-19 23:12:04 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: immediately insert a lost-event on multiple lost packets There is a problem with the code today, where a single timer will be scheduled for a series of lost packets, and then if the first packet in that series arrives, it will cause a rescheduling of that timer, going from a "multi"-timer to a single-timer, causing a lot of the packets in that timer to be unaccounted for, and creating a situation in where the jitterbuffer will never again push out another packet. This patch solves the problem by instead of scheduling those lost packets as another timer, it instead asks to have that lost-event pushed straight out. This very much goes with the intent of the code here: These packets are so desperately late that no cure exists, and we might as well get the lost-event out of the way and get on with it. This change has some interesting knock-on effect being presented in later commits. It completely removes the concept of "already-lost", so that is why that test has been disabled in this commit, to be removed later. 2020-03-19 23:03:50 +0100 Havard Graff * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: rework large-gap tests Make sure to set the time the buffer is supposed to arrive at, so as not to trigger an artificial situation. 2020-03-19 12:17:22 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: refactor lost_timeout code Split it up in code related to the timer, (do_lost_timeout) and code to insert a lost-item/event and update private jitterbuffer-variables. 2019-10-18 17:43:36 +0200 Havard Graff * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/rtptimerqueue.c: * tests/check/meson.build: test/check: split out rtptimerqueue-tests in a separate file 2020-02-05 09:56:23 +0100 Dmitriy Purgin * ext/qt/qtplugin.pro: gstqmlgl: Link to opengl32.lib on MinGW 2020-03-19 23:51:47 +0900 Seungha Yang * gst/isomp4/gstqtmux.c: qtmux: Fix build warning gstqtmux.c(644): warning C4133: '=': incompatible types - from 'gboolean (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)' to 'GstFlowReturn (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)' 2020-03-19 23:05:49 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Reset cleanly for reuse Reset the splitmuxsink completely when changing states so that it can be reused. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1241 2020-02-17 22:37:10 -0600 Zebediah Figura * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: mpegaudioparse: Use a constant bit rate to convert between time and bytes if possible. This should result in no worse accuracy than the base parse element, and may result in better accuracy. In particular, the number of bytes processed at any given point, as accumulated by baseparse, can be only accurate to (1 / # of frames) bytes per second, and if we try to seek immediately after pausing the pipeline to a large offset, this small inaccuracy can propagate to something noticeable. The use case that prompted this patch is a 45-minute MPEG-1 layer 3 file, which has a constant bit rate but no seek tables. Trying to seek the pipeline immediately after pauisng it, without the ACCURATE flag, to a location 41 minutes in, yields a location that is, even with , still audibly incorrect. This patch yields a much closer position, no longer audibly incorrect, and likely within a frame of the most correct position. 2020-03-04 22:10:40 +0100 Mathieu Duponchelle * gst/isomp4/gstqtmux.c: qtmux: fix renegotiation check By the time sink_event is called, the pad's current caps have already been updated. To address this, implement sink_event_pre_queue, and check if the pad can be renegotiated there. Fixes #707 2020-03-12 20:34:47 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmuxsink: Decouple keyframe request and the decision for fragmentation Split the decision for keyframe request and fragmentation in order to ensure periodic keyframe request. 2020-02-26 18:29:06 +1100 Matthew Waters * ext/qt/gstqtglutility.cc: * ext/qt/gstqtoverlay.cc: * ext/qt/qtglrenderer.cc: * ext/qt/qtglrenderer.h: * ext/qt/qtitem.cc: * tests/examples/qt/qmloverlay/main.cpp: * tests/examples/qt/qmloverlay/overlay2.qml: * tests/examples/qt/qmloverlay/qmloverlay.qrc: qt: add a qml overlay filter element [part 2] It takes a qml scene description and renders it using a possible input stream. Currently supported on GLX and WGL. Follow up to (as that MR had an old version of the commit): - https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/475 - 4778d7166a02caf793df4f845dc35b6933d87c81: qt: add a qml overlay filter element 2020-02-26 18:29:06 +1100 Matthew Waters * ext/qt/gstplugin.cc: * ext/qt/gstqtglutility.cc: * ext/qt/gstqtglutility.h: * ext/qt/gstqtoverlay.cc: * ext/qt/gstqtoverlay.h: * ext/qt/meson.build: * ext/qt/qtglrenderer.cc: * ext/qt/qtglrenderer.h: * tests/examples/qt/meson.build: * tests/examples/qt/qmloverlay/main.cpp: * tests/examples/qt/qmloverlay/main.qml: * tests/examples/qt/qmloverlay/meson.build: * tests/examples/qt/qmloverlay/overlay.qml: * tests/examples/qt/qmloverlay/qmloverlay.qrc: qt: add a qml overlay filter element It takes a qml scene description and renders it using a possible input stream. Currently supported on GLX and WGL. 2020-02-25 21:47:14 +1100 Matthew Waters * ext/qt/gstqsgtexture.cc: * ext/qt/qtitem.cc: qt: don't always activate/deactivate our GstGLContext Techincally it is enough to activate at the beginning and then forget. 2020-02-04 19:43:52 +1100 Matthew Waters * tests/examples/qt/meson.build: * tests/examples/qt/qmlsink-dynamically-added/.gitignore: * tests/examples/qt/qmlsink-dynamically-added/main.cpp: * tests/examples/qt/qmlsink-dynamically-added/main.qml: * tests/examples/qt/qmlsink-dynamically-added/meson.build: * tests/examples/qt/qmlsink-dynamically-added/play.pro: * tests/examples/qt/qmlsink-dynamically-added/qmlsink.qrc: test/qml: add an dynamically adding qmlglsink element The example shows how to add qmlglsink to an already running pipeline with pre-existing OpenGL elements. 2020-02-04 19:40:45 +1100 Matthew Waters * ext/qt/gstqtsink.cc: qmlglsink: propagate the context up the the application Allows the application to be notified of the OpenGL context creation. 2020-02-03 15:59:34 +1100 Matthew Waters * ext/qt/qtitem.cc: qtitem: fix leak of caps 2020-03-15 19:28:18 +0100 Niels De Graef * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.h: wavpack: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:26:18 +0100 Niels De Graef * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp9dec.h: * ext/vpx/gstvp9enc.h: * ext/vpx/gstvpxdec.h: * ext/vpx/gstvpxenc.h: vpx: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:22:00 +0100 Niels De Graef * ext/twolame/gsttwolamemp2enc.h: twolame: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:20:49 +0100 Niels De Graef * ext/taglib/gstapev2mux.h: * ext/taglib/gstid3v2mux.h: taglib: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:18:39 +0100 Niels De Graef * ext/speex/gstspeexdec.h: * ext/speex/gstspeexenc.h: speex: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:16:22 +0100 Niels De Graef * ext/soup/gstsouphttpclientsink.h: soup: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:14:17 +0100 Niels De Graef * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:11:52 +0100 Niels De Graef * ext/raw1394/gst1394clock.h: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gstdv1394src.h: * ext/raw1394/gsthdv1394src.h: raw1394: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:06:50 +0100 Niels De Graef * ext/qt/gstqtsink.h: * ext/qt/gstqtsrc.h: qt: Use G_DECLARE_FINAL_TYPE 2020-03-15 19:00:18 +0100 Niels De Graef * ext/pulse/pulsedeviceprovider.h: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.h: pulse: Use G_DECLARE_FINAL_TYPE 2020-03-15 18:54:33 +0100 Niels De Graef * ext/mpg123/gstmpg123audiodec.h: mpg123: Use G_DECLARE_FINAL_TYPE 2020-03-15 18:52:57 +0100 Niels De Graef * ext/libpng/gstpng.h: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.h: libpng: Use G_DECLARE_FINAL_TYPE 2020-03-15 18:49:53 +0100 Niels De Graef * ext/libcaca/gstcacasink.h: * ext/libcaca/gstcacatv.h: libcaca: Use G_DECLARE_FINAL_TYPE 2020-03-15 18:40:28 +0100 Niels De Graef * ext/lame/gstlamemp3enc.h: lame: Use G_DECLARE_FINAL_TYPE 2020-03-14 17:52:38 +0100 Niels De Graef * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.h: jack: Use G_DECLARE_FINAL_TYPE 2020-03-14 17:43:50 +0100 Niels De Graef * ext/gtk/gstgtkbasesink.h: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.h: gtk: Use G_DECLARE_FINAL_TYPE 2020-03-13 18:47:49 +0100 Niels De Graef * ext/gdk_pixbuf/gstgdkpixbufdec.h: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: * ext/gdk_pixbuf/gstgdkpixbufsink.h: gdk_pixbuf: Use G_DECLARE_FINAL_TYPE 2020-03-13 18:42:38 +0100 Niels De Graef * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.h: * ext/flac/gstflactag.h: flax: Use G_DECLARE_FINAL_TYPE 2020-03-13 18:39:38 +0100 Niels De Graef * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.h: dv: Use G_DECLARE_FINAL_TYPE 2020-03-12 19:24:57 +0100 Niels De Graef * ext/cairo/gstcairooverlay.h: cairo: Use G_DECLARE_FINAL_TYPE 2020-03-12 19:20:42 +0100 Niels De Graef * ext/aalib/gstaasink.h: * ext/aalib/gstaatv.h: aalib: Use G_DECLARE_FINAL_TYPE 2020-03-12 16:55:44 +0000 Tim-Philipp Müller * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: add minimal vp8/vp9 rtp payloading/depayloading test 2018-10-19 16:17:17 +0200 Stian Selnes * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9pay.c: rtpvp8pay, rtpvp9pay: fix caps leak in set_caps() 2020-03-12 11:22:56 +0100 Edward Hervey * gst/videomixer/videomixer2.c: videomixer: Don't leak peer caps 2020-02-11 16:19:15 -0300 Thibault Saunier * docs/gst_plugins_cache.json: * gst/multifile/gstimagesequencesrc.c: * gst/multifile/gstimagesequencesrc.h: * gst/multifile/gstmultifile.c: * gst/multifile/meson.build: imagesequencesrc: Cleanup and add some features * Implement the GstURIHandlerInterface * Rework the locking * Implement backward seeking handling * Generate documentation 2016-04-10 02:25:32 +0000 Fabian Orccon * gst/multifile/gstimagesequencesrc.c: * gst/multifile/gstimagesequencesrc.h: Add an imagesequencesrc element to stream sequence of images See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/121 2020-03-05 08:55:44 -0800 Gordon Hart * sys/v4l2/gstv4l2src.c: v4l2src: decrease gst_v4l2src_create log verbosity Lower the verbosity of the 'sync' log message emitted each buffer from gst_v4l2src_create down to LOG(6) from INFO(4). This brings the logging behavior of v4l2src closer to the GStreamer guidelines, which recommend the INFO level be reserved for rare or one-off messages. 2020-03-10 17:19:46 +0800 yychao * gst/isomp4/qtdemux.c: qtdemux: Add support for AC4 The caps received from qtdemux for AC-4 content are audio/x-gst-fourcc-ac_4 Based on patch by: Savinderjit Kaur Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/413 2020-03-10 21:07:12 +1100 Matthew Waters * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: handle reconfigure events on the srcpad 2020-03-05 22:47:16 +1100 Matthew Waters * gst/imagefreeze/gstimagefreeze.c: imagefreeze: properly ignore setting caps failures Ignore the return value of gst_pad_set_caps() so that setcaps will set a framerate that is usable. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/705 2020-03-05 22:45:32 +1100 Matthew Waters * gst/imagefreeze/gstimagefreeze.c: imagefreeze: don't fail sending sticky events downstream They will be repropagated anyway. 2020-03-09 23:31:09 +0100 Markus Ebner * gst/videocrop/gstvideocrop.c: videocrop: Add support for Y41B and Y42B 2020-03-09 23:25:03 +0100 Markus Ebner * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: Add support for Y444 - Refactored the planar transform method to support all video formats that are stored planar, independent of the used subsampling - Added support for Y444 2020-03-09 23:23:50 +0100 Markus Ebner * gst/videocrop/gstvideocrop.c: videocrop: Use G_VALUE_INIT to initialize GValues 2020-02-28 19:35:34 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Configure JPEG chroma-siting for YUV formats 2020-02-06 09:23:24 +0100 Ognyan Tonchev * gst/rtp/gstbuffermemory.c: * gst/rtp/gstbuffermemory.h: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/meson.build: * tests/check/elements/rtph264.c: rtph26x: Use gst_memory_map() instead of gst_buffer_map() in avc mode gst_buffer_map () results in memcopying when a GstBuffer contains more than one GstMemory and when AVC (length-prefixed) alignment is used. This has quite an impact on performance on systems with limited amount of resources. With this patch the whole GstBuffer will not be mapped at once, instead each individual GstMemory will be iterated and mapped separately. 2019-11-26 15:08:20 +0100 Milian Wolff * ext/qt/gstqtgl.h: qmlgl: ensure Qt defines GLsync to fix compile on some platforms By explictly including QtGui/qopengl.h we force the code path that defines GLsync in the Qt-specific way. Without that, some platforms failed to compile the qmlgl plugin, since neither Qt nor gstreamer defined GLsync then, leading to e.g.: ``` make[4]: Entering directory '/.../gst-plugins-good-1.16.1/ext/qt' CXX libgstqmlgl_la-qtitem.lo In file included from gstqtgl.h:32, from qtitem.h:27, from qtitem.cc:28: /.../usr/include/gstreamer-1.0/ gst/gl/gstglfuncs.h:93:17: error: expected identifier before ‘*’ token ret (GSTGLAPI *name) args; ^ /.../usr/include/gstreamer-1.0/ gst/gl/glprototypes/sync.h:27:1: note: in expansion of macro ‘GST_GL_EXT_FUNCTION’ GST_GL_EXT_FUNCTION (GLsync, FenceSync, ^~~~~~~~~~~~~~~~~~~ ``` 2020-03-02 13:50:55 +0100 Havard Graff * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtptwcc.c: * gst/rtpmanager/rtptwcc.h: rtptwcc: make RTPTWCCManager a GObject 2020-03-04 11:17:16 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix stalling when resetting timers When calling gst_rtp_jitter_buffer_reset you pass in a seqnum. This is considered the starting-point for a new stream. However, the old behavior would unref this buffer, basically lying to the thread that is pushing out buffers saying that it can expect this buffer, when it would never arrive. The resulting effect being no more buffer pushed out of the jitterbuffer, and it would buffer incoming data indefinitely. By instead inserting the buffer in the gap_packets queue, the _reset() function will take responsibility for using that as the first buffer of the new stream. Fixes #703 2020-02-21 02:14:11 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: * gst/multifile/gstsplitmuxsrc.c: * tests/check/elements/splitmux.c: splitmux: Avoid negative DTS In order to concatenate fragments, splitmuxsrc offsets the start of each fragment PTS to 0 to align it with the previous file. This means that DTS can go negative for the first fragment, with really bad results. Add a fixed offset to outgoing timestamp ranges to avoid that. 2020-03-04 03:43:51 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Remove warning in the log for mono video Vanilla mono video was generating a spurious warning into the debug log that's just misleading. Handle mono caps explicitly to avoid the warning. 2020-01-27 12:29:18 +0530 Guillaume Desmottes * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: add alternate support In this mode each field is carried using its own buffer. Allow deinterlace to negotiate caps with the Interlaced feature and adjust the algorithm fetching lines. Fix #620 2020-02-03 13:08:39 +0530 Guillaume Desmottes * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: add wrapper to get field lines from history No semantic change so far, will be used to implement alternate support. 2020-02-04 16:48:21 +0530 Guillaume Desmottes * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: stop checking line index boundaries The LINE2() macro already prevents out of bound indexes using CLAMP_HI() and CLAMP_LOW(). 2020-01-20 12:30:12 +0530 Guillaume Desmottes * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: fix video info on output frames Output frames used to have their interlace mode set to the same one as the input. This breaks their field and comp heights when deinterlacing an alternate stream. 2020-01-14 14:51:07 +0530 Guillaume Desmottes * gst/deinterlace/gstdeinterlace.c: deinterlace: use output caps to compute buffer size In interlace-mode=alternate the input buffers have half the size of the output ones as each field has its own buffer. 2020-02-29 08:10:56 -0500 Jennifer Berringer * gst/audioparsers/gstflacparse.c: flacparse: fix broken reordering of flac metadata Each FLAC metadata block starts with a flag denoting whether it is the last metadata block. The existing flacparse code moves any existing VORBISCOMMENT block to immediately follow the STREAMINFO block without changing any block's last-metadata-block flag. If no VORBISCOMMENT block exists, it created one with the last-metadata-block flag set to true. This results in gstflacdec sometimes giving bad headers to libflac when trying to play perfectly valid FLAC files depending on the file's metadata ordering. Depending on the contents of the other metadata blocks, current versions of libflac may or may not return FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER when given this broken metadata. This is most noticeable with files that have a large cover art image attached where VORBISCOMMENT is the very last metadata block with no PADDING afterwards. This patch changes that behavior so that: 1. For FLAC files that already have a VORBISCOMMENT block, the metadata order is preserved. 2. For FLAC files that do not have a VORBISCOMMENT block, the generated dummy VORBISCOMMENT is placed immediately after STREAMINFO and inherits the last-metadata-block flag from STREAMINFO. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/484 2020-02-27 14:50:51 +0900 Yeongjin Jeong * tests/check/elements/flvmux.c: tests: flvmux: Instead of using the testclock, just send eos event for drain When using the testclock for determining clock in test, it is sometimes observed that the clock entry is not registered in time by the aggregator. So deadlock occurs between the aggregator and the test thread. 2020-02-28 14:23:51 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Try to infer useful header values for raw audio if the sound sample descriptions contain zero values 2020-02-28 14:00:51 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Also use the enda atom for determining endianess of in32, fl32 and fl64 formats Previously it was only used for in24. 2020-02-28 13:59:42 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix up header information for various fixed-format raw audio formats Sometimes the headers contain useless, wrong or zero values for e.g. the sample size with these formats. There's only a single valid value for them so let's set these instead. 2020-02-28 13:59:06 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Don't print "unhandled type" warnings for various other raw audio fourccs 2020-02-28 13:57:37 +0200 Sebastian Dröge * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Add some more raw audio fourccs to the header instead of duplicating them 2020-02-25 21:14:54 +0530 Nirbheek Chauhan * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Don't use glib format modifiers with sscanf We do not have a way to know the format modifiers to use with string functions provided by the system. G_GUINT64_FORMAT and other string modifiers only work for glib string formatting functions. We cannot use them for string functions provided by the stdlib. See: https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description ``` ../gst/rtpmanager/gstrtpjitterbuffer.c: In function 'gst_jitter_buffer_sink_parse_caps': ../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: unknown conversion type character 'l' in format [-Werror=format=] || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1) ^~~~~~~~~~ In file included from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32, from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32, from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib.h:30, from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27, from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/rtp/gstrtpbuffer.h:27, from ../gst/rtpmanager/gstrtpjitterbuffer.c:108: /home/nirbheek/cerbero/build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here #define G_GUINT64_FORMAT "llu" ^ ../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: too many arguments for format [-Werror=format-extra-args] || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1) ^~~~~~~~~~ ``` See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/379 2020-02-24 15:25:07 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Add support for 8k resolutions in prefill mode with ProRes 2020-02-25 11:06:43 +0200 Sebastian Dröge * gst/rtpmanager/rtptimerqueue.c: rtpjitterbuffer: Include string.h for memcpy() / memset() Usually something else is pulling it in somehow already, but not on Windows. 2020-02-24 13:06:27 +0000 Håvard Graff * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: fix crash when no extension-header present for twcc 2020-02-21 09:34:30 +0100 Johan Bjäreholt * gst/matroska/matroska-mux.c: matroska-mux: Fix incorrect rounding of timestamps Previously we saved the buffer_timestamp straight into mux->cluster_time. Since the cluster time saved into the file does not have as high precision as GstClockTime depending on the timecodescale the rounding of relative_timestamp was invalid as mux->cluster_time which it was calculated relative to was not equal to the cluster time written to the matroska file. Example of "mkvinfo -v" of how it looks before and after this change in an scenario where previously timestamps got out of order because of this issue. Notice the timestamp of the SimpleBlock right before and right after the Cluster now being in order. The consequence of this however is that the cluster timestamp is not necessarily the same as the timestamp of the first buffer in the cluster however (in case it's rounded up). Before | + SimpleBlock (track number 1, 1 frame(s), timecode 126.922s = 00:02:06.922) | + Frame with size 432 | + SimpleBlock (track number 2, 1 frame(s), timecode 126.933s = 00:02:06.933) | + Frame with size 329 | + SimpleBlock (track number 2, 1 frame(s), timecode 126.955s = 00:02:06.955) | + Frame with size 333 |+ Cluster | + Cluster timecode: 126.954s | + Cluster previous size: 97344 | + SimpleBlock (key, track number 1, 1 frame(s), timecode 126.954s = 00:02:06.954) | + Frame with size 61239 | + SimpleBlock (track number 2, 1 frame(s), timecode 126.975s = 00:02:06.975) | + Frame with size 338 After | + SimpleBlock (track number 1, 1 frame(s), timecode 135.456s = 00:02:15.456) | + Frame with size 2260 | + SimpleBlock (track number 2, 1 frame(s), timecode 135.468s = 00:02:15.468) | + Frame with size 332 | + SimpleBlock (track number 2, 1 frame(s), timecode 135.490s = 00:02:15.490) | + Frame with size 335 |+ Cluster | + Cluster timecode: 135.489s | + Cluster previous size: 158758 | + SimpleBlock (key, track number 1, 1 frame(s), timecode 135.490s = 00:02:15.490) | + Frame with size 88070 | + SimpleBlock (track number 2, 1 frame(s), timecode 135.511s = 00:02:15.511) | + Frame with size 336 2020-02-19 15:59:19 +1100 Jake Barnes * ext/soup/gstsouphttpsrc.c: souphttpsrc: Fix cookies property Disable session sharing and cookie jar when cookies property is set. The cookie jar actually replaces or removes any existing Cookie header set on the message, so the cookies property was effectively being ignored. There doesn't appear to be a way to inject the cookies into the jar without having to specify matching domains etc., so it's not possible to simulate the old behaviour of unconditionally sending the cookies with all messages, besides simply disabling the cookie jar. 2020-02-20 09:06:10 +0100 Stefano Buora * gst/rtsp/gstrtspsrc.c: rtspsrc: remove useless function calls Comparing gst_rtspsrc_loop_interleaved and gst_rtspsrc_loop_udp, and investigating on timeout issues, it sounds like a piece of code has been originally copied from udp to the interleaved one. The timeout variable is never used inside the interleaved one. No side effect has been seen in the removed function calls. The debug message removed is pointless as the timeout used is "src->tcp_timeout" that is fixed. The presence of the two timeout drove my team in investigating if the reference to the tcp_timeout was correct (it is). Hence we removed the misleading reference to the local timeout variable. 2020-02-20 13:43:13 +1100 Matthew Waters * gst/rtpmanager/gstrtpbin.c: rtpbin: fix typo setting max-dropout/misorder-time we were setting the max-dropout-time to the value of the max-misorder-time which by default has a factor of 30 difference in value. 2020-02-19 20:27:54 +0900 Seungha Yang * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Parse VP Codec Configuration Box The VP Codec Configuration Box (vpcC) contains vp9 profile and colorimetry information. Especially the profile information might be useful for downstream to select capable decoder element. 2020-02-18 18:36:36 +0900 Yeongjin Jeong * tests/check/elements/flvmux.c: tests: flvmux: Add test for rollover timestamp The timestamps that exceed uint32 maximum value should be handled to rollover. 2020-02-18 14:58:00 +0900 Yeongjin Jeong * gst/flv/gstflvmux.c: flvmux: Support rollover in timestamp For live streams, if we keep the stream for a long time, the timestamp will be larger than max_uint32. In that case, timestamp should be handled as a rollover timestamp rather than a backward timestamp. 2020-02-17 15:03:28 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: don't use the timer-object after JBUF_UNLOCK It could have been freed (rtp_timer_free) in the meantime. 2019-06-29 18:06:11 +0200 Havard Graff * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/meson.build: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: * gst/rtpmanager/rtptwcc.c: * gst/rtpmanager/rtptwcc.h: * tests/check/elements/rtpsession.c: rtpmanager: Google Transport-Wide Congestion Control RTP Extension Generating and parsing the RTCP-messages described in: https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 2020-02-14 10:08:05 +0000 Håvard Graff * gst/rtpmanager/gstrtpfunnel.c: * tests/check/elements/rtpfunnel.c: rtpfunnel: various cleanups * Organize GstRtpFunnelPad and GstRtpFunnel separately * Use G_GNUC_UNUSED instead of (void) casts * Don't call an event "caps" * Use semicolons after GST_END_TEST (helps gst-indent) 2020-01-29 23:51:45 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Merge sample tables for raw audio streams with one container sample per audio sample Instead of having chunks with one sample per raw audio sample, have chunks with a single sample that contains lots of raw audio samples. If necessary these are still split again later when reading the stream. With this we are allocating a lot less memory for the parsed sample tables and can play files that previously triggered our limit of 200MB for the sample table. For example, one file here would previously allocate 3.5GB for the sample table and now only allocates 70KB. 2020-01-13 11:55:42 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Add a minimum buffer size for raw audio to not output one buffer per frame Outputting 48000 buffers per second is not a good idea performance-wise. If a container sample is less than 1024 raw audio frames, combine multiple samples to get at least 1024 raw audio samples as long as they're stored contiguous in the file. For the other direction, if a container sample contains more than 4096 samples there is already code for splitting them up. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692750 2020-02-11 21:52:41 +0100 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: fix requested range When the server replies with a range "now-", it is presumed to be a "live" stream and we should request a similar range. This was the case prior to my refactoring to make use of gst_rtsp_range_to_string in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7, this commit restores the behaviour for that case. 2017-07-13 13:49:07 +0200 Mikhail Fludkov * gst/rtpmanager/gstrtpptdemux.c: * tests/check/elements/rtpptdemux.c: * tests/check/meson.build: rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps Refactoring to remove duplicate code and add test 2017-03-16 20:57:54 +0100 Stian Selnes * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: Fix debug to use GST_DEBUG_OBJECT 2016-09-14 16:49:26 +0200 Mikhail Fludkov * gst/rtpmanager/gstrtpbin.c: rtpbin: use max-streams on rtpssrcdemux The proper way of capping on max-streams is to do it in rtpssrcdemux. This patch uses the newly introduced property on rtpssrcdemux. Previous behavior would not prevent rtpssrcdemux spawning new pads for every new ssrc and potentialy causing performance trouble during teardown. 2017-01-18 14:32:03 +0000 John Bassett * gst/rtpmanager/gstrtpssrcdemux.c: * tests/check/elements/rtpssrcdemux.c: rtpssrcdemux: Handle RTCP APP packets Fix crash when processing RTCP APP packets. 2017-01-12 16:05:59 +0000 John Bassett * gst/rtpmanager/gstrtpssrcdemux.c: * tests/check/elements/rtpssrcdemux.c: rtpssrcdemux: Bad RTP/RTCP packet is not fatal When used for processing bundled media streams within rtpbin the rtpssrcdemux element may receive bad RTP and RTCP packets, these should not be treated as a fatal error. 2016-09-14 16:41:02 +0200 Mikhail Fludkov * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: * tests/check/elements/rtpssrcdemux.c: rtpssrcdemux: introduce max-streams property The property is useful against atacks when the sender changes SSRC for every RTP packet. The property with the same name introduced in rtpbin was not enough, because we still can end up with thousands of pads allocated in rtpssrcdemux. 2020-02-10 14:22:47 +0100 Havard Graff * tests/check/elements/rtpssrcdemux.c: rtpssrcdemux: fix test warnings 2020-02-07 10:03:49 +0100 Alexander Lapajne * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix for segmentation fault when handling set/get_parameter requests gstrtspsrc uses a queue, set_get_param_q, to store set param and get param requests. The requests are put on the queue by calling get_parameters() and set_parameter(). A thread which executs in gst_rtspsrc_thread() then pops requests from the queue and processes them. The crash occured because the queue became empty and a NULL request object was then used. The reason that the queue became empty is that it was popped even when the thread was NOT processing a get parameter or set parameter command. The fix is to make sure that the queue is ONLY popped when the command being processed is a set parameter or get parameter command. 2019-09-27 16:52:06 -0400 Olivier Crête * gst/rtpmanager/rtpsource.c: * tests/check/elements/rtpsession.c: rtpsession: Add test for packet rate maths 2019-09-10 19:03:02 +0100 olivier.crete@collabora.com * gst/rtpmanager/rtpstats.c: rtpstats: Base the packet rate average on the packet rate itself Do this so that the average update speed is in time instead of varying based on the actual packet arrival rate. 2019-09-10 18:59:02 +0100 olivier.crete@collabora.com * gst/rtpmanager/rtpstats.c: rtpstats: Don't save the ts & seqnum if the avg is not updated This makes it update correctly when you have more than one packet per frame. 2020-02-05 12:48:45 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: map GST_VIDEO_FORMAT_BGR15 The GstVideoFormat to v4l2 conversion was missing for BGR15. 2020-02-05 12:00:00 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: fix crash on invalid caps gst_v4l2_object_set_format_full() was returning FALSE without setting an error. Caller code (gst_v4l2src_fixate()) was then derefing a NULL pointer when trying to handle the error. 2020-01-27 16:00:30 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Include actual sink element in the fragment-opened/closed messages If not configuring the sinks via the "location" property this can be useful to know for which sink the fragment was actually opened/closed, especially if finalization of the fragments is happening asynchronously. 2020-01-29 12:05:07 +0100 Juergen Werner * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: fix scaling from RTP-time to NTP-time The scaling was inverse. 2020-01-27 23:59:05 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: * tests/check/elements/rtprtx.c: rtprtxsend: allow generic input caps When connected to an upstream rtpfunnel element, payload-type, ssrc and clock-rate will not be present in the received caps. rtprtxsend can already deal with only the clock rate being present there, a new property is exposed to allow users to provide a payload-type -> clock-rate map, this enables the use of the max-size-time property for bundled streams. 2020-01-27 15:17:27 -0800 Julien Isorce * ext/vpx/gstvpxenc.c: vp8enc/vp8enc: set 1 for the default value of VP8E_SET_STATIC_THRESHOLD In Google webrtc, the setting VP8E_SET_STATIC_THRESHOLD is set to 1 (except when the content is known to be static very often in which case it is set to 100, i.e. when sharing screen with Google Hangouts). The cpu usage drops a lot when using 1 for above setting because it allows the encoder to skip static/low content blocks. The current 0 default value uses too much cpu and confuses the user regarding the cpu usage expectations. User expects vp8enc to use low cpu by default. Documentation of VP8E_SET_STATIC_THRESHOLD: https://github.com/webmproject/libvpx/blob/master/vpx/vp8cx.h#L188 chromium/webrtc: https://chromium.googlesource.com/external/webrtc/+/b484ec0082948ae086c2ba4142b4d2bf8bc4dd4b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc#822 Closes #58 2020-01-27 17:16:02 -0500 Nicolas Dufresne * ext/jpeg/gstjpegdec.c: jpegdec: Check return value of gst_buffer_map() Without this check, the element will crash instead of returning an error. 2020-01-27 15:52:42 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Check the correct sink class for the existence of the "location" property 2020-01-13 11:58:12 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Always prefer information from v1/v2 sound sample description over sample description entry ffmpeg is doing the same and various files in the wild have bogus information in the sample description if the same information is also duplicated afterwards in the v1/v2 sound sample desription. Previously we only did this for non-raw audio due to https://bugzilla.gnome.org/show_bug.cgi?id=374914 but this specific file is already worked around differently. It still works after this change. Also remove ad-hoc GST_READ_DOUBLE_BE re-implementation and move the switch for legacy audio formats after reading all the sample descriptions as we want to override the values from there. 2020-01-13 20:02:58 +0200 Sebastian Dröge * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: Add support for >2 raw audio channels For this case write a WAVEFORMATEXTENSIBLE header and also reorder the raw audio channels to the AVI channel order if needed. 2020-01-13 20:07:01 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Fix writing of the channel mask with >2 channels The channel position is an enum but the conversion code assumed it's a mask. Convert accordingly. 2020-01-10 16:30:33 +0100 Kristofer Björkström * gst/rtp/gstrtph265pay.c: * tests/check/elements/rtph265.c: rtph265pay: TID for NALU type 48 was always set to 7 A typo bug: | instead of & resulted in TID alwasy being set to 7 for the aggregated NALU of type 48 2020-01-10 14:54:26 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Add support for replacing the output buffer By default imagefreeze will still reject new buffers after the first one and immediately return GST_FLOW_EOS but the new allow-replace property allows to change this. Whenever updating the buffer we now also keep track of the configured caps of the buffer and from the source pad task negotiate correctly based on the potentially updated caps. Only the very first time negotiation of a framerate with downstream is performed, afterwards only the caps themselves apart from the framerate are updated. 2020-01-09 18:43:02 +0000 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Fix race on pad reconnection Elements emitting frames through several srcpads should use a flow combiner to aggregate the chain returns and therefore only return GST_FLOW_NOT_LINKED to upstream when all the downstream pads have received GST_FLOW_NOT_LINKED. In addition to that, in order to handle pads being relinked downstream, the flow combiner should be reset in response to RECONFIGURE events. This ensures that a both srcpads process a chain operation before a GST_FLOW_NOT_LINKED can be propagated upstream (which would usually stop the pipeline). Otherwise, in a configuration with two srcpads, only one linked at a time, after the relink the element could chain data through the now unlinked pad and the flow combiner would resolve as GST_FLOW_NOT_LINKED (stopping the pipeline) just because the now linked pad has not been chained yet to update the flow combiner. This patch adds handling of RECONFIGURE events to qtdemux. Also, since this event handling causes the flow combiner to be used from a thread other than the qtdemux streaming thread, usages of the flow combiner has been guarded by the object lock. 2020-01-07 01:20:24 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix assertion failure on set_property() GValue might have null object. (gst-inspect-1.0:10304): GStreamer-CRITICAL ... gst_object_ref_sink: assertion 'object != NULL' failed 2020-01-03 15:16:02 +0100 Daniel Molkentin * gst/videocrop/gstvideocrop.c: videocrop: allow properties to be animated by GstController 2019-12-24 08:24:51 -0500 Aaron Boxer * gst/rtsp/gstrtspsrc.c: rtspsrc: improved handling of control concatenation with base Also, `control_url` variable has been renamed to `control_path`, as it is actually a path. 2019-12-06 12:34:15 -0500 Aaron Boxer * gst/rtsp/gstrtspsrc.c: rtspsrc: append aggregate control string to base URL before query string Appending control string to end of query changes meaning of query string Fixes #650 2019-12-28 23:01:19 +0000 Eric Marks * ext/aalib/gstaasink.c: * ext/aalib/gstaatv.c: * ext/aalib/gstaatv.h: * ext/aalib/meson.build: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacatv.c: * ext/libcaca/gstcacatv.h: * ext/libcaca/meson.build: aasink & cacasink: add filter aatv & cacatv Add transform filter capabilities to aasink and cacasink in the form of new elements aatv and cacatv. 2019-06-06 11:03:34 +0200 Niels De Graef * gst/alpha/gstalpha.h: * gst/alpha/gstalphacolor.h: alpha: Cleanup using G_DECLARE_FINAL_TYPE We started depending on GLib 2.44, so we can clean up all the GObject boilerplate macros. 2019-12-18 16:07:18 +0100 Stéphane Cerveau * ext/shout2/gstshout2.c: * gst/multipart/multipartmux.c: * sys/ximage/gstximagesrc.c: good: use of g_value_dup_string Use helper method to get string from GValue. 2019-12-19 23:48:09 +0100 Havard Graff * gst/rtpmanager/gstrtpbin.c: * tests/check/elements/rtpbin.c: rtpbin: fix shutdown crash in rtpbin The key is to make sure the jitterbuffer is set to NULL *before* the ptdemux. The race that existed would basically happen when ptdemux had reached READY, and the jitterbuffer would then push a buffer, triggering a new pad with a new payloadtype being added and ghosted to the rtpbin itself. However, the srcpad of the ptdemux would now be inactive, and all the sticky-event pushed on it would be swallowed, not allowing any to reach the ghost-pad. Then the buffer in-flight would come to the ghostpad, and we would assert that a buffer arrived before the necessary events. By simply re-ordering the state-changes, we ensure that there will be no buffer racing into the ptdemux while its state is being changed, and the problem disappears completely. Notice also that there is not point in disconnecting the signals on the ptdemux before this point, since we need the push-thread to settle down before we can do this in a non-racy way. 2019-09-12 14:22:10 -0600 Aaron Boxer * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid seek DISCONT when only rate changes in same direction Not setting DISCONT avoids a noticable delay when seeking with only rate changing, in the same direction as current rate. 2019-12-10 18:13:11 -0500 Olivier Crête * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Remove deprecated GTimeVal GTimeVal won't work past 2038 2019-12-10 17:13:45 -0500 Olivier Crête * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: Remove deprecated GTimeVal 2019-12-18 12:19:27 +0200 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Add support for S24LE and S32LE raw audio avidemux already handles this correctly. 2019-12-16 21:07:08 +0200 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Allow muxing v210 video into AVI avidemux already handles this. 2019-12-16 18:43:44 +0200 Vivia Nikolaidou * gst/flv/gstflvdemux.c: flvdemux: Don't replace video codec data when we receive a PAR Receiving a pixel-aspect-ratio should trigger a caps change, but not replace the existing video codec tag 2019-12-12 20:20:35 +0100 Mathieu Duponchelle * gst/isomp4/gstqtmux.c: qtmux: protect access to GstElement.sinkpads 2019-12-03 15:30:06 +0100 Mathieu Duponchelle * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * tests/check/elements/qtmux.c: qtmux: port to GstAggregator 2019-12-16 13:03:51 +0100 Joakim Johansson * gst/rtsp/gstrtspsrc.c: gstrtspsrc: Add missing lock on free set_get_param_q Otherwise is it possible to get a crash in gst_rtspsrc_set_parameter. 2019-12-12 18:53:00 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Increment fragment_id even if no fragment location was provided Applications might handle locations and generally configuration of the sink by themselves instead of having splitmuxsink set the location on the sink. Nonetheless it makes sense to increment the fragment_id that is passed to the signal so that applications know which fragment is requested. 2019-12-12 10:59:35 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Use the last DTS for the metadata timestamp This avoids creating a timestamp regression during a stream. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/429 2019-12-11 17:30:50 +0100 Mathieu Duponchelle * gst/isomp4/qtdemux.c: qtdemux: send GAP events for lagging audio and video streams too The logic is taken straight from matroskademux, see 77403d0afee635f2de6c2e53a23e1f50ad0d00fa 2019-12-10 23:48:35 +0900 Seungha Yang * gst/flv/gstflvmux.c: * meson.build: flvmux: Use thread-safe gmtime_r if available gmtime on *nix is not thread-safe. 2019-12-05 14:58:40 +0000 Stéphane Cerveau * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: provides a start-index property Allow to change the fragment-id start index. 2019-12-03 11:36:07 +0100 Philipp Zabel * ext/qt/meson.build: qmlglsink: fix build on EGL platform without X11 headers If Mesa is built without X11 headers, building against Mesa EGL headers requires a dependency on egl.pc, to define MESA_EGL_NO_X11_HEADERS. This fixes a build error when compiling ext/qt/gstqtglutility.cc: In file included from /usr/include/EGL/egl.h:39, from /usr/include/gstreamer-1.0/gst/gl/egl/gstegl.h:44, from ../gst-plugins-good-1.16.1/ext/qt/gstqtglutility.cc:43: /usr/include/EGL/eglplatform.h:124:10: fatal error: X11/Xlib.h: No such file or directory 2019-12-04 01:03:49 +0000 Tim-Philipp Müller * gst/rtp/gstrtpjpegdepay.c: rtpjpegdepay: outputs framed jpeg Add parsed=true to output caps, as we always output whole frames, timestamped and all. Means also that the output can be decoded by avdec_mjpeg wihout plugging an extra parser (which has no rank). 2019-12-03 13:47:22 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Correct metadata handling in file and stream mode In file mode, only push one onMetaData at the start of the stream. In stream mode, always push complete onMetaData. They get replaced, not merged. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418 2019-12-03 13:46:09 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Don't calculate duration in streamable mode There's no header to rewrite, so the duration is left unused. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418 2016-11-30 15:55:01 +0100 Havard Graff * gst/rtp/gstrtpL16depay.c: rtpL16depay: don't crash if data is not modulo channels*width 2019-12-02 19:00:45 +0000 Tim-Philipp Müller * meson.build: * pkgconfig/gstreamer-plugins-good-uninstalled.pc.in: * pkgconfig/meson.build: pkgconfig: remove gst-plugins-good-1.0-uninstalled.pc This was never installed and it was only used by the uninstalled autotools dev environment to locate the -good plugins for use in unit tests in gstreamer modules higher up the stack. It is no longer needed now that we no longer have an autotools build. 2017-10-10 15:45:28 +0200 Håvard Graff * pkgconfig/meson.build: meson.build: use join_paths() on prefix So that "/" are correct on Windows. 2017-06-30 09:48:58 +0200 Havard Graff * gst/rtp/gstrtpopuspay.c: rtpopuspay: use baseclass allocator for buffers That way we get some of the meta -> rtp-extension goodies. 2019-11-29 20:46:26 +0900 Seungha Yang * ext/vpx/gstvp9dec.c: vp9dec: Fix broken 4:4:4 8bits decoding VPX_IMG_FMT_I444 pixel format with sRGB colorspace means GBR data. Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/651 2019-10-18 17:45:43 +0200 Havard Graff * tests/check/elements/rtpsession.c: rtpsession: add test for requesting FIR after having requested PLI 2019-11-26 15:00:18 +0100 Havard Graff * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: make test more stable 2019-11-29 14:23:49 +0100 Havard Graff * gst/rtpmanager/gstrtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: add locking for clear-pt-map ...or it will segfault from time to time... 2018-05-31 10:29:43 +0200 Linus Svensson * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska-read-common.c: matroskamux: Add property to set DateUTC Add a property that makes it possible for an application to set the DateUTC header field in matroska files. This is useful for live feeds, where the DateUTC header can be set to a UTC timestamp, matching the beginning of the file. Needs gstreamer!323 Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/481 2018-05-31 11:20:36 +0200 Linus Svensson * gst/matroska/ebml-ids.h: * gst/matroska/ebml-read.c: * gst/matroska/ebml-write.c: * gst/matroska/matroska-mux.c: matroskamux: Use nanosecond precision for DateUTC DateUTC is specified with nanosecond precision in matroska, make use of that. 2018-10-17 02:28:13 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Queue number of allocated buffers to capture Before we do streamon, we queue all capture buffers by calling resurrect. When the driver supports CREATE_BUFS, this would lead to buffers being allocated till the maximum of 32 is reached. Instead, we now save the number of allocated buffers and queue this amount. 2019-11-19 14:23:48 +0100 Jan Alexander Steffens (heftig) * gst/matroska/matroska-mux.c: matroskamux: Pass the right size to gst_collect_pads_add_pad We were lucky that GstMatroskamuxPad is larger than GstMatroskaPad. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/393 2019-11-18 13:27:42 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Workaround bad TRY_FMT colorimetry implementation libv4l2 reset the colorpace to 0 and does not do any request to the driver. This yields an invalid colorspace which currently cause a negotiation failure. This workaround by ignoring bad values during the TRY_FMT step. 2019-11-04 17:18:30 +0800 aogun * gst/audioparsers/gstaacparse.c: aacparse: fix wrong offset of adts channel 2019-10-07 12:45:00 +0900 Seungha Yang * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't take lock during posting message An application might try to access splitmuxsink from sync message handler by g_object_{get,set} which takes lock also. In general, we don't take lock around message handler. 2019-09-12 15:21:24 -0400 Scott Kanowitz * ext/jpeg/gstjpegdec.c: jpegdec: Fix incorrect logic in EOI tag detection This change fixes the reversed logic in the EOI tag detection code. 2019-08-26 08:03:24 +0200 Niels De Graef * ext/cairo/gstcairooverlay.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: * gst/rtp/rtpstorage.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpsession.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * sys/v4l2/tuner.c: * sys/v4l2/tunerchannel.c: Don't pass default GLib marshallers for signals By passing `NULL` to `g_signal_new` instead of a marshaller, GLib will actually internally optimize the signal (if the marshaller is available in GLib itself) by also setting the valist marshaller. This makes the signal emission a bit more performant than the regular marshalling, which still needs to box into `GValue` and call libffi in case of a generic marshaller. Note that for custom marshallers, one would use `g_signal_set_va_marshaller()` with the valist marshaller instead. 2019-11-14 17:33:08 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Check the exit condition after executing timers The do_expected_timeout() function may release the JBUF_LOCK, so we need to check if nothing wanted the timer thread to exit after this call. The side effect was that we may endup going back into waiting for a timer which will cause arbitrary delay on tear down (or deadlock when test clock is used). Fixes #653 2019-11-14 17:20:51 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Check exit condition immediately after JBUF_WAIT JBUF_WAIT_QUEUE drops the JBUF_LOCK, which means the stop condition for the chain function may have changed (change_state to NULL). Check this immediately after the wait so that we don't delay shutting down. 2019-11-12 17:28:22 -0500 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: videocrop: Also update the coordinate when in-place This update is needed when the output caps is not changed (e.g. we are moving a viewport around). Fixes #669 2019-11-11 13:19:08 -0500 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: videocrop: Don't always re-run the allocation query When in-place, running an allocation is not useful since videocrop is not implicated in the allocation. So only force the allocation query for the case it was in passthrough. This is needed since the change in the crop region will likely pull us out of this mode. For the case we where neither in passthrough or in-place, the allocation query is already ran by the baseclass, so nothing special is needed. This fixes performance issues when changing the crop region per frame. This was reproduced using videocrop2-test. 2019-11-11 13:18:52 -0500 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: videocrop: Cleanup spurious assignment These are just writing the same thing a second time. 2018-11-07 09:00:02 +0100 Michael Olbrich * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: don't overwrite the last valid line If the the height is not a multiple of the macro block size then the memory of the last line is reused for all extra lines. This is no problem if the last line is duplicated properly. However, if the extra lines are not initialized properly during encoding, then the last visible line is overwritten with undefined data. Use a extra buffer to avoid this problem. 2019-11-07 12:28:58 +0100 Stéphane Cerveau * gst/multifile/gstsplitmuxsink.c: splitmuxsink: add fakesink support fakesink does not support "location" property and was generating a warning. 2018-12-12 19:07:39 +0300 Sergey Nazaryev * gst/udp/gstmultiudpsink.c: multiudpsink: don't lose scope_id 2019-11-05 21:41:55 +0530 Nirbheek Chauhan * ext/vpx/meson.build: vpx: Error out if enabled and no features found Seee: https://gitlab.freedesktop.org/gstreamer/cerbero/issues/200 2019-05-25 21:19:21 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2object: update match_buffer_layout() debug messages It's no longer used only to try importing buffers. 2019-05-23 10:49:39 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2object: try matching buffer layout from downstream Ask v4l2 to produce buffers matching the buffer layout requested downstream. 2019-05-21 10:31:46 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2object: factor out gst_v4l2_object_match_buffer_layout() No semantic change. 2019-10-20 12:17:25 +0200 Havard Graff * gst/rtpmanager/rtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: make sure not to drop packets based on skew One of the jitterbuffers functions is to try and make sense of weird network behavior. It is quite unhelpful for the jitterbuffer to start dropping packets itself when what you are trying to achieve is better network resilience. In the case of a skew, this could often mean the sender has restarted in some fashion, and then dropping the very first buffer of this "new" stream could often mean missing valuable information, like in the case of video and I-frames. This patch simply reverts back to the old behavior, prior to https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/commit/8d955fc32b552b2db933c67f3cfa31d987f36b81 and includes the simplest test I could write to demonstrate the behavior, where a single packet arrives "perfectly", then a 50ms gap happens, and then two more packets arrive in perfect order after that. # Conflicts: # tests/check/elements/rtpjitterbuffer.c 2019-04-17 12:40:22 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2transform.c: v4l2transform: use alignments from upstream when importing on sink Try configuring the v4l2 output with the alignments from upstream when importing its buffers. This allows us to support importing with non-standard strides and/or heights if supported by the driver. 2019-04-17 12:25:14 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2object: add support for vertical padding when importing buffers We were already supporting horizontal padding by setting bytesperline to the buffer stride but not vertical one. We are now updating the format height with the padded height and crop to the actual video resolution if needed. 2019-04-17 11:46:10 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2object: fix debug message if driver rejects stride The 'want' and 'got' strides were inversed. 2019-04-15 11:43:41 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: improve logs when importing buffers Log strides and offsets from upstream. Also fix a typo. 2019-10-29 14:05:48 +0000 James Cowgill * sys/v4l2/gstv4l2videodec.c: v4l2videodec: ensure pool exists before orphaning it In commit e2ff87732d0b ("v4l2videodec: support orphaning") support for orphaning the capture buffer pool was added when the format is renegotiated. However, the commit forgot to check that a pool existed before doing this. This is needed because it's possible for the format to be renegotiated before a capture pool is allocated, which would result in trying to orphan a NULL pool and lead to a NULL pointer dereference. Fix this by checking a pool exists first. If the pool doesn't exist, there are no buffers to be reclaimed, so skip the allocation query in that case. 2019-10-25 22:03:18 +1100 Matthew Waters * ext/qt/qtwindow.cc: qmlglsrc: read from the back buffer when use-default-fbo = TRUE glReadBuffer(GL_COLOR_ATTACHMENT0) on the default framebuffer (0) is invalid GL API usage and would result in a GL error being thrown. 2019-10-25 21:47:01 +1100 Matthew Waters * ext/qt/gstqtsrc.cc: qmlglsrc: fix vertical flip matrix Some time ago libgstgl defined the majorness of matrices it uses. The majorness used by qmlglsrc was incompatible with the libgstgl. 2019-07-30 12:07:18 +0200 Patricia Muscalu * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Fix memory leak while pushing fragmented data The memory leak occurs in the case when the buffer has been added to the fragment_buffers array of the current pad and never been sent because of the push failure of the previous buffers: moof or mdat header or fragmented buffer(s). 2019-10-11 14:20:15 +0200 Edward Hervey * gst/debugutils/cpureport.c: * gst/debugutils/cpureport.h: * gst/debugutils/progressreport.c: * gst/debugutils/progressreport.h: * gst/flv/gstflvmux.c: * gst/isomp4/atoms.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-mux.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: * gst/udp/gstmultiudpsink.c: * sys/v4l2/gstv4l2src.c: good: Avoid usage of deprecated API GTimeval and related functions are now deprecated in glib. Replacement APIs have been present since 2.26 2019-07-15 07:46:56 +0200 Javier Celaya * sys/osxaudio/meson.build: osxaudio: misspelled dependency When building osxaudio, the required 'AudioToolbox' dependency is misspelled as 'AudioToolBox', which crashes the build with error: ld: framework not found AudioToolBox 2019-06-09 00:43:00 +0100 Tim-Philipp Müller * .gitignore: * .gitmodules: * Makefile.am: * README: * autogen.sh: * common: * configure.ac: * docs/.gitignore: * ext/Makefile.am: * ext/aalib/Makefile.am: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/flac/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/gtk/Makefile.am: * ext/jack/.gitignore: * ext/jack/Makefile.am: * ext/jpeg/Makefile.am: * ext/lame/Makefile.am: * ext/libcaca/Makefile.am: * ext/libpng/Makefile.am: * ext/mpg123/Makefile.am: * ext/pulse/Makefile.am: * ext/qt/.gitignore: * ext/qt/Makefile.am: * ext/raw1394/.gitignore: * ext/raw1394/Makefile.am: * ext/shout2/Makefile.am: * ext/soup/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/.gitignore: * ext/taglib/Makefile.am: * ext/twolame/Makefile.am: * ext/vpx/Makefile.am: * ext/wavpack/Makefile.am: * gst/Makefile.am: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/.gitignore: * gst/audiofx/Makefile.am: * gst/audioparsers/Makefile.am: * gst/auparse/.gitignore: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/.gitignore: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debugutils/Makefile.am: * gst/deinterlace/Makefile.am: * gst/dtmf/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/.gitignore: * gst/equalizer/Makefile.am: * gst/flv/Makefile.am: * gst/flx/Makefile.am: * gst/goom/.gitignore: * gst/goom/Makefile.am: * gst/goom2k1/.gitignore: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/imagefreeze/Makefile.am: * gst/interleave/Makefile.am: * gst/isomp4/Makefile.am: * gst/law/Makefile.am: * gst/level/.gitignore: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/monoscope/.gitignore: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/rtsp/.gitignore: * gst/rtsp/Makefile.am: * gst/shapewipe/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/.gitignore: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/.gitignore: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/.gitignore: * gst/wavparse/Makefile.am: * gst/y4m/Makefile.am: * m4/.gitignore: * m4/Makefile.am: * m4/README: * m4/a52.m4: * m4/aalib.m4: * m4/as-ffmpeg.m4: * m4/as-liblame.m4: * m4/as-slurp-ffmpeg.m4: * m4/check-libheader.m4: * m4/freetype2.m4: * m4/glib.m4: * m4/gst-alsa.m4: * m4/gst-artsc.m4: * m4/gst-fionread.m4: * m4/gst-ivorbis.m4: * m4/gst-matroska.m4: * m4/gst-sdl.m4: * m4/gst-shout2.m4: * m4/gst-sid.m4: * m4/gtk.m4: * m4/libfame.m4: * m4/ogg.m4: * m4/vorbis.m4: * pkgconfig/.gitignore: * pkgconfig/Makefile.am: * po/.gitignore: * po/Makevars: * po/POTFILES: * sys/Makefile.am: * sys/directsound/Makefile.am: * sys/oss/.gitignore: * sys/oss/Makefile.am: * sys/oss4/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxvideo/Makefile.am: * sys/v4l2/Makefile.am: * sys/waveform/Makefile.am: * sys/ximage/Makefile.am: * tests/Makefile.am: * tests/check/.gitignore: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/generic/.gitignore: * tests/check/pipelines/.gitignore: * tests/examples/Makefile.am: * tests/examples/audiofx/.gitignore: * tests/examples/audiofx/Makefile.am: * tests/examples/cairo/.gitignore: * tests/examples/cairo/Makefile.am: * tests/examples/equalizer/.gitignore: * tests/examples/equalizer/Makefile.am: * tests/examples/gtk/.gitignore: * tests/examples/gtk/Makefile.am: * tests/examples/jack/Makefile.am: * tests/examples/level/.gitignore: * tests/examples/level/Makefile.am: * tests/examples/qt/qmlsink/.gitignore: * tests/examples/qt/qmlsrc/.gitignore: * tests/examples/rtp/.gitignore: * tests/examples/rtp/Makefile.am: * tests/examples/rtsp/Makefile.am: * tests/examples/shapewipe/.gitignore: * tests/examples/shapewipe/Makefile.am: * tests/examples/spectrum/.gitignore: * tests/examples/spectrum/Makefile.am: * tests/examples/v4l2/.gitignore: * tests/examples/v4l2/Makefile.am: * tests/files/Makefile.am: * tests/icles/.gitignore: * tests/icles/Makefile.am: Remove autotools build system 2019-10-13 12:46:58 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: fix wrong type cast Follow-up to commit 1b752c0f !361 2019-09-25 12:36:32 +0000 HuQian * sys/v4l2/gstv4l2object.c: is a typo here? gstv4l2object.c 2019-10-11 12:27:12 +0000 Kevin Song * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Check stop in flush() to avoid race condition. Backward playback will drain and flush every frame. Stop playback when backward playback have race condition between exit thread and streaming thread flush. Add one check to avoid it. Fixes #639 2019-10-11 10:33:20 +0800 Fuwei Tang * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: fix type conversion errors 2019-09-02 08:27:35 -0400 Aaron Boxer * NEWS: * docs/gst_plugins_cache.json: * ext/dv/gstdvdemux.c: * ext/flac/gstflactag.c: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gtk/gstgtkbasesink.c: * ext/jack/gstjackaudioclient.c: * ext/jpeg/Makefile.am: * ext/pulse/pulsesink.c: * ext/qt/qtwindow.cc: * ext/raw1394/gstdv1394src.h: * ext/taglib/gstid3v2mux.cc: * ext/wavpack/gstwavpackenc.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: * gst/audiofx/gstscaletempo.c: * gst/audiofx/gstscaletempoplugin.c: * gst/autodetect/gstautodetect.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/tvtime/sse.h: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopBottom.inc: * gst/deinterlace/tvtime/tomsmocomp/StrangeBob.inc: * gst/deinterlace/tvtime/tomsmocomp/WierdBob.inc: * gst/deinterlace/tvtime/vfir.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/flv/gstflvdemux.c: * gst/flv/gstindex.c: * gst/interleave/deinterleave.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_lang.c: * gst/level/gstlevel.c: * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/monoscope/monoscope.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/patternspec.c: * gst/replaygain/replaygain.h: * gst/rtp/README: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/rtpstorage.c: * gst/rtp/rtpulpfeccommon.c: * gst/rtp/rtpulpfeccommon.h: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtsp/README: * gst/rtsp/gstrtspsrc.c: * gst/spectrum/gstspectrum.h: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/README: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * hooks/pre-commit.hook: * m4/aalib.m4: * m4/freetype2.m4: * m4/glib.m4: * m4/gst-fionread.m4: * m4/gst-matroska.m4: * m4/gst-sdl.m4: * m4/gst-shout2.m4: * m4/gtk.m4: * m4/libfame.m4: * m4/ogg.m4: * m4/vorbis.m4: * sys/oss4/oss4-audio.c: * sys/oss4/oss4-soundcard.h: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxvideo/osxvideosink.m: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/v4l2_calls.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: * sys/ximage/ximageutil.h: * tests/check/elements/jpegdec.c: * tests/check/elements/level.c: * tests/check/elements/qtmux.c: * tests/check/elements/rgvolume.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/rtpred.c: * tests/check/elements/rtprtx.c: * tests/check/elements/rtpsession.c: * tests/check/elements/rtpstorage.c: * tests/check/elements/splitmux.c: * tests/check/pipelines/simple-launch-lines.c: * tests/examples/cairo/cairo_overlay.c: * tests/examples/gtk/glliveshader.c: * tests/examples/rtp/client-rtpaux.c: * tests/examples/v4l2/camctrl.c: documentation: fix a number of typos 2019-10-04 20:31:56 +0000 Simon Arnling Bååth * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: gstrtpjitterbuffer: Custom messages when dropping packets This commit adds custom element messages for when gstrtpjitterbuffer drops an incoming rtp packets due to for example arriving too late. Applications can listen to these messages on the bus which enables actions to be taken when packets are dropped due to for example high network jitter. Two properties has been added, one to enable posting drop messages and one to set a minimum time between each message to enable throttling the posting of messages as high drop rates. 2019-09-03 16:46:30 -0400 Thibault Saunier * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Specify REDIRECT information in error message There are in the wild (mp4) streams that basically contain no tracks but do have a redirect info[0], in which case, we won't be able to expose any pad (there are no tracks) so we can't post anything but an error on the bus, as: - it can't send EOS downstream, it has no pad, - posting an EOS message will be useless as PAUSED state can't be reached and there is no sink in the pipeline meaning GstBin will simply ignore it The approach here is to to add details to the ERROR message with a `redirect-location` field which elements like playbin handle and use right away. [0]: http://movietrailers.apple.com/movies/paramount/terminator-dark-fate/terminator-dark-fate-trailer-2_480p.mov 2019-09-26 18:39:48 -0400 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Cancel timers instead of just unlocking loop thread When the queue is full (and adding more packets would risk a seqnum roll-over), the best approach is to just start pushing out packets from the other side. Just pushing out the packets results in the timers being left hanging with old seqnums, so it's safer to just execute them immediately in this case. It does limit the timer space to the time it takes to receiver about 32k packets, but without extended sequence number, this is the best RTP can do. This also results in the test no longer needed to have timeouts or timers as pushing packets in drives everything. Fixes #619 2019-09-27 14:04:28 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Optimize offset update As we are applying the same offset over all timers, there timer ordering won't change, so we can safely skip time-reordering. 2019-09-27 16:21:22 -0400 Nicolas Dufresne * gst/rtpmanager/rtptimerqueue.c: rtptimerqueue: Optimize reschedule optations This basically add ability to choose between inserting from head, tail or in-place in order to try and minimize the distance to walk through in the timer queue. This removes an overhead we had seen on high drop rate. 2019-09-27 14:04:03 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix a typo in comment 2019-07-02 15:52:25 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Don't use stats timer on the timers queue The timer passed to update_timers may be from the stats timer. At the moment, we could endup rescheduling (reusing) that timer onto the normal timer queue, unschedul it as if it was from the normal timer queue or duplicate it into the stats timer queue again. This was protected before as the with the fact the stats timer didn't have a valid idx. 2019-06-21 14:08:26 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Update timers on ts-offset changes As the offset is already applied now, we need to update and reschedule all timers each time the offset is changed. I'm not sure who expect this to be retro-actively applied, but there was a unit test for it. 2019-06-20 15:59:48 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: No need to wake the timer thread on head changes If the jitterbuffer head change, there is no need to systematically wakeup the timer thread. The timer thread will be waken up on if an earlier timeout has been pushed. This prevent some more spurious wakeup when the system is loaded. As a side effect, cranking the clock may set the clock at an earlier position. 2019-06-18 19:07:29 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtptimerqueue.h: rtpjittterbuffer: Port timers array to RtpTimerQueue In this patch we now make use of the new RtpTimerQueue instead of the old GArray. This required a lot of changes all over the place, some of the important changes are that `timer->timeout` is no longer a PTS but the actual timeout. This was required to get the RtpTimerQueue sorting right. The applied offset is saved as `timer->offset`, this allow retreiving back the PTS when needed. The clockid updates only happens once per incoming packet. If the currently schedule timer is before the earliest timer in the queue, we no longer wakeup the thread. This way, if other timers get setup in the meantime, this will reduce the number of wakup. The timer loop code has been mostly rewritten, though the behaviour of running the lost timers first has been kept (even though there is no test to show what would be the side effect of doing this differently). Fixes #608 2019-06-14 14:29:36 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjittterbuffer: Port from TimerQueue to RtpTimerQueue 2019-06-13 17:08:31 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtptimerqueue.h: rtpjitterbuffer: Port use the new RtpTimer structure First iteration toward porting to the new timer queue. 2019-06-12 09:59:31 -0400 Nicolas Dufresne * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/meson.build: * gst/rtpmanager/rtptimerqueue.c: * gst/rtpmanager/rtptimerqueue.h: * tests/check/Makefile.am: * tests/check/elements/rtpjitterbuffer.c: * tests/check/meson.build: rtptimerqueue: Consolidate a data structure for timers Implement a single timer queue for all timers. The goal is to always use ordered queues for storing timers. This way, extracting timers for execution becomes O(1). This also allow separating the clock wait scheduling from the timer itself and ensure that we only wake up the timer thread when strictly needed. The knew data structure is still O(n) on insertions and reschedule, but we now use proximity optimization so that normal cases should be really fast. The GList structure is also embeded intot he RtpTimer structure to reduce the number of allocations. 2019-06-10 16:46:05 -0400 Nicolas Dufresne * tests/check/elements/rtpjitterbuffer.c: tests: jitterbuffer: Demacroify some helpers There is no reason for these to be macros anymore. This makes the test helper much more readable. 2019-06-06 14:44:27 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: Move item structure outside of the element This moves the RtpJitterBufferStructure type, alloc, free into rtpjitterbuffer.c/h implementation. jitterbuffer.c strictly rely on the fact this structure is compatible with GList, and so it make more sense to keep encapsulate it. Also, anything that could possibly reduce the amount of code in the element is a win. In order to support that move, a function pointer to free the data was added. This also allow making the free function option when flushing the jitterbuffer. 2019-06-06 13:09:29 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Constify timer pointers where possible This helps understanding which function modify the Timerdata and which one does not. This is not always obvious from thelper name considering recalculate_timer() does not. 2019-09-27 08:46:22 +0200 Philipp Zabel * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2mpeg2codec.c: * sys/v4l2/gstv4l2mpeg2codec.h: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/meson.build: v4l2: Add MPEG-2 profile and level support Add support for V4L2 MPEG-2 decoders reporting supported profiles and levels. 2019-09-23 14:34:20 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: add support for ABGR, xBGR, RGBA, and RGBx formats Map them to the new V4L2_PIX_FMT_{BGRA32,BGRX32,RGBA32,RGBX32} pixel formats. 2019-09-23 14:10:15 +0200 Philipp Zabel * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: v4l2: update kernel headers to latest from media tree Update to the latest installed headers (output of make headers_install) from the media tree, keeping the slight modifications to the includes. This includes typo fixes in enum v4l2_mpeg_video_multi_slice_mode, MPEG-2 level and profile enums, new FWHT and H.264 Qp controls, new RGB(A) formats, and new continuous bytestream and dynamic resolution format flags. 2017-12-19 18:23:16 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: add request-jitterbuffer signal This can be used to pass the threadsharing jitterbuffer from gst-plugins-rs for example. 2019-09-23 18:46:16 +1000 Matthew Waters * gst/isomp4/qtdemux.c: build: fix werror build with newer gcc In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:55, from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/tag/tag.h:25, from ../gst/isomp4/qtdemux.c:56: In function ‘qtdemux_inspect_transformation_matrix’, inlined from ‘qtdemux_parse_trak’ at ../gst/isomp4/qtdemux.c:10676:5, inlined from ‘qtdemux_parse_tree’ at ../gst/isomp4/qtdemux.c:14210:5: ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstinfo.h:645:5: error: ‘%s’ directive argument is null [-Werror=format-overflow=] 645 | gst_debug_log ((cat), (level), __FILE__, GST_FUNCTION, __LINE__, \ | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 646 | (GObject *) (object), __VA_ARGS__); \ | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstinfo.h:1062:35: note: in expansion of macro ‘GST_CAT_LEVEL_LOG’ 1062 | #define GST_DEBUG_OBJECT(obj,...) GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_DEBUG, obj, __VA_ARGS__) | ^~~~~~~~~~~~~~~~~ ../gst/isomp4/qtdemux.c:10294:5: note: in expansion of macro ‘GST_DEBUG_OBJECT’ 10294 | GST_DEBUG_OBJECT (qtdemux, "Transformation matrix rotation %s", | ^~~~~~~~~~~~~~~~ ../gst/isomp4/qtdemux.c: In function ‘qtdemux_parse_tree’: ../gst/isomp4/qtdemux.c:10294:64: note: format string is defined here 10294 | GST_DEBUG_OBJECT (qtdemux, "Transformation matrix rotation %s", | ^~ 2019-09-18 18:31:27 +0300 Sebastian Dröge * gst/isomp4/atoms.c: qtmux: Use the new helper functions for mapping the colr atom values to colorimetry 2019-09-18 18:29:27 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Use the new helper functions for mapping the colr atom values to colorimetry 2019-09-10 22:44:20 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: docs: update plugin cache 2019-09-10 22:43:49 +0200 Mathieu Duponchelle * gst/smpte/barboxwipes.c: smpte: don't register transition types twice 2019-09-08 20:43:17 -0400 Doug Nazar * gst/alpha/gstalpha.c: alpha: Fix one_over_kc calculation On arm/aarch64, converting from float directly to unsigned int uses a different opcode and negative numbers result in 0. Cast to signed int first. 2019-07-31 16:17:36 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmux: Add muxer-pad-map property Add a property which explicitly maps splitmuxsink pads to the muxer pads they should connect to, overriding the implicit logic that tries to match pads but yields arbitrary names. 2019-07-26 02:21:59 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: In async mode, retain previous muxer pad names. When running in async-finalize mode, request new pads from the muxer using the same names as old pads, instead of letting the muxer assign new ones based on the pad template name. 2019-07-26 02:13:31 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Mark split-* signals as action signals. Doc fixes. Add the G_SIGNAL_ACTION flag to the split-* signals on splitmuxsink, and make some improvements to their docstrings 2019-08-29 22:11:02 +0900 Seungha Yang * gst/isomp4/gstqtmux.c: qtmux: Fix incompatible type warning with MSVC gstqtmux.c(5582): warning C4133: 'function': incompatible types - from 'GstVideoMultiviewFlags *' to 'guint *' 2019-09-02 16:33:05 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: fix git diff indentation 2019-08-30 22:42:58 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: normalize variable to boolean 2019-08-29 21:29:34 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: clip output segment on accurate seeks The output segment is only used in ONVIF mode. The previous behaviour was to output a segment computed from the Range response sent by the server. In ONVIF mode, servers will start serving from the appropriate synchronization point (keyframe), and the Range in response will start at that position. This means rtspsrc can now perform truly accurate seeks in that mode, by clipping the output segment to the values requested in the seek. The decoder will then discard out of segment buffers and playback will start without artefacts at the exact requested position, similar to the behaviour of a demuxer when an accurate seek is requested. 2019-08-30 14:00:26 +1000 Matthew Waters * ext/vpx/gstvpxenc.c: vpx: fix macos werror build ../ext/vpx/gstvpxenc.c:1723:49: error: format specifies type 'long' but the argument has type 'vpx_codec_pts_t' (aka 'long long') [-Werror,-Wformat] ", gst frame pts: %" G_GINT64_FORMAT, pkt->data.frame.pts, pts); ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~ /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:1065:96: note: expanded from macro 'GST_TRACE_OBJECT' #define GST_TRACE_OBJECT(obj,...) GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_TRACE, obj, __VA_ARGS__) ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~ /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:646:31: note: expanded from macro 'GST_CAT_LEVEL_LOG' (GObject *) (object), __VA_ARGS__); \ ^~~~~~~~~~~ ../ext/vpx/gstvpxenc.c:1723:70: error: format specifies type 'long' but the argument has type 'vpx_codec_pts_t' (aka 'long long') [-Werror,-Wformat] ", gst frame pts: %" G_GINT64_FORMAT, pkt->data.frame.pts, pts); ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~ /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:1065:96: note: expanded from macro 'GST_TRACE_OBJECT' #define GST_TRACE_OBJECT(obj,...) GST_CAT_LEVEL_LOG (GST_CAT_DEFAULT, GST_LEVEL_TRACE, obj, __VA_ARGS__) ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~ /Library/Frameworks/GStreamer.framework/Versions/1.0/include/gstreamer-1.0/gst/gstinfo.h:646:31: note: expanded from macro 'GST_CAT_LEVEL_LOG' (GObject *) (object), __VA_ARGS__); \ ^~~~~~~~~~~ 2019-08-30 13:37:59 +1000 Matthew Waters * sys/osxvideo/cocoawindow.m: osxvideosink: call superclass in reshape Fixes macos werror build ../sys/osxvideo/cocoawindow.m:437:1: error: method possibly missing a [super reshape] call [-Werror,-Wobjc-missing-super-calls] } ^ 2019-08-23 18:56:01 +0200 Mathieu Duponchelle * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/lame/gstlamemp3enc.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/shapewipe/gstshapewipe.c: docstrings: port ulinks to markdown links 2019-08-10 12:33:46 +0100 Tim-Philipp Müller * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: replaygain: fix up doc links to defunct replaygain.org website Fixes #624 2019-08-22 00:18:51 +0900 Seungha Yang * ext/soup/gstsouphttpsrc.c: souphttpsrc: Fix incompatible type build warning gstsouphttpsrc.c(2191): warning C4133: '=': incompatible types - from 'guint (__cdecl *)(GType)' to 'GstURIType (__cdecl *)(GType)' 2019-08-19 11:07:56 +0100 Tim-Philipp Müller * ext/vpx/gstvpxdec.c: * ext/vpx/meson.build: vpx: bump libvpx requirement to 1.5.0 Was released in Nov 2015. 2019-08-19 11:03:00 +0100 Tim-Philipp Müller * ext/vpx/meson.build: vpx: avoid confusing meson configure output when checking for vpx versions Used to print: |Run-time dependency vpx found: YES 1.7.0 |Message: libvpx provides VP8 encoder interface (vpx_codec_vp8_cx_algo) |Message: libvpx provides VP8 decoder interface (vpx_codec_vp8_dx_algo) |Message: libvpx provides VP9 encoder interface (vpx_codec_vp9_cx_algo) |Message: libvpx provides VP9 decoder interface (vpx_codec_vp9_dx_algo) |Dependency vpx found: YES (cached) |Dependency vpx found: NO found '1.7.0' but need: '>=1.8.0' |Run-time dependency vpx found: NO (tried pkgconfig and cmake) We can check the version of the found dep in a way that doesn't produce this confusing output. 2019-08-19 07:30:17 +0000 Amr Mahdi * gst/wavparse/gstwavparse.c: wavparse: Fix push mode ignoring audio with a size smaller than segment buffer In push mode (streaming), if the audio size is smaller than segment buffer size, it would be ignored. This happens because when the plugin receives an EOS signal while a single audio chunk that is less than the segment buffer size is buffered, it does not flush this chunk. The fix is to flush the data chunk when it receives an EOS signal and has a single (first) chunk buffered. How to reproduce: 1. Run gst-launch with tcp source ``` gst-launch-1.0 tcpserversrc port=3000 ! wavparse ignore-length=0 ! audioconvert ! filesink location=bug.wav ``` 2. Send a wav file with unspecified data chunk length (0). Attached a test file ``` cat test.wav | nc localhost 3000 ``` 3. Compare the length of the source file and output file ``` ls -l test.wav bug.wav -rw-rw-r-- 1 amr amr 0 Aug 15 11:07 bug.wav -rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav ``` The expected length of the result of the gst-lauch pipeline should be the same as the test file minus the headers (44), which is ```3564 - 44 = 3520``` but the actual output length is ```0``` After the fix: ``` ls -l test.wav fix.wav -rw-rw-r-- 1 amr amr 3520 Aug 15 11:09 fix.wav -rwxrwxr-x 1 amr amr 3564 Aug 15 11:06 test.wav ``` 2019-08-12 18:56:34 +0300 Sebastian Dröge * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: rtpvp8depay: Add property for waiting until the next keyframe after packet loss If VP8 is not encoded with error resilience enabled then any packet loss causes very bad artefacts when decoding and waiting for the next keyframe instead improves user experience considerably. 2019-08-06 22:27:40 -0400 Nicolas Dufresne * sys/v4l2/ext/types-compat.h: v4l2: Fix type compatibility issue with glibc 2.30 From now on, we will use linux/types.h on Linux, and use typedef of the various flavour of BSD. Fixes #635 2019-08-07 18:29:25 -0400 Mathieu Duponchelle * tests/check/gst-plugins-good.supp: valgrind: suppress Cond error coming from gnutls taken from https://salsa.debian.org/debian/flatpak/commit/fb4a8dda211c4bc036781f2b0d706266e95ce068 2019-07-10 22:07:05 +0300 Mart Raudsepp * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroska: Provide audio lead-in for some lossy formats Various audio formats require an audio lead-in to decode it properly. Most parsers would take care of it, but when a container like matroska is involved, the demuxer handles the seeking and without its own lead-in handling would never even pass the lead-in data to the parser. This commit provides an initial implementation of that for audio/mpeg, audio/x-ac3 and audio/x-eac3 by calculating the worst case lead-in time needed from known samplerate, potential lead-in frames need and the maximum blocksize possible for the format (as we don't parse that out exactly in matroskademux) and seeking that much earlier in case of accurate seeks. This is especially important for NLE use-cases with GES. If accurate seeking to a position that happens to have a video keyframe, it'll go back to the previous keyframe than needed, but with typical video files that's the best we can do anyway without falling back to scanning the clusters, as typically only keyframes are indexed in Cueing Data. If the media doesn't have a CUE, then we bisect for the cluster to seek to with the same modified time as well in case of accurate seeking, ensuring sufficient lead-in. This code path is typically hit only with (suboptimal) audio-only matroska files, e.g. when created with ffmpeg, which doesn't add a CUE for audio-only mkv muxing. 2019-03-11 15:15:12 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for invalid packets in buffer list Upstream elements can send all kinds of data in a buffer list, so cover the case of an invalid RTP packet mixed with valid RTP packets. 2019-03-11 15:12:03 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for multiplexed RTP and RTCP RTP and RTCP packets can be muxed together on the same channel (see RFC5761) and can arrive in the same buffer list. The GStreamer rtpsession element support RFC5761, so add a test to cover this case for buffer lists too. 2019-03-11 15:09:27 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for different timestamps in buffer list Buffers with different timestamps (e.g. packets belonging to different frames) can arrive together in the same buffer list, Add a test to cover this case. 2019-03-12 15:24:26 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add function to check timestamp 2019-04-02 18:02:19 +0200 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test about reordered or duplicated seqnums 2019-04-02 17:52:54 +0200 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for lange jump in seqnums with recovery 2019-04-02 17:50:35 +0200 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for large jump in sequence numbers 2019-04-02 17:47:27 +0200 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for wrapping sequence numbers 2019-03-11 15:07:08 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for permissible gap in sequence numbers 2019-03-11 15:03:31 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for the case of failed probation When a new source fails to pass the probation period (i.e. new packets have non-consecutive sequence numbers), then no buffer shall be pushed downstream. Add a test to validate this case. 2019-03-12 15:23:16 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add function to check sequence number 2019-04-03 14:46:35 +0200 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add test to verify that receiving stats are correct Add a test to verify that stats about received packets are correct when using buffer lists in the rtpsession receive path. Split get_session_source_stats() in two to be able to get stats from a GstRtpSession object directly. 2019-02-27 16:17:57 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add a test for buffer lists on the recv path 2019-02-27 17:03:44 +0100 Antonio Ospite * gst/rtpmanager/gstrtpsession.c: rtpsession: add support for buffer lists on the recv path The send path in rtpsession processes the buffer list along the way, sharing info and stats between packets in the same list, because it assumes that all packets in a buffer list are from the same frame. However, in the receiving path packets can arrive in all sorts of arrangements: - different sources, - different frames (different timestamps), - different types (multiplexed RTP and RTCP, invalid RTP packets). so a more general approach should be used to correctly support buffer lists in the receive path. It turns out that it's simpler and more robust to process buffers individually inside the rtpsession element even if they come in a buffer list, and then reassemble a new buffer list when pushing the buffers downstream. This avoids complicating the existing code to make all functions buffer-list-aware with the risk of introducing regressions, To support buffer lists in the receive path and reduce the "push overhead" in the pipeline, a new private field named processed_list is added to GstRtpSessionPrivate, it is set in the chain_list handler and used in the process_rtp callback; this is to achieve the following: - iterate over the incoming buffer list; - process the packets one by one; - add the valid ones to a new buffer list; - push the new buffer list downstream. The processed_list field is reset before pushing a buffer list to be on the safe side in case a single buffer was to be pushed by upstream at some later point. NOTE: The proposed modifications do not change the behavior of the send path. The process_rtp callback is called in rtpsource.c by the push_rtp callback (via source_push_rtp) only when the source is not internal. So even though push_rtp is also called in the send path, it won't end up using process_rtp in this case because the source would be internal in the send path. The reasoning from above may suggest a future refactoring: push_rtp might be split to better differentiate the send and receive path. 2019-08-07 10:01:34 -0400 Doug Nazar * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-parse.c: matroska: Handle interlaced field order 2019-08-07 12:09:46 +0000 Amr Mahdi * gst/wavparse/gstwavparse.c: wavparse: Fix ignoring of last chunk in push mode In push mode (streaming), if the last audio payload chunk is less than the segment rate buffer size, it would be ignored since the plugin waits until it has at least segment rate bufer size of audio. The fix is to introduce a flushing flag that indicates that no more audio will be available so that the plugin can recognize this condition and flush the data is has even if it is less than the desired segment rate buffer size. 2019-08-06 16:27:37 +0200 Robert Tiemann * ext/soup/gstsouphttpsrc.c: souphttpsrc: Log any error returned by soup_session_send() 2019-08-07 11:42:21 +0900 luke.lin * gst/isomp4/qtdemux.c: qtdemux: enlarge the maximal atom size For 8K content, frame size is over 25MB, and cause the negotiation failure. Enlarge the limitation of QTDEMUX_MAX_ATOM_SIZE to 32MB. 2019-07-27 04:05:01 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: expose and implement is-live property This is useful to support the ONVIF case: when is-live is set to FALSE and onvif-rate-control is no, the client can control the rate of delivery and arrange for the server to block and still keep sending when unblocked, without requiring back and forth PAUSE / PLAY requests. This enables, amongst other things, fast frame stepping on the client side. When is-live is FALSE, we don't use a manager at all. This case was actually already pretty well handled by the current code. The standard manager, rtpbin, is simply no longer needed in this case. Applications can instantiate a downloadbuffer after rtspsrc if needed. 2019-07-27 04:03:44 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: reset_time when flush stopping 2019-07-12 22:33:08 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: expose and implement onvif-mode property Refactor the code for parsing and generating the Range, taking advantage of existing API in GstRtspTimeRange. Only use the TCP protocol in that mode, as per the specification. Generate an accurate segment when in that mode, and signal to the depayloader that it should not generate its own segment, through the "onvif-mode" field in the caps, see for more information. Translate trickmode seek flags to their ONVIF representation Expose an onvif-rate-control property 2019-07-01 20:38:20 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: improve handling of rate in seeks 2019-07-31 21:55:16 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtpfunnel.c: rtpfunnel: forward correct segment when switching pad Forwarding a single segment event from the pad that first gets chained is incorrect: when that first event was sent by an element such as x264enc, with its offset start, we end pushing out of segment buffers for the other pad(s). Instead, everytime the active pad changes, forward the appropriate segment event. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1028 2019-08-05 19:35:36 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Use new GstRTSPMessage API to set message body from a buffer directly 2019-04-04 13:17:34 +0200 Antonio Ospite * gst/rtpmanager/rtpsource.c: rtpsource: fix receiver source stats to consider previously queued packets When it is not clear yet if a packet relative to a source should be pushed, the packet is put into a queue, this happens in two cases: - the source is still in probation; - there is a large jump in seqnum, and it is not clear what the cause is, future packets will help making a guess. In either case stats about received packets are not updated at all; and even if they were, when init_seq() is called it resets all receiver stats, effectively loosing any possible stat about previously received packets. Fix this by taking into account the queued packets and update the stats when calling init_seq(). 2019-04-09 10:46:39 +0200 Antonio Ospite * gst/rtpmanager/rtpsource.c: rtpsource: clarify meaning of the octets-sent and octets-received stats The octets-send and octets-received stats count the payload bytes excluding RTP and lower level headers, clarify that in the documentation. 2019-04-04 13:16:36 +0200 Antonio Ospite * gst/rtpmanager/rtpsource.c: rtpsource: expose field bytes_received in RTPSourceStats Since commit c971d1a9a (rtpsource: refactor bitrate estimation, 2010-03-02) bytes_received filed in RTPSourceStats is set but then never used again, expose it so that it can be used by user code to verify how many bytes have been received. 2019-06-21 17:46:36 +0200 Antonio Ospite * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.h: rtpmanager: consider UDP and IP headers in bandwidth calculation According to RFC3550 lower-level headers should be considered for bandwidth calculation. See https://tools.ietf.org/html/rfc3550#section-6.2 paragraph 4: Bandwidth calculations for control and data traffic include lower-layer transport and network protocols (e.g., UDP and IP) since that is what the resource reservation system would need to know. Fix the source data to accommodate that. Assume UDPv4 over IP for now, this is a simplification but it's good enough for now. While at it define a constant and use that instead of a magic number. NOTE: this change basically reverts the logic of commit 529f443a6 (rtpsource: use payload size to estimate bitrate, 2010-03-02) 2019-08-01 15:02:23 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Use empty-array safe way to cleanup GPtrArray Fix assertion fail GLib-CRITICAL **: g_ptr_array_remove_range: assertion 'index_ < rarray->len' failed 2019-08-01 14:28:04 +0000 Marc Leeman * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: rtpmp4vpay: config-interval -1 send at idr adjust/port from rtph264pay and allow sending the configuration data at every IDR The payloader was stripping the configuration data when the config-interval was set to 0. The code was written in such a way !(a > 0) that it stripped the config when it was set at -1 (send config_data as soon as possible). This resulted in some MPEG4 streams where no GOP/VOP-I was detected to be sent out without configuration. 2019-07-27 14:21:34 -0400 Doug Nazar * gst/matroska/matroska-demux.c: matroskademux: Ignore crc32 element while peeking at cluster. 2019-07-25 21:21:26 +0530 Guillaume Desmottes * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: gtkglsink: fix crash when widget is resized after element destruction Prevent _size_changed_cb() to be called after gtkglsink has been finalized. Fix #632 2019-07-26 02:45:51 +0200 Mathieu Duponchelle * gst/isomp4/qtdemux.c: qtdemux: fix reverse playback EOS conditions In reverse playback, we don't want to rely on the position of the current keyframe to decide a stream is EOS: the last GOP we push will start with a keyframe, which position is likely to be outside of the segment. Instead, let the normal seek_to_previous_keyframe mechanism do its job, it works just fine. 2019-07-23 01:42:02 +0200 Mathieu Duponchelle * gst/isomp4/qtdemux.c: qtdemux: fix key unit seek corner case If a key unit seek is performed with a time position that matches the offset of a keyframe, but not its actual PTS, we need to adjust the segment nevertheless. For example consider the following case: * stream starts with a keyframe at 0 nanosecond, lasting 40 milliseconds * user does a key unit seek at 20 milliseconds * we don't adjust the segment as the time position is "over" a keyframe * we push a segment that starts at 20 milliseconds * we push a buffer with PTS == 0 * an element downstream (eg rtponviftimestamp) tries to calculate the stream time of the buffer, fails to do so and drops it 2019-07-25 15:08:54 +0300 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Don't dereference NULL input state if we have no caps in TIME segments Simply assume that the JPEG frame is not going to be interlaced instead of crashing. 2019-07-22 10:28:50 +0200 Knut Andre Tidemann * gst/rtp/gstrtpopuspay.c: rtp: opuspay: fix memory leak in gst_rtp_opus_pay_setcaps. The src caps were never dereferenced, causing a memory leak. 2019-07-12 20:51:44 +0200 Mathieu Duponchelle * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: implement support for trickmode interval When the seek event contains a (newly-added) trickmode interval, and TRICKMODE_KEY_UNITS was requested, only let through keyframes separated with the required interval 2019-07-17 19:12:19 +0530 Nirbheek Chauhan * docs/meson.build: meson: Don't generate doc cache when no plugins are enabled Fixes gst-build with -Dauto-features=disabled 2019-07-15 23:24:05 +0900 Seungha Yang * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: Port to color_{primaries,transfer,matrix}_to_iso ... and remove duplicated code. 2019-05-25 22:08:05 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmuxsink: add the ability to mux auxilliary video streams The primary video stream is used to select fragment cut points at keyframe boundaries. Auxilliary video streams may be broken up at any packet - so fragments may not start with a keyframe for those streams. 2019-06-11 23:17:30 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Add video_%d pad template. splitmuxsrc actually supports multiple video pads. Make that clear, especially since it was already creating pads named "video_0" etc. 2019-07-09 23:12:45 +0200 Mathieu Duponchelle * gst/isomp4/qtdemux.c: qtdemux: fix conditions for end of segment in reverse playback The time_position field of the stream is offset by the media_start of its QtDemuxSegment compared to the start of the GstSegment of the demuxer, take it into account when making comparisons. 2019-07-09 23:06:12 +0900 Seungha Yang * gst/matroska/matroska-demux.c: matroskademux: Fix mismatched transfer characteristic TransferCharacteristics(18) should be ARIB STD-B67 (HLG) See https://www.webmproject.org/docs/container/#TransferCharacteristics Also map more color primaries indexes which have been handled by matroska-mux. 2019-07-09 19:49:57 +0900 Seungha Yang * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2: Remove misleading comments gst_pad_template_new() does not take ownership of the caps 2019-01-23 18:27:06 -0500 Olivier Crête * tests/check/elements/rtpsession.c: rtp session: Add test for collision loopback detection Ignore further collisions if the remote SSRC change with ours, it's probably because someone is sending us back the packets we send out. 2019-01-23 18:14:23 -0500 Olivier Crête * tests/check/elements/rtpsession.c: rtpsession tests: Add test for third-party collision detection Add tests to validate the code that ignores the same packets coming from 2 different sources (an third-party collision). 2019-01-23 17:19:15 -0500 Olivier Crête * tests/check/elements/rtpsession.c: rtpsession: Add test for collision on incoming packets Make sure that the collision is properly detected on incoming packets. 2019-01-23 17:09:27 -0500 Olivier Crête * tests/check/elements/rtpsession.c: rtpsession test: Verify that on-ssrc-collision message is emitted 2019-01-23 16:58:22 -0500 Olivier Crête * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Also send conflict event when sending packet If the conflict is detected when sending a packet, then also send an upstream event to tell the source to reconfigure itself. Also ignore the collision if we see more than one collision from the same remote source to avoid problems on loops. 2019-04-15 16:32:03 -0700 Song Bing * sys/v4l2/gstv4l2transform.c: v4l2transform: set right buffer count. Set right buffer count to avoid one buffer. 2019-06-27 19:47:41 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: * tests/check/elements/rtph265.c: rtph265pay: Also immediately send packet if it is a suffix NAL Immediately send packet if it contains any suffix NAL, this is required in case they come after the VCL nal to not have to wait until the next frame. 2019-06-27 19:46:01 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Don't drop second byte of NAL header At least keep 2 bytes per NAL even if the second one is 0, the second byte of the NAL header could very well be 0. 2019-06-26 16:42:44 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: rtph26xpay: Avoid print when there is no bundle at end of packet 2019-06-26 16:25:01 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtph264.c: * tests/check/elements/rtph265.c: rtph26xpay: Wait until there is a VCL or suffix NAL to send With unit tests. 2019-06-19 17:16:03 -0400 Olivier Crête * tests/check/elements/rtph265.c: rtph265pay test: Add unit tests for aggregation 2019-06-18 19:07:38 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtph265.c: rtph265pay: Implement Aggregation packets Align with rtph264pay 2019-06-18 15:03:09 -0400 Olivier Crête * tests/check/elements/rtph264.c: rtph264pay test: Add unit tests for aggregation 2019-06-18 13:45:15 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Report latency when in maximal aggregation mode 2019-06-17 11:31:53 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * tests/check/elements/rtph264.c: rtph264pay: Default to not adding latency when aggregating Send the bundle as soon as there is one VCL unit in the packet at the end of an incoming buffer. The DELTA_UNIT flag is not reliable, so ignore it. 2019-06-14 16:54:23 -0400 Olivier Crête * tests/check/elements/rtp-payloading.c: rtp-payloading test: Fix working to 1.0 buffers instead of groups 2019-06-13 18:07:35 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: rtph265pay: Replace fragmentation while-loop with for-loop Align with rtph264pay 2019-06-13 17:42:05 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Rename payload_len to max_fragment_size Align to rtph264pay 2019-06-13 17:30:08 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Clean up _payload_nal Move determining whether we need to fragment at all into the fragmenter. Align with rtph264pay 2019-06-13 17:23:26 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Extract sending fragments into _payload_nal_fragment Align with rtph264pay 2019-06-13 16:22:57 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Extract sending a single packet into _payload_nal_single Align with rtph264pay 2019-06-13 16:14:31 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Define and use FU_A_TYPE_ID Align with rtph264pay 2019-06-13 16:08:37 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Use snake_case variables Align with rtph264pay 2019-06-13 16:04:39 -0400 Olivier Crête * gst/rtp/gstrtph265pay.c: rtph265pay: Clean up whitespace and syntax Align with rtph264pay 2018-07-03 19:39:25 +0200 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtph264.c: rtph264pay: Support STAP-A bundling Add a new property "do-aggregate"* to the H.264 RTP payloader which enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled, packets are bundled instead of sent immediately, up until the MTU size. Bundles also end at access unit boundaries or when packets have to be fragmented. *: The property-name is kept generic since it might apply more widely, e.g. STAP-B or MTAP. [1]: https://tools.ietf.org/html/rfc6184#section-5.7 Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434 2018-11-05 17:15:39 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Fix delta-unit/discont handling when injecting SPS/PPS Apply the wanted delta-unit and discont to the first packet; following packets for this frame are always delta units and not discont. 2018-11-05 19:03:45 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Replace fragmentation while-loop with for-loop 2018-11-05 18:57:38 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Calculate the right max_fragments 2018-11-05 18:36:35 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Rename payload_len to max_fragment_size 2018-11-05 18:34:40 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Clean up _payload_nal_fragment 2018-11-05 18:06:19 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Clean up _payload_nal Move determining whether we need to fragment at all into the fragmenter. 2018-11-05 18:04:13 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Clean up _payload_nal_single 2018-11-05 17:55:23 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Extract sending fragments into _payload_nal_fragment 2018-11-05 17:49:52 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Extract sending a single packet into _payload_nal_single 2018-11-05 17:10:03 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Define and use FU_A_TYPE_ID 2018-11-05 17:07:06 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Use snake_case variables 2018-11-05 17:04:14 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Clean up whitespace and syntax 2019-06-06 16:05:31 -0400 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: Unlock output if the queue is full 2019-06-29 23:17:28 -0600 Thomas Bluemel * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Ignore unsolicited rtx packets. If an rtx packet arrives that hasn't been requested (it might have been requested from prior to a reset), ignore it so that it doesn't inadvertently trigger a clock skew. 2019-06-29 23:16:44 -0600 Havard Graff * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Add unit test for unsolicited rtx affecting skew 2019-06-13 15:45:28 -0600 Thomas Bluemel * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Only calculate skew or reset if no gap. In the case of reordered packets, calculating skew would cause pts values to be off. Only calculate skew when packets come in as expected. Also, late RTX packets should not trigger clock skew adjustments. Fixes #612 2019-07-02 21:21:05 +0300 Mart Raudsepp * gst/isomp4/qtdemux.c: qtdemux: Provide a 30 frames lead-in for MP3 mpegaudioparse suggests MP3 needs 10 or 30 frames of lead-in (depending on mpegaudioversion, which we don't know here), thus provide at least 30 frames lead-in for such cases as a followup to commit cbfa4531ee5ef. 2019-05-24 10:31:39 -0400 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: max-dropout-time gets cast to int32 So any value over MAXINT32 gets considered as negative and is silently ignored. 2019-07-02 13:00:32 +0200 Mathieu Duponchelle * gst/isomp4/qtdemux.c: qtdemux: do_seek can never be called with a NULL event 2019-07-01 22:38:41 +0200 Mathieu Duponchelle * gst/isomp4/qtdemux.c: qtdemux: only adjust segment time when adjusting segment start We ended up setting segment.time to segment.position when doing reverse playback, which is obviously wrong. 2019-07-01 13:54:13 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: unref the event in element seek handler 2019-06-29 00:25:26 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: handle seek event on the element Without this, the user has to wait for rtspsrc to have sent a PLAY request and exposed its pads before seeking it. 2019-06-26 18:03:29 -0400 Nicolas Dufresne * gst/udp/gstmultiudpsink.c: multiudpsink: Add missing socket.h include Without this include, macro like SO_BINDTODEVICE is not visible and associated feature gets out-compiled. This also affects the support for SO_SNDBUF. 2019-06-24 17:35:15 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Clear new_tags if sending metadata in header This avoids sending an additional metadata object right after the headers. 2018-06-13 14:55:29 -0700 Song Bing * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Fix drain() function return type Return right type for drain() function. 2019-06-24 14:28:39 +0300 Mart Raudsepp * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: add back segment clipping to parsers that have lost it The pre_push_frame default clipping behaviour was introduced in 2010 with commit 30be03004e82 and modified with commit 4163969a2422 in 2011, when most parsers didn't implement a pre_push_frame yet. Not having it meant that clipping was done by default. Those that did implement a pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag adjusted as part of the 2011 refactor work. All other parsers got a pre_push_frame vfunc implementation only in 2013, but seem to have forgot to keep the clipping behaviour, as was done automatically when a pre_push_frame implementation doesn't exist for the parser. aacparse lost it with commit 91d4abcea in July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting in commits 6f89b430e, d2ab5199b, 29f2cae12, 753d3c23a and 292780574. 2019-06-24 09:42:31 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2codec.c: v4l2: fix compiler warning due to c99-ism 2019-06-19 14:28:28 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/flvmux.c: test: flvmux: Test changing caps with one sinkpad These tests segfault without the preceding crash fix. 2019-06-19 14:08:06 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/flvmux.c: test: flvmux: Use gst_harness_sink_push_many And check its return value. 2019-06-19 12:31:46 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Simplify an if-else chain Merge the identical branches and turn the condition around to make it easier to read. 2019-06-19 12:28:22 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvmux.c: flvmux: Avoid crash when changing caps without both streams mux->video_pad and mux->audio_pad can be NULL if the corresponding pad has not been requested. 2019-06-12 15:57:48 +0300 Sebastian Dröge * gst/rtp/gstrtpgstpay.c: rtpgstpay: Send caps anyway if caps are pending in the adapter but are different from the new ones Otherwise it can happen that we receive a caps event, then another caps event and only then buffers. We would then send out the first caps event in the stream but mark buffers with the caps version of the second caps event. 2019-06-12 14:57:24 +0300 Sebastian Dröge * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: rtpgstdepay: Only store the current caps and drop old caps immediately Otherwise it can happen that we already collected 7 caps, miss the 8th caps packet (packet loss) and then re-use the 1st caps for the following buffers instead of the 8th caps which will likely cause errors further downstream unless both caps are accidentally the same. Keeping old caps around does not seem to have any value other than potentially causing errors. We would always receive new caps whenever they change (even if they were previous ones) and it's very unlikely that they happen to be exactly the same as the previous ones. Also after having received new caps or a buffer with a next caps version, no buffers with old caps version will arrive anymore. 2019-06-15 02:00:43 +1000 Jan Schmidt * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Clear clock master before unreffing Make sure to clear any master clock on the media_clock before unreffing it to release the timer callback that's updating the clock and keeping it reffed. 2019-06-16 11:07:31 +1000 Jan Schmidt * gst/matroska/matroska-ids.c: matroska: Initialise a video_context field to satisfy valgrind Clear the mastering_display_info_present field explicitly after reallocating the track context into a video context to avoid uninitialised warnings in valgrind 2019-06-14 17:34:31 -0400 Thibault Saunier * gst/multifile/gstmultifilesink.c: docs: Fix link to strings We can't link to #gchar* this way. 2019-06-14 00:17:22 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: jitterbuffer: unset DTS on output buffers 2019-05-22 21:40:52 +0200 Mathieu Duponchelle * gst/multifile/gstsplitmuxsink.c: splitmuxsink: set the same seqnum on flush_start / flush_stop It's currently not made mandatory by aggregator, but it might eventually be, and is more consistent behaviour See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/977 2019-06-13 11:55:04 +0200 Mikhail Fludkov * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: late packets shouldn't affect PTS of the following packet If, say, a rtx-packet arrives really late, this can have a dramatic effect on the jitterbuffer clock-skew logic, having it being reset and losing track of the current dts-to-pts calculations, directly affecting the packets that arrive later. This is demonstrated in the test, where a RTX packet is pushed in really late, and without this patch the last packet will have its PTS affected by this, where as a late RTX packet should be redundant information, and not affect anything. 2019-06-12 10:47:39 +0200 Mikhail Fludkov * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix rtx delay calulation when large packet spacing 2016-11-24 18:18:01 +0100 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps This patch corrects the delay set on EXPECTED timers that are added when processing gaps. Previously the delay could be too small so that 'timout + delay' was much less than 'now', causing the following retries to be scheduled too early. (They were sent earlier than rtx-retry-timeout after the previous timeout.) 2018-11-20 16:11:12 +0100 Havard Graff * gst/rtpmanager/rtpstats.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: don't try and calculate packet-rate if seqnum are jumping Turns out that the "big-gap"-logic of the jitterbuffer has been horribly broken. For people using lost-events, an RTP-stream with a gap in sequencenumbers, would produce exactly that many lost-events immediately. So if your sequence-numbers jumped 20000, you would get 20000 lost-events in your pipeline... The test that looks after this logic "test_push_big_gap", basically incremented the DTS of the buffer equal to the gap that was introduced, so that in fact this would be more of a "large pause" test, than an actual gap/discontinuity in the sequencenumbers. Once the test was modified to not increment DTS (buffer arrival time) with a similar gap, all sorts of crazy started happening, including adding thousands of timers, and the logic that should have kicked in, the "handle_big_gap_buffer"-logic, was not called at all, why? Because the number max_dropout is calculated using the packet-rate, and the packet-rate logic would, in this particular test, report that the new packet rate was over 400000 packets per second!!! I believe the right fix is to don't try and update the packet-rate if there is any jumps in the sequence-numbers, and only do these calculations for nice, sequential streams. 2019-06-12 11:16:22 +0200 Havard Graff * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix unused variables 2019-06-12 02:42:42 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Protect initial pad configuration with the object lock gst_splitmux_src_activate_part() configures the pad information before starting the pad task, but occasionally the changes it makes to the pad are not seen in the pad task because they're not protected by the right locking. Use the pad's object lock to protect those variables. 2019-06-12 01:42:20 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Restart pad task on a reconfigure On a reconfigure event, restart streaming on the pad so that switching tracks in playbin works cleanly 2019-06-11 18:40:09 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: splitmuxsrc: Use an RW lock instead of a mutex to protect the pad list Fix a deadlock around the pads list by using an RW lock to allow simultaneous readers. The pad list doesn't really changes except at startup and shutdown. 2019-06-11 23:18:24 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Ignore duplicate seeks Use the seqnum to ignore duplicated seek events. 2019-05-29 09:20:07 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Improve debug output Make the debug output less confusing by not mentioning a src pad when doing calculations on the sink pad side. Improve debug around why a GOP is considered overflowing a fragment 2019-05-29 09:20:07 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Give internal queues useful names Makes debug output more useful 2019-06-05 23:13:33 +0300 Mart Raudsepp * gst/isomp4/qtdemux.c: qtdemux: Provide a 2 frames lead-in for audio decoders AAC and various other audio codecs need a couple frames of lead-in to decode it properly. The parser elements like aacparse take care of it via gst_base_parse_set_frame_rate, but when inside a container, the demuxer is doing the seek segment handling and never gives lead-in data downstream. Handle this similar to going back to a keyframe with video, in the same place. Without a lead-in, the start of the segment is silence, when it shouldn't, which becomes especially evident in NLE use cases. 2019-05-28 20:14:49 +0300 Mart Raudsepp * gst/isomp4/qtdemux.c: qtdemux: remove indent exception and reindent As the indent disabling isn't playing along for a following fix, remove the indent disabling and reindent in a way that doesn't look too stupid. 2019-03-08 14:43:20 +0000 Philippe Normand * sys/v4l2/gstv4l2h264codec.c: v4l2: Fix H.264 level 3 string representation The string_to_level function handles "3" so the level_to_string function should do the same, to prevent caps negotiation issues. 2019-03-04 11:05:29 +0000 Philippe Normand * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2codec.c: * sys/v4l2/gstv4l2codec.h: * sys/v4l2/gstv4l2fwhtenc.c: * sys/v4l2/gstv4l2h263enc.c: * sys/v4l2/gstv4l2h264codec.c: * sys/v4l2/gstv4l2h264codec.h: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2h264enc.h: * sys/v4l2/gstv4l2h265codec.c: * sys/v4l2/gstv4l2h265codec.h: * sys/v4l2/gstv4l2h265enc.c: * sys/v4l2/gstv4l2h265enc.h: * sys/v4l2/gstv4l2jpegenc.c: * sys/v4l2/gstv4l2mpeg4codec.c: * sys/v4l2/gstv4l2mpeg4codec.h: * sys/v4l2/gstv4l2mpeg4enc.c: * sys/v4l2/gstv4l2mpeg4enc.h: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2videoenc.h: * sys/v4l2/gstv4l2vp8codec.c: * sys/v4l2/gstv4l2vp8codec.h: * sys/v4l2/gstv4l2vp8enc.c: * sys/v4l2/gstv4l2vp8enc.h: * sys/v4l2/gstv4l2vp9codec.c: * sys/v4l2/gstv4l2vp9codec.h: * sys/v4l2/gstv4l2vp9enc.c: * sys/v4l2/gstv4l2vp9enc.h: * sys/v4l2/meson.build: v4l2: Profile and level probing support for encoders and decoders There used to be some profile/level support in encoders. This code was moved to GstV4l2Codecs and is now also used for decoders. The caps templates for the H.264, H.265, MPEG4, VP8 and VP9 encoders and decoders should now reflect the profiles and levels advertised by the kernel. 2019-06-03 16:21:12 -0400 Aaron Boxer * gst/matroska/matroska-mux.c: matroskamux: fix typo in property description 2019-06-04 13:39:00 -0400 Nicolas Dufresne * tests/check/gst-plugins-good.supp: supp: Ignore leaks caused by shout/sethostent sethostent() seems to be using a global state and we endup with leaks from that API when called through shout_init(). We had the option to only ignore the shout case, but the impression is that if we have shout and another sethostend user, as it's a global state, we may endup with a different stack trace for the same leak. So in the end, we just ignore memory allocated by sethostent in general. 2019-04-30 17:28:25 -0400 Thibault Saunier * ext/pulse/pulsedeviceprovider.c: pulse-device: Hide the alsa device provider if we provide alsa devices 2019-05-21 15:25:03 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpssrcdemux.c: * tests/check/elements/rtpssrcdemux.c: rtpssrcdemux: Avoid taking streamlock out-of-band In this change we now protect the internal srcpads list using the stream lock and limit usage of the internal stream lock to preventing data flowing on the other src pad type while creating and signalling the new pad. This fixes a deadlock with RTPBin shutdown lock. These two locks would end up being taken in two different order, which caused a deadlock. More generally, we should not rely on a streamlock when handling out-of-band data, so as a side effect, we should not take a stream lock when iterating internal links. 2019-05-27 18:08:54 +0900 Damian Hobson-Garcia * sys/v4l2/gstv4l2object.c: v4l2object: Orphan buffer pool on object_stop if supported Use V4L2 buffer orphaning, on recent kernels so that the device can be restarted immediately with a new buffer pool during renogatiation. 2019-05-30 13:12:31 +0900 Damian Hobson-Garcia * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Free orphaned allocator resources when buffers are released Allocator resources cannot be freed when a buffer pool is orphaned while its buffers are in use. They should, however, be freed once those buffers are no longer needed. This patch disposes of any buffers belonging to an orphaned pool as they are released, and makes sure that the allocator is cleaned up when the last buffer is returned. 2019-05-30 11:13:07 +0900 Damian Hobson-Garcia * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: return TRUE when buffer pool orphaning succeeds When trying to orphan a buffer pool, successfully return and unref the pool when the pool is either successfully stopped or orphaned. Indicate failure and leave the pool untouched otherwise. 2019-05-31 23:04:11 +0200 Niels De Graef * configure.ac: * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsrc.c: * meson.build: meson: Bump minimal GLib version to 2.44 This means we can use some newer features and get rid of some boilerplate code using the G_DECLARE_* macros. As discussed on IRC, 2.44 is old enough by now to start depending on it. 2018-09-05 21:10:51 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Use size of first closed caption buffer in prefill mode It must be accurate for all samples to work in Final Cut properly, so the best we can do is to assume that all samples are the same as the first. Bigger samples are truncated, smaller samples are padded. 2019-05-29 22:06:58 +0200 Mathieu Duponchelle * docs/meson.build: * ext/lame/gstlamemp3enc.c: * ext/mpg123/gstmpg123audiodec.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/twolame/gsttwolamemp2enc.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: * gst/level/gstlevel.c: * gst/rtp/gstrtpL8depay.c: * gst/rtp/gstrtpL8pay.c: * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: * gst/spectrum/gstspectrum.c: * sys/v4l2/gstv4l2object.c: doc: remove xml from comments 2019-05-29 11:02:26 +0100 Tim-Philipp Müller * docs/gst_plugins_cache.json: docs: update plugins cache And add gtk+ and qt plugins 2019-05-29 10:58:40 +0100 Tim-Philipp Müller * ext/dv/meson.build: * ext/gtk/meson.build: * ext/qt/meson.build: * sys/osxaudio/meson.build: * sys/osxvideo/meson.build: * sys/waveform/meson.build: dv, gtk, qt, osxaudio, osxvideo, waveform: add to plugins list Makes sure the paths for these plugins are included in the uninstalled plugin paths list. And also for the docs. Fixes #604 2019-04-18 15:31:00 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: Add new property to offset all streams to start at zero This takes the timestamp of the earliest stream and offsets it so that it starts at 0. Some software (VLC, ffmpeg-based) does not properly handle Matroska files that start at timestamps much bigger than zero. Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/449 2019-05-28 14:13:56 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: rtpmp4gdepay: don't spam debug log for broken ADTS-in-RTP AAC Print warning only once. 2019-05-22 18:06:04 +0300 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Only set running time on finalizing sink element when in async-finalize mode There is only a single sink element in async-finalize mode, and we would keep the running time from previous fragments set in that case. As we don't ever set the running time for the very last fragment on EOS, this would mean that the closing time reported for the very last fragment is the same as the closing time of the previous fragment. 2015-03-26 13:08:32 -0400 Nicolas Dufresne * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove uneeded keep-alive hack The rtsp connection code has been fixed now. https://bugzilla.gnome.org/show_bug.cgi?id=744209 2019-05-26 17:46:06 +0300 Vivia Nikolaidou * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Print GstClockTimeDiff as GST_STIME_FORMAT 2019-05-25 19:45:02 +0200 Mathieu Duponchelle * docs/gst_plugins_cache.json: doc: update plugin cache 2019-05-25 17:25:02 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: the documentation for GstVideoMixer2Pad is not exposed 2019-05-25 16:56:32 +0200 Mathieu Duponchelle * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/soup/gstsouphttpsrc.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/multifile/gstmultifilesrc.c: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/spectrum/gstspectrum.c: doc: fix element section documentations Element sections were not rendered anymore after the hotdoc port, fixing this revealed a few incorrect links. 2019-02-19 12:15:19 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpbin.c: rtpbin: Improve RTPStorage action signal documentation This is a tiny clarification as the storage was loosely named "storage". This change clarify that the storage is specificaly used for received RTP packets. This is unlike the storage found in rtprtxsend that stores a backlog of sent RTP packets. 2019-05-05 22:16:36 +0900 Seungha Yang * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: Add BT2020_10, PQ and HLG transfer functions The direct use of newly added transfer functions 2019-05-23 12:38:06 +0300 Sebastian Dröge * ext/aalib/meson.build: aasink: Generate pkg-config file for the plugin 2019-05-22 11:01:17 +0900 Seungha Yang * gst/multifile/gstmultifilesink.c: multifilesink: Fix documentation of max-file-duration property The max-file-duration property works with max-duration mode 2019-05-14 17:36:14 -0400 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Always keep at least one NACK on early RTCP We recently added code to remove outdate NACK to avoid using bandwidth for packet that have no chance of arriving on time. Though, this had a side effect, which is that it was to get an early RTCP packet with no feedback into it. This was pretty useless but also had a side effect, which is that the RTX RTT value would never be updated. So we we stared having late RTX request due to high RTT, we'd never manage to recover. This fixes the regression by making sure we keep at least one NACK in this situation. This is really light on the bandwidth and allow for quick recover after the RTT have spiked higher then the jitterbuffer capacity. 2019-05-16 09:14:19 -0400 Thibault Saunier * docs/meson.build: docs: Stop building the doc cache by default Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36 2019-05-13 22:53:59 -0400 Thibault Saunier * docs/gst_plugins_cache.json: docs: Update plugins documentation cache 2019-04-23 12:28:23 -0400 Thibault Saunier * ext/soup/gstsouputils.c: * gst/goom/flying_stars_fx.c: * gst/goom/goom_tools.h: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpmux.h: * sys/v4l2/gstv4l2object.c: doc: Fix some docstrings 2018-10-22 11:39:55 +0200 Thibault Saunier * Makefile.am: * configure.ac: * docs/Makefile.am: * docs/all_index.md: * docs/gst_api_version.in: * docs/gst_plugins_cache.json: * docs/index.md: * docs/meson.build: * docs/plugins/.gitignore: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/gst-plugins-good-plugins.types: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-directsound.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-osxaudio.xml: * docs/plugins/inspect/plugin-osxvideo.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-waveform.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * docs/random/ChangeLog-0.8: * docs/random/PORTED_09: * docs/sitemap.txt: * docs/version.entities.in: * ext/aalib/meson.build: * ext/cairo/meson.build: * ext/flac/meson.build: * ext/gdk_pixbuf/meson.build: * ext/jack/meson.build: * ext/jpeg/meson.build: * ext/lame/meson.build: * ext/libcaca/meson.build: * ext/libpng/meson.build: * ext/mpg123/meson.build: * ext/pulse/meson.build: * ext/raw1394/meson.build: * ext/shout2/meson.build: * ext/soup/meson.build: * ext/speex/meson.build: * ext/taglib/meson.build: * ext/twolame/meson.build: * ext/vpx/meson.build: * ext/wavpack/meson.build: * gst/alpha/meson.build: * gst/apetag/meson.build: * gst/audiofx/meson.build: * gst/audioparsers/meson.build: * gst/auparse/meson.build: * gst/autodetect/meson.build: * gst/avi/meson.build: * gst/cutter/meson.build: * gst/debugutils/meson.build: * gst/deinterlace/meson.build: * gst/dtmf/meson.build: * gst/effectv/meson.build: * gst/equalizer/meson.build: * gst/flv/meson.build: * gst/flx/meson.build: * gst/goom/filters.c: * gst/goom/meson.build: * gst/goom2k1/meson.build: * gst/icydemux/meson.build: * gst/id3demux/meson.build: * gst/imagefreeze/meson.build: * gst/interleave/meson.build: * gst/isomp4/meson.build: * gst/law/meson.build: * gst/law/mulaw-conversion.c: * gst/level/meson.build: * gst/matroska/meson.build: * gst/monoscope/meson.build: * gst/multifile/meson.build: * gst/multipart/meson.build: * gst/replaygain/meson.build: * gst/rtp/meson.build: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/meson.build: * gst/rtsp/meson.build: * gst/shapewipe/meson.build: * gst/smpte/meson.build: * gst/spectrum/meson.build: * gst/udp/meson.build: * gst/videobox/meson.build: * gst/videocrop/meson.build: * gst/videofilter/meson.build: * gst/videomixer/meson.build: * gst/wavenc/meson.build: * gst/wavparse/meson.build: * gst/y4m/meson.build: * meson.build: * meson_options.txt: * sys/directsound/meson.build: * sys/oss/meson.build: * sys/oss4/meson.build: * sys/v4l2/meson.build: * sys/ximage/meson.build: doc: Port documentation to hotdoc 2018-11-12 08:05:45 -0300 Thibault Saunier * gst/isomp4/gstqtmux.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: Mark some properties as DOC_SHOW_DEFAULT 2018-10-22 11:39:24 +0200 Thibault Saunier * ext/aalib/gstaasink.c: * ext/cairo/gstcairooverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/audiofx/gstscaletempo.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/deinterlace/gstdeinterlace.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/flv/gstindex.c: * gst/flx/gstflxdec.c: * gst/goom/filters.c: * gst/goom/goom_config.h: * gst/goom/goom_filters.h: * gst/goom/goom_plugin_info.h: * gst/goom/gstgoom.c: * gst/goom/ifs.c: * gst/goom/sound_tester.h: * gst/goom2k1/filters.h: * gst/goom2k1/goom_core.h: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-conversion.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpL24pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpklvpay.c: * gst/rtp/gstrtpstreamdepay.c: * gst/rtp/gstrtpstreampay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpsession.c: * gst/rtsp/gstrtpdec.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/tuner.c: * sys/v4l2/tunerchannel.c: * sys/v4l2/tunernorm.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: docs: Port all docstring to gtk-doc markdown 2019-05-02 22:14:35 -0700 Thiago Santos * gst/rtsp/gstrtspsrc.c: rtspsrc: do not try to send EOS with invalid seqnum The second udpsrc (rtcp) might not have seen the segment event if it was not enabled or if rtcp is not available on the server. So if the application tries to send an EOS event it will try to set an invalid seqnum to the event. 2019-04-24 13:54:12 -0400 Nicolas Dufresne * gst/rtpmanager/rtpsource.c: rtpsource: Add more information to probation warning 2019-04-24 13:47:54 -0400 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Call on-new-ssrc earlier Right now, we may call on-new-ssrc after we have processed the first RTP packet. This prevents properly configuring the source as some property like "probation" are copied internally for use as a decreasing counter. For this specific property, it prevents the application from disabling probation on auxiliary sparse stream. Probation is harmful on sparse streams since the probation algorithm assume frequent and contiguous RTP packets. 2019-02-19 13:34:49 +0900 Seungha Yang * gst/matroska/matroska-mux.c: matroskamux: Write MasteringMetadata and Max{CLL,FALL} Enable muxing with HDR meta data if upstream provided it 2019-02-18 23:28:50 +0900 Seungha Yang * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: matroskademux: Add support parsing HDR metadata Set SMPTE ST 2086 mastering-display-metadata and content-light-level to caps, if any 2019-02-19 18:27:23 +0900 Seungha Yang * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska-read-common.h: matroska: Remove white space 2019-05-01 10:00:51 +0300 Sebastian Dröge * gst/rtp/gstrtpvrawdepay.c: rtprawdepay: Don't get rid of the buffer pool on FLUSH_STOP We expect there to be a pool as long as the caps are known and FLUSH_STOP is not resetting the caps. Getting rid of the pool would cause assertions. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/584 2019-02-08 10:09:17 +0100 Danny Smith * gst/rtpmanager/gstrtpbin.c: rtpbin: Free storage when freeing session 2019-04-25 21:52:42 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Fix typo in error message 2019-04-25 11:19:06 +0300 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Only set the DISCONT flag on the first buffer after segment start 2019-04-24 02:38:32 +0900 okuoku * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: Use jack_free(3) to release ports Port objects acquired with jack_get_ports() need to be freed with jack_free(3), not stdlib free(). On Windows, Jack may be linked against different libc than GStreamer libraries so free()ing port objects directly might cause crash because of libc mismatch. 2019-04-23 10:10:01 +0100 Philippe Normand * gst/audiofx/gstscaletempo.c: scaletempo: Advertise interleaved layout in caps templates Scaletempo doesn't support non-interleaved layout. Not explicitely stating this would trigger critical warnings and a caps negotiation failure when scaletempo is used as playbin audio-filter. Patch suggested by George Kiagiadakis . Fixes #591 2019-04-21 20:12:28 +0900 Seungha Yang * gst/matroska/meson.build: meson: matroska: Ensure header dependency not only library Library existence does not guarantee header. 2018-11-13 13:48:11 +0100 Robert Rosengren * gst/udp/gstmultiudpsink.c: multidupsink: Use gst_net_utils_set_socket_tos for QoS DSCP Util function in net library exists for setting QoS DSCP on socket, hence use it to simplify code. 2019-04-19 10:27:38 +0100 Tim-Philipp Müller * README: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * meson.build: Back to development === release 1.16.0 === 2019-04-19 00:23:16 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * README: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.16.0 2019-04-19 00:23:16 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2019-04-19 00:23:14 +0100 Tim-Philipp Müller * po/el.po: * po/zh_CN.po: Update translations 2019-04-18 17:14:18 +0200 Benjamin Sigonneau * ext/qt/qtplugin.pro: qmlglsink: fix compilation with Qt >= 5.5 on Windows As of Qt >= 5.5, qmake do not link to opengl32 by default anymore. This commit adds opengl32.lib to the .pro file so that the plugin can be build using QtCreator on Windows. 2019-04-17 15:48:26 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Build qt plugin in C++11 mode explicitly This works implicitly most of the time, but we need to set it explicitly for building with Android. 2019-04-16 14:35:06 +0530 Guillaume Desmottes * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2: fix use after free when handling events The sink_event parent function may consume the event so we shouldn't use it after having calling it. === release 1.15.90 === 2019-04-11 00:26:58 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.15.90 2019-04-11 00:26:58 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2019-04-09 23:51:22 +0100 Tim-Philipp Müller * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: rtpulpfecdec,enc: unbreak plugin gtk-doc build in autotools Fix doc chunks to not use that syntax for links that have the url as description, it will be put verbatim into the xml/*.xml file and then the expat parser will throw a syntax error like: File "../../common/mangle-db.py", line 71, in main() File "../../common/mangle-db.py", line 69, in main patch (details.replace("-details", ""), os.path.basename(details)) File "../../common/mangle-db.py", line 20, in patch doc = xml.dom.minidom.parse(related) File "/usr/lib/python2.7/xml/dom/minidom.py", line 1918, in parse return expatbuilder.parse(file) File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 924, in parse result = builder.parseFile(fp) File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 207, in parseFile parser.Parse(buffer, 0) xml.parsers.expat.ExpatError: not well-formed (invalid token): line 84, column 7 2019-04-08 11:35:34 +0200 Antonio Ospite * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: preserve GST_BUFFER_FLAG_DISCONT on the first outputted buffer If the incoming frame buffer has GST_BUFFER_FLAG_DISCONT set this should be preserved and set for the first output buffer too, like other payloaders do. Spotted with gst-validate-1.0 when adding integration tests for rtpsession, a minimal test to reproduce the issue is: $ gst-validate-1.0 videotestsrc num-buffers=1 ! rtpvrawpay ! identity ! fakesink Starting pipeline Pipeline started warning : Buffer didn't have expected DISCONT flag333 speed: 1.000000 /> Detected on Detected on Detected on Description : Buffers after SEGMENT and FLUSH must have a DISCONT flag Issues found: 1 =======> Test PASSED (Return value: 0) 2019-03-22 12:42:14 -0400 Olivier Crête * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: rtpulpfec*: Replace github URIs with gitlab.fdo ones 2019-03-21 17:01:11 -0400 Olivier Crête * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpredenc.c: rtpred*: Add example pipelines 2019-03-21 16:48:37 -0400 Olivier Crête * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: rtpulpfec*: Improve documentation 2019-03-20 18:31:48 -0400 Olivier Crête * gst/rtp/gstrtpstorage.c: * gst/rtp/gstrtpulpfecdec.c: rtpstorage + rtpulpfecdec: Get the storage using a query as fallback This allows it to be used using gst-launch for easier testing. 2019-03-19 06:22:29 -0700 Dan Kegel * sys/osxvideo/Makefile.am: * sys/osxvideo/meson.build: osxvideo: fix mac os 10.14 build lockFocusIfCanDraw is deprecated in mac os 10.14. Apple suggests a different way to do what that does, but for now, just suppress the deprecation. There's no way to disable just that deprecation, so shut them all down. OpenGL is also deprecated in mac os 10.14. There is a gentle way to turn off just those deprecations (GL_SILENCE_DEPRECATION), but since this commit turns them all off, that's moot. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/577 2019-04-07 12:00:49 -0400 Nicolas Dufresne * tests/check/elements/rtpsession.c: test: rtpsession: Verify on-sending-nacks callback 2019-03-27 16:19:15 -0400 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Allow overriding NACK packet creation This introduce a new signal on RTSession, on-sending-nacks is emited right before the list of seqnums to be nacked are processed and transformed into FB Nack. This allow implementing custom nacks handling through another mechanism with APP feedback. 2018-11-20 02:45:04 +0100 Mathieu Duponchelle * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * tests/check/elements/rtpsession.c: rtpsession: Add disable-sr-timestamp property The Onvif Streaming Spec, in section 6.11, mandates that when Rate-Control is disabled potential RTCP packets shall have their timestamps set to 0. 2019-03-05 20:57:44 +0100 Philipp Zabel * sys/v4l2/ext/types-compat.h: v4l2: remove __user define from types-compat.h Remove the now unused __user define. 2019-03-05 20:53:47 +0100 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: use opRGB colorspace and xfer func defines AdobeRGB defines have been renamed to opRGB in the kernel headers, use the new names. 2019-01-24 16:12:13 +0100 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2videodec: support orphaning Recent kernels allow REQBUFS(0) on a queue that still has buffers in use (mmapped or exported via dmabuf), orphaning all buffers on the queue. If this is supported, the v4l2videodec element does not have to send a drain request downstream. 2019-01-24 16:12:13 +0100 Philipp Zabel * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: support orphaning Now that the v4l2allocator allows orphaning the V4L2 buffer queue, add support for orphaning in the v4l2bufferpool. gst_v4l2_buffer_pool_orphan can be used as a replacement for gst_v4l2_buffer_pool_stop, without having to wait for buffers to be returned to the pool. 2019-01-24 16:12:13 +0100 Philipp Zabel * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: v4l2allocator: support orphaning Recent kernels allow REQBUFS(0) on a queue that still has buffers in use (mmapped or exported via dmabuf), orphaning all buffers on the queue. Orphaning the allocator causes it to release all buffers with REQBUFS(0), even if they are still in use. An orphaned allocator can only be stopped. It can not be restarted or create new buffers. 2019-01-24 15:36:49 +0100 Philipp Zabel * sys/v4l2/ext/v4l2-common.h: * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: v4l2: update kernel headers to latest from media tree Update to the latest installed headers (output of make headers_install) from the media tree, keeping the slight modifications to the includes. This includes new HEVC controls, the AdobeRGB -> opRGB rename, a new capabilities field for v4l2_requestbuffers and v4l2_create_buffers, new 32-bit YUV formats, and request_fd changes. 2019-04-03 14:13:49 -0400 Nicolas Dufresne * ext/shout2/gstshout2.c: shout2: Fix leak on error in start 2019-03-29 22:48:53 -0400 Nicolas Dufresne * tests/check/elements/rtpsession.c: test: rtpsession: Test FB Nack packing We used to split the NACK if a smaller seqnum of a range of seqnum was submited. This test also make sure that the three operations (append, prepend, update) works properly. 2019-03-29 22:34:47 -0400 Nicolas Dufresne * tests/check/elements/rtpsession.c: test: rtpsession: Test handling of NACK surplus This test verify that NACKs that didn't fit in one packet are properly filtered and inserted into the following pipeline. 2019-03-25 13:42:25 -0400 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsession: Send as many nack seqnum as possible In order to do that, we now split the nacks registration from the actual FB nack packet construction. We then try and add as many FB Nacks as possible into the active packets and leave the remaining seqnums in the RTPSource. In order to avoid sending outdated NACK later on, we save the seqnum calculated deadline and cleanup the outdated seqnums before the next RTCP send. Fixes #583 2018-04-30 10:54:19 +0200 John Bassett * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Fix race when sending PLI, FIR and NACK packets Calling rtp_session_send_rtcp before marking the source as requiring a pli/fir/nack meant the rtcp_thread could be scheduled and start running before the source was updated. This meant the request would not be sent early but instead was transmitted with the next regular RTCP packet. Add test for nack generation. 2019-03-29 16:49:37 -0400 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: rtpsession: Fix early rtcp time comparision If the current time is equal to the early rtcp time deadline, there is no need to schedule a timer. This ensure that immediate feedback is really immediate and simplify implementing unit tests with the test clock, which stops perfectly on the timeout time. This fix has been extracted from Pexip feature patch called "rtpsession: Allow instant transmission of RTCP packets" 2019-01-24 11:54:49 +0100 Guillaume Desmottes * sys/v4l2/gstv4l2src.c: v4l2src: preserve features when fixating caps The caps features were lost when sorting caps structures in gst_v4l2src_fixate(). This was breaking alternate as GST_CAPS_FEATURE_FORMAT_INTERLACED was removed from the caps. 2018-11-13 21:23:30 +0100 Mathieu Duponchelle * gst/rtp/gstrtpgstpay.c: rtpgstpay: Set DELTA_UNIT flag when appropriate When used in combination with a rtponviftimestamp element downstream, forwarding this flag ensures it gets correctly serialized in the ONVIF header extension. 2019-04-03 16:42:26 +0200 Antonio Ospite * gst/rtpmanager/gstrtpjitterbuffer.c: docs: fix typo s/abonormally/abnormally/ 2019-04-03 16:38:56 +0200 Antonio Ospite * gst/debugutils/gsttaginject.c: * gst/goom2k1/gstgoom.c: * gst/monoscope/gstmonoscope.c: * gst/rtp/README: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpsource.c: * gst/smpte/gstsmpte.c: docs: fix typo s/incomming/incoming/ 2019-04-03 16:34:22 +0200 Antonio Ospite * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpvrawpay.c: rtp: fix indentation after G_DEFINE_TYPE A missing colon after G_DEFINE_TYPE declaration was confusing gst-indent and causing problem in the pre-commit hook. Add the missing colon and fix the following function declaration to follow the normal GStreamer style. 2019-03-07 15:34:03 +0100 Antonio Ospite * gst/rtpmanager/gstrtpsession.c: rtpsession: fix comment to refer to buffers instead of groups One comments in gst_rtp_session_chain_send_rtp_common() is referring to groups in a buffer list, however this concept of "group" comes from GStreamer 0.10 and does not exist anymore in GStreamer 1.0, so update the comment to refer to buffers instead. 2019-03-06 09:52:45 +0100 Antonio Ospite * gst/rtpmanager/rtpsource.c: rtpsource: add comment to explain why probation queue is not always cleared 2019-04-02 12:51:04 +0200 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: add test to verify that stats are correct Add a test to verify that stats about sent and received packets are correct even when using buffer lists. NOTE: the newly introduced get_session_source_stats() selects the desired source (sender or receiver) by filtering them by type (using the get_sender parameter) rather than by ssrc because this simplifies the code and it's good enough for testing purposes as there is usually one source per type in the test setup. Filtering by ssrc would have required handling asynchronous signals like "on-new-sender-ssrc", with the relative locking, just to retrieve the actual ssrc of the sender. 2019-03-05 13:43:12 +0100 Antonio Ospite * gst/rtpmanager/rtpsource.c: rtpsource: fix stats about received packets The update_receiver_stats() function is called also when sending packets in rtp_source_send_rtp(), and sending packets may happen using a buffer list rather than individual buffers. So update the stats using the actual number of packets sent. NOTE: this is fine for the receive path too (rtp_process_send_rtp) because the receive path does not support buffer lists and pinfo->packets would always be equal to 1 in this case. 2019-03-11 10:08:21 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: move buffer list creation next to its validation The tests create a buffer list and then use the chain_list callback to verify that the correct packets have been pushed. Move the creation and validation code next to each other so that the reader can more easily understand what is going on. While at it add some comments to introduce the two related functions. 2019-03-06 19:27:01 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: set the chain_list function directly in the test The helper function set_chain_function does not really do anything useful, remove it. 2019-03-06 19:19:03 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: make check_packet more flexible Make it possible to differentiate between the position in the list and the packet index in the global structures in check_packet, in some future case the list may change, in case some element removes a buffer from the list, and the two indices may not coincide. 2019-03-05 12:47:29 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: factor out a function to create packets buffers 2019-03-04 11:27:33 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: check if the chain_list function has been called Make the test more useful to verify that the chain list function has actually been called. 2019-02-27 12:27:21 +0100 Antonio Ospite * tests/check/elements/rtpbin_buffer_list.c: test: rtpbin_buffer_list: port to GStreamer 1.0 Port the rtpbin_buffer_list test to GStreamer 1.0 and re-enable it. Some other changes include: - the check on the caps has been moved from the buffer level to the pad level; - remove underscore prefix from static functions names, this is not idiomatic in C and rarely used in the other tests; - the unused header_buffer variable has been removed; - check_group() has been renamed to check_packet() because in GStreamer 1.0 there is no concept of "group" anymore, the comments have also been updated to reflect this. 2019-04-01 18:20:53 +0100 Tim-Philipp Müller * tests/check/elements/jpegdec.c: tests: jpegdec: bump discoverer timeout for valgrind Tests might take a bit longer, esp. when run under valgrind and/or they're running on the CI with other things going on, so let's just bump the timeout to something higher and let the test runner time us out if needed. 2019-04-01 18:20:28 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Only ensure that moc is available on Linux On other OSes, it's not possible to have qmake or the qt5 pkg-config files and not have moc, and `moc` will not be in `PATH`, so this only causes problems. 2019-03-21 18:24:43 -0400 Olivier Crête * gst/rtp/rtpstoragestream.c: rtpstorage: Limit the queue size Limit to the queue size in case there is no arrival time or in case there is a huge flood of packets. 2019-03-18 15:30:54 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Request the FEC decoder even if ignore-pt is set 2019-03-18 15:27:21 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Factor out the code that exposes the src pad 2019-03-22 02:08:01 -0400 Olivier Crête * gst/rtp/gstrtpreddec.c: rtpreddec: Add some more debug prints 2019-03-21 17:32:18 -0400 Olivier Crête * gst/rtp/rtpstorage.c: rtpstorage: Issue warning if request by size if 0 If the size is 0, then nothing will ever be in the storage, if a request is received, it generally implies a misconfigured pipeline. 2019-03-21 17:24:42 -0400 Olivier Crête * gst/rtp/gstrtpstorage.c: * gst/rtp/rtpstorage.c: * gst/rtp/rtpstoragestream.c: rtpstorage: Add more debug messages 2019-03-21 17:12:53 -0400 Olivier Crête * gst/rtp/gstrtpstorage.c: * gst/rtp/rtpstorage.c: * gst/rtp/rtpstoragestream.c: * gst/rtp/rtpstoragestream.h: * tests/check/Makefile.am: * tests/check/elements/rtpstorage.c: * tests/check/meson.build: rtpstorage: Make debug category available to sub objects 2019-03-21 17:12:33 -0400 Olivier Crête * gst/rtp/gstrtpstorage.c: rtpstorage: Add debug funcptr to chain function 2019-03-22 12:01:01 +0100 Julian Bouzas * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: flac: report latency in flacenc and flacdec The FLAC specification states that the data is processed in blocks, regardless of the number of channels. Thus, The latency can be calculated using the blocksize and rate. For example a 1 second block sampled at 44.1KHz has a blocksize of 44100 2019-03-22 23:36:42 +0000 Tim-Philipp Müller * tests/examples/rtsp/test-onvif.c: examples: rtsp: fix compiler warning "control reaches end of non-void function" 2019-03-22 15:07:56 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Remove set but not use running-time 2019-03-19 09:50:04 -0400 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: rtpmanager: Register chain functions to debug 2019-02-27 15:49:13 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpbin.c: rtpbin: Allow reusing the sender AUX bin This is needed for the case you don't know in advance all the sessions you will be using, but would like to place all the related AUX element in the same GstBin. As per current implementation, each time an sender AUX bin is requested and returned, RTPBin will walk the src pads and create sessions for these pads. In the current implementation, if a src pad already have a sessions, it returns an error and stops. As a side effect, if an AUX bin is reused in a following AUX bin request, it can only work if the pads are created on the last request. This change simply relax the restriction in order to keep walking, and just ensure that all newly created pads have a sessions. 2018-06-25 17:49:07 +0200 Philipp Zabel * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: set GstVideoCodecFrame sync point flag The V4L2 elements already set the delta unit buffer flag when dequeueing the buffer, but gst_video_encoder_finish_frame overwrites it from the passed codec frame's sync point flag. Set the flag correctly. 2018-08-23 11:47:14 +0300 George Kiagiadakis * gst/rtpmanager/gstrtpsession.c: gstrtpsession: improve stats about rtx requests 2019-03-20 15:45:35 -0400 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Improve looging of not found RTX packet When an RTX packet is not found, display a message that say if the packet have not arrived yet or if it was already removed from the RTX packet queue. 2018-08-09 16:40:26 +0300 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Remove unused rtp_session_create_source 2019-03-21 11:17:08 +0000 Tim-Philipp Müller * meson.build: meson: add -Wno-unused also to C++ args when gst debug system is disabled And check if argument is supported instead of just passing it blindly, and make meson code slightly cleaner, centralising the argument setting in one place. 2019-03-10 19:30:50 +0000 Piotr Drąg * po/LINGUAS: Update LINGUAS 2019-03-19 12:31:35 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Don't pass zero to denominator for framerate Need to respect return of gst_video_guess_framerate() to ensure non-zero denominator. This patch is to fix below error with an abnormal (but has valid frame) file. (gst-play-1.0:17940): GStreamer-CRITICAL **: passed '0' as denominator for `GstFraction' 2019-03-05 09:43:47 +0000 Philippe Normand * sys/v4l2/gstv4l2fwhtenc.c: * sys/v4l2/gstv4l2h263enc.c: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2h265enc.c: * sys/v4l2/gstv4l2jpegenc.c: * sys/v4l2/gstv4l2mpeg4enc.c: * sys/v4l2/gstv4l2vp8enc.c: * sys/v4l2/gstv4l2vp9enc.c: v4l2: Set Hardware classifier on encoders 2019-02-27 11:56:20 +0000 Philippe Normand * sys/v4l2/gstv4l2videodec.c: v4l2: Set Hardware classifier on video decoders 2019-03-01 14:58:24 +0100 Philipp Zabel * sys/v4l2/gstv4l2transform.c: v4l2transform: don't segfault if flushed without pools The v4l2output and v4l2capture v4l2objects can have pool == NULL if they have been stopped before. 2019-02-07 11:58:19 +0000 Charlie Turner * gst/isomp4/qtdemux.c: qtdemux: Find mp4a esds atoms in protected streams sample description tables. This problem was found in Test. 2 of the YouTube 2018 EME tests[1]. The code was accidentally not finding an mp4a's esds atom in the sample description table when the stream was encrypted. It assumed that if the stream is protected, then only an enca atom will be found here. What happens with YouTube is they often provide protected content with a few seconds of clear content, and then switch to the encrypted stream. The failure case here was an incorrect codec_data field being sent into aacparse. The advertisement of stereo audio @ 44.1kHz for the mp4a (unprotected) stream was incorrect. As usual, the esds contained the real values here which were mono at 22050 Hz. Here's what the MP4 tree looks like for these types of files, demonstrating why the code was making a wrong assumption (or maybe YouTube is being unusual), [ftyp] size=8+16 ... [moov] size=8+1571 ... [trak] size=8+559 ... [stsd] size=12+234 entry-count = 2 [enca] size=8+147 channel_count = 2 sample_size = 16 sample_rate = 44100 [esds] size=12+27 ... ... [mp4a] size=8+67 channel_count = 2 sample_size = 16 sample_rate = 44100 [esds] size=12+27 ... In addition to fixing this, the checks for esds atoms in mp4a and mp4v have been made symmetrical. While I haven't seen a test case for video with the same problem, it seemed better to make the same checks. This also fixes a crash reported from another user[2], they also noted the asymmetry with mp4v and mp4a. [1] https://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2018.html?test_type=encryptedmedia-test [2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/398 2019-03-15 10:41:20 +0100 Andreas Frisch * gst/flv/gstflvmux.c: flvmux: Fix scale of time values in warning message 2019-03-15 09:18:00 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't remove udpsrc/sink from rtspsrc if they were not added to it This can happen in various error cases that could happen between the creation of the element in question and the adding to the rtspsrc. It causes an ugly critical warning right now but is otherwise harmless. 2019-03-13 14:00:10 +0100 Antonio Ospite * tests/check/elements/imagefreeze.c: test: imagefreeze: add test for the num-buffers property 2019-03-13 13:03:44 +0100 Antonio Ospite * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: add a num-buffers property The imagefreeze element can be handy for benchmarking downstream elements because it re-uses the same buffer memory and introduces less overhead compared to always creating new frames with videotestsrc. However it's not possible to make imagefreeze send EOS when using gst-launch-1.0. Add a num-buffers property to make it look more like a source in the above scenario. 2019-03-12 16:52:45 +0100 Guillaume Desmottes * gst/matroska/matroska-mux.c: matroskamux: add support for new color primaries 2019-03-07 11:24:38 +0100 Philipp Zabel * sys/v4l2/gstv4l2sink.c: v4l2sink: fix pool-less allocation query handling This fixes a critical warning if the last-sample property is enabled: (gst-launch-1.0:391): GStreamer-CRITICAL **: 01:12:57.428: gst_object_unref: assertion 'object != NULL' failed If the allocation query does not contain any allocation pools, gst_query_parse_nth_allocation_pool will leave the local pool, min, and max variables undefined, so check the array length first. If pool is NULL, do not call gst_object_unref. 2019-03-08 11:03:31 +0900 Seungha Yang * tests/examples/meson.build: meson: Build v4l2 example only if v4l2 plugin was built Otherwise v4l2 example will be built with MSVC 2019-03-07 12:38:41 +0100 Antonio Ospite * ext/dv/gstdvdemux.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsrc.c: docs: fix typos s/recieve/receive/ 2019-02-28 12:32:51 +0100 Antonio Ospite * gst/rtpmanager/rtpsource.c: rtpsource: fix documentation of rtp_source_send_rtp parameters In commit 28e5f9098 (rtpbin: use PacketInfo for the sender, 2013-09-13) the rtp_source_send_rtp signature changed but the documentation was not adjusted to match the new one. Update the documentation to match the function signature. 2019-03-06 12:59:52 +0100 Antonio Ospite * gst/rtpmanager/rtpsession.c: rtpsession: fix typo in a comment, s/SESSION_LOCK/RTP_SESSION_LOCK/ Fix a typo in a comment, mainly to avoid confusing autocompletion in text editors. 2019-02-27 16:45:54 +0100 Antonio Ospite * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: fix typos and update parameters names in comments Some functions now accept a generic 'gpointer data' parameter because they can work either on a single buffer or a buffer list. However the comments were still referring to the old 'GstBuffer *buffer' parameter, so update the comments to match the actual functions signature. 2019-03-06 16:28:34 +0100 Antonio Ospite * gst/rtpmanager/rtpstats.h: rtpstats: fix some fields names in the RTPSourceStats documentation Fix documentation of RTPSourceStats to use the actual fields names. 2019-03-06 17:40:12 +0000 Mathieu Duponchelle * gst/rtp/gstrtpulpfecdec.c: rtpulpfdecdec: only put recovered packet back into storage if not recovered from there 2019-03-06 17:38:03 +0000 Mathieu Duponchelle * gst/rtp/gstrtpulpfecdec.c: rtpulpfecdec: fix buffer leak when packet is recovered from storage Exposed by rtpulpfecdec_recovered_from_storage test. 2019-03-06 17:35:58 +0000 Tim-Philipp Müller * tests/check/elements/rtpulpfec.c: tests: rtpulpfec: fix buffer leak in unit test This freed wrapped memory instead of the GstMemory or buffer. 2019-03-06 17:33:23 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: fix caps leak Exposed by rtp_h264depay_bytestream() unit test. 2019-03-06 17:28:57 +0000 Tim-Philipp Müller * tests/check/elements/rtpjitterbuffer.c: tests: rtpjitterbuffer: fix leaks in new test_push_eos() test 2019-03-06 17:26:23 +0000 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/meson.build: tests: states: blacklist gtk sinks for state change test gtk_init() throws GLib-GIO-WARNING **: unknown schema extension 'd' unrelated to our test environment. 2019-03-06 17:26:03 +0000 Tim-Philipp Müller * tests/check/elements/.gitignore: * tests/examples/rtp/.gitignore: tests: .gitignore more test and example binaries 2019-03-05 15:26:45 +1100 Matthew Waters * ext/gtk/gstgtkglsink.c: * ext/gtk/gtkgstglwidget.c: gtkgl: Also try retrieving an EGL context from Gdk with X11 Some embedded platforms will use EGL instead of GLX within the X11 ecosystem. 2019-03-04 09:07:30 +0000 Tim-Philipp Müller * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * meson.build: Back to development 2019-02-25 11:23:56 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: fix AV1 caps when there's no codec_data There is no "byte-stream" format for AV1 in Matroska, this was probably cargo-culted from H.264. codec_data / CodecPrivate is now mandatory for AV1 in Matroska[*], but there are sample files out there which don't have it (e.g. some Elecard ones). [*] https://github.com/Matroska-Org/matroska-specification/blob/master/codec/av1.md#codecprivate-1 2019-02-28 08:52:28 +0000 Tim-Philipp Müller * tests/meson.build: meson: don't build icles when tests are disabled They are manual tests, so let them be controlled via the tests option. 2019-02-27 15:39:12 +0100 Marc Leeman * gst/rtpmanager/rtpsource.c: rtpsource: small spell correct === release 1.15.2 === 2019-02-26 11:47:29 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.15.2 2019-02-26 11:47:29 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2019-02-26 11:47:25 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/fur.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update translations 2019-02-22 12:22:04 -0300 Mauro Carvalho Chehab * sys/v4l2/gstv4l2object.c: v4l2: accept Bayer as possible input/output for V4L2 codecs A V4L2 transform codec may input/output data on Bayer format. Add support for that. 2019-02-22 12:22:44 -0300 Mauro Carvalho Chehab * sys/v4l2/v4l2_calls.c: v4l2: fix a typo on a debug message at v4l2_calls suppored -> supported 2019-02-25 19:08:08 +1100 Matthew Waters * sys/v4l2/gstv4l2videodec.c: v4l2dec: also remove the colorimetry and chroma-site fields If a different format is chosen, then these values are incorrect. 2019-02-22 16:02:12 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpsession.c: rtpsession: Fix EOS forwarding So far we assumed that if all sources are bye, this meant we needed to send an EOS on the RTCP sink. The problem is that this case may happens if we only had one internal source and it detected a collision. So now we limit the EOS forwarding to when there is a send_rtp_sink pad and that this pad has received EOS. We don'tcheck the recv_rtp_sink since the code does not wait for the bye to be send before sending EOS to the RTCP src pad. 2019-02-25 01:12:56 +1100 Jan Schmidt * gst/wavparse/gstwavparse.c: wavparse: Declare support for RF64 RF64 encode support was added to wavenc quite some time ago, but not declared in wavparse. It seems wavparse can decode it though, so add it to the sink pad. The RF64 support was added in https://bugzilla.gnome.org/show_bug.cgi?id=735627 2019-02-12 18:28:40 -0500 Nicolas Dufresne * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtp: Add property to disable RTCP reports per internal rtpsource This is useful when implementing custom retransmission mechanism like RIST to prevent RTCP from being produces for the retransmitted SSRC. This would also be used in general for various purpose when customizing an RTP base pipeline. 2019-02-12 18:26:21 -0500 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Emit on-new-sender-ssrc for RTX ssrc also 2019-01-15 18:04:09 -0500 Olivier Crête * tests/check/elements/rtpjitterbuffer.c: rtp jitterbuffer test: Test for queue filling 2019-01-11 17:53:43 -0500 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: Limit size to 2^15 packets If it goes over 2^15 packets, it will think it has rolled over and start dropping all packets. So make sure the seqnum distance is not too big. But let's not limit it to a number that is too small to avoid emptying it needlessly if there is a spurious huge sequence number, let's allow at least 10k packets in any case. 2019-02-11 11:33:32 -0500 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: There is no automatic reorder threshold 2019-01-30 10:47:49 -0300 Thibault Saunier * ext/pulse/pulsedeviceprovider.c: pulse: Post DEVICE_CHANGED on modification 2018-11-26 13:48:56 -0300 Thibault Saunier * ext/pulse/pulsedeviceprovider.c: * ext/pulse/pulsedeviceprovider.h: pulse: Mark default devices as "default" 2019-02-08 16:10:25 +0000 Ilya Smelykh * gst/flv/gstflvmux.c: flvmux: Use 8kHz sample rate for alaw/mulaw audio 2019-02-07 09:54:31 +0000 Ilya Smelykh * gst/flv/gstflvdemux.c: flvdemux: set sample rate to 8KHz for G.711 audio 2019-02-08 13:59:19 +0200 Vivia Nikolaidou * gst/isomp4/gstqtmux.c: qtmux: Only write timecode trak for video Recent changes in ccextractor were attaching timecode meta to the closed caption track. We shouldn't write timecode information for the closed caption trak. 2019-02-05 22:14:18 +0100 Jan Alexander Steffens (heftig) * configure.ac: * ext/vpx/gstvpxdec.c: * ext/vpx/meson.build: vpx: Fix build against libvpx 1.8 The deprecated debug visualizer was removed. 2019-02-05 19:41:51 +0530 Nirbheek Chauhan * tests/check/elements/rtph264.c: * tests/check/elements/rtph265.c: * tests/check/elements/rtpulpfec.c: misc: Fix warnings on Cerbero's mingw (gcc 4.7) error: this decimal constant is unsigned only in ISO C90 [-Werror] 2019-02-06 14:43:18 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Deal with not being able to convert a format to caps It is possible that PulseAudio adds formats that are not yet supported in pulsesink, and in those cases, we want to gracefully skip them rather than cause an assert on a NULL caps. 2019-01-17 09:22:18 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Remove trailing '\n' in debug 2019-02-05 15:27:49 +1100 Matthew Waters * ext/qt/gstqtgl.h: qmlgl: Fix opengl header guard changes again Reapply 3d708a5bfa8961cc37671bc3226976dfc9ba50ad in the correct place after the iOS additions. 2019-02-02 02:29:10 +0100 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: use the correct segment seqnum 2019-02-02 02:26:47 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: use the correct segment seqnum 2019-02-02 02:24:01 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: rtpsession: use the correct segment seqnum 2019-01-26 10:35:31 -0300 Thibault Saunier * gst/flv/gstflvdemux.c: flvdemux: Do not error out if the first added and chained pad is not linked And let it the oportunity to get its other pad linked Example: ``` $ gst-launch-1.0 uridecodebin uri=file:///home/thiblahute/gst-validate.save/gst-integration-testsuites/testsuites/../medias/defaults/flv/819290236.flv caps=audio/x-raw expose-all-streams=FALSE ! fakesink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error. Additional debug info: ../subprojects/gst-plugins-good/gst/flv/gstflvdemux.c(2760): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: streaming stopped, reason not-linked (-1) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... ``` 2019-01-16 23:54:25 -0800 Christopher Snowhill * gst/matroska/webm-mux.c: webmmux: allow resolutions above 4096 Modify the caps string to allow width and height greater than 4096. There is no need to restrict it since the matroska format allows the width and height values to be up to eight bytes long, and this also applies to the webm subset of the format. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/550 2019-02-01 14:27:11 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: qmlgl plugin iOS definitions Tested with cross-ios-arm64 and cross-ios-x86, since those two are the only archs shipped with the official Qt binaries. 2019-02-01 14:27:11 +0530 Nirbheek Chauhan * ext/qt/gstqtgl.h: qt: Don't define GLsync inside gstglfuncs.h This was originally added for fixing conflicting definitions between Android and Qt, but times have changed and now this breaks the build on iOS: [...]/OpenGLES.framework/Headers/ES3/gl.h:1006:48: error: unknown type name 'gst_qt_GLsync' GL_API void GL_APIENTRY glGetSynciv (GLsync sync, GLenum pname, GLsizei bufSize, GLsizei* length, GLint* values) OPENGLES_DEPRECATED(ios(7.0, 12.0), tvos(9.0, 12.0)); ^ ../ext/qt/gstqtgl.h:49:16: note: expanded from macro 'GLsync' #define GLsync gst_qt_GLsync ^ 6 errors generated. Instead, we simply avoid defining GLsync ourselves if we're using Qt. 2019-02-01 14:27:11 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Fix indentation in qt plugin and add a FIXME comment 2019-01-26 21:02:27 -0500 Nicolas Dufresne * gst/rtp/gstrtph265depay.c: rtph265depay; Fix handling of marker on aggregated packet When multiple nals are aggrgated, the marker bit should be associated only with the last NAL of the packet. Otherwise we may break rendering in with AU alignment. 2019-01-26 21:01:08 -0500 Nicolas Dufresne * gst/rtp/gstrtph264depay.c: rtph264depay: Fix handling or marker on STAP-A Only forward the marker for the last NAL of the STAP-A. Otherwise each NAL endup being assumed to be a full frame which may break rendering. Fixes 557 2019-01-27 09:19:00 -0500 Nicolas Dufresne * tests/check/elements/rtph265.c: test: h265depay: Add todo for testing aggregate packets with marker We are missing a sample to test this, but a fix has been made, so add a todo. 2019-01-26 20:42:40 -0500 Nicolas Dufresne * tests/check/elements/rtph264.c: test: rtph264depay: Check handling of STAP-A marker Related to #557 2019-01-31 15:23:43 +0530 Nirbheek Chauhan * tests/check/meson.build: meson: orc-test is not required This is especially never available on iOS. 2019-01-30 19:44:01 +0900 Seungha Yang * meson.build: * tests/check/meson.build: meson: Add support orc fallback Allow fallback to orc subproject if any. Additionally 'dependencies' keyword is removed from find_library, because it's invalid keyword for find_library. 2019-01-17 21:06:54 +0100 Mathieu Duponchelle * ext/gdk_pixbuf/gstgdkpixbufdec.c: gdkpixbufdec: always output a TIME segment It makes no sense for a decoder to output a BYTES segment, and many elements one would plug downstream of a video decoder assume the segments they receive are in TIME format, for example this fixes: gst-validate-1.0 filesrc location=opacity01.svg ! gdkpixbufdec ! \ videobalance ! videoconvert ! fakesink In that case, videobalance was emitting an assertion when trying to call gst_object_sync_values() 2019-01-29 12:12:51 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Add macOS definitions for qmlgl plugin Tested with Cerbero. 2019-01-29 12:12:51 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Fix building of qmlgl plugin on Android Needs gnustl for C++ STL support, which is the GNU STL on Android API 19 and older, and is a wrapper for the llvm-libc++ STL on newer APIs. QtGui C++ templates use GL functions, so GLESv2 is needed at link time 2019-01-24 16:21:12 -0500 Vincent Penquerc'h * gst/interleave/deinterleave.c: deinterleave: Allow switching between 1 channel configs regardless of whether they're positioned, since positioning with a 1 channel stream doesn't change anything. 2019-01-22 11:45:49 +0530 Nirbheek Chauhan * configure.ac: configure.ac: Fix Qt Android integration The Qt Android integration is now signalled with HAVE_QT_ANDROID See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/86 2018-12-18 14:46:25 -0500 Xavier Claessens * ext/soup/meson.build: Meson: fallback to libsoup subproject 2019-01-22 12:52:25 +0000 Tim-Philipp Müller * meson.build: meson: detect opengl api from -base .pc files correctly There was a mismatch between the .pc files generated by autotools and by meson that would lead to meson not detecting that opengl api is available even though it is, if -base was built with autotools. The mismatch has now been rectified in -base, so we need to update for that. This is mostly for consistency, this problem didn't seem to affect anything in -good. See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/871 2019-01-22 09:51:33 +0000 Tim-Philipp Müller * sys/ximage/meson.build: meson: ximage: check for XShmAttach() Fixes FIXME. 2019-01-22 09:32:31 +0000 Tim-Philipp Müller * meson_options.txt: * sys/ximage/meson.build: meson: add options for ximagesrc xshm, xfixes, xdamage checks And rename x11 option to ximagesrc. Fixes #553 2019-01-21 11:53:53 +0200 George Kiagiadakis * ext/qt/README.md: qmlgl: add README.md with information on building for non-linux platforms with qmake 2019-01-19 15:46:41 +0100 George Kiagiadakis * ext/qt/meson.build: qmlgl: meson: fix theoretical support for building for android The android code path is slightly different than the EGLFS one, so I added previously a HAVE_QT_ANDROID define for use with qmake. Here I also add it in meson, although I expect nobody will ever use meson to build this, as it's complicated. 2019-01-19 15:37:45 +0100 George Kiagiadakis * ext/qt/qtplugin.pro: qmlgl: qmake: add support for MacOS target 2019-01-19 15:21:43 +0100 George Kiagiadakis * ext/qt/qtplugin.pro: qmlgl: qmake: remove cerbero's include dir from the include path pkg-config should do it's job here, this is unnecessary and implies using cerbero 2019-01-19 15:19:26 +0100 George Kiagiadakis * ext/qt/gstqtgl.h: * ext/qt/qtplugin.pro: qmlgl: qualify Qt includes with their module and remove module include dir from the .pro file it is perfectly legal to use the style of includes with Qt and it avoids the need for having the module's include dir in the include path 2019-01-19 15:10:09 +0100 George Kiagiadakis * ext/qt/qtplugin.pro: qmlgl: qmake: don't link against QtWidgets, it's not used 2019-01-19 15:07:44 +0100 George Kiagiadakis * ext/qt/gstqtglutility.cc: * ext/qt/qtplugin.pro: qmlgl: qmake: fix building for android 2019-01-19 02:39:32 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Generate pkg-config file for qmlgl plugin 2019-01-17 16:26:56 +0100 Victor Toso * tests/check/elements/rtp-payloading.c: tests: rtp-payloading avoid -Wmaybe-uninitialized More false positives as both of them are initialized in the line before they are used, wrapped with fail_unless() check. 2019-01-17 16:19:40 +0100 Victor Toso * tests/check/elements/matroskamux.c: tests: matroskamux avoid -Wmaybe-uninitialized False positive for the three variables but some warnings like: ../tests/check/elements/matroskamux.c:875:10: warning: 'chapters_offset' may be used uninitialized in this function [-Wmaybe-uninitialized] *index = chapters_offset; ~~~~~~~^~~~~~~~~~~~~~~~~ The above is false positive as there is a gboolean to check if it was initialized or not (found_chapters_declaration). 2018-05-28 14:39:53 +0530 Arun Raghavan * ext/pulse/pulseutil.c: pulse: Fix format info to caps conversion for PCM 2019-01-18 12:27:34 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: Revert "pulsesrc: Move to extended stream API" This reverts commit 4d67d1bd16bcf25acf89d8acd952badcd5b9a657. Using the extended API for the capture path depends on a fix in PulseAudio (https://gitlab.freedesktop.org/pulseaudio/pulseaudio/merge_requests/49). Until then, let's go back to the standard API. Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/552 2019-01-18 14:41:14 +0530 Nirbheek Chauhan * ext/qt/meson.build: meson: Search for qmake-qt5 before qmake The canonical name for the binary is qmake-qt5, and qmake is the generic name that can also be a qt4 qmake. 2019-01-17 15:30:25 +0100 Guillaume Desmottes * sys/v4l2/gstv4l2.c: v4l2: mark caps from probe as MAY_BE_LEAKED 2019-01-15 18:06:11 +0100 Guillaume Desmottes * sys/v4l2/gstv4l2transform.c: v4l2transform: fix cdata caps leaks The cdata structure was freed but not its caps. It was already done in gst_v4l2_video_dec_subclass_init() and gst_v4l2_video_enc_subclass_init(). === release 1.15.1 === 2019-01-17 01:59:28 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.15.1 2019-01-17 01:59:28 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2019-01-17 01:59:18 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/fur.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update translations 2019-01-16 14:11:44 +0200 Sebastian Dröge * ext/gtk/gtkgstglwidget.c: gtk/gl: Only unbind buffers/vertex attrib arrays if we can't directly bind the vertex array to 0 Binding the vertex array to 0 will unbind everything else already. In the previous order older versions of the Intel GL driver caused errors to be printed for every single call when disabling the vertex attrib arrays after binding the vertex array to 0. 2019-01-16 00:57:46 +0000 Tim-Philipp Müller * tests/check/meson.build: meson: enable tests for orc code 2018-11-29 16:07:08 +0100 Patrick Radizi * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: send GstRTSPSrcTimeout message on timeout The GstRTSPSrcTimeout message is sent by the rtspsrc when it receives the on-timeout signal from rtpsession. This can be used by an application for error handling. https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/499 2019-01-09 17:52:28 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Handle the encoder metadata the same as metadatacreator And store it in our ENCODER tag. 2019-01-09 17:48:36 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Add encoder metadata to the header And also add a property for setting this. By default it has the same value as the metadatacreator metadata. Various software is using encoder instead of metadatacreator, others are using them both for different purposes. As such it's useful to have support for setting both here. 2018-05-28 14:41:05 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulsesrc: Move to extended stream API This is needed as a precursor to allowing capture of IEC61937 formats. We now also need to include the channel map while converting format info to caps so that a correct channel mask is generated for pulsesrc's caps. 2019-01-09 16:27:16 +0100 Jan Alexander Steffens (heftig) * tests/check/elements/rtph265.c: test: rtph265pay: Verify we only mark the last fragment 2019-01-09 16:24:54 +0100 Jan Alexander Steffens (heftig) * tests/check/elements/rtph265.c: test: rtph265pay: Use a bigger test frame The existing frame's last slice is too small to be used for fragmentation tests. 2019-01-09 15:59:16 +0100 Jan Alexander Steffens (heftig) * tests/check/elements/rtph264.c: test: rtph264pay: Verify we only mark the last fragment 2019-01-09 16:25:36 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph265pay.c: rtph265pay: Only mark the last fragment of an AU Commit e721071dcac9f231e5e10b4bb31323658a6cdd1a removed the check for the end of fragmentation. As a result, all fragments of an AU's last NALU were marked. 2019-01-09 15:56:51 +0100 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: rtph264pay: Only mark the last fragment of an AU Commit 4add820cce278213ede3d5fce427ea92e0619b6f removed the check for the end of fragmentation. As a result, all fragments of an AU's last NALU were marked. Potential fix for https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/540 2019-01-09 11:48:52 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Refactor part preparation code and remove "prepared" signal from reader helper object We don't need a special signal anymore but can directly work with async-done 2019-01-09 11:42:36 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: splitmuxsrc: Implement state change asynchronously instead of blocking Blocking in change_state() is a recipe for disaster, even more so if we wait for another thread that also calls into various element API and could then lead to deadlocks on e.g. the state lock. 2019-01-05 23:10:46 +0400 Marc-André Lureau * ext/pulse/pulsesrc.c: pulsesrc: fix checking for invalid stream index PA_INVALID_INDEX, the default value, is unfortunately !0. Setting the volume before the stream is created will put the ring buffer in error state. Unfortunately, that's what spice-gtk does. 2018-12-20 12:14:46 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Split CEA608 buffers correctly so that each output buffer represents a single frame 2018-12-20 11:45:36 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Refactor buffer pushing into its own function 2018-12-20 11:31:58 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Extract CEA608 framerate from the (first) video stream EA608 closed caption tracks are a bit special in that each sample can contain CCs for multiple frames, and CCs can be omitted and have to be inferred from the duration of the sample then. As such we take the framerate from the (first) video track here for CEA608 as there must be one CC byte pair for every video frame according to the spec. For CEA708 all is fine and there is one sample per frame. 2018-12-30 21:55:28 +0900 Seungha Yang * tests/check/meson.build: tests: Enable more unit tests on Windows 2018-12-30 21:54:44 +0900 Seungha Yang * tests/check/elements/audioamplify.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiopanorama.c: * tests/check/elements/avimux.c: * tests/check/elements/avisubtitle.c: * tests/check/elements/capssetter.c: * tests/check/elements/level.c: * tests/check/elements/matroskamux.c: * tests/check/elements/multifile.c: * tests/check/elements/qtdemux.h: * tests/check/elements/qtmux.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/shapewipe.c: * tests/check/elements/spectrum.c: * tests/check/elements/splitmux.c: * tests/check/elements/udpsrc.c: * tests/check/elements/videobox.c: * tests/check/elements/videocrop.c: * tests/check/elements/videofilter.c: * tests/check/elements/videomixer.c: * tests/check/elements/wavpackparse.c: * tests/check/elements/y4menc.c: * tests/check/generic/states.c: tests: Remove pointless unistd.h include 2018-12-26 20:27:58 +0900 Seungha Yang * gst/matroska/matroska-demux.c: matroskademux: Don't leak allocated index memory Don't forget to free returned memory from _search_pos() 2018-12-25 15:31:44 +0100 Tim-Philipp Müller * tests/files/Makefile.am: tests: dist new rtph265.rtp file Fixes make distcheck. 2018-12-25 14:51:38 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: * gst/audiofx/gststereo.c: * gst/audiofx/meson.build: audiofx: add stereo element which was moved from -bad to build Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/457 2018-12-25 13:07:23 +0100 Tim-Philipp Müller Move stereo plugin from -bad https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/457 2018-12-22 17:55:51 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: Offset correction for track language code parsing The duration field being a uint64, is stored in 8 bytes, not 4. So the offset of the following field, language code, needs to be updated accordingly so that the parsed language code is not garbage. 2018-12-21 10:59:22 +0100 Juan Navarro * gst/rtsp/gstrtspsrc.c: rtspsrc: Accept NULL for "port-range" property The documentation of "port-range" implies that passing NULL should be valid, but currently it is not. Without this check, the sscanf() call will crash. 2018-12-19 14:28:54 +0100 Mathieu Duponchelle * docs/plugins/gst-plugins-good-plugins.signals: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtpbundle.c: * tests/check/meson.build: * tests/examples/rtp/.gitignore: * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-rtpbundle.c: * tests/examples/rtp/meson.build: * tests/examples/rtp/server-rtpbundle.c: Revert "rtpbin: receive bundle support" This reverts commit dcd3ce9751cdef0b5ab1fa118355f92bdfe82cb3. This functionality was implemented for gstopenwebrtc, but it turned out this was not actually needed for webrtc bundling support, as shown in webrtcbin. It also doesn't correspond to any standards. This is an API break, but nothing should actually depend on this, at least not for its initial purpose. Changes in rtpbin.c were reverted manually, to preserve some refactoring that had occurred in the original commit. Fixes #537 2018-12-19 11:36:37 -0500 Nicolas Dufresne * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: rtph264pay/rtph265pay: Fix use after free We can't assume a buffer that has been pushed in the adapter is still valid. This fixes a use after free detect when running test on jenkins. 2018-12-19 22:51:11 +0900 KimTaeSoo * tests/check/pipelines/tagschecking.c: tagschecking: Use gst_message_parse_warning in case of GST_MESSAGE_WARNING Bus message handler of tags checking unit test uses gst_message_parse_error() in case of GST_MESSAGE_ERROR and GST_MESAGE_WARNING. If gst_message_parse_error() is called in case of GST_MESSAGE_WARNING, assert occurs. So modified to use gst_message_parse_warning() in case of GST_MESSAGE_WARNING. 2018-12-19 09:51:10 -0500 Nicolas Dufresne * tests/check/Makefile.am: test: rtph264/265: Add libgstrtp in auto-tool makefile 2018-12-18 12:43:30 -0500 Nicolas Dufresne * tests/check/Makefile.am: * tests/check/elements/rtph265.c: * tests/check/meson.build: * tests/files/h265.rtp: test: rtph265: Copy and port tests from rtph264 This copy and port all the relevant tests from rtph264. 2018-12-14 17:54:36 -0500 Nicolas Dufresne * tests/check/elements/rtph264.c: test: rtph264depay: Check the marker is converted to flag 2018-12-14 17:53:17 -0500 Nicolas Dufresne * tests/check/elements/rtph264.c: test: rtph264depay: Check that EOS drains the depayloaded In AU mode, the depayloader may have accumulated NALs, test that these NALs are drained and not dropped. 2018-12-14 15:30:21 -0500 Nicolas Dufresne * tests/check/elements/rtph264.c: test: rtph264pay: Add tests for marker bit Test that marker bit is transferred when input buffer has the marker flag set but also that it's set whenever the payloader receives complete AU. 2018-12-13 15:57:24 -0500 Nicolas Dufresne * tests/check/elements/rtph264.c: test: rtph264pay: Verify slices timestamp This test make sure that timestamps are properly transfered to each NALU. 2018-12-04 16:06:15 -0500 Nicolas Dufresne * tests/check/elements/rtph264.c: * tests/check/meson.build: test: rtph264pay: Add reserved nals test 2018-12-18 13:16:44 -0500 Nicolas Dufresne * gst/rtp/gstrtph265pay.c: rtph265pay: Don't wait for next nal when input is aligned This is the same as what was done on rtph264pay in the patch d5d28055c1e816e90e8c2d1151816b0c3e760ff3 2018-12-18 12:53:15 -0500 Nicolas Dufresne * gst/rtp/gstrtph265depay.c: rtph265depay: Drain on EOS event 2018-12-18 12:50:40 -0500 Nicolas Dufresne * gst/rtp/gstrtph265depay.c: rtph265depay: Factor out the code that push This will be needed to implement draining on EOS. 2018-12-17 16:48:53 -0500 Nicolas Dufresne * gst/rtp/gstrtph264depay.c: rtph264depay: Drain on EOS event 2018-12-14 18:19:42 -0500 Nicolas Dufresne * gst/rtp/gstrtph264depay.c: rtph264depay: Factor out the code that push This will be needed to implement draining on EOS. 2018-12-14 15:51:51 -0500 Nicolas Dufresne * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: rtph26xpay: Remove unused IS_ACCESS_UNIT macro This macro is not longer used. It was secretly checking if that nal was a slice, and confusingly name to that one may think it was checking if the nal is an AUD. 2018-10-03 14:14:17 -0400 Nicolas Dufresne * gst/rtp/gstrtph265pay.c: rtph265pay: Fix reading timestamps from adapter The code was reading the timestamp from the adapter before pushing the new buffer into it. As a side effect, if the adapter was empty, we'd end up using an older timestamp. In alignment=au, it means that all timestamp was likely one frame in the past, while in alignment=nal, with multiple slices per frame, the first slice would have the timestamp of the previous one. 2018-10-03 13:46:08 -0400 Nicolas Dufresne * gst/rtp/gstrtph265depay.c: rtph265pay: Forward the marker bit as buffer flag We have a buffer flag to represent the marker bit (when present). Forward this bit by setting the buffer flag accordingly. 2018-10-03 13:44:56 -0400 Nicolas Dufresne * gst/rtp/gstrtph265pay.c: rtph265pay: Properly set the marker bit The marker bit is used for efficient decoding. The assumption that it should be set on the AUD is wrong, since the AUD is conceptually starts the frame, while the marker is to indicate the end. So properly set the marker bit as soon as we know we are ending an AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER flag. 2018-09-25 11:49:52 -0400 Nicolas Dufresne * gst/rtp/gstrtph264pay.c: rtph264pay: Fix reading timestamps from adapter The code was reading the timestamp from the adapter before pushing the new buffer into it. As a side effect, if the adapter was empty, we'd end up using an older timestamp. In alignment=au, it means that all timestamp was likely one frame in the past, while in alignment=nal, with multiple slices per frame, the first slice would have the timestamp of the previous one. 2018-09-24 15:31:12 -0400 Nicolas Dufresne * gst/rtp/gstrtph264pay.c: rtph264pay: Properly set the marker bit The marker bit is used for efficient decoding. The assumption that it should be set on the AUD is wrong, since the AUD is conceptually starts the frame, while the marker is to indicate the end. So properly set the marker bit as soon as we know we are ending an AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER flag. 2018-09-24 15:27:41 -0400 Nicolas Dufresne * gst/rtp/gstrtph264depay.c: rtph264depay: Forward the marker bit as buffer flag We have a buffer flag to represent the marker bit (when present). Forward this bit by setting the buffer flag accordingly. 2018-09-21 20:22:43 +0000 Nicolas Dufresne * gst/rtp/gstrtph264pay.c: rtph264pay: Protect against use of reserved NAL types Don't allow external encoder to use one of the reserved NAL type implicated in NAL aggreation. These out-of-spec NAL types, if passed from the outside world will lead to an invalid RTP payload being created. 2018-12-07 21:46:12 +0900 Seungha Yang * meson.build: * tests/check/meson.build: * tests/meson.build: tests: Enable unit test on Windows Allow run some unit tests on Windows. * Remove hardcoded path separator in whitelist env for Meson to choose OS-specific separator automatically (i.e., ';' for windows and ':' for *nix) * Add dependency explicitly for some test cases, otherwise plugins couldn't be loaded on uninstalled environment of Windows. 2018-12-18 20:39:40 +0900 Seungha Yang * meson.build: * tests/check/meson.build: meson: Prefer to use join_paths() over '/' ... to avoid mixing '/' and '\' in a path string on Windows. 2018-12-17 18:04:37 +0000 Jonny Lamb * tests/check/elements/rtpulpfec.c: rtpulpfec: stop and start the harness when setting error-after gstreamer!55 makes some changes to how the `error-after` counter works which breaks this test. This change makes the test not rely on the ability to alter `error-after` at runtime and explicitly stops and starts the harness before pushing data. An alternative would be to add another argument to `harness_rtpulpfecdec` to set `error-after` on construction but that's slightly more long-winded. so I went for this approach instead. Fixes #532, even though that's already closed. 2018-12-17 18:59:34 +0100 Mathieu Duponchelle * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtpaux.c: * tests/check/meson.build: tests: remove rtpaux test The initial mission statement for this test was: * demonstrate usage of the request-aux-* signals in rtpbin * test the rtx elements We have examples that serve the first use case, and better (harnessed) tests for the second use case. This test is slow and racy, it served its purpose but can now be removed. Fixes #533 2018-12-17 19:18:43 +0100 Nicola Murino * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: check difference in time from the last socket read before changing blocksize If the pipeline consumes the data slower than the available network speed, for example because sync=true, is useless to increase the blocksize and reading in too big blocksizes can cause the connection to time out Closes #463 2018-08-08 09:27:09 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: Avoid code duplication The function gst_v4l2_object_add_interlace_mode() has repeating code so it's best use a loop instead. That will make it easy and simple to add additional interlace modes in a following patch. 2018-06-27 23:20:33 +0200 Zeeshan Ali * sys/v4l2/gstv4l2object.c: v4l2: Make use of gst_video_interlace_mode_to_string() Instead of a custom map to translate the interlace modes to strings, let's make use of the base API provided. 2018-12-17 13:45:36 +0100 Nicola Murino * sys/osxaudio/gstosxcoreaudio.c: osxcoreaudio: fix typo kAudioFormatFlagIsSignedInteger is a format flags Closes #394 2018-12-17 09:33:39 +0100 Edward Hervey * ext/qt/gstqtgl.h: qtgl: Handle OPENGL header guard changes In 2018 khronos changed the gl header guards. If we don't detect this properly we would end up with plenty of symbol redifinition (since we would be importing twice the "same" header). Instead detect if the "newer" header was already included and if so define the "old" define to avoid this situation Fixes #523 2018-12-10 17:34:03 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: isomp4: Replace GST_VIDEO_CAPTION_TYPE_CEA608_IN_CEA708_RAW with CEA608_S334_1A For the demuxer we have to select line offset 0 for the time being as this information is not passed over MOV. 2018-12-13 20:45:23 -0500 Olivier Crête * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer tests: Validate the number of buffers 2018-12-13 19:17:43 -0500 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Run all timers immediately on EOS When the EOS event is received, run all timers immediately and avoid pushing the EOS downstream before this has been run. This ensures that the lost packet statistics are accurate. 2018-12-13 19:16:11 -0500 Olivier Crête * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer test: Stop jitterbuffer before pads to avoid race The teardown of the pads checks the refcount, but there are timers inside the jitterbuffer that can push things, so if we're not lucky, things could be pushed while the pads are being shut down. Putting the jitterbuffer to NULL first avoids this. 2018-11-22 10:41:29 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Stop waiting after EOS After EOS is received, it is pointless to wait for further events, specially waiting on timers. This patches fixes two cases where we could wait instead of returning GST_FLOW_EOS and trigger a spin of the loop function when EOS is queued, regardless if this EOS is the queue head or not. 2018-10-27 13:41:46 +0200 Jochen Henneberg * ext/flac/gstflacdec.c: flacdec: Use new channel count for audio info 2018-10-27 13:36:16 +0200 Jochen Henneberg * ext/flac/gstflacdec.c: flacdec: Caps may have changed on FLAC metadata change If the decoder signals metadata change we need to update the output format and negotiate with downstream elements. 2018-10-27 13:28:56 +0200 Jochen Henneberg * ext/flac/gstflacdec.c: flacdec: Reset decoder on set_format() Any call to set_format() could mean that the stream type changed so we reset the decoder and mark got_headers FALSE. 2018-12-05 18:42:55 +0100 Jochen Henneberg * gst/audioparsers/gstflacparse.c: flacparse: On sink caps change restart parser Draining the parser is not enough here, on caps change we need to reset it so it is ready to accept new caps. 2018-12-04 18:50:51 +0100 Jochen Henneberg * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: Update pad caps if inline caps change If the inlined caps change while using the same CV we need to update the source pad caps. 2018-12-14 12:21:58 +0900 Justin Kim * sys/osxvideo/meson.build: osxvideo: meson: Add dependencies by using appleframeworks Otherwise, it fails to link. gst-build#13 2018-12-07 19:09:30 +0200 Sebastian Dröge * ext/cairo/gstcairooverlay.c: cairooverlay: Optimize premultiplication/unpremultiplication loops Pull in video frame fields into local variables. Without this the compiler must assume that they could've changed on every use and read them from memory again. This reduces the inner loop from 6 memory reads per pixels to 4, and the number of writes stays at 3. 2018-12-05 19:37:13 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Put framerate into the closedcaption caps if it can be calculated from the stream Using the same calculation used for video streams. 2018-12-05 19:31:25 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Set timescale of closedcaption tracks to the one of the main video track 2018-12-05 17:24:13 -0300 Thibault Saunier * common: Automatic update of common submodule From ed78bee to 59cb678 2018-11-19 18:20:52 +0000 Maciej Wolny * gst/flv/gstflvmux.h: * sys/v4l2/gstv4l2allocator.h: Remove duplicate declarations This causes 'redefinition of typedef ...' errors on GCC 4.5.3 2018-11-30 23:56:12 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: rtpssrcdemux: fix uninstalled autotools build and distcheck 2018-11-30 19:29:30 +0100 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: set need_segment after a second moov stream.segment should be updated with the values of the current edit list, also when a new `moov` is received. Unfortunately this was not being the case because of an early return. As a consequence of this bugs, no end of movie clipping was being performed on the new moov and no segment event was being emitted. When performing stream switching (e.g. in MSE) the new moov may have a different edit list. This is often the case when switching between baseline H.264 (which lacks B-frames) and more demanding profiles. For this reason it's important to emit a new segment in order to be able to get matching stream times. 2018-11-29 22:42:34 +0100 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Initialize QtDemuxStream.segment in its constructor This patch moves the initialization of QtDemuxStream.segment from gst_qtdemux_add_stream() to _create_stream(). This ensures the segment is always initialized when the stream is created. Otherwise the segment format is left as GST_FORMAT_UNDEFINED in the case were a track is reparsed and qtdemux_reuse_and_configure_stream() is called instead of gst_qtdemux_add_stream(). (See qtdemux_expose_streams() in the non streams-aware case.) 2018-11-29 13:48:33 +0100 Miguel Paris * gst/rtpmanager/rtpsession.c: rtpsession: properly handle rtcp_feedback_retention_window - Consider GST_CLOCK_TIME_NONE as not to be used. - Complete "rtcp-feedback-retention-window" property getter/setter implementation. 2018-11-29 13:02:53 +0100 Miguel Paris * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsource: properly prune RTCP packets out of feedback_retention_window Closes #522 2018-11-29 13:01:44 +0100 Miguel Paris * gst/rtpmanager/rtpsource.c: rtpsource: properly compare buffer PTSs 2018-11-29 12:58:18 +0100 Miguel Paris * gst/rtpmanager/rtpsource.c: rtpsource: retain_rtcp_packet: warning if invalid running_time 2018-11-29 12:55:38 +0100 Miguel Paris * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: properly set the running_time for rtcp packet info 2018-11-29 14:54:06 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Rename confusingly name lock macros This is an extra internal recurisve lock use to avoid having to take both sink pad streams lock all the time. This patch renamed it INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream GST_PAD API. 2018-11-28 17:14:11 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Hold on internal stream lock while pushing sticky This reverts "6f3734c305 rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock" and actually hold on the internal stream lock. This prevents in some needed case having a second streaming thread poping in and messing up event ordering. 2018-11-27 17:10:57 -0500 Nicolas Dufresne * tests/check/Makefile.am: * tests/check/elements/rtpssrcdemux.c: * tests/check/meson.build: test: rtpssrcdemux: Test event forwarding This the first unit test of this element. It adds a test that verify that events are forwarded correctly. 2015-11-04 12:52:17 +0100 Matej Knopp * gst/matroska/matroska-demux.c: matroskademux: fix handling of MS ACM audio Pass riff codec-data as strf, not strd, which is where gst_riff_create_audio_caps() expects the WAVEFORMATEXTENSIBLE data. https://bugzilla.gnome.org/show_bug.cgi?id=757583 Fixes #234 2018-11-28 05:52:16 +0200 Jordan Petridis * gst/matroska/matroska-demux.c: * gst/rtp/gstrtpg722pay.c: * gst/rtpmanager/gstrtpmux.c: * gst/udp/gstudpsrc.c: * sys/v4l2/gstv4l2jpegenc.c: * tests/check/elements/rtpmux.c: * tests/check/elements/rtpsession.c: Run gst-indent through the files This is required before we enabled an indent test in the CI. https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33 2018-11-26 08:10:24 -0300 Thibault Saunier * gst/videocrop/gstaspectratiocrop.c: aspectcropration: Fix potential unref of NULL pointer 2018-11-25 11:31:11 -0300 Thibault Saunier * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstaspectratiocrop.h: aspectcropratio: Set caps from the streaming thread on property changes Otherwise it might lead to deadlocks See https://gitlab.gnome.org/GNOME/pitivi/issues/2259 Closes #518 2018-11-23 14:01:35 -0500 Nicolas Dufresne * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Forward serialized events to all pads While forwarding serialized event, we use gst_pad_forward() function. In the forward callback (GstPadForwardFunction) we always return TRUE. Returning true there will stop the dispatching procedure. As a side effect, only one events is receiving the events. This breaks when sending EOS from the applicaiton, it also breaks the latency tracer. 2018-11-24 19:13:28 +0900 Seungha Yang * meson.build: meson: Specify encoding to UTF-8 when building with MSVC Use build arguments consistent with core and -base. This can also remove noisy "C4819" warning of non-us locale MSVC. 2018-11-21 15:11:00 -0500 Xavier Claessens * meson.build: Check for zlib header 2018-11-21 18:53:39 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: v4l2: Properly fix Android build The previous patch did not even compile on any possible platform or C standard. That commit also didn't have a proper commit message. Android ships Linux with a different signature for ioctl. They first released an ioctl with int as request type, and later "fixed" it by adding an override with unsign, which is still not matching Linux and BSD implementation which uses unsigned long int. 2018-11-21 16:11:02 -0500 Xavier Claessens * sys/v4l2/gstv4l2object.h: Fix ioctl() signature on Android 2018-10-09 16:43:08 -0400 Xavier Claessens * meson.build: Fix zlib detection when there is no pkg-config file 2018-11-19 20:05:39 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: Expose the correct max rate that we support PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it supports. We were previously exposing a maximum rate of INT_MAX, which is incorrect, but worked because nothing was really using a rate greater than 384000 kHz. While playing DSD data, we hit a case where there might be very high sample rates (>1MHz), and pulsesink fails during stream creation with such streams because it erroneously advertises that it supports such rates. Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in the caps string. Instead, we fix up the rate to what we actually support whenever we use our macro caps. 2018-11-14 08:57:55 +0100 Alicia Boya García * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: Defer seeks received before GST_MATROSKA_READ_STATE_DATA This patch enables matroskademux to receive seeks before it reaches GST_MATROSKA_READ_STATE_DATA. Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/514 This also enables receiving seeks in the element READY state. When such a seek is received, it is stored to be later handled when GST_MATROSKA_READ_STATE_DATA is reached. 2018-10-16 12:38:46 +0200 Linus Svensson * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: rtpsession: Implement reset Reset RTPSession when rtpsession changes state from PAUSED to READY. Without this change, a stored last_rtptime in RTPSource could interfere with RTP timestamp generation in RTCP Sender Report. Fixes #510 2018-11-06 15:05:54 +0100 Linus Svensson * tests/check/elements/rtpsession.c: rtpsession: test: Plug memory leak 2018-11-13 00:37:11 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpfunnel.c: * gst/rtpmanager/gstrtpfunnel.h: rtpfunnel: Stop using G_DECLARE_FINAL_TYPE Fixes #516 2018-11-12 13:42:29 +0200 Jordan Petridis * .gitlab-ci.yml: Add Gitlab CI configuration This commit adds a .gitlab-ci.yml file, which uses a feature to fetch the config from a centralized repository. The intent is to have all the gstreamer modules use the same configuration. The configuration is currently hosted at the gst-ci repository under the gitlab/ci_template.yml path. Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29 2018-10-18 22:23:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Read driver selected interlace mode If there was no interlace-mode field in the caps. Read back the value selected by the driver. This way, if the driver does not support progressive, then it will automatically negotiate the returned mode unless this mode is not supported by GStreamer. This method was already used for colorimetry. Just like colorimetry, the interlace mode is not longer probed by v4l2src dues to performance issues. Fixes #511 2018-05-17 21:58:25 +1000 Matthew Waters * gst/matroska/matroska-demux.c: matroska: implement preliminary support for the bitrate query Return the size / total duration as a ballpark estimate. https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60 2018-05-17 21:53:56 +1000 Matthew Waters * gst/isomp4/qtdemux.c: isomp4: add preliminary support for the bitrate query Return the upstream size over the duration as a first estimate. https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60 2018-11-06 23:02:21 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Sink jitterbuffer/storage before passing as parameters to signals Otherwise signal handlers from bindings will take ownership of them as they are still floating, and we won't own a reference inside rtpbin anymore. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/515 2018-10-27 18:00:52 +0100 Havard Graff * tests/check/elements/flvmux.c: flvmux: Test that timestamps are always increasing Decreasing timestamps break rtmpsink. With contributions from Olivier Crête. https://bugzilla.gnome.org/show_bug.cgi?id=796382 2018-10-27 19:27:12 +0100 Olivier Crête * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Force timestamps to always be increasing https://bugzilla.gnome.org/show_bug.cgi?id=796382 2018-11-05 05:36:26 +0000 Matthew Waters * .gitmodules: Update common submodule location Remove the git directory 2018-11-05 12:16:46 +0800 Haihao Xiang * .gitmodules: * gst-plugins-good.doap: Clone the code from gitlab This fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/513 2018-11-01 20:37:12 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Ignore corrupted CTTS box If ctts (CompositionOffsetBox) has larger sample_offset (offset between PTS and DTS) than (2 * duration) of the stream, assume the ctts box to be corrupted and ignore the box. https://bugzilla.gnome.org/show_bug.cgi?id=797262 2018-10-23 09:45:36 +0100 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: Implement SEGMENT query https://bugzilla.gnome.org/show_bug.cgi?id=797313 2018-10-23 09:42:21 +0100 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Implement SEGMENT query https://bugzilla.gnome.org/show_bug.cgi?id=797313 2018-10-28 17:12:59 +0000 Olivier Crête * gst/dtmf/gstdtmfsrc.c: dtmfsrc: Declare output as interleaved This element doesn't support planar audio yet. 2018-10-28 14:09:21 +0000 Nirbheek Chauhan * tests/icles/meson.build: meson: Add some missing test dependencies Without these dependencies, the enumtype may not be generated when the test is built, which will cause a compile failure. 2018-10-28 14:07:54 +0000 Nirbheek Chauhan * tests/check/meson.build: meson: Cleanup old FIXMEs that relied on meson changes 2018-10-16 17:28:00 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Allow changing the SDES at runtime Make it possible to modify the SDES in a packet at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=763502 2018-03-01 17:25:07 +0100 Alicia Boya García * gst/isomp4/atoms.c: * gst/isomp4/gstqtmux.c: qtmux: round to nearest when computing mehd and tkhd duration This fixes a bug where in some files mehd.fragment_duration is one unit less than the actual duration of the fragmented movie, as explained below: mehd.fragment_duration is computed by scaling the end timestamp of the last frame of the movie in (in nanoseconds) by the movie timescale. In some situations, the end timestamp is innacurate due to lossy conversion to fixed point required by GstBuffer upstream. Take for instance a movie with 3 frames at exactly 3 fps. $ gst-launch-1.0 -v videotestsrc num-buffers=3 \ ! video/x-raw, framerate="(fraction)3/1" \ ! x264enc \ ! fakesink silent=false dts: 999:59:59.333333334, pts: 1000:00:00.000000000, duration: 0:00:00.333333333 dts: 999:59:59.666666667, pts: 1000:00:00.666666666, duration: 0:00:00.333333334 dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333 The end timestamp is calculated by qtmux in this way: end timestamp = last frame DTS + last frame DUR - first frame DTS = = 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 = = 0:00:00.999999999 qtmux needs to round this timestamp to the declared movie timescale, which can ameliorate this distortion, but it's important that round-neareast is used; otherwise it would backfire badly. Take for example a movie with a timescale of 30 units/s. 0.999999999 s * 30 units/s = 29.999999970 units A round-floor (as it was done before this patch) would set fragment_duration to 29 units, amplifying the original distorsion from 1 nanosecond up to 33 milliseconds less than the correct value. The greatest distortion would occur in the case where timescale = framerate, where an entire frame duration would be subtracted. Also, rounding is added to tkhd duration computation too, which potentially has the same problem. https://bugzilla.gnome.org/show_bug.cgi?id=793959 2018-05-16 14:15:13 +0200 Marc Leeman * gst/udp/gstudpsrc.c: udpsrc: print information about bind_error socket error In some cases, a bind error occurs during operation. Printing the information about the problem is critical for finding the conflict https://bugzilla.gnome.org/show_bug.cgi?id=797340 2018-10-17 12:58:08 +0200 Johan Bjäreholt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-read-common.c: matroska-demux: Fix caps memleak https://bugzilla.gnome.org/show_bug.cgi?id=797326 2018-10-11 09:24:53 +0900 Wonchul Lee * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: fix typo resurect to resurrect https://bugzilla.gnome.org/show_bug.cgi?id=797273 2018-10-18 12:29:00 +0530 Amit Pandya * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2h265enc.c: * sys/v4l2/gstv4l2h265enc.h: * sys/v4l2/meson.build: v4l2videoenc: Add HEVC support Add HEVC encoder support. https://bugzilla.gnome.org/show_bug.cgi?id=797141 2018-10-19 17:37:28 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: vl42allocator: Don't dup exported dmabufs We can now use the new GstFAllocator to ask the allocator not to close the wrapped FD. This way the dup is no longer needed. 2018-10-19 17:14:15 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Don't dup imported DMABuf FD There is no specific needs to duplicate the FD. Unlike the exportation, we don't depend on code that will call close. This will make debugging easyer since the traced FD will match the exporter. 2018-10-23 13:04:34 +0200 Johan Bjäreholt * gst/matroska/matroska-ids.c: matroska-ids: Fix uninitialized memory in contexts https://bugzilla.gnome.org/show_bug.cgi?id=797327 2018-10-19 17:02:11 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Add property for providing a threshold after which we create an edit list for gaps at the start https://bugzilla.gnome.org/show_bug.cgi?id=797290 2018-10-22 12:21:54 +0100 Sebastian Dröge * gst/isomp4/atoms.c: qtmux: Correctly set tkhd width/height to the display size It was previously set to the display aspect ratio, e.g. 4x3, 16x9, etc. but should be set to the display size. This is a regression from e655d47dfce1652630fe8ff5fb6be56370087004 (1.5.1) and was correct before that. https://bugzilla.gnome.org/show_bug.cgi?id=797318 2018-10-21 11:15:15 +0900 Yeongjin Jeong * tests/check/elements/flvmux.c: tests: flvmux: Fix pushing invalid audio caps in tests Previous commit created caps with incorrect aac codec data that did not match the audio channel. https://bugzilla.gnome.org/show_bug.cgi?id=797256 2018-10-20 00:10:04 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Fix build with GLib versions < 2.54 g_ptr_array_find_with_equal_func was introduced in glib 2.54 which is a higher version than our minimum required one. https://bugzilla.gnome.org/show_bug.cgi?id=797239 2018-10-17 13:52:20 +0200 Havard Graff * tests/check/elements/rtpsession.c: rtpsession: fix up GHashTable-behavior dependent tests GHashTable iteration order changed in recent GLib, and tests were relying on that. https://mail.gnome.org/archives/desktop-devel-list/2018-October/msg00016.html 2018-10-07 20:07:39 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Don't switch active streams and old streams ... ... before the old streams is not exposed yet for MSS stream. In case of DASH, newly configured streams will be exposed whenever demux got moov without delay. Meanwhile, since there is no moov box in MSS stream, the caps will act like moov. Then, there is delay for exposing new pads until demux got the first moof. So, following scenario is possible only for MSS but not for DASH, STREAM-START -> CAPS -> (configure stream but NOT EXPOSED YET) -> STREAM-START-> CAPS (configure stream again). In above scenario, we can reuse old stream without any stream reconfigure. https://bugzilla.gnome.org/show_bug.cgi?id=797239 2018-10-07 16:43:34 +0900 Seungha Yang * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Use GPtrArray to store QtDemuxStream structure GPtrArray has less overhead than linked list and the length also can be auto updated by using it. https://bugzilla.gnome.org/show_bug.cgi?id=797239 2018-10-07 16:50:45 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Make QtDemuxStream refcounted structure This a prework for porting GPtrArray. Refcounting will help the use of g_ptr_array_new_with_free_func() with QtDemuxStream structure https://bugzilla.gnome.org/show_bug.cgi?id=797239 2018-10-06 20:19:40 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Make function foreach method friendly https://bugzilla.gnome.org/show_bug.cgi?id=797239 2018-07-26 15:25:06 -0400 Olivier Crête * gst/isomp4/qtdemux.c: qtdemux: Only set width/height in caps if they're non-0 If they are not valid, then let a downstream parser complete them. https://bugzilla.gnome.org/show_bug.cgi?id=796878 2018-08-16 12:07:30 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: fix misleading debug line 2018-06-22 16:00:11 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: Avoid warning when reporting about decryptors https://bugzilla.gnome.org/show_bug.cgi?id=796652 2018-10-17 14:15:33 +0100 Tim-Philipp Müller * gst/audiofx/meson.build: * gst/deinterlace/meson.build: * gst/videobox/meson.build: * gst/videomixer/meson.build: meson: Replace empty configuration_data() with copy keyword Use 'copy' keyword to avoid meson warning message. Note that 'copy' keyword in configure_file() is available since meson 0.47.0 https://bugzilla.gnome.org/show_bug.cgi?id=797298 2018-10-16 15:42:12 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Do not hardcode frames_of_daily_jam Apart from the obvious drawbacks of hardcoding, the drawback here was that, if we subtracted 2 frames (instead of 2.6) from the target running time, we'd request the next keyframe a bit too far into the future, which would make our files split at the wrong position. https://bugzilla.gnome.org/show_bug.cgi?id=797293 2018-10-02 19:32:47 +0300 Vivia Nikolaidou * gst/isomp4/gstqtmux.c: qtmux: Allow up to 1% of frame rate for lateness https://bugzilla.gnome.org/show_bug.cgi?id=797290 2018-09-18 13:15:06 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtpfunnel.c: rtpfunnel: fix shutdown By disposing of the ssrc_to_pad map in finalize instead of dispose. 2017-10-18 11:14:36 +0200 Havard Graff * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpfunnel.c: * gst/rtpmanager/gstrtpfunnel.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/meson.build: * tests/check/Makefile.am: * tests/check/elements/rtpfunnel.c: * tests/check/meson.build: Initial commit of GstRtpFunnel For funneling together rtp-streams into a single session. Use-cases include multiplexing and bundle. 2018-10-12 22:33:15 +0900 Yeongjin Jeong * tests/check/elements/flvdemux.c: tests: flvdemux: Add new test for channel detect using aac codec-data https://bugzilla.gnome.org/show_bug.cgi?id=797275 2018-10-11 16:36:17 +0900 Yeongjin Jeong * gst/flv/gstflvdemux.c: flvdemux: Use aac codec-data to adjust channels if needed Flv does not support various channels in AAC stream format, for example flvdemux detect an audio channels of 2(stereo) when the AAC really is 1(mono). https://bugzilla.gnome.org/show_bug.cgi?id=797275 2018-10-11 14:31:20 +0900 Yeongjin Jeong * tests/check/elements/flvmux.c: tests: flvmux: Add new test for caps change after starting to write headers https://bugzilla.gnome.org/show_bug.cgi?id=797256 2018-10-05 17:16:26 +0900 Yeongjin Jeong * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Don't refuse caps changes after starting to write headers in streamable mode. Flv does support changing the stream type and stream properties after the headers were started to be written, and for example H264 codec_data changes can be supported. https://bugzilla.gnome.org/show_bug.cgi?id=797256 2018-10-11 13:55:01 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix if condition in drop-frame timecode wrap-around Was previously: if ( x | y && a == b). Changed it into if ((x & y) && (a == b)). 2018-10-09 16:39:11 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Subtract daily jam offset when day wraps around For drop-frame framerates, when the expected next max timecode wraps around at the end of the day, we have to subtract the offset of the daily jam, otherwise we end up with a duration that's a few frames too long. https://bugzilla.gnome.org/show_bug.cgi?id=797270 2017-09-25 14:30:13 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: * tests/check/elements/rtpmux.c: rtpmux: respect downstream "timestamp-offset" in caps. https://bugzilla.gnome.org/show_bug.cgi?id=795162 2016-06-07 14:38:19 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * tests/check/elements/rtpmux.c: rtpmux: cleanup ssrc-handling code a bit And add some better logging. https://bugzilla.gnome.org/show_bug.cgi?id=795162 2016-05-04 11:48:04 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: rtpmux: protect against NULL caps Due to state-changes deactivating the pad from another thread, this can happen. https://bugzilla.gnome.org/show_bug.cgi?id=795162 2015-07-22 09:47:22 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: * tests/check/elements/rtpmux.c: rtpmux: property should overrule both upstream and downstream https://bugzilla.gnome.org/show_bug.cgi?id=762213 https://bugzilla.gnome.org/show_bug.cgi?id=795162 2018-10-08 20:45:08 +0100 Tim-Philipp Müller * meson.build: meson: use new 'python' module instead of deprecated https://github.com/mesonbuild/meson/pull/4169 2018-10-08 20:35:15 +0100 Tim-Philipp Müller * tests/examples/gtk/meson.build: meson: only build gtk gl examples if gst-gl was found And fix typo in glliveshader example binary name. 2018-10-03 16:17:22 +0200 Peter Körner * gst/multifile/gstsplitmuxsink.c: splitmuxsink: accept pads named 'sink' on the muxer, handle static pads as well https://bugzilla.gnome.org/show_bug.cgi?id=797241 2018-09-25 17:44:15 +0300 Sebastian Dröge * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: cairooverlay: Don't map input buffers if we just attach the overlay as meta https://bugzilla.gnome.org/show_bug.cgi?id=797091 2018-09-25 17:02:26 +0300 Sebastian Dröge * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: cairooverlay: Add overlay as meta to the buffers if we can This requires that downstream supports it and draw-on-transparent-surface is enabled. https://bugzilla.gnome.org/show_bug.cgi?id=797091 2018-09-25 15:34:40 +0300 Sebastian Dröge * ext/cairo/gstcairooverlay.c: cairooverlay: Pre-multiply and un-premultiply alpha in case of ARGB32 Cairo expects pre-multiplied alpha, we work on un-premultiplied alpha. https://bugzilla.gnome.org/show_bug.cgi?id=797091 2018-09-25 15:31:20 +0300 Sebastian Dröge * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: cairooverlay: Add property for drawing on a transparent surface and then blending This allows us to use the GstVideoOverlayComposition API and correctly handle pre-multiplied alpha, while also only doing the alpha conversion once instead of twice for the whole frame. At a later point we can attach the meta to the buffer instead of blending ourselves if downstream supports that. https://bugzilla.gnome.org/show_bug.cgi?id=797091 2018-10-03 17:34:49 +0200 Thibault Saunier * gst/matroska/matroska-demux.c: * gst/matroska/matroska-read-common.c: matroskdemux: do not use MapInfo.data after unmapping And minor gst-indenting 2018-09-30 19:28:07 +0200 Yacine Bandou * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: Add the WebM encrypted content support in matroskademux This commit: 1. Reads the WebM and Matroska ContentEncryption subelements. 2. Creates a GST_PROTECTION event for each ContentEncryption, which will be sent before pushing the first source buffer. The DRM system id field in this event is set to GST_PROTECTION_UNSPECIFIED_SYSTEM_ID, because it isn't specified neither by Matroska nor by the WebM spec. 3. Reads the protection information of encrypted Block/SimpleBlock and extracts the IV and the partitioning format (subsamples). 4. Creates the metadata protection for each encrypted Block/SimpleBlock, with those informations: KeyID (extracted from ContentEncryption element), IV and partitioning format. 5. Adds a new caps for WebM encrypted content named "application/x-webm-enc", with the following new fields: "encryption-algorithm": The encryption algorithm used. values: "None", "DES", "3DES", "Twofish", "Blowfish", "AES". "encoding-scope": The field that describes which Elements have been modified. Values: "frame", "codec-data", "next-content". "cipher-mode": The cipher mode used in the encryption. Values: "None", "CTR". https://bugzilla.gnome.org/show_bug.cgi?id=765275 2018-09-26 17:43:05 +0300 John Nikolaides * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Added a split-at-running-time action signal The video file can now be split at an arbitrary time, given by the user as an argument to the action signal. https://bugzilla.gnome.org/show_bug.cgi?id=787922 2018-09-21 19:47:44 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: rtpmp4gdepay: detect broken senders who send AAC with ADTS frames Strip ADTS headers if we detect any, apparently some Sony cameras send AAC with ADTS headers. We could also change the stream-format in the output caps, but that would be unexpected to pipeline builders and would not exactly be backwards compatible. 2018-09-21 18:17:25 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: factor out pushing of output buffer 2018-09-26 13:29:42 +0300 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Allow ANY capsfeatures 2018-09-26 00:06:09 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-video4linux2.xml: docs: update for git master 2018-06-22 12:05:17 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: PIFF track encryption box support The PIFF track encryption box is a UUID box containing the default encryption values that should be used for PIFF sample encryption. https://bugzilla.gnome.org/show_bug.cgi?id=796647 2018-09-24 11:45:46 +0200 Nicola Murino * sys/osxaudio/gstosxcoreaudio.c: osxaudio: add support for parsing more channel layouts ... ... and fallback to gst_audio_info_set_format for not yet supported layouts. Fix audio playback on iOS 12. Based on patch from Byron Schiel https://bugzilla.gnome.org/show_bug.cgi?id=796919 2018-09-22 17:22:46 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: turn impossible condition into an assert qtdemux_update_streams() is only ever called after checking `qtdemux->streams_aware` is TRUE. There is no need to check for that condition again. `qtdemux->streams_aware` is only modified when the demuxer is hard-resetted, which is mutually exclusive with demuxing, so it cannot be modified during the call. https://bugzilla.gnome.org/show_bug.cgi?id=797191 2018-09-21 22:24:02 +0200 Alicia Boya García * gst/matroska/matroska-demux.c: matroskademux: Emit no-more-pads after parsing Tracks Currently matroskademux does not emit no-more-pads until the first Cluster is parsed, even though the Tracks have already been parsed and from that point on there can be no more tracks. This is important in MSE because the browser needs to know when the MSE initialization segment has been completely parsed so that it can expose the tracks to the user. Some applications depend on this been done before they feed frames to the demuxer. As a consequence, historically WebKit has relied on hacks such as listening to the `pad-added` event, which made impossible to support multiple tracks in the same file. Let's fix that. https://bugzilla.gnome.org/show_bug.cgi?id=797187 2018-09-21 20:38:02 +0200 Alicia Boya García * gst/matroska/matroska-demux.c: matroskademux: Parse successive Tracks elements This patch allows matroskademux to parse a second Tracks element, erroring out if the tracks are not compatible (different number, type or codec) and emitting new caps and tag events should they have changed. https://bugzilla.gnome.org/show_bug.cgi?id=793333 2018-09-21 16:23:57 +0200 Alicia Boya García matroskademux: Refactor track parsing out from adding tracks This splits gst_matroska_demux_add_stream() into: * gst_matroska_demux_parse_stream(): will read the Matroska bytestream and fill a GstMatroskaTrackContext. * gst_matroska_demux_parse_tracks(): will check there are no repeated tracks. * gst_matroska_demux_add_stream(): creates and sets up the pad for the track. https://bugzilla.gnome.org/show_bug.cgi?id=793333 2017-11-30 20:44:23 +0100 Alicia Boya García * gst/matroska/matroska-demux.c: matroskademux: Allow Matroska headers to be read more than once This is necessary for MSE, where a new MSE initialization segment may be appended at any point. These MSE initialization segments consist of an entire WebM file until the first Cluster element (not included). [1] Note that track definitions are ignored on successive headers, they must match, but this is not checked by matroskademux (look for `(!demux->tracks_parsed)` in the code). Source pads are not altered when the new headers are read. This patch has been splitted from the original patch from eocanha in [2]. [1] https://www.w3.org/TR/mse-byte-stream-format-webm/ [2] https://bug334082.bugzilla-attachments.gnome.org/attachment.cgi?id=362212 https://bugzilla.gnome.org/show_bug.cgi?id=793333 2018-08-16 21:42:37 +0200 Mathieu Duponchelle * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Implement split-after The behaviour of split-now is to output the current GOP after starting a new file. The newly-added split-after signal will output the current GOP to the old file if possible once a new GOP is opened. https://bugzilla.gnome.org/show_bug.cgi?id=796982 2018-09-20 12:12:55 +0900 Seungha Yang * gst/flv/gstflvmux.c: flvmux: Don't leak codec_data buffer Use gst_buffer_replace() to prevent buffer leak https://bugzilla.gnome.org/show_bug.cgi?id=797179 2018-09-18 18:13:52 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Set Closed Caption track width/height to that of the first video track Otherwise software like Premiere or Final Cut Pro won't like our files. https://bugzilla.gnome.org/show_bug.cgi?id=797111 2018-09-19 11:45:59 +0100 Tim-Philipp Müller * meson.build: * meson_options.txt: meson: add glib-checks option to disable API guards and such We want this enabled by default, also in releases, but people may want to disable this for performance-critical workloads or on embedded devices. 2018-09-19 11:45:00 +0100 Tim-Philipp Müller * meson_options.txt: meson: fix missing closing bracket in option descriptions 2018-09-06 20:10:30 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Initialize caption track language code to 0 instead of "und" Without this, Final Cut considers it "non-standard" and 0 (english) is a good default for closed captions. https://bugzilla.gnome.org/show_bug.cgi?id=797111 2018-09-13 03:16:32 +0000 Song Bing * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Add HEVC decoder support https://bugzilla.gnome.org/show_bug.cgi?id=771686 2018-09-13 02:35:39 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Move capture probe after input format is set This is to support Amlogic CODEC driver which does not provide a full list of formats when the driver is initially opened. GStreamer does not strictly need this full list initially, but only later, in order to negotiate with downstream if multiple format can be selected. With this change, we will no longer probe twice the device, since the probed list can be directly used for negotation. 2018-09-11 16:46:34 -0300 Ezequiel Garcia * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2: Add a debug message beforing waiting for codec stop Add a debug message right before waiting for the driver. This is useful in order to debug drivers without a properly implemented decoder or encoder stop command. 2018-09-10 13:18:45 -0300 Ezequiel Garcia * sys/v4l2/gstv4l2.c: v4l2: Add a debug message indicating probe operation It's useful to see the v4l2 element running the probe operation, to confirm it's turned on and working. 2018-09-10 13:18:30 -0300 Ezequiel Garcia * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2jpegenc.c: * sys/v4l2/gstv4l2jpegenc.h: * sys/v4l2/meson.build: v4l2: Add JPEG encoding support This commit adds the support for V4L JPEG stateful encoders. 2018-09-10 16:20:52 -0300 Ezequiel Garcia * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2fwhtenc.c: * sys/v4l2/gstv4l2fwhtenc.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/meson.build: v4l2: Add FWHT codec support The recently added vicodec (virtual codec) V4L driver uses the Fast Walsh-Hadamard Transform for encoding and decoding. Add support for it. 2018-09-12 21:28:24 -0400 Nicolas Dufresne * sys/v4l2/ext/v4l2-common.h: * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: v4l2: Sync kernel header with linuxtv tree This notably add HEVC and FWHT support, and VP8/9 profiles are now an enumeration and their control exposed as a menu. 2018-09-12 17:24:00 +0300 Vivia Nikolaidou * gst/isomp4/gstqtmux.c: qtmux: Allow up to 1 trak timescale unit of lateness in prefill mode For 59.94 FPS, it's common to set 60000 as timescale. For that timescale, if the audio is late by as little as 0:00:00.000016666 (definitely less than one audio sample), lateness gets rounded to 1. Added a safeguard that allows lateness up to 1 sample with the specific trak's timescale, to make sure that values less than e.g. one audio sample won't break the prefill mode. What will happen in this case is that the audio will get squeezed back to the video's timestamp, which in practice means that the audio will be 0.000016666 seconds early (with the patch). https://bugzilla.gnome.org/show_bug.cgi?id=797133 2018-09-10 20:20:39 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix indentation 2018-09-11 00:18:32 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Protect against zero PAR num/demu This fixes an assertion when the driver implement CROPCAP but does not set the PAR. 2018-09-12 00:52:19 +0100 Tim-Philipp Müller * gst/audioparsers/gstwavpackparse.c: wavpackparse: fix handling of correction streams Accept wavpack correction streams (.wvc) on sink pad, so that wavpackparse can also be used to packetise correction streams. Fix parsing of subblock ID tags - the higher bits are flags and are not part of the ID. This resulted in correction blocks not being recognised properly and the output not having the right (correction) caps. 2018-09-07 18:47:22 +0530 Nirbheek Chauhan * ext/speex/meson.build: meson: Explicitly pass -DWIN32 while building speex The speex headers assume that WIN32 will always be defined when building on Windows, but this is only true by default on MinGW. Always set it explicitly. 2018-09-06 13:13:19 +0900 Seungha Yang * gst/flv/gstflvmux.c: flvmux: Don't omit streamheader from caps on downstream reconfigure The reconfigured downstream elements (e.g., dynamically added sink element) most likely require the flv streamheader https://bugzilla.gnome.org/show_bug.cgi?id=797089 2018-09-05 16:11:00 -0700 Martin Kelly * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: don't store used UIDs Currently, whenever we generate a 128-bit UID, we store it in a list and return 0 if we ever encounter a collision. This is so mathematically improbable that it's not worth checking for, so we can save memory and time by not tracking the UID. Even if a collision happened, a list of only 10 UIDs would be unlikely to detect it. This article has a good description of how improbable a collision is: https://en.wikipedia.org/wiki/Universally_unique_identifier#Collisions https://bugzilla.gnome.org/show_bug.cgi?id=797086 2018-09-06 20:06:10 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/gstqtmux.c: qtmux: Use existing helper function to create "und" language code 2018-09-05 20:15:57 +0530 Nirbheek Chauhan * ext/meson.build: meson: Don't skip plugins that don't build with MSVC We now have options for all plugins, so we will just disable these in the cerbero recipe instead. These require external deps, so they won't affect gst-build either. 2018-09-03 16:04:33 +0530 Nirbheek Chauhan * ext/mpg123/gstmpg123audiodec.h: mpg123: Remove ssize_t fallback, not needed anymore The mpg123 headers now contain a definition for ssize_t and building with MSVC fails because of a redefinition for ssize_t 2018-07-31 12:52:36 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Keep sample data from the current fragment only (push mode) This patch clears the sample table whenever the demuxing of a new fragment begins. This avoids increasing memory usage for long videos. This behavior was already present when upstream_format_is_time; this patch extends it to all push mode operation (e.g. Media Source Extensions). https://bugzilla.gnome.org/show_bug.cgi?id=796899 2018-09-01 09:30:23 +0530 Nirbheek Chauhan * meson.build: * sys/osxaudio/meson.build: meson: Fix osxaudio build on iOS Must define HAVE_IOS, and use appleframeworks dependency to ensure the right frameworks are picked up. 2018-08-22 19:23:53 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Only offer MMAP/DMABUF pool The propose allocation was offering a pool even in DMABUF_IMPORT or USERPTR mode. These pool are internal only. 2018-08-22 17:51:52 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2transform.h: v4l2transform: Add "disable-passthrough" property This allow forcing going through the transform driver even if there isn't an conversion happening. This is usedful when the m2m driver can be used to adapt the type of memory between two drivers. 2018-08-31 14:25:09 +0300 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Reset frame/tc/caption pointer to NULL after moving it in the history 2018-08-29 09:51:42 +0200 Edward Hervey * gst/rtp/gstrtpmp4vpay.c: rtpmp4vpay: Increase ranking Both rtpmp4vpay and rtpmp4gpay support MPEG4 elementary streams. But the most supported variant is the video-specific one (rtpmp4vpay), therefore increase the rank of that one so that auto-plugging of payloaders for MPEG4 elementary streams ends up picking that one and not the generic one. 2018-08-15 12:53:34 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: implement keyframe search also without cluster prev size If we have cluster prev size (GStreamer muxer will write it by default), we can go back to the previous cluster efficiently, but if we don't then just search backwards until we find a cluster ebml identifier, like we do when searching for clusters in the bisection loop. 2018-08-15 12:14:24 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: make max backtrack distance for keyframe search configurable Add property instead of hardcoding it in the code. In some scenarios such as CCTV variable fps and extra long GOPs are used to minimise storage space, for example. In those cases there might not be any keyframes for many minutes, so provide a property to override the max allowed distance. https://bugzilla.gnome.org/show_bug.cgi?id=790696 2018-08-15 11:49:57 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: set limit how much to backtrack to find a keyframe If we seek without an index and land on a cluster that starts with a delta frame. https://bugzilla.gnome.org/show_bug.cgi?id=790696 2018-08-15 11:25:21 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: no need to search for keyframes for intra-only streams If the video streams are all I-frame only then we don't need to look for a cluster with a keyframe, we can just assume there will be one. https://bugzilla.gnome.org/show_bug.cgi?id=790696 2018-08-15 01:10:32 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: figure out if we have prev_size when starting up This is useful to know in case someone initiates a seek or direction change before we reach the second cluster. 2018-08-08 12:37:54 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: try to ensure keyframe when seeking without index When seeking in pull mode without an index (because there is no index or the file is still being written to) we bisect to find the right cluster to jump to. However, it's possible the cluster we found doesn't start with a keyframe, which leads to decoding errors, so if we know that the found cluster starts with a delta frame try to scan back to previous clusters until we find one that starts with a keyframe or we are back at the beginning. Theoretically it's possible that all clusters but the first one do not start with a keyframe and the keyframes are in the middle of clusters, but this is extremely unusual, so we will cover this case with a basic sanity check. This problem is especially problematic with content recorded with dynamic GOP and FPS, where long GOP lengths and low FPS may cause a large set of clusters to lack key frames. Playback would then be started on a non-keyframe cluster, and the large number of such frames would make the content impossible to decode fo a long stretch of time. Based on patch by: Mats Lindestam https://bugzilla.gnome.org/show_bug.cgi?id=790696 2017-01-18 10:27:38 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: extract cluster prevsize if available This is useful for reverse playback/trickmodes without an index, and will also be useful in the seek handler if we need to scan back to find a cluster that starts with a keyframe. https://bugzilla.gnome.org/show_bug.cgi?id=790696 2018-07-25 19:27:01 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Validate stride/offset when importing This will prevent situation where buffer size allow importing but rendering goes wrong due to a miss-match in expected stride and offset. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-08-01 13:07:52 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Add a method to try and import buffers This method will check if a buffer, base on it's video meta, can be imported. It will also try and adapt the request stride in case this is the only that miss-match. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-08-01 12:07:20 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Trace the buffer index we import to https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-07-25 22:16:59 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Fix typo in error message https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-07-24 12:07:22 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Only queue buffer if preparation worked The preparation code imports the buffer, doing bunch of validation. Only queue the buffer in the driver if the importation worked. This way we don't rely on the driver to validate. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-07-24 12:05:45 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Only allow DMABuf export for STREAMING device DMABuf exportation requires mmap, which requires STREAMING capabilities. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-07-13 14:42:21 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Activate the other pool first This change has no effect. We will need to acquire a buffer from the pool later in order to validate / adapt with the video alignment for the downstream buffers. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-07-09 15:33:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Simplify format handling Always initially use try_format(), delaying set_format() to when the allocation is being negotiated. This avoid having two code paths, and will be help adding support for properly importing buffers of specific strides and offsets. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-08-23 22:57:35 +0200 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: matroska: fix handling of FlagInterlaced This is an enum not a boolean, and a value of 2 signals that the video is progressive, but we would mistakenly set interlace-mode=mixed on the output caps. https://bugzilla.gnome.org/show_bug.cgi?id=787206 2018-08-09 15:14:05 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: complete colorspace info in debug log The desired colorimetry is logged with all parameters (colorpsace, range, matrix, and transfer function), but of the values actually set by the driver, only colorspace is logged. Complete the debug log message to display all colorimetry parameters: Desired colorspace is 8:1:1:1 Got format of 640x480, format YU12, nb planes 1, colorspace 8 -> Desired colorspace is 8:1:1:1 Got format of 640x480, format YU12, nb planes 1, colorspace 8:0:0:0 https://bugzilla.gnome.org/show_bug.cgi?id=796940 2018-08-09 15:12:57 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: fix typo in comment https://bugzilla.gnome.org/show_bug.cgi?id=796940 2018-08-09 15:08:59 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: improve colorspace handling for JPEG sources gstjpegdec sets 1:4:0:0 colorimetry (full range BT.601 YCbCr encoding with unknown primaries and unknown transfer function). This currently gets translated to bt601 or bt709 depending on resolution. Both cases result in a negotiation failure: ERROR: from element /GstPipeline:pipeline0/v4l2video0convert:v4l2video0convert0: Device '/dev/video0' does not support 1:4:0:0 colorimetry Improve the guessing game by selecting JPEG colorimetry (JPEG colorspace with sRGB transfer function) under these specific conditions, and loosen the matching so that 1:4:0:0 input gets accepted if the device is actually configured to 1:4:7:1 (V4L2_PIX_FMT_JPEG default). https://bugzilla.gnome.org/show_bug.cgi?id=796940 2018-08-09 17:24:35 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: stop V4L2 from zeroing extended colorimetry for non-mplane Setting the priv field to a magic value stops V4L2 core from zeroing the extended colorimetry fields quantization, ycbcr_enc, and xfer_func for non-mplane queues. https://bugzilla.gnome.org/show_bug.cgi?id=796940 2018-08-19 15:39:16 +0200 Zeeshan Ali * sys/v4l2/gstv4l2object.c: v4l2: Remove a trailing whitespace Otherwise, the latest gst-indent check doesn't pass. 2018-08-18 21:08:55 +0100 Tim-Philipp Müller * meson.build: * meson_options.txt: meson: add options to disable gobject cast checks and glib asserts ... and define G_DISABLE_DEPRECATED for development versions, like we do in autotools. 2018-08-18 21:01:52 +0100 Tim-Philipp Müller * REQUIREMENTS: * ext/jpeg/meson.build: meson: find libjpeg via pkg-config This effectively (but optionally) requires libjpeg-turbo which ships with a .pc file and is what pretty much everyone these days uses anyway for libjpeg, so shouldn't be a problem hopefully. https://bugzilla.gnome.org/show_bug.cgi?id=796947 2018-08-17 17:35:43 -0400 Nicolas Dufresne * gst/udp/gstudpsrc.c: udpsrc: Fix build when SO_RCVBUFFORCE is not defined This shoudl fix the mingw build. 2018-08-17 14:17:39 -0400 Nicolas Dufresne * gst/udp/gstudpsrc.c: udpsrc: Balance Linux value of get/set_rcvbuf On Linux, the kernel returns twice the size as it will allocate extra space for accouting. We devides this value by two in order to ensure that get/set value now match. This fixes the set buffer size validation and allow having a nice warning when the size if surpassed and the process does not have CAP_NET_ADMIN capabilities. https://bugzilla.gnome.org/show_bug.cgi?id=727067 2018-08-17 14:05:04 +0200 Guillaume Desmottes * gst/udp/gstudpsrc.c: updsrc: set udp buffer size forcibly The udp buffer size is limited to a maximum of around 100K. Some apps need to set the force bufsize for their own operation. Use the SO_RCVBUFFORCE option in order to override the rmem_max limit of linux kernel. Require user to have the CAP_NET_ADMIN privilege to work. Original patch from Kyungnam Bae https://bugzilla.gnome.org/show_bug.cgi?id=727067 2018-08-17 13:59:00 +0200 Guillaume Desmottes * gst/udp/gstudpsrc.c: udpsrc: factor out gst_udpsrc_get_rcvbuf() No semantic change. https://bugzilla.gnome.org/show_bug.cgi?id=727067 2018-08-17 19:11:21 +0530 Nirbheek Chauhan * ext/libcaca/meson.build: * meson_options.txt: meson: Rename caca option to libcaca All options must match the plugin directory name. 2018-08-17 18:56:54 +0530 Nirbheek Chauhan * meson_options.txt: * sys/directsound/meson.build: * sys/meson.build: * sys/osxaudio/meson.build: * sys/osxvideo/meson.build: * sys/waveform/meson.build: meson: Add build files for osxaudio, osxvideo, waveform osxaudio is for macOS and iOS osxvideo is for macOS waveform is for Windows 2018-08-17 14:44:26 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-multifile.xml: docs: update for changes in master 2018-08-17 11:45:47 +0100 Tim-Philipp Müller * tests/examples/Makefile.am: examples: dist qt examples https://bugzilla.gnome.org/show_bug.cgi?id=796968 2018-08-17 00:27:59 +0530 Nirbheek Chauhan * meson.build: * meson_options.txt: * tests/meson.build: meson: Add an option for tests This is needed because we don't always have gstreamer-check available, for instance inside Cerbero on iOS. 2018-08-16 18:55:29 +0200 Mathieu Duponchelle * gst/isomp4/gstqtmux.c: * gst/multifile/gstsplitmuxsink.c: mp4 robust muxing: improve documentation and logging 2018-07-03 23:11:56 +0200 Jan Alexander Steffens (heftig) * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: rtph26*pay: Update param set timestamp even if parameters unchanged rtph264pay and rtph265pay skip updating the parameter set timestamp if the units they see contain no new configuration. This can result in them injecting duplicate parameters. https://bugzilla.gnome.org/show_bug.cgi?id=796748 2018-08-15 13:43:53 +0200 Ulf Olsson * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add support for SET_PARAMETER and GET_PARAMETER using signals https://bugzilla.gnome.org/show_bug.cgi?id=792131 2018-08-15 02:28:20 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't leak old muxer/sink in async mode Make sure to clear the reference taken earlier in the function when switching muxer/sink asynchronously so they don't leak 2018-08-15 02:10:25 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Fix reference counting loop The stream context was holding a reference to the internal queue and pads, with pad probes that were in turn holding references to the stream context. This lead to a leak if the request pads weren't explicitly released. https://bugzilla.gnome.org/show_bug.cgi?id=796893 2018-08-11 16:45:25 +0800 Roland Jon * gst/audioparsers/gstaacparse.c: aacparse: fix codec_data buffer leak https://bugzilla.gnome.org/show_bug.cgi?id=740101 2018-08-02 16:12:45 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Handle closed captions as subtitle streams 2018-08-02 08:40:17 +0200 Iñigo Huguet * sys/v4l2/gstv4l2object.c: v4l2src: fix first input used is always used next times The input from an v4l2 device that was used the first time was remembered for next times, and set again always the pipeline is set to READY state. This was making that users wasn't able to select a different input without having to create a new pipeline. This patch makes that v4l2src element forget previous used input when going to NULL state, so it will check again for the current selected input when going again to READY state. Users can change to NULL state, select a new input with a VIDIOC_S_INPUT ioctl and change to PLAYING again. https://bugzilla.gnome.org/show_bug.cgi?id=796908 2018-08-02 13:40:09 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: The sample size we have to reserve is 256+8 bytes for the header for CDP packets 2018-08-02 12:27:45 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Properly allocate 256 bytes per CDP packet in prefill mode Instead of allowing 256 but only pre-allocating 100. 2018-08-02 12:27:17 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: Revert "qtmux: Allow for CDP packets up to 320 bytes" This reverts commit 5eed1d49bdb7e7a632c7135656c482ed38a6ac2a. 255 is actually the maximum, there's a bug if more is arriving. 2018-08-01 16:50:03 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Allow for CDP packets up to 320 bytes Apparently they can be bigger than 256 bytes sometimes. 2018-07-13 22:31:04 -0400 Nicolas Dufresne * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9pay.c: rtppayload: Fix VP8/VP9/OPUS dual encoding name handling All these were copy pasted and would lead to assertion when chained with rtpmux. This commit rewrite the negotiation with downstream. This also drop the fallback to ancient names if the pad is unlinked. This was completly arbitrary decision that made no sense. https://bugzilla.gnome.org/show_bug.cgi?id=796809 2018-08-01 12:06:23 +1000 Matthew Waters * ext/qt/gstqtgl.h: qt: Ensure GL headers are included Otherwise there may be no valid typedef of GLsync. ... /usr/include/gstreamer-1.0/gst/gl/gstglfuncs.h:93:24: note: in definition of macro 'GST_GL_EXT_FUNCTION' ret (GSTGLAPI *name) args; ^~~~ /usr/include/gstreamer-1.0/gst/gl/glprototypes/sync.h:33:23: error: 'GLsync' has not been declared (GLsync sync)) ^~~~~~ ... https://bugzilla.gnome.org/show_bug.cgi?id=796879 2018-08-01 03:18:58 +0530 Nirbheek Chauhan * sys/oss4/meson.build: meson: Fix oss4 header checks Otherwise, oss4 ends up getting built when force-disabled. 2018-08-01 01:10:49 +0530 Nirbheek Chauhan * sys/v4l2/meson.build: meson: Fix missing variable in v4l2 build 2018-07-31 12:47:47 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtdemux: Don't assert in prefill mode if a track has no samples at all Just write it with a duration of 0, no samples, etc. 2018-07-31 12:33:54 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Don't assert if a file does not have any active streams ** (gst-play-1.0:9113): CRITICAL **: 12:31:54.360: qtdemux_is_streams_update: assertion 'qtdemux->active_streams != NULL' failed 2018-07-30 13:33:28 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Reserve 256 bytes for CDP packets in pre-fill mode 92 is sometimes too small and compared to the wasted space for other codecs 256 bytes is small (and should be the maximum CDP packet size) 2018-07-25 07:35:28 +0530 Nirbheek Chauhan * ext/aalib/meson.build: * ext/cairo/meson.build: * ext/dv/meson.build: * ext/flac/meson.build: * ext/gdk_pixbuf/meson.build: * ext/gtk/meson.build: * ext/jack/meson.build: * ext/jpeg/meson.build: * ext/lame/meson.build: * ext/libcaca/meson.build: * ext/libpng/meson.build: * ext/mpg123/meson.build: * ext/pulse/meson.build: * ext/qt/meson.build: * ext/raw1394/meson.build: * ext/shout2/meson.build: * ext/soup/meson.build: * ext/speex/meson.build: * ext/taglib/meson.build: * ext/twolame/meson.build: * ext/vpx/meson.build: * ext/wavpack/meson.build: * gst/matroska/meson.build: * gst/meson.build: * meson.build: * meson_options.txt: * sys/directsound/meson.build: * sys/meson.build: * sys/oss/meson.build: * sys/oss4/meson.build: * sys/v4l2/meson.build: * sys/ximage/meson.build: * tests/examples/qt/qmlsink/meson.build: * tests/examples/qt/qmlsrc/meson.build: * tests/icles/meson.build: * tests/meson.build: meson: Add feature options for all plugins Checks for GL, Qt5, and C++ are still automagic. FIXMEs have been added for these so they can be fixed later. https://bugzilla.gnome.org/show_bug.cgi?id=795107 2018-07-25 17:15:53 +0300 Vivia Nikolaidou * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.h: deinterlace: Closed caption pass-through Pass through closed caption data when deinterlacing. When two deinterlaced frames are created for the same interlaced frame (e.g. fields=all), the second of the two frames will have no closed caption data. Also fixed memory leaks related to timecode meta pass-through. https://bugzilla.gnome.org/show_bug.cgi?id=796876 2018-07-25 18:37:48 -0400 Olivier Crête * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Implement muxing of AV1 into MP4 files According to https://aomediacodec.github.io/av1-isobmff/ 2018-07-25 17:09:06 -0400 Olivier Crête * gst/matroska/matroska-mux.c: matroskamux: Put codec_data as CodecPrivate for AV1 2018-07-25 17:08:53 -0400 Olivier Crête * gst/matroska/matroska-mux.c: matroskamux: Accept muxing AV1 2018-07-25 16:51:38 -0400 Olivier Crête * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux_types.c: qtdemux: Recognize more AV1 atoms 2018-07-25 16:39:18 -0400 Olivier Crête * gst/matroska/matroska-demux.c: matroskademux: Extract codec_data for AV1 According to https://github.com/Matroska-Org/matroska-specification/blob/av1-mappin/codec/av1.md 2018-07-25 14:31:39 -0400 Olivier Crête * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Extract AV1 codec_data and put it in the caps Also extract the presentation-delay and put it in the caps. 2018-07-25 10:43:11 -0400 Olivier Crête * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_types.c: qtdemux: Add initial support for AV1 demuxing Following the spec at https://aomediacodec.github.io/av1-isobmff/ 2018-07-27 00:41:57 +1000 Jan Schmidt * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add a small configurable teardown delay This causes rtspsrc to send a teardown and wait on PAUSED->READY transition, with a configurable delay. Otherwise, typically teardown never gets sent in playbin / uridecodebin where the transition back to NULL happens too quickly. The timeout is set to 100ms default. https://bugzilla.gnome.org/show_bug.cgi?id=751994 2018-07-26 16:43:28 +0300 Sebastian Dröge * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: rtpgstpay: Add support for force-keyunit events This triggers immediate re-sending of the configuration data in-band. https://bugzilla.gnome.org/show_bug.cgi?id=796877 2018-07-13 19:45:19 +0300 Sebastian Dröge * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: rtp: Use running_time instead of PTS for config-interval calculations PTS can start again from a different offset while the running time is increasing. The only thing that matters here is the running time. https://bugzilla.gnome.org/show_bug.cgi?id=796807 2018-07-19 22:48:34 -0400 Nicolas Dufresne * tests/examples/gtk/meson.build: * tests/examples/meson.build: example: Build GTK and GTK GL example code 2018-07-19 17:31:03 +0200 Michael Olbrich * gst/rtp/gstrtpL8pay.c: rtpL8pay: don't try to modify a read-only structure Just remove the code. It's not doing anything useful anyways. The modified caps are the result of a caps query, so either not used afterwards of a reference to some internal caps of another element that should not be modified. https://bugzilla.gnome.org/show_bug.cgi?id=796837 2018-07-17 08:23:54 +0200 Iñigo Huguet * ext/qt/gstqtgl.h: qmlgl: Fix conflicting declaration of type GLsync for non-android https://bugzilla.gnome.org/show_bug.cgi?id=796821 2018-07-16 19:03:39 +0300 Vivia Nikolaidou * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.h: deinterlace: Timecode pass-through When it is trivial to pass-through a timecode, by only removing the "interlaced" flag, do pass-through. Otherwise, double the fps_n and adjust the "frames" field. https://bugzilla.gnome.org/show_bug.cgi?id=796818 2018-07-17 00:03:19 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmux: Improve handling of repeated timestamps When handling input with timestamps that repeat, sometimes splitmuxsink would get confused and ignore a keyframe. The logic in question is a holdover from before the cmd queue moved the file cutting to the multiqueue output side and made it deterministic, so it's no longer needed on the input here. https://bugzilla.gnome.org/show_bug.cgi?id=796773 2018-07-17 01:33:55 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: Revert "splitmuxsrc: Make sure events are writable" This reverts commit 3ac5430311b20f30814cdabf5724fb687748bb5b. There's no need to make a freshly created event writable, and the other half of this patch was already fixed and pushed in f2f15a1 2018-07-16 23:43:29 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Make sure events are writable Before setting the seqnum on events sent downstream, make sure they are writable. 2018-07-13 16:51:24 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Validate that capture buffers were queued When the pool is started, we allocate and release buffer, expecting the pool release-buffer handler to queue them. Though, as we rely on release function, there is no direct way to detect that this process didn't work. To check this, validate that the number of queued buffer is the same as the number of allocated buffers. This allow returning an error when buffer importation was refused by the driver. https://bugzilla.gnome.org/show_bug.cgi?id=583890 2018-07-13 16:02:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Only return eos for M2M devices This will avoid sending EOS on v4l2src when a driver sends an empty buffers. This case would be a bug in the driver, but yet the camera should keep running. This also removes the check for corrupted buffers, as this check is already done later. https://bugzilla.gnome.org/show_bug.cgi?id=794842 2018-07-13 15:58:36 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2.c: * sys/v4l2/v4l2-utils.h: * sys/v4l2/v4l2_calls.c: v4l2: Add a macro to check for M2M https://bugzilla.gnome.org/show_bug.cgi?id=794842 2018-07-13 14:41:13 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix userptr importation The length passed to the driver was always 0 instead of the size of the memory. This would fail validation in videobuf2. 2018-07-12 15:11:39 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Remove duplicate check We were calling gst_v4l2_is_buffer_valid() before and inside gst_v4l2_buffer_pool_qbuf() as we needed to access the group. The second check failed since the writability of the buffer get inherited from the GstMemory, which lead to pipeline failure. As we cannot avoid the extra ref, it would be racy otherwise, just pass the group to _dbuf() so it does not have to call gst_v4l2_is_buffer_valid() again. https://bugzilla.gnome.org/show_bug.cgi?id=796692 2017-08-25 11:58:12 +0200 Havard Graff * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * tests/check/elements/rtpsession.c: rtpsession: Don't start the RTCP thread until it's needed Always wait with starting the RTCP thread until either a RTP or RTCP packet is sent or received. Special handling is needed to make sure the RTCP thread is started when requesting an early RTCP packet. We want to wait with starting the RTCP thread until it's needed in order to not send RTCP packets for an inactive source. https://bugzilla.gnome.org/show_bug.cgi?id=795139 2018-07-11 12:21:44 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: v4l2src: Try to avoid TRY_FMT when camera is streaming Some camera firmware crash is TRY_FMT is called during streaming. As a side effect. This try and detect that the same format as currently running is about to be tried, and skip renegotiation. https://bugzilla.gnome.org/show_bug.cgi?id=796789 2018-07-09 13:59:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Protect double calls to set_format() In some cases, set_format() may get called twice before the output format is set. Running an allocation query in this case is both not needed and will cause assertion due tot he NULL caps. 2018-07-08 20:08:18 -0400 Thibault Saunier * gst/audiofx/gstscaletempo.c: scaletempo: Mark as Audio in classification 2018-07-06 15:21:33 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Store and propagate SEGMENT sequence numbers * When receiving a segment in TIME, use that seqnum * Only reset the stored sequence number when doing HARD reset (and not when we get a FLUSH event from upstream) 2018-07-01 15:27:32 -0400 Michael Tretter * sys/v4l2/gstv4l2transform.c: v4l2transform: Implement stable element name The first converter to be found will now gain the name v4l2convert. Other converters will be named after the m2m dev node end point they are attached to. https://bugzilla.gnome.org/show_bug.cgi?id=784958 2018-06-13 17:39:57 +0100 Philippe Normand * gst/matroska/matroska-demux.c: matroskademux: Set subtitle tag title from TrackName field GUI applications can then use the title tag to set menu items or labels representing the track. https://bugzilla.gnome.org/show_bug.cgi?id=796567 2018-06-28 19:08:35 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Only renegotiate with upstream When the decoder get linked further, it will receive a renegotiation event from downstream. This case is not supported and should be ignored. This fixes issues when this encoder is used inside an GstRtspServer pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=796525 2018-06-09 23:58:01 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: rework segment event pushing, again This patch aims at fixing the recent regressions in the adaptive test suite. All segment pushing in push mode is now done with gst_qtdemux_check_send_pending_segment(), which is idempotent and handles both edit lists cases and cases where the upstream TIME segments have to be sent directly. Fragmented files that start with a non-zero tfdt are also taken into account, but their handling has been vastly simplified: now they are handled as implicit default seeks so there is no need to extend the GstSegment formulas as was being done before. qtdemux->segment.duration is no longer modified when upstream_format_is_time, respecting in this way the durations provided by dashdemux and fixing bugs in reverse playback tests where mangled durations appeared in the emitted segments. https://bugzilla.gnome.org/show_bug.cgi?id=752603 2018-06-17 02:01:59 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Don't send EOS during upstream reverse playback Upstream driving elements such as dashdemux often do reverse playback by feeding qtdemux with the fragments containing the requested playback range in reverse order. But the requested playback range stop may be somewhere in the middle of a fragment. In that case, a naive pts >= segment.stop condition may declare end of segment prematurely when demuxing this first fragment. This used not to happen because there were places in moov parsing where segment.stop was overwritten to GST_CLOCK_TIME_NONE even if upstream_format_is_time -- resulting in this case in a segment with rate < 0 and stop == -1 and hence not triggering the EOS check, but that was likely an accident. This patch modifies the EOS check to take this case into account, not sending EOS when upstream_format_is_time if rate < 0. This fixes adaptive.dash.playback.seek_end_live.DASHIF_livestream_testpic_2s https://bugzilla.gnome.org/show_bug.cgi?id=752603 2018-02-06 13:51:14 +0100 Peter Seiderer * sys/v4l2/gstv4l2transform.c: v4l2transform: fold property set/get PROP_OUTPUT_IO_MODE case into default https://bugzilla.gnome.org/show_bug.cgi?id=796714 2018-06-22 14:56:31 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Don't set colorimetry on capture The colorimetry will be set along with the raw format and those fields will then be copied from sink to src caps by the gst encoder. https://bugzilla.gnome.org/show_bug.cgi?id=791471 2018-06-27 16:57:29 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Really always set colorimetry This fixes patch dd1c5aed656e07e3dad01f83410f3af16cfb14cf which pretended to always set colorimetry but the patch was incomplete. This is again best effort considering the spec says that for CAPTURE you may only read this value. 2018-06-26 15:04:39 +0200 Michael Tretter * sys/v4l2/gstv4l2videodec.c: v4l2videodec: do not call streamon while pool is flushing gst_v4l2_buffer_pool_flush() executes streamoff for the output, but streamoff->streamon for the capture of the decoder. gst_v4l2_buffer_pool_streamon() on capture assumes that is able to resurrect the buffers from the pool, but acquiring buffers fails if the buffer pool is still flushing. The decoder needs to stop flushing the pools before calling gst_v4l2_buffer_pool_flush() to restart the v4l2 device. Otherwise starting the decoding thread might fail, because there are no buffers in the capture pool. This fixes a regression that was introduced in 97985a335c78 ("v4l2videodec: Add dynamic resolution change support"). https://bugzilla.gnome.org/show_bug.cgi?id=796681 2018-06-25 16:03:17 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: use S_SELECTION instead of S_CROP in gst_v4l2_object_set_crop The S_CROP call doesn't work on mem2mem output queues. Use the S_SELECTION call to set the crop rectangle and only fall back to S_CROP for ancient kernels. This will allow v4l2videoenc to set the coded size on the output queue via S_FMT and then set the visible size via the crop rectangle, as required by the V4L2 codec API. https://bugzilla.gnome.org/show_bug.cgi?id=796672 2018-06-27 13:46:00 +0000 Marian Mihailescu * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: activate capture pool after output pool Some drivers need output buffers set before capture buffers. CODA cannot set output format if capture is streaming. Exynos MFC fails on output STREAMON if capture is already streaming. This patch delays capture activation until output is configured and streaming https://bugzilla.gnome.org/show_bug.cgi?id=796693 2018-06-23 23:44:19 +0200 Tim-Philipp Müller * ext/gtk/gtkgstglwidget.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: Update for g_type_class_add_private() deprecation in recent GLib https://gitlab.gnome.org/GNOME/glib/merge_requests/7 2018-06-20 10:03:59 +0200 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Protect input stream with lock This was the last remaining place where modifying/unreffing the input stream was not protected by the lock https://bugzilla.gnome.org/show_bug.cgi?id=796639 2018-06-18 12:13:48 +0300 Sebastian Dröge * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Make sure events are writable before setting their seqnum 2018-05-28 15:19:52 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Drop truncated frames Drop truncated frames regardless if they have the ERROR flag or not. Truncated frame causes video frame map failure in many elements including cluttersink, glupload etc. 2018-04-02 12:59:33 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Try return input buffer soon In this patch we use a non-blocking poll in order to return all input buffers (buffers from v4l2-output queue). This prevent holding too long on upstreaming buffer in importing. https://bugzilla.gnome.org/show_bug.cgi?id=794904 2018-06-07 13:56:03 +1000 Matthew Waters * ext/qt/meson.build: * tests/examples/qt/qmlsink/meson.build: * tests/examples/qt/qmlsrc/meson.build: qt: also check for un-suffixed moc e.g. Qt windows installer doesn't have suffixes 2018-06-06 11:44:33 -0400 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Do not set INVALID seqnum on events 2018-06-01 22:47:10 +0900 Seungha Yang * tests/check/elements/qtdemux.c: tests: qtdemux: Add checking exposed segment event https://bugzilla.gnome.org/show_bug.cgi?id=796480 2018-06-01 21:08:10 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Forward upstream time-format segment without mapping Sample table based segment event (genereted by qtdemux) could break presentation timeline. For example, qtdemux should not modify upstream time format segment (e.g., adaptivedemux use case) https://bugzilla.gnome.org/show_bug.cgi?id=796480 2018-04-19 08:14:47 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Seek handling is always done with a valid event Remove the checks 2018-06-06 07:46:54 +0200 Edward Hervey * gst/wavparse/gstwavparse.c: wavparse: Don't set invalid seqnum on events Some codepath will call gst_wavparse_perform_seek without an event and therefore without a valid seqnum 2018-05-25 12:28:04 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Clarify field name about stream-encryption-system This field is actually only informatory and the user can potentially choose something else. EME tests in WebKit testsuite actually doesn't take it into and force another encryption system to be used, and expects to be given the occasion to do so. This basically also reverts 3e063703b3a51b8aaa7f75f36c4660c583a60e93. 2018-05-28 11:01:42 -0700 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: mark segment as sent after pushing when moov is received Otherwise we would try to send it a second time if the same moov is received or in any other situation that might trigger segment sending. https://bugzilla.gnome.org/show_bug.cgi?id=752603 2018-05-28 10:59:14 -0700 Thiago Santos * tests/check/elements/qtdemux.c: tests: qtdemux: Avoid using data beyond array and improve error msg Makes it easier to debug the failures as well as prevents problems reading out of bounds data. 2018-05-16 20:16:44 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't open the device in get property This is both racy and inefficient. This function is still missing some locking which will be address in later patch. https://bugzilla.gnome.org/show_bug.cgi?id=796185 2018-05-27 20:29:47 +0100 Tim-Philipp Müller * tests/check/elements/rtpstorage.c: * tests/check/elements/rtpulpfec.c: tests: rtpstorage: fix potential crashes / test failures on 32-bit Pass 64 bits to g_object_set() for 64-bit integer properties like rtpstorage's "size-time" property. https://bugzilla.gnome.org/show_bug.cgi?id=796429 2018-05-13 21:59:49 -0700 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: do not update segment.stop is it is not a valid time Otherwise it overflows and starts having a meaningful and wrong value. https://bugzilla.gnome.org/show_bug.cgi?id=752603 2016-04-26 16:54:30 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: offset edts segments by the min timestamp of the stream Otherwise if the stream is starting at timestamp=X it would wait 'X' to start playing. https://bugzilla.gnome.org/show_bug.cgi?id=752603 2016-04-26 14:34:16 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: rework segment event pushing Instead of always keeping a safe segment (start=0) event from the beginning, delay the creation of this event to when we really know the timestamp of the first sample. This is important to properly start fragmented streams that we might join in the middle or to play isolated fragment files that might have an advanced tfdt. https://bugzilla.gnome.org/show_bug.cgi?id=752603 2018-05-25 10:49:21 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Do not unref a NULL stream_tags stream->stream_tags is reset to NULL once we expose the stream and these have been consumed, we need to check that when cleaning up the stream. 2018-05-25 10:17:29 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Do not run the preferred decryptor context query if no decryptor avalaible Ultimately this avoids a segfault as the code expect a non NULL array here. 2018-03-30 17:03:13 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Allow edit lists on fragmented files on push mode Fragmented files often use elst.duration=0 which before ee78825eaef2c5fffac7d6c5526fe18cec6b3eef was wrongly interpreted as having no frames. Since that issue has now been fixed, there is no reason to disable edit lists in fragmented files. This commit enables them, therefore producing correct stream time for files containing edit lists. https://bugzilla.gnome.org/show_bug.cgi?id=793058 2018-05-24 12:58:00 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: fix computation of first_duration for fragmented files in push mode Since ca068865c391e87932b1268d0c675be233dd2ffe the duration of the first frame is not used for estimating the frame rate. For this purpose, stream->first_duration was initialized with the duration of the first frame. In fragmented files, this was previously done by peeking the first moof, but that can only be done in pull mode. Fortunately, we don't really need to do that, at least with the current design: When we are estimating the frame rate we already have the sample table, regardless of the scheduling mode and whether the file is fragmented or not, so we can obtain first_duration there much more reliably. This fixes frame rate estimation for fragmented files in push mode. https://bugzilla.gnome.org/show_bug.cgi?id=796384 2017-06-13 17:42:55 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmuxsink: Added new async-finalize mode This mode is useful for muxers that can take a long time to finalize a file. Instead of blocking the whole upstream pipeline while the muxer is doing its stuff, we can unlink it and spawn a new muxer+sink combination to continue running normally. This requires us to receive the muxer and sink (if needed) as factories, optionally accompanied by their respective properties structures. Also added the muxer-added and sink-added signals, in case custom code has to be called for them. https://bugzilla.gnome.org/show_bug.cgi?id=783754 2018-05-23 19:00:48 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Don't send gaps bigger than 1 second (now in push mode too) This applies the same workaround to gaps that is being used in pull mode. https://bugzilla.gnome.org/show_bug.cgi?id=778426 2018-05-23 20:08:56 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Properly handle edit list in push mode If there are empty segments in edit list, demux should adjust "accumulated_base" to apply it into running time. https://bugzilla.gnome.org/show_bug.cgi?id=778426 2018-05-22 22:14:03 +0200 Mathieu Duponchelle * gst/matroska/matroska-mux.c: matroska-mux: write colorimetry This is a straightforward translation of 5dd39d8, can be trivially checked by running: gst-launch-1.0 -v videotestsrc ! video/x-raw, colorimetry=2:4:7:1 ! \ matroskamux ! matroskademux ! fakesink and verifying that the colorimetry is correctly preserved. https://bugzilla.gnome.org/show_bug.cgi?id=796344 2018-03-31 17:19:03 +0200 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: fix buggy duration in edits with duration=0 in fragmented files without a mehd https://bugzilla.gnome.org/show_bug.cgi?id=794858 2018-05-23 13:14:27 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpsession.h: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstmultiudpsink.c: docs: fix typos 2018-03-31 18:42:47 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Clarify variable name As defined by spec, use "empty edit". It's more straightforward. https://bugzilla.gnome.org/show_bug.cgi?id=778426 2017-06-21 17:59:21 +0200 Xabier Rodriguez Calvar * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: add context for a preferred protection qtdemux selected the first system corresponding to a working GStreamer decryptor. With this change, before selecting that decryptor, qtdemux will check if it has context (a preferred decryptor id) and if not, it will request it. The request includes track-id, available key system ids for the available decryptors and even the events so that the init data is accessible. [eocanha@igalia.com: select the preferred protection system even if not available] Test "4. ClearKeyVideo" in YouTube leanback EME conformance tests 2016 for H.264[1] uses a media file[2] with cenc encryption which embeds 'pssh' boxes with the init data for the Playready and Widevine encryption systems, but not for the ClearKey encryption system (as defined by the EMEv0.1b spec[3] and with the encryption system id defined in [4]). Instead, the ClearKey encryption system is manually selected by the web page code (even if not originally detected by qtdemux) and the proper decryption key is dispatched to the decryptor, which can then decrypt the video successfully. [1] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2016.html?test_type=encryptedmedia-test&webm=false [2] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/media/car_cenc-20120827-86.mp4 [3] https://dvcs.w3.org/hg/html-media/raw-file/eme-v0.1b/encrypted-media/encrypted-media.html#simple-decryption-clear-key [4] https://www.w3.org/Bugs/Public/show_bug.cgi?id=24027#c2 https://bugzilla.gnome.org/show_bug.cgi?id=770107 2017-05-20 16:55:40 +0000 Enrique Ocaña González * gst/isomp4/qtdemux.c: qtdemux: also push buffers without encryption info instead of dropping them Test "17. PlayReadyH264Video" in YouTube leanback EME conformance tests 2016 for H.264[1] uses a media file[2] with cenc encryption whose first two 'moof' boxes have no encryption information (no 'saiz' and 'saio' boxes). Those boxes are actually not encrypted and the current qtdemux implementation was just dropping them, breaking the test use case. This patch detects those kind of situations and just lets the unencrypted buffers pass. Of course, this needs some collaboration by the decryptors, which should also do the same and not to try to decrypt those clear buffers. [1] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/2016.html?test_type=encryptedmedia-test&webm=false [2] http://yt-dash-mse-test.commondatastorage.googleapis.com/unit-tests/media/oops_cenc-20121114-142.mp4 https://bugzilla.gnome.org/show_bug.cgi?id=770107 2018-05-21 11:49:08 +0100 Tim-Philipp Müller * meson.build: meson: use cdata.set_quoted() in more places 2018-05-21 11:46:59 +0100 Tim-Philipp Müller * meson.build: * meson_options.txt: meson: add 'nls' option to disable translations And enable by default. Was implicitly disabled because ENABLE_NLS was not defined. 2016-02-09 14:00:00 -0800 Andre McCurdy * ext/taglib/gstid3v2mux.cc: id3v2mux: ensure valid sentinal for gst_structure_get() gst_structure_get() is declared with G_GNUC_NULL_TERMINATED, ie __attribute__((__sentinel__)), which means gcc will generate a warning if the last parameter passed to the function is not NULL (where a valid NULL in this context is defined as zero with any pointer type). The C code callers to gst_structure_get() within gst-plugins-good use the C NULL definition (ie ((void*)0)), which is a valid sentinel. However gstid3v2mux.cc uses the C++ NULL definition (ie 0L), which is not a valid sentinel without an explicit cast to a pointer type. Upstream-Status: Pending Signed-off-by: Andre McCurdy 2016-02-03 18:12:38 -0800 Andre McCurdy * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: raw1394: avoid including directly Note from Edward Hervey: Patch from git.yoctoproject.org musl libc generates warnings if is included directly. Upstream-Status: Pending Signed-off-by: Andre McCurdy 2018-02-23 13:38:32 +0100 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux_parse_segments: remove superfluous variable https://bugzilla.gnome.org/show_bug.cgi?id=793751 2018-04-23 13:29:30 -0400 Olivier Crête * gst/flv/gstflvmux.c: flvmux: Remove custom get_next_time implementation GstAggregator now does the same thing in the simple implementation. https://bugzilla.gnome.org/show_bug.cgi?id=795486 2018-05-15 11:50:30 +0100 Havard Graff * tests/check/elements/rtpsession.c: rtpsession: Add tests for PLI and FIR https://bugzilla.gnome.org/show_bug.cgi?id=795139 2018-04-30 08:41:19 +0200 Havard Graff * gst/rtpmanager/gstrtpsession.c: rtpsession: make "clear-pt-map" action signal actually work Needed for PLI + FIR unit tests in follow-up commit. https://bugzilla.gnome.org/show_bug.cgi?id=795139 2016-10-06 16:08:38 +0200 Mikhail Fludkov * gst/rtpmanager/rtpsession.c: rtpsession: Avoid unnecessary copy of stats structure The code before copied GstStructure twice. The first time inside gst_value_set_structure and the second time in g_value_array_append. Optimized version does no copies, just transfers ownership to GValueArray. It takes advantage of the fact that array has already enough elements preallocated and the memory is zero initialized. https://bugzilla.gnome.org/show_bug.cgi?id=795139 2018-05-15 10:35:09 +0100 Tim-Philipp Müller * gst/replaygain/gstrgvolume.c: Revert "BugFix : Change peak value to normalize audio file with fallback gain" This reverts commit 36e49fd6f872f0b3f33083107a55fb7f671a47d0. Breaks unit test, someone needs to investigate if it's the patch's fault or if the test needs adjusting/updating. https://bugzilla.gnome.org/show_bug.cgi?id=673970 2016-12-13 10:13:52 +0100 Stian Selnes * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Drop packet if trying to send from non-internal source If obtain_internal_source() returns a source that is not internal it means there exists a non-internal source with the same ssrc. Such an ssrc collision should be handled by sending a GstRTPCollision event upstream and choose a new ssrc, but for now we simply drop the packet. Trying to process the packet further will cause it to be pushed usptream (!) since the source is not internal (see source_push_rtp()). https://bugzilla.gnome.org/show_bug.cgi?id=795139 2018-05-14 00:29:24 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: tag disabled streams with FLAG_UNSELECT So they're never picked as default, only by explicit user action. https://bugzilla.gnome.org/show_bug.cgi?id=690911 2018-05-14 21:06:55 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Print expected/actual values in debug log on mismatch in prefill mode This helps debugging a lot. 2018-04-10 18:05:47 +0200 Havard Graff * gst/rtpmanager/rtpsession.c: * tests/check/Makefile.am: * tests/check/elements/rtpsession.c: rtpsession: Try media_ssrc if no src can be found for PLI sender_ssrc Some RTP stacks out there does not set the sender_ssrc. In order to be more robust, try to lookup the media_ssrc before dropping the PLI. https://bugzilla.gnome.org/show_bug.cgi?id=795139 2017-08-25 11:59:00 +0200 Mikhail Fludkov * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Fix on-feedback-rtcp race If there is an external source which is about to timeout and be removed from the source hashtable and we receive feedback RTCP packet with the media ssrc of the source, we unlock the session in rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal allowing rtcp timer to kick in and grab the lock. It will get rid of the source and rtp_session_process_feedback will be left with RTPSource with ref count 0. The fix is to grab the ref to the RTPSource object in rtp_session_process_feedback. https://bugzilla.gnome.org/show_bug.cgi?id=795139 2017-11-27 10:56:47 +0100 Stian Selnes * gst/rtpmanager/rtpsession.c: rtpsession: Add missing lock around sess->ssrcs iteration https://bugzilla.gnome.org/show_bug.cgi?id=795139 2017-08-25 11:22:47 +0200 John-Mark Bell * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: do not emit RBs for internal senders. These are the sources we send from, so there is no reason to report receive statistics for them (as we do not receive on them, and the remote side has no knowledge of them). https://bugzilla.gnome.org/show_bug.cgi?id=795139 2018-04-10 18:22:57 +0200 Havard Graff * tests/check/elements/rtpsession.c: tests: rtpsession: fix indentation https://bugzilla.gnome.org/show_bug.cgi?id=795139 2018-05-12 08:03:28 +0200 Edward Hervey * sys/v4l2/gstv4l2videodec.c: v4l2: Fix typo in debug messages It's a decoder, not an encoder :) https://bugzilla.gnome.org/show_bug.cgi?id=795941 2018-03-22 18:00:37 +0100 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Added caption_%u pad template For closed-caption-enabled muxers (e.g. qtmux) 2018-05-10 13:57:30 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Initialize riff library Avoids debugging message issues. Also just use the main riff header 2018-05-08 20:31:41 +0900 Seungha Yang * tests/check/elements/qtdemux.c: * tests/check/elements/qtdemux.h: tests: qtdemux: Add test for stream change Add test case to verify track-id change and stream change https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 20:30:18 +0900 Seungha Yang * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Protect _expose_streams() from flush event Flush during stream change can break autoplugging or the flush event could be dropped. https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 20:26:41 +0900 Seungha Yang * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Try to expose whenever got new moov or new stream-start Whenever got new moov or new stream-start, demux will try to expose new pad by following rule. Comparing stream-id in the current moov with previous one, then * If matched stream-id is found from previous one, reuse existing pad (most common case) * Otherwise, expose new pad with new stream-start * No more used stream will be freed https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 20:10:39 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Remove duplication of initializing member variables Most initialization of variables in gst_qtdemux_init() are duplicated in gst_qtdemux_reset() function. https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 20:09:10 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Create stream whenever got new moov Whenever demux got moov, demux will create new stream. Only exception is duplicated track-id in a moov box. In that case the first stream will be accepted. This patch is pre-work for rework of moov handling. https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 19:57:11 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Store stream-id to manage streams In order to figure out stream change such as track-id change or stream-id change, demux will store stream-id per QtDemuxStream structure. https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 19:39:02 +0900 Seungha Yang * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Use GList to manage QtDemuxStream * Move to GList from static array * Logging track-id instead of array index. It's more meaningful. https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 18:44:15 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Adjust the number of args of some functions To be used with g_list_free_full in the next patch https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-05-08 18:22:58 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Add parentheses in macro https://bugzilla.gnome.org/show_bug.cgi?id=684790 2018-03-19 23:36:13 +0100 Marinus Schraal * gst/isomp4/qtdemux.c: isomp4: Use full date time if available The ©day tag contains a full date time, use it for the DATE_TIME tag instead of just the DATE tag. This overrules the unreliable qt creation time. https://bugzilla.gnome.org/show_bug.cgi?id=731029 2018-03-15 22:59:39 +1100 Jan Schmidt * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix doc comment markers 2016-02-02 18:58:24 +0200 Kyrylo Polezhaiev * gst/icydemux/gsticydemux.c: icydemux: avoid timestamp field initialisation for tag event This field is not used and will be removed in 2.0 API. https://bugzilla.gnome.org/show_bug.cgi?id=761462 2014-10-05 15:51:18 +0200 Matej Knopp * gst/audioparsers/gstdcaparse.c: dcaparse: do not accept header with invalid channel count https://bugzilla.gnome.org/show_bug.cgi?id=737928 2018-05-05 19:27:24 +0530 Nirbheek Chauhan * meson.build: * meson_options.txt: * sys/v4l2/meson.build: meson: Update option names to omit disable_ and with- prefixes Also yield common options to the outer project (gst-build in our case) so that they don't have to be set manually. 2012-04-12 09:53:24 +0200 Anthony Violo * gst/replaygain/gstrgvolume.c: BugFix : Change peak value to normalize audio file with fallback gain https://bugzilla.gnome.org/show_bug.cgi?id=673970 2018-05-05 16:32:59 +0200 Tim-Philipp Müller * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: don't use buffer lists if everything fits into one buffer People might use very large mtu sizes where every payload fits into a single output packet. https://bugzilla.gnome.org/show_bug.cgi?id=795758 2018-04-04 15:50:55 +0200 Kirill Marinushkin * configure.ac: configure: Fix hard-coded enabled v4l2 probe on Linux/ARM Currently, enable_v4l2_probe is hard-coded to "yes" on linux, platforms arm and aarch64. This even overrides the --disable-v4l2-probe argument. As a result, it is impossible to disable v4l2_probe. It becomes a problem for use-cases, when startup time is critical, because the v4l2_probe feature increases the initialization time. This commit makes the v4l2_probe feature configurable. On linux, platforms arm and aarch64, the default value is still "yes". But now it can be disabled by the --disable-v4l2-probe argument. https://bugzilla.gnome.org/show_bug.cgi?id=795200 2018-04-23 11:26:12 -0400 Olivier Crête * gst/flv/gstflvmux.c: flvmux: Don't wake up the muxer unless there is data https://bugzilla.gnome.org/show_bug.cgi?id=795332 2018-04-23 11:19:18 -0400 Olivier Crête * gst/flv/gstflvmux.c: flvmux: Save the current position in the output segment https://bugzilla.gnome.org/show_bug.cgi?id=795332 2018-04-19 17:53:51 -0400 Olivier Crête * gst/flv/gstflvmux.c: * tests/check/elements/flvmux.c: flvmux: Wait for caps from both srcs before writing header Wait for caps on all pads to start writing data even when source is live. Includes unit test by Havard Graff that simulates it. https://bugzilla.gnome.org/show_bug.cgi?id=794722 2018-04-13 13:29:06 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/v4l2_calls.c: v4l2: rely on gst_v4l2_dup() to set no_initial_format and keep_aspect gst_v4l2_dup() will now take care of setting v4l2capture->no_initial_format and keep_aspect instead of doing it manually. Fix a typo as keep_aspect was set twice on v4l2output but never on v4l2capture. https://bugzilla.gnome.org/show_bug.cgi?id=795028 2018-04-24 14:06:10 -0400 Xavier Claessens * ext/cairo/meson.build: * ext/dv/meson.build: * ext/flac/meson.build: * ext/gdk_pixbuf/meson.build: * ext/gtk/meson.build: * ext/jack/meson.build: * ext/jpeg/meson.build: * ext/lame/meson.build: * ext/libpng/meson.build: * ext/mpg123/meson.build: * ext/pulse/meson.build: * ext/shout2/meson.build: * ext/soup/meson.build: * ext/speex/meson.build: * ext/taglib/meson.build: * ext/twolame/meson.build: * ext/vpx/meson.build: * ext/wavpack/meson.build: * gst/alpha/meson.build: * gst/apetag/meson.build: * gst/audiofx/meson.build: * gst/audioparsers/meson.build: * gst/auparse/meson.build: * gst/autodetect/meson.build: * gst/avi/meson.build: * gst/cutter/meson.build: * gst/debugutils/meson.build: * gst/deinterlace/meson.build: * gst/dtmf/meson.build: * gst/effectv/meson.build: * gst/equalizer/meson.build: * gst/flv/meson.build: * gst/flx/meson.build: * gst/goom/meson.build: * gst/goom2k1/meson.build: * gst/icydemux/meson.build: * gst/id3demux/meson.build: * gst/imagefreeze/meson.build: * gst/interleave/meson.build: * gst/isomp4/meson.build: * gst/law/meson.build: * gst/level/meson.build: * gst/matroska/meson.build: * gst/monoscope/meson.build: * gst/multifile/meson.build: * gst/multipart/meson.build: * gst/replaygain/meson.build: * gst/rtp/meson.build: * gst/rtpmanager/meson.build: * gst/rtsp/meson.build: * gst/shapewipe/meson.build: * gst/smpte/meson.build: * gst/spectrum/meson.build: * gst/udp/meson.build: * gst/videobox/meson.build: * gst/videocrop/meson.build: * gst/videofilter/meson.build: * gst/videomixer/meson.build: * gst/wavenc/meson.build: * gst/wavparse/meson.build: * gst/y4m/meson.build: * meson.build: * sys/directsound/meson.build: * sys/v4l2/meson.build: * sys/ximage/meson.build: Meson: Generate pc file for all plugins in good https://bugzilla.gnome.org/show_bug.cgi?id=794568 2018-04-25 10:58:41 +0100 Tim-Philipp Müller * meson.build: meson: use -Wl,-Bsymbolic-functions where supported Just like the autotools build. 2018-04-25 10:37:40 +0200 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Read caption from input buffer And not from unallocated output buffer GstMapInfo CID #1435131 2018-02-07 11:00:18 +0100 Edward Hervey * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/gstqtmuxmap.h: isomp4: qtmux: Add Closed Caption support Supports CEA 608 and CEA 708 CC streams Also supports usage in "Robust Prefill" mode if the incoming caption stream is constant (i.e. there is one incoming CC buffer for each video frame). https://bugzilla.gnome.org/show_bug.cgi?id=606643 2018-02-06 15:38:00 +0100 Edward Hervey * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: isomp4: Make 'gmhd' atom usage more generic Only the 'gmin' atom is required. Any other entry within it are optional. https://bugzilla.gnome.org/show_bug.cgi?id=606643 2018-04-22 10:40:19 -0300 Thibault Saunier * ext/jpeg/gstjpegenc.c: jpegenc: Accept sof-marker=4 sof-marker is 4 when input is in the RGB colorspace. https://bugzilla.gnome.org/show_bug.cgi?id=795463 2018-04-02 16:06:35 +0200 Mathieu Duponchelle * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecdec.h: * tests/check/elements/rtpulpfec.c: ulpfecdec: output perfect seqnums ULP FEC, as defined in RFC 5109, has the protected and protection packets sharing the same ssrc, and a different payload type, and implies rewriting the seqnums of the protected stream when encoding the protection packets. This has the unfortunate drawback of not being able to tell whether a lost packet was a protection packet. rtpbasedepayload relies on gaps in the seqnums to set the DISCONT flag on buffers it outputs. Before that commit, this created two problems: * The protection packets don't make it as far as the depayloader, which means it will mark buffers as DISCONT every time the previous packets were protected * While we could work around the previous issue by looking at the protection packets ignored and dropped in rtpptdemux, we would still mark buffers as DISCONT when a FEC packet was lost, as we cannot know that it was indeed a FEC packet, even though this should have no impact on the decoding of the stream With this commit, we consider that when using ULPFEC, gaps in the seqnums are not a reliable indicator of whether buffers should be marked as DISCONT or not, and thus rewrite the seqnums on the decoding side as well to form a perfect sequence, this obviously doesn't prevent the jitterbuffer from doing its job as the ulpfec decoder is downstream from it. https://bugzilla.gnome.org/show_bug.cgi?id=794909 2018-04-17 17:57:16 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * tests/examples/rtsp/test-onvif.c: Revert "rtspsrc: Fix up sendonly/recvonly attribute handling" This reverts commit af273b4de9eb292c0b6af63665e10ca015895902. While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say the opposite, just like the ONVIF standard. Let's follow those RFCs as we're doing RTSP here, and add a property at a later time if needed to switch to the SDP RFC behaviour. https://bugzilla.gnome.org/show_bug.cgi?id=793964 2018-04-16 21:27:47 +0300 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Drain the parser when a CAPS event is received After a CAPS event, in theory a new stream can start and it might start with the FLAC headers again. We can't detect FLAC headers in the middle of the stream, so we drain the parser to be able to detect either FLAC headers after the CAPS event or the continuation of the previous stream. This fixes for example gst-launch-1.0 audiotestsrc num-buffers=200 ! flacenc ! c. \ audiotestsrc num-buffers=200 freq=880 ! flacenc ! c. \ concat name=c ! rtpgstpay ! udpsink host=127.0.0.1 port=5000 gst-launch-1.0 udpsrc multicast-group=127.0.0.1 port=5000 \ caps=application/x-rtp,media=application,clock-rate=90000,encoding-name=X-GST ! \ rtpgstdepay ! flacparse ! flacdec ! audioconvert ! pulsesin 2018-04-16 10:52:56 +0100 Tim-Philipp Müller * README: * common: Automatic update of common submodule From 3fa2c9e to ed78bee 2018-04-05 16:05:12 +1000 Matthew Waters * ext/meson.build: * ext/qt/gstqtglutility.cc: * ext/qt/meson.build: * tests/examples/meson.build: * tests/examples/qt/meson.build: * tests/examples/qt/qmlsink/CMakeLists.txt: * tests/examples/qt/qmlsink/meson.build: * tests/examples/qt/qmlsink/play.pro: * tests/examples/qt/qmlsink/qmlsink.qrc: * tests/examples/qt/qmlsrc/grabqml.pro: * tests/examples/qt/qmlsrc/meson.build: * tests/examples/qt/qmlsrc/qmlsrc.qrc: meson: add build files for the qml plugin Tested on linux with X11/wayland and semi-tested on Windows. Windows crashes on item destruction however this is better than nothing. Fix up some win32 build issues on the way with mismatched {} and G_STMT_{START,END} 2018-04-13 23:02:26 +0200 Mathieu Duponchelle * tests/check/elements/flvmux.c: flvmux test: refactor looped test. Looping the test 500 times to only execute the test once every 33 times means we inited and deinited gstreamer 467 times for no reason at all, which was annoying when running the test with valgrind. 2018-04-13 23:01:20 +0200 Mathieu Duponchelle * gst/flv/gstflvmux.c: flvmux: unref return of aggregator_pad_peek_buffer We ended up leaking every single buffer going through the muxer, which is far from ideal 2018-04-13 22:49:43 +0200 Mathieu Duponchelle * gst/isomp4/gstqtmux.c: qtmux: Fix leak gst_qt_mux_can_renegotiate () gets called everywhere following that pattern: return gst_qt_mux_can_renegotiate (ref(self)); This means the reference must be released both in the success and failure cases, it was only done in the success case. 2018-04-13 22:44:14 +0200 Mathieu Duponchelle * gst/flv/gstflvmux.c: flvmux: aggregate should not push EOS itself Instead it is expected to return GST_FLOW_EOS, and let the base class handle that. 2018-04-13 21:19:02 +0200 Mathieu Duponchelle * tests/check/gst-plugins-good.supp: valgrind supps: ignore gnutls leaking a certificate After investigating, we do dispose of the TLS connections appropriately in the souphttpsrc test, which in turn calls gnutls_deinit, but certificates get leaked anyway. 2018-04-13 20:35:24 +0200 Mathieu Duponchelle * tests/check/elements/souphttpsrc.c: souphttpsrc test: free g_get_current_dir return 2018-04-13 20:31:07 +0200 Mathieu Duponchelle * tests/check/gst-plugins-good.supp: valgrind supps: bring getaddrinfo suppression from -base 2018-04-13 20:28:35 +0200 Mathieu Duponchelle * tests/check/gst-plugins-good.supp: valgrind supps: ignore more twolame conditional moves 2018-04-13 17:37:47 +0200 Mathieu Duponchelle * tests/check/elements/rtpulpfec.c: rtpulpfec tests: Fix leaks 2018-02-16 23:40:50 +0100 Alicia Boya García * gst/matroska/matroska-demux.c: matroskademux: Add comment about Opus clipping https://bugzilla.gnome.org/show_bug.cgi?id=793523 2018-04-11 20:28:00 +0000 Whoopie * sys/v4l2/gstv4l2object.c: v4l2object: Disable DMABuf for emulated formats libv4l2 does not prevent exporting DMABuf even when emulated formats are in use. As a side effect, userspace ends up with buffers of the original formats which will cause issues. https://bugzilla.gnome.org/show_bug.cgi?id=795097 2018-04-08 20:42:16 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Only use BT2020_12 for BT2020 v4l2 colorspace BT2020_12 is not represented in V4L2, so drivers providing full colority for BT2020 will set V4L2_XFER_FUNC_709 transfer function. To fix the issue, we bump this to BT2020_12 if the resoltion is 4K, but we should only do that if the colorspace is BT2020 to start with, otherwise it's not possible to use normal BT709 for 4K 8bit formats. 2018-04-08 13:43:56 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Always set the colorimetry in S_FMT So far we were only setting colorimetry for OUTPUT devices (v4l2sink or m2m sink pad). This prevented selecting through caps negotiation the colorimetry for CAPTURE devices (v4l2src or m2m src pad). This is rarely selectable, but trying is harmless. 2018-04-11 21:41:58 +0200 Sebastian Dröge * gst/monoscope/gstmonoscope.c: monoscope: Only fixate pixel-aspect-ratio if the field exists 2018-04-11 17:54:38 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't send fragment-opened-closed message if the reference ctx is NULL It can happen during teardown that the reference context becomes NULL. In that case, trying to send the fragment-opened-closed message would lead to a crash. 2018-04-11 09:12:09 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Run gst_iterator_foreach() as long as it returns GST_ITERATOR_RESYNC CID 1434160 2018-04-11 08:51:32 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Fix comparision for extra caption atom We want to make sure we have *enough* data for the potential 2nd caption atom. CID #1434161 2018-04-11 08:42:54 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Handle bogus caption samples Corrupted files could potentially have multiple cdat/cdt2 atoms in a sample entry, which is unclear how to handle. Ignore repeated ones. CID #1434162 CID #1434159 2018-04-10 21:15:48 +0200 Sebastian Dröge * gst/monoscope/gstmonoscope.c: monoscope: Fixate pixel-aspect-ratio too and make sure the final caps are completely fixated Otherwise e.g. this fails with assertions: gst-launch-1.0 audiotestsrc ! audioconvert ! monoscope ! videoconvert ! \ videoscale ! video/x-raw,width=800,height=600 ! ximagesink 2018-03-08 10:10:01 +0100 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Add comments and doc about prefill mode 2018-02-06 14:36:50 +0100 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Refactor pad re-negotiation code It was similar for all pads https://bugzilla.gnome.org/show_bug.cgi?id=606643 2018-01-31 15:10:03 +0100 Edward Hervey * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: qtdemux: Detect and expose CEA 608/708 Closed Caption tracks https://bugzilla.gnome.org/show_bug.cgi?id=606643 2018-04-04 01:48:44 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtprtxsend.c: rtxsend: fix wrong memory layout assumption The code responsible for creating retransmitted buffers assumed the stored buffer had been created with rtp_buffer_new_allocate when copying the extension data, which isn't necessarily the case, for example when the rtp buffers come from a udpsrc. https://bugzilla.gnome.org/show_bug.cgi?id=794958 2018-04-02 23:04:06 +0200 Mathieu Duponchelle * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: new signal "get-storage" Similar to the get-session and get-internal-session signals, we expose a get-storage signal in addition to the get-internal-storage signal to give access to the actual element for applications that need to set properties on the element, in particular "size-time" https://bugzilla.gnome.org/show_bug.cgi?id=794910 2018-03-29 19:19:21 +0300 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Add new reset-muxer property With this the muxer is not set to NULL after each segment but instead only flush events are sent to it to reset the EOS state. As a result, the muxer will keep stream state and e.g. mpegtsmux will keep the packet continuity counter continuous between segments as needed by hlssink2. https://bugzilla.gnome.org/show_bug.cgi?id=794816 2018-04-02 12:48:50 +0100 Tim-Philipp Müller * tests/icles/Makefile.am: * tests/icles/meson.build: * tests/icles/v4l2src-test.c: tests: remove broken and now pointless v4l2src-test This tests APIs that don't exist any longer and also doesn't work at all, and was last touched in a meaningful way in 2006. 2018-03-21 00:19:37 +0900 Seungha Yang * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: Fix unknown type name ‘off_t’ error Fix following build error gstv4l2object.h:197:17: error: unknown type name ‘off_t’ gint fd, off_t offset); ^ https://bugzilla.gnome.org/show_bug.cgi?id=794533 2017-05-25 03:44:39 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: reject segment seeks https://bugzilla.gnome.org/show_bug.cgi?id=784681 2018-02-13 11:50:05 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Handle variant of vorbis in mp4 Comes from gpac apparently. The codec_data uses the same packing mechanism as matroska. https://bugzilla.gnome.org/show_bug.cgi?id=738244 2018-03-22 15:20:47 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Check sample count is valid in PIFF parsing The value stored in cenc_aux_sample_count wasn't in sync with the parsing code that followed which checks whether all entries are valid and present. Only write the actual sample count when we know for sure. CID #1427087 2018-03-04 15:14:08 +0100 Carlos Rafael Giani * configure.ac: * ext/qt/gstqtglutility.cc: qt: Get EGL native display from QPA if platform header is available https://bugzilla.gnome.org/show_bug.cgi?id=792378 2018-03-06 02:14:34 +0100 Petr Kulhavy * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: switch to using a buffer pool This exposes a new property, mtu, which is used to determine the initial size of buffers from the buffer pool. If received data exceeds this, the element gracefully handles that in a manner similar to what we had previously: a large memory gets filled and reallocated at the next call to "fill". The default size is set to 1500, which should cover most use cases. With contributions from Mathieu Duponchelle https://bugzilla.gnome.org/show_bug.cgi?id=772841 2016-11-15 09:39:31 +0100 Petr Kulhavy * gst/udp/gstudpsrc.h: udpsrc: optimize GstUdpSrc object for cache performance Optimize GstUdpSrc for cache performance. Move the hot properties, which are used by the read function, to the top: @used_socket, @addr, @cancellable, @skip_first_bytes, @timeout, @retrieve_sender_address. Remove the unused property @ttl. Where needed reorder so that holes are avoided (the 64-bit @timeout) https://bugzilla.gnome.org/show_bug.cgi?id=772841 2018-03-05 12:48:15 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix seeking on streams with frame reordering The samples table is sorted by DTS, not PTS. As such we can only get the correct result when using a binary search on it, if we search for the DTS. Also if we only ever search for the frame, where the following frame is the first one with a PTS after the search position, we will generally stop searching too early if frames are reordered. In forwards playback this is not really a problem (after the decoder reordered the frames, clipping is happening), in reverse playback it means that we can output one or more frames too few as we stop too early and the decoder would never receive it. https://bugzilla.gnome.org/show_bug.cgi?id=782118 2018-03-20 11:36:32 +0200 Sebastian Dröge * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/rtpstoragestream.c: * tests/check/elements/rtpred.c: * tests/check/elements/rtpulpfec.c: rtp: Fix compilation with non-C99 compilers By moving variable declarations out of loop headers. 2018-03-20 09:24:19 +0000 Tim-Philipp Müller * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * meson.build: Back to development === release 1.14.0 === 2018-03-19 20:18:22 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.14.0 2018-03-19 20:18:22 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2018-03-19 18:39:08 +0000 Tim-Philipp Müller * gst/rtp/gstrtpulpfecdec.c: rtpulpfecdec: fix build with older gcc As on Ubuntu Trusty. https://bugzilla.gnome.org/show_bug.cgi?id=794493 2018-03-19 10:58:28 +0200 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Allow splitting at exactly the time/bytes threshold 76e458a119926424e9dd5acf3210a592a314d713 changed the conditions from "queued > threshold" to "queued >= threshold", which broke hlssink2 and resulting in too small fragments being created although keyframes would be at *exactly* the configured threshold. https://bugzilla.gnome.org/show_bug.cgi?id=794440 2018-03-17 20:29:35 +0000 Tim-Philipp Müller * gst/rtp/rtpulpfeccommon.h: rtpulpfec: fix unconditional use of __attribute__ ((packed)) Fix compilation with MSVC. We still assume that attribute is supported by all other relevant compilers, which seems to be the case since we haven't had any complaints about similar code in rtpsbcpay. 2018-03-17 13:04:47 +0000 Tim-Philipp Müller * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/rtpulpfeccommon.c: rtpulpfec: don't use non-portable notation for 64-bit int constants Use GLib macro instead, even if it's a bit unwieldy. 2018-03-17 12:55:57 +0000 Tim-Philipp Müller * gst/rtp/gstrtpulpfecdec.c: rtpulpfecdec: don't use __builtin_ctzll unconditionally Fixes build with MSVC, and possibly other compilers too. === release 1.13.91 === 2018-03-13 19:16:42 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.13.91 2018-03-13 19:16:42 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2018-03-12 13:21:08 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: docs: rtpbin: add some Since markers for new properties 2018-03-10 18:57:38 +0530 Nirbheek Chauhan * sys/directsound/meson.build: meson: Add deviceprovider changes to directsoundsink These were missed when they were added to Makefile.am 2018-03-08 10:12:16 +0100 Michael Tretter * configure.ac: configure.ac: enable largefile support if possible https://bugzilla.gnome.org/show_bug.cgi?id=793103 2018-03-07 14:16:02 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: Fix support for 32bit mmap https://bugzilla.gnome.org/show_bug.cgi?id=793103 === release 1.13.90 === 2018-03-03 22:19:36 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.13.90 2018-03-03 22:19:36 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Update docs 2018-03-01 18:24:33 -0500 Olivier Crête * gst/flv/gstflvmux.c: * tests/check/elements/flvmux.c: flvmux: Duration & unit tests The muxed buffers will not carry the duration of the incoming buffers. https://bugzilla.gnome.org/show_bug.cgi?id=793457 2018-03-01 17:15:02 -0500 Olivier Crête * gst/flv/gstflvmux.c: flvmux: Set PTS based on running time https://bugzilla.gnome.org/show_bug.cgi?id=793457 2018-03-01 18:13:20 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Ignore sendonly/recvonly attributes unless a backchannel is configured This works around a bug in various ONVIF cameras that implement the attributes the wrong way around. They still won't work with a backchannel but at least normal playback will work for the time being. It restores pre-1.14 behaviour where we would fail to preroll on any SDP that lists a recvonly stream. For 1.16 a better solution should be found. The problem here is that the ONVIF spec has the meaning of the two attributes the wrong way around in the examples, compared to RFC4566. https://bugzilla.gnome.org/show_bug.cgi?id=793715 2018-03-01 18:16:24 +0100 Mathieu Duponchelle * meson.build: meson: enable more warnings https://bugzilla.gnome.org/show_bug.cgi?id=793961 2018-03-01 00:34:20 +0100 Mathieu Duponchelle * gst/flv/gstflvmux.c: Port to latest GstAggregator segment API The aggregator segment is now exposed on the src pad https://bugzilla.gnome.org/show_bug.cgi?id=793945 2018-03-01 15:34:13 +0530 Nirbheek Chauhan * sys/directsound/gstdirectsoundplugin.c: directsoundsink: Downgrade rank to match directsoundsrc in -bad As stated in commit c2956036b8da4b8f22a63a4f5a254be03e870aa6 in -bad, the wasapi elements are now better than directsound, and should be preferred if they are available. For a later release, once the elements have more testing, we can consider moving them to -good. 2018-02-28 19:21:53 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Only mark new clusters as keyframe if they start on a keyframe or we're muxing only audio Based on a patch by Nicola Murino https://bugzilla.gnome.org/show_bug.cgi?id=792775 2018-02-28 19:19:10 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Clip maximum cluster duration to the maximum possible value Only up to timescale * G_MAXINT16 is possible as cluster duration, which is already higher than our default value. Using higher values would cause overflows and broken files. Based on the investigation by Nicola Murino https://bugzilla.gnome.org/show_bug.cgi?id=792775 2018-02-26 13:03:59 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroska-mux: Refuse caps changes after starting to write headers Matroska does not support changing the stream type and stream properties after the headers were started to be written, and for example H264 codec_data changes can't be supported. https://bugzilla.gnome.org/show_bug.cgi?id=782949 2018-02-27 16:33:53 +0100 Mathieu Duponchelle * tests/check/elements/rtpred.c: tests: fix redenc tests The default of the allow-no-red-blocks property was changed in a previous commit, thus breaking the test assumptions 2018-02-27 13:13:49 +0000 Tim-Philipp Müller * gst/rtp/rtpulpfeccommon.c: rtp: fix another debug log printf format warning on 32-bit systems rtpulpfeccommon.c:432:27: error: format ‘%lx’ expects argument of type ‘long unsigned int’, but argument 10 has type ‘guint64 {aka long long unsigned int}’ https://bugzilla.gnome.org/show_bug.cgi?id=793732 2018-02-26 17:02:52 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: provide example usage for ignored-payload-types 2018-02-26 16:53:08 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpptdemux.c: rtpbin, rtpptdemux: Add missing Since markers 2018-02-26 15:57:28 +0100 Mathieu Duponchelle * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtpstorage.c: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/gstrtpulpfecenc.h: FEC elements: document, remove irrelevant properties The ulpfecenc "mux-seq" and "ssrc" properties were initially added because the element did more than implement ULPFEC. As it was decided that FLEXFEC would be implemented in a separate element, both properties are now unneeded and confusing. Change the default for the ulpfecenc multi-packet property, as it is expected that most users of this element will be protecting video streams. Change the default property for the rtpredenc allow-no-red-blocks property, as it should also be its default mode of operation. https://bugzilla.gnome.org/show_bug.cgi?id=793843 2018-02-24 20:05:05 +0100 Mathieu Duponchelle * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: do not warn when caps were not yet received It is expected that when connecting to a stream that has already started, the caps will only arrive at the interval specified on rtpgstpay, we shouldn't be warning as this is a normal mode of operation. https://bugzilla.gnome.org/show_bug.cgi?id=793798 2018-02-22 21:53:40 +0100 Arnaud Bonatti * gst/rtp/gstrtpulpfecdec.c: rtpulpfec: fix debug log printf format warning on 32-bit platforms https://bugzilla.gnome.org/show_bug.cgi?id=793732 2018-02-22 14:58:12 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-rtp.xml: * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtpstorage.c: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: docs: hook up new RTP FEC elements https://bugzilla.gnome.org/show_bug.cgi?id=792696 2018-02-22 14:57:58 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update for git master 2018-02-22 10:54:02 +0000 Tim-Philipp Müller * .gitignore: * tests/check/elements/.gitignore: .gitignore more test binaries 2018-02-21 20:46:10 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: also dist new fec test header file 2018-02-21 20:44:26 +0000 Tim-Philipp Müller * gst/rtp/Makefile.am: rtp: dist new header files Fixes make distcheck 2018-02-21 18:52:44 +0000 Tim-Philipp Müller * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpstorage.c: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/rtpulpfeccommon.c: * gst/rtp/rtpulpfeccommon.h: rtp: fec: fix build with gstreamer debug log system disabled 2018-02-21 19:59:04 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: do no assume sink caps are non NULL 2018-02-21 18:51:17 +0100 Mathieu Duponchelle * tests/check/Makefile.am: check: Fix ulpfec test build The test name was updated but not the build definition 2017-11-28 06:02:05 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: Expose FEC support signals Also slightly refactor complete_session_src https://bugzilla.gnome.org/show_bug.cgi?id=792696 2017-11-17 03:52:03 +0100 Mikhail Fludkov * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpreddec.c: * gst/rtp/gstrtpreddec.h: * gst/rtp/gstrtpredenc.c: * gst/rtp/gstrtpredenc.h: * gst/rtp/gstrtpstorage.c: * gst/rtp/gstrtpstorage.h: * gst/rtp/gstrtpulpfecdec.c: * gst/rtp/gstrtpulpfecdec.h: * gst/rtp/gstrtpulpfecenc.c: * gst/rtp/gstrtpulpfecenc.h: * gst/rtp/meson.build: * gst/rtp/rtpredcommon.c: * gst/rtp/rtpredcommon.h: * gst/rtp/rtpstorage.c: * gst/rtp/rtpstorage.h: * gst/rtp/rtpstoragestream.c: * gst/rtp/rtpstoragestream.h: * gst/rtp/rtpulpfeccommon.c: * gst/rtp/rtpulpfeccommon.h: * tests/check/Makefile.am: * tests/check/elements/packets.h: * tests/check/elements/rtpred.c: * tests/check/elements/rtpstorage.c: * tests/check/elements/rtpulpfec.c: * tests/check/meson.build: rtp: Implement ULPFEC (RFC 5109) We expose a set of new elements: * ULPFEC encoder / decoder * A storage element, which should be placed before jitterbuffers, and is used to store packets in order to attempt reconstruction after the jitterbuffer has sent PacketLost events * RED encoder / decoder (RFC 2198), these are necessary to use FEC in webrtc, as browsers will propose and expect ulpfec packets to be wrapped in red packets With contributions from: Mathieu Duponchelle Sebastian Dröge https://bugzilla.gnome.org/show_bug.cgi?id=792696 2017-11-28 01:11:54 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: rtpptdemux: Add ignored-payload-types property Packets with these payload types will be dropped. A use case for this is FEC, where we want FEC packets to go through the jitterbuffer, but not be output by rtpbin. https://bugzilla.gnome.org/show_bug.cgi?id=792696 2017-11-20 18:08:38 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: Add ssrc to output caps It may be useful downstream https://bugzilla.gnome.org/show_bug.cgi?id=792696 2018-02-21 11:12:10 +0100 Arnaud Bonatti * ext/gtk/gstgtkbasesink.c: gtk: fix compiler warning with recent glib https://bugzilla.gnome.org/show_bug.cgi?id=793688 2018-02-21 11:35:33 +1100 Matthew Waters * ext/qt/gstqtglutility.cc: qt: don't use libEGL functions when we don't link to libEGL Use the provided wrapper available from libgstgl. https://bugzilla.gnome.org/show_bug.cgi?id=793547 2018-02-18 21:38:13 +0100 Sebastian Dröge * gst/monoscope/gstmonoscope.c: * gst/monoscope/gstmonoscope.h: monoscope: Forward the SEGMENT event from the chain function Otherwise we'll break the event order and forward the SEGMENT event before sending a CAPS event. 2018-02-16 12:25:29 +0000 James Stevenson * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix missing read property of backchannel Add missing read property code for backchannel https://bugzilla.gnome.org/show_bug.cgi?id=793507 2018-02-16 09:42:59 +0000 Tim-Philipp Müller * tests/examples/rtsp/meson.build: examples: rtsp: fix meson build take 2 2018-02-16 11:30:01 +0200 Sebastian Dröge * tests/examples/rtsp/meson.build: rtsp: Fix meson.build of the example 2018-01-26 16:33:21 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Implement ONVIF backchannel support via TCP 2017-10-13 18:05:54 +0300 Nirbheek Chauhan * configure.ac: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: * tests/examples/Makefile.am: * tests/examples/meson.build: * tests/examples/rtsp/Makefile.am: * tests/examples/rtsp/meson.build: * tests/examples/rtsp/test-onvif.c: rtspsrc: Implement ONVIF backchannel support Set backchannel=onvif to enable, and use the 'push-backchannel-sample' action signal with the correct stream id. 2018-02-16 01:49:57 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: splitmuxsrc: Improve not-linked handling. Don't report not-linked unless all pads have returned not-linked. 2018-02-15 19:44:19 +0000 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * meson.build: Back to development === release 1.13.1 === 2018-02-15 17:06:10 +0000 Tim-Philipp Müller * NEWS: * configure.ac: * gst-plugins-good.doap: * meson.build: Release 1.13.1 2018-02-15 17:05:23 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gtk.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mpg123.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-qmlgl.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-twolame.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update plugin docs 2018-02-15 13:32:20 +0000 Tim-Philipp Müller * po/bg.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/fr.po: * po/hr.po: * po/hu.po: * po/nb.po: * po/nl.po: * po/pl.po: * po/ru.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: po: update translations 2018-02-14 16:38:07 +0100 Patrick Radizi * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: allow timestamps to move backwards The original solution for #784002 incorrectly assumed that timestamps may not move backwards and changed timestamps that did so. https://bugzilla.gnome.org/show_bug.cgi?id=784002 2018-02-15 00:58:38 +0000 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: * gst/flv/gstindex.c: * sys/v4l2/gstv4l2src.c: docs: remove pointless Since: 0.10.x markers 2017-09-27 16:01:35 +0200 Alban Bedel * gst/rtp/gstrtpvorbisdepay.c: rtpvorbisdepay: fix unbounded memory usage All received configurations are parsed and added to a list, this lead to an unbounded memory usage. As the configuration is resent every second this quickly lead to a large memory usage. Add a check to only add the config if it is not already available in the list. This fix only handle the typical case of a well behaved stream, a malicious server could still send many useless configurations to raise the client memory usage. 2018-02-12 18:41:41 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-qmlgl.xml: docs: add qt plugin https://bugzilla.gnome.org/show_bug.cgi?id=754094 2018-02-12 18:34:16 +0000 Tim-Philipp Müller * configure.ac: * ext/Makefile.am: * ext/meson.build: * tests/examples/meson.build: qt: hook up to build https://bugzilla.gnome.org/show_bug.cgi?id=754094 2018-02-12 18:13:17 +0000 Tim-Philipp Müller Move qt plugin from -bad https://bugzilla.gnome.org/show_bug.cgi?id=754094 2018-02-12 15:44:35 +0000 Tim-Philipp Müller * configure.ac: configure: fix build with --disable-external 2018-02-10 20:31:49 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-gtk.xml: docs: add moved gtk plugin to docs 2018-02-10 20:28:46 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-deinterlace.xml: docs: update for git master 2018-02-12 11:02:12 +0000 Tim-Philipp Müller * ext/gtk/meson.build: * ext/meson.build: * meson.build: * tests/examples/meson.build: gtk: hook up to meson build 2018-02-10 13:20:43 +0000 Tim-Philipp Müller * configure.ac: * ext/Makefile.am: * ext/gtk/Makefile.am: * tests/examples/Makefile.am: * tests/examples/gtk/.gitignore: * tests/examples/gtk/Makefile.am: gtk: hook up to autotools build 2018-02-10 12:49:36 +0000 Tim-Philipp Müller Move gtk plugin from -bad https://bugzilla.gnome.org/show_bug.cgi?id=754094 2018-02-09 11:26:56 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Fix comment typo in previous commit 2018-02-09 11:20:38 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: More 'meta' atom parsing fixes Turns out everybody is doing it their own way, so peek into the meta atom itself to figure out which spec it is following 2018-02-02 13:51:49 +0200 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Add support for muxing svmi atom for stereoscopic video information https://bugzilla.gnome.org/show_bug.cgi?id=793120 2018-02-09 08:59:56 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Minor cleanup Just move variables to the blocks where they are used. That function is massive, could do with some splitting up for readability :( 2018-02-09 08:54:05 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Cope with difference between QTFF and ISO BMFF specs The 'meta' atom is defined differently in QTFF and BMFF, so try to guess which spec the current stream applies to by looking at the major file type. 2018-02-09 08:35:52 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: isomp4: Make 'hdlr' atom dump more flexible The smallest possible is 24 (and not 25) bytes. The last "name" field can according to QTFF specifications not be present at all. The parser will handle this fine and so will the rest of the qtdemux code. 2018-02-09 08:35:25 +0100 Edward Hervey * gst/audiofx/audiopanoramaorc-dist.c: * gst/deinterlace/tvtime-dist.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videomixer/videomixerorc-dist.c: Update ORC files 2018-02-08 19:09:45 +0000 Tim-Philipp Müller * meson.build: meson: make version numbers ints and fix int/string comparison WARNING: Trying to compare values of different types (str, int). The result of this is undefined and will become a hard error in a future Meson release. 2017-10-01 18:21:26 +0200 Jérôme Laheurte * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideosink: fix build on macOS versions < 12.0 Use value instead of version macro when testing for mac OS version, since the define for the newer version may not be defined when compiling against older versions. https://bugzilla.gnome.org/show_bug.cgi?id=788402 2018-02-07 20:15:00 +1100 Matthew Waters * ext/qt/gstqtglutility.cc: qt: don't #include platform specific gstglcontext_*.h headers They aren't public headers 2018-02-04 11:47:05 +0100 Tim-Philipp Müller * configure.ac: * tests/check/Makefile.am: autotools: use -fno-strict-aliasing where supported https://bugzilla.gnome.org/show_bug.cgi?id=769183 2017-12-04 20:12:40 +0900 Justin Kim * gst/isomp4/gstqtmux.c: * gst/multifile/gstsplitmuxsink.c: qtmux: send stream warning when refusing video caps If codec_data is changed, the stream is no longer valid. Rather than keeping running when refusing new caps, this patch send a warning to the bus. Also fix up splitmuxsink to ignore this warning while changing caps. https://bugzilla.gnome.org/show_bug.cgi?id=790000 2017-11-29 21:30:11 +0900 Justin Kim * gst/rtp/gstrtph264depay.c: rtph264depay: update output caps regardless format `codec_data` should be transfered if any information of SPS/PPS is changed. https://bugzilla.gnome.org/show_bug.cgi?id=790000 2018-01-31 19:11:16 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_types.c: isomp4: Add gmhd/gmin debugging * gmhd is a container, mark it as such so we can see/dump what is contained within * Add dumping for the Base Media Information atom (gmin) 2015-09-23 10:01:32 +0200 Matthieu Crapet * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: add snapshot property Like pngenc, automatically send an EOS message. Example of bin: appsrc ! jpegenc snapshot=true ! filesink location=out.jpg This is especially useful for limited/slow hardware. Otherwise calling gst_video_convert_sample() is a better option (internally uses videoconvert and videoscale). https://bugzilla.gnome.org/show_bug.cgi?id=755453 2018-01-31 15:02:50 +0000 Philippe Normand * gst/interleave/interleave.c: interleave: fix memory leak of GAP buffers https://bugzilla.gnome.org/show_bug.cgi?id=793067 2018-01-31 11:38:35 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: qtdemux_dump: Demote verbose logging to TRACE level 2018-01-31 11:22:23 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: qtdemux: Re-enable full debug logging of stsz entries No idea why it was disabled (was the case since 2007) 2018-01-30 20:34:32 +0000 Tim-Philipp Müller * ext/taglib/meson.build: * meson.build: meson: use -fno-strict-aliasing where supported https://bugzilla.gnome.org/show_bug.cgi?id=769183 2017-12-12 00:14:02 +0900 Seungha Yang * gst/isomp4/qtdemux.h: qtdemux: Remove white space at end of line https://bugzilla.gnome.org/show_bug.cgi?id=791483 2017-12-12 00:11:24 +0900 Seungha Yang * gst/isomp4/Makefile.am: * gst/isomp4/gstisoff.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux_debug.h: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_lang.c: * gst/isomp4/qtdemux_types.c: qtdemux: Apply qtdemux debug category to gstisoff .. instead of the use of default debug category. And, make new header to declare the debug category https://bugzilla.gnome.org/show_bug.cgi?id=791483 2018-01-25 00:46:57 +0000 Tim-Philipp Müller * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: properly set total duration on outgoing segment We would accidentally pass through the duration value from the demuxer from a single fragment, which causes problems when feeding the stream from splitmuxsrc to rtsp-server. Streaming would stop after one fragment due to that. https://bugzilla.gnome.org/show_bug.cgi?id=792861 2018-01-25 00:42:52 +0000 Tim-Philipp Müller * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: don't respond to duration query with CLOCK_TIME_NONE total_duration is initialised to CLOCK_TIME_NONE, not 0, so check for that as well in order not to return an invalid duration to a duration query. Doesn't fix anything particular observed in practice, just seemed inconsistent. 2018-01-25 20:48:42 +0100 Alicia Boya García * gst/isomp4/qtdemux.c: qtdemux: Add more prose to the comment of gst_qtdemux_find_sample() https://bugzilla.gnome.org/show_bug.cgi?id=792910 2011-02-09 12:48:00 +0000 Oleksij Rempel * ext/vpx/gstvpxdec.c: vpx: add VP8_DEBUG_TXT_* flags for postprocessing https://bugzilla.gnome.org/show_bug.cgi?id=641399 2018-01-25 21:22:10 +0200 Sebastian Dröge * sys/directsound/gstdirectsoundsink.h: directsoundsink: Add missing \ in multi-line #define 2018-01-22 15:07:38 +0200 Sebastian Dröge * sys/directsound/Makefile.am: * sys/directsound/gstdirectsounddevice.c: * sys/directsound/gstdirectsounddevice.h: * sys/directsound/gstdirectsoundplugin.c: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: directsoundsink: Add support for a DeviceProvider https://bugzilla.gnome.org/show_bug.cgi?id=792782 2018-01-23 18:37:09 +0000 Tim-Philipp Müller * gst/multifile/gstmultifilesrc.c: multifilesrc: fix up uri handler a little Fix path escaping when creating URI from location in get_uri(). Return FALSE with an error when URI can't be parsed in set_uri(). https://bugzilla.gnome.org/show_bug.cgi?id=783581 2017-06-15 13:37:28 +0200 Dimitrios Katsaros * gst/multifile/gstmultifilesrc.c: multifilesrc: implement uri handler With this patch we can now provide a set of files created by multifilesink as a source for uri elements. e.g. gst-launch-1.0 playbin uri=multifile://img%25d.ppm Note that for the %d pattern you need to replace % with %25. This is to be compliant with URL naming standards. https://bugzilla.gnome.org/show_bug.cgi?id=783581 2018-01-19 15:05:26 +0200 Vivia Nikolaidou * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Make sure timecode uses the same timescale as video Don't blindly derive it from the frame rate, but try to get the per-pad configured timescale first (if it exists) https://bugzilla.gnome.org/show_bug.cgi?id=792680 2018-01-18 18:36:27 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Allow configuring trak timescale per pad/trak It generally makes not much sense to configure it for all pads/traks at once as this value is usually different for each of them. As such, add a new property on the pads in addition to the existing property on the whole muxer. https://bugzilla.gnome.org/show_bug.cgi?id=792649 2018-01-23 09:46:32 +0000 Tim-Philipp Müller * gst/flv/gstflvmux.c: Update for renamed aggregator pad API https://bugzilla.gnome.org/show_bug.cgi?id=791204 2018-01-22 12:24:18 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix up sendonly/recvonly attribute handling We can't handle recvonly streams, sendonly streams are perfectly fine. The direction is the one from the point of view of the SDP offerer (i.e. the RTSP server), and a recvonly stream would be one where the server expects us to send media. RFC 3264, section 5.1: If the offerer wishes to only send media on a stream to its peer, it MUST mark the stream as sendonly with the "a=sendonly" attribute. This is mixed up in the ONVIF streaming specification examples, but actual implementations and conformance tools seem to not care at all about the attributes. https://bugzilla.gnome.org/show_bug.cgi?id=792376 2017-11-11 13:49:22 +0900 paul.kim * ext/soup/gstsouphttpsrc.c: souphttpsrc: Reset retry_count to 0 when GST_FLOW_FLUSHING If a lot of seek method is called very quickly, sometimes data reading and do_request occurs while seek flush event is occurring and error occurs because retry_count reaches to the max. Thus, reset retry_count if flush occurs after do_request and read_buffer. https://bugzilla.gnome.org/show_bug.cgi?id=790199 2018-01-18 15:09:04 +0100 Jan Alexander Steffens (heftig) * tests/check/elements/aacparse.c: tests: aacparser: Test that short raw frames don't get concatenated https://bugzilla.gnome.org/show_bug.cgi?id=792644 2018-01-18 14:23:07 +0100 Jan Alexander Steffens (heftig) * gst/audioparsers/gstaacparse.c: aacparse: When parsing raw input, accept frames of any size Raw AAC streams might have very small frames, e.g. 6 byte frames when encoding silence. These frames are then smaller than aacparse's default min_frame_size of 10 bytes (ADTS_MAX_SIZE). When passthrough is disabled or aacparse has to output ADTS, GstBaseParse will concatenate these short frames to the following frame before handling them to aacparse, which processes each input buffer as a single frame, producing bad output. To avoid this problem, set the min_frame_size to 1 when receiving a raw stream. https://bugzilla.gnome.org/show_bug.cgi?id=792644 2017-05-02 21:24:06 -0300 Adrián Pardini * ext/shout2/gstshout2.c: shout2send: print actual username in debug log out https://bugzilla.gnome.org/show_bug.cgi?id=782093 2018-01-15 18:13:37 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtpbin.c: * tests/check/elements/rtpbin.c: rtpbin: fix leak of elements requested by signals When the signal returns a floating reference, as its return type is transfer full, we need to sink it ourselves before passing it to gst_bin_add (which is transfer floating). This allows us to unref it in bin_remove_element later on, and thus to also release the reference we now own if the signal returns a non-floating reference as well. As we now still hold a reference to the element when removing it, we also need to lock its state and setting it to NULL before unreffing it Also update the request_aux_sender test. https://bugzilla.gnome.org/show_bug.cgi?id=792543 2018-01-17 11:10:37 +0100 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: fix division by 0 for complex video formats So complex video formats have 0 as pstride. Don't try to divide the stride in such cases. https://bugzilla.gnome.org/show_bug.cgi?id=792596 2018-01-17 11:08:25 +0100 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: display stride and width values if stride is too small https://bugzilla.gnome.org/show_bug.cgi?id=792596 2018-01-16 13:19:29 +0000 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: document unit of "max-file-duration" property 2018-01-12 12:21:37 +0100 Florent Thiéry * gst/udp/gstudpsrc.c: udpsrc: fix typo in documentation https://bugzilla.gnome.org/show_bug.cgi?id=792458 2018-01-12 09:53:37 +0100 Peter Seiderer * sys/v4l2/gstv4l2videodec.c: v4l2videodec: add property set/get PROP_CAPTURE_IO_MODE error handling https://bugzilla.gnome.org/show_bug.cgi?id=791841 2018-01-12 09:46:30 +0100 Peter Seiderer * sys/v4l2/gstv4l2videodec.c: v4l2videodec: fold property set/get PROP_OUTPUT_IO_MODE case into default https://bugzilla.gnome.org/show_bug.cgi?id=791841 2018-01-12 09:49:14 +0100 Peter Seiderer * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: add property set/get PROP_CAPTURE_IO_MODE error handling https://bugzilla.gnome.org/show_bug.cgi?id=791841 2018-01-12 09:44:03 +0100 Peter Seiderer * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: fold property set/get PROP_OUTPUT_IO_MODE case into default https://bugzilla.gnome.org/show_bug.cgi?id=791841 2018-01-11 10:44:18 +0100 Peter Seiderer * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: fix capture-io-mode property get https://bugzilla.gnome.org/show_bug.cgi?id=791841 2018-01-11 17:47:39 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Maintain downstream caps order The g_list_insert_sorted() will behave like prepend when the compare function returns 0. In our case, we want to maintain the order hence append. This fixes this issue and improve the sorting algorithm to make a 10x10 prefered over 10x200 with a preference of 10x8 (and similar cases which was badly handled). This fixes generally fixes issue were a sub-optimal format / size is picked. https://bugzilla.gnome.org/show_bug.cgi?id=792435 2017-12-21 23:02:30 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Also re-enabled paused task When we only run _finish(), the task is never stopped externally, instead it's only paused from the inside. We still want to restart it in this case. 2018-01-08 15:23:24 +0100 Mathieu Duponchelle * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: flush flac decoder on lost sync. This to allow the decoder to start searching for a new frame again. https://bugzilla.gnome.org/show_bug.cgi?id=791473 2017-12-21 22:56:51 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Call stop on object before renegotiation Otherwise renegotiation fails as we are still streaming. https://bugzilla.gnome.org/show_bug.cgi?id=791338 2017-12-21 22:55:49 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Remove dead code gst_v4l2_object_stop() will free and nullify the pool, so the following if will never be true. https://bugzilla.gnome.org/show_bug.cgi?id=791338 2017-12-21 22:29:06 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Delay capture pool activation This is support CODA driver which prevents setting the output format if the capture is streaming. https://bugzilla.gnome.org/show_bug.cgi?id=791338 2017-12-13 20:23:46 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Add dynamic resolution change support This implements a "big hammer" reallocation method. We effectively drain and stop both side of the decoder and restart. This though is the most generic method. This change should enable on most drivers adaptive streaming. https://bugzilla.gnome.org/show_bug.cgi?id=752962 2017-12-30 01:52:13 +0000 Tim-Philipp Müller * meson.build: meson: zlib is not actually a hard requirement 2017-09-28 18:00:38 -0300 Ezequiel Garcia * ext/jpeg/gstjpegdec.c: jpeg: Fixup frames without an EOI marker Some cameras fail to send an end-of-image marker (EOI) and can't be properly decoded by either JPEG or libjpeg. This commit parses the frame, making sure it has an EOI. If there isn't one, the EOI gets added to the buffer. A similar fixup is done in the rtpjpegdepay element, and it makes sense to do it in jpegdec as well. Signed-off-by: Ezequiel Garcia https://bugzilla.gnome.org/show_bug.cgi?id=791988 2017-12-26 13:50:24 +0100 Tim-Philipp Müller * meson.build: meson: skip translations if gettext is not available 2017-12-24 13:14:06 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-rtp.xml: docs: add rtpL8pay/depay to docs 2017-12-24 13:11:00 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-rtp.xml: docs: update for recent changes 2015-05-15 17:00:26 +0100 Tim Allen * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL8depay.c: * gst/rtp/gstrtpL8depay.h: * gst/rtp/gstrtpL8pay.c: * gst/rtp/gstrtpL8pay.h: * gst/rtp/meson.build: rtp: add L8 audio support 2017-12-23 12:45:17 +0100 Mark Nauwelaerts * gst/udp/gstudpsrc.c: udpsrc: fix typo in multicast join error message 2017-12-23 12:44:31 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: also proxy multicast-iface property to RTCP udpsrc 2015-11-02 00:41:28 +0100 Sebastian Rasmussen * gst/udp/gstmultiudpsink.c: multiudpsink: don't try to set IPV6_TCLASS on IPV4 sockets Avoids ERROR log message. https://bugzilla.gnome.org/show_bug.cgi?id=757449 2015-11-02 00:41:28 +0100 Sebastian Rasmussen * tests/check/Makefile.am: * tests/check/elements/udpsink.c: tests: udpsink: add check that sets QoS on IPv4/6 sockets https://bugzilla.gnome.org/show_bug.cgi?id=757449 2017-12-22 10:21:28 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2deviceprovider.c: v4l2deviceprovider: Don't do slow probes This is problematic in the current design at it seriously slow down startup of applications. As of now, no known application uses the colorimetry and the interlace-modes for anything (the two fields that won't be probed). So let's disable it, in the long term we'll try and find a way to interact with the provider so applicaiton could opt-in these slow probing methods for more advance configuration. 2017-12-22 10:15:48 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't redefine mmap64 On Linux, there exist a case where mmap64 is already a define to mmap, so avoid the redefine warning here. 2017-12-19 17:37:58 +0800 Ting-Wei Lan * configure.ac: * meson.build: * sys/v4l2/gstv4l2object.c: v4l2object: Don't use mmap64 if off_t is 64-bit The difference between mmap and mmap64 is the type of 'offset' argument. mmap64 always uses a 64-bit interger as offset, while mmap uses off_t, whose size can vary on different operating systems or architectures. However, not all operating systems support mmap64. Fortunately, although FreeBSD only has mmap, its off_t is always 64-bit regardless of architectures, so we can simply use mmap when sizeof(off_t) == 8. https://bugzilla.gnome.org/show_bug.cgi?id=791779 2017-12-22 09:17:04 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: Revert "v4l2object: Use mmap64 to match libv4l2 signature" This reverts commit b61bba48488c0a627d90f04cc9917d8c4f3f0d9b. 2017-12-19 17:37:58 +0800 Ting-Wei Lan * configure.ac: * meson.build: * sys/v4l2/gstv4l2object.c: v4l2object: Check for mmap64 before using it mmap64 is not available on FreeBSD. https://bugzilla.gnome.org/show_bug.cgi?id=791779 2017-12-20 15:23:26 -0500 Vincent Penquerc'h * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flv: flvmux ported to the GstAggregator This makes it possible to create a flv file from a live source and not stop when there are packet drops. https://bugzilla.gnome.org/show_bug.cgi?id=782920 2017-12-19 16:47:52 -0500 Olivier Crête * gst/udp/gstmultiudpsink.c: multiudpsink: Call gst_base_sink_wait_preroll on unlock This means that packets will not be lost on fast pause/playing cycles. Also refactor the code a little to simplify it. https://bugzilla.gnome.org/show_bug.cgi?id=774945 2017-12-19 16:22:52 -0500 Olivier Crête * tests/examples/gtk/Makefile.am: gtk example: Fix cflags in Makefile.am 2017-12-19 15:46:52 -0500 Olivier Crête * gst/udp/gstmultiudpsink.c: multiudpsink: Remove unused variable 2017-12-19 13:03:28 +0000 Tim-Philipp Müller * ext/gtk/gtkgstglwidget.c: gtk: don't include uninstalled header 2017-12-17 20:54:06 +0000 Tim-Philipp Müller * ext/qt/Makefile.am: gl: update plugins to use GstGL from -base 2017-12-17 20:54:06 +0000 Tim-Philipp Müller * ext/gtk/Makefile.am: * ext/gtk/meson.build: * tests/examples/gtk/Makefile.am: gl: update plugins to use GstGL from -base 2017-12-19 11:57:52 +0100 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix two leaks * gst_event_new_stream_start() does not take ownership of the stream_id * the pipeline_request_id string that is created was not being freed 2017-12-07 22:08:42 -0500 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: videocrop: Add GstVideoCropMeta support If downstream supports this meta, it will add or update it from the GstBuffer in-place rather then copying. https://bugzilla.gnome.org/show_bug.cgi?id=791453 2017-12-13 09:22:17 +0000 Sean DuBois * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/webm-mux.c: Add AV1 to matroska plugin https://bugzilla.gnome.org/show_bug.cgi?id=784160 2017-12-15 14:48:09 +0100 fengalin * gst/matroska/matroska-mux.c: * gst/matroska/matroska-read-common.c: * tests/check/elements/matroskademux.c: * tests/check/elements/matroskamux.c: matroska: fix memory leaks due to toc related updates https://bugzilla.gnome.org/show_bug.cgi?id=790686 2017-12-15 11:40:13 +0200 Sebastian Dröge * tests/check/elements/matroskamux.c: matroskamux: Fix various memory leaks in the unit test https://bugzilla.gnome.org/show_bug.cgi?id=790686 2017-12-14 19:05:36 +0100 fengalin * tests/check/elements/matroskademux.c: * tests/check/elements/matroskamux.c: matroska-mux: migrate test to gst_harness ... following the guide lines from Håvard Graff (see https://gstconf.ubicast.tv/videos/moar-better-tests/). https://bugzilla.gnome.org/show_bug.cgi?id=790686 2017-12-01 18:17:06 +0100 fengalin * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: * tests/check/elements/matroskademux.c: * tests/check/elements/matroskamux.c: matroska: re-activate and update TOC support TOC support in mastroskamux has been deactivated for a couple of years. This commit updates it to recent GstToc evolutions and introduces toc unit tests for both matroska-mux and matroska-demux. There are two UIDs for Chapters in Matroska's specifications: - The ChapterUID is a mandatory unsigned integer which internally refers to a given chapter. Except for title & language which use dedicated fields, this UID can also be used to add tags to the Chapter. The tags come in a separate section of the container. - The ChapterStringUID is an optional UTF-8 string which also uniquely refers to a chapter but from an external perspective. It can act as a "WebVTT cue identifier" which "can be used to reference a specific cue, for example from script or CSS". During muxing, the ChapterUID is generated and checked for unicity, while the ChapterStringUID receives the user defined UID. In order to be able to refer to chapters from the tags section, we maintain an internal Toc tree with the generated ChapterUID. When demuxing, the ChapterStringUIDs (if available) are assigned to the GstTocEntries UIDs and an internal toc mimicking the toc is used to keep track of the ChapterUIDs and match the tags with the appropriate GstTocEntries. https://bugzilla.gnome.org/show_bug.cgi?id=790686 2017-12-14 18:28:00 +0200 Sebastian Dröge * tests/examples/v4l2/v4l2src-renegotiate.c: v4l2src: Fix compiler error in example caused by re-declaring `index` ../tests/examples/v4l2/v4l2src-renegotiate.c:57:13: error: ‘index’ redeclared as different kind of symbol static gint index = 0; ^ 2017-12-14 14:49:01 +1100 Matthew Waters * common: Automatic update of common submodule From e8c7a71 to 3fa2c9e 2017-12-13 14:39:47 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/v4l2_calls.c: v4l2object: Use a debug object for tracing This way we can pass the pad name instead of the element for tracing which helps identifying which v4l2object is used withing M2M element like decoder, encoder and transform. For the reference, pads are name :. 2017-12-13 12:06:21 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Push a GAP event if there's a second *or more* And not "more than a second" 2017-12-13 11:35:37 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Don't push GAP event if first buffer is within 1s If we saw empty segments, we previously unconditionally pushed a GAP event downstream regardless of the duration of that empty segment. In order to avoid issues with initial negotiation of downstream elements (which would negotiate to something before receiving any data due to that initial GAP event), check if there's at least a second of difference (like we do for other GAP-related checks in qtdemux) before deciding to push a GAP event downstream. 2017-12-13 10:21:17 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Don't set pared=True on underspecified audio/mpeg This *really* needs to go through a parser to figure out what the exact content type is. 2017-12-11 15:27:08 -0600 Michael Catanzaro * gst/equalizer/gstiirequalizer.c: equalizer: Fix -Wincompatible-pointer-types warning This is caused by the new type propagation for g_object_ref. https://bugzilla.gnome.org/show_bug.cgi?id=791494 2017-12-09 16:15:24 +0000 Tim-Philipp Müller * tests/check/elements/.gitignore: tests: ignore rtph264 test binary 2017-08-25 15:19:37 +0300 George Kiagiadakis * tests/check/elements/udpsrc.c: tests: udpsrc: verify the correct amount of bytes is sent to the socket https://bugzilla.gnome.org/show_bug.cgi?id=786799 2017-08-25 14:59:06 +0300 George Kiagiadakis * tests/check/elements/udpsrc.c: tests: udpsrc: ensure test won't timeout if the buffers are already received Sometimes all the buffers are received before the time we lock the check_mutex, in which case g_cond_wait will wait forever for another one. Just check if this is the case before waiting. https://bugzilla.gnome.org/attachment.cgi?id=358397 2017-08-25 14:45:52 +0300 George Kiagiadakis * tests/check/elements/udpsrc.c: tests: udpsrc: fix test_udpsrc to actually run and fix locking Previously this would silently be skipped because 1600 != 1400 and there is no assertion on this call. Also unlock check_mutex after use. https://bugzilla.gnome.org/show_bug.cgi?id=786799 2017-09-21 18:23:54 +0300 John Nikolaides * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: added a "split now" action signal Now, the video file can be split at an arbitrary time chosen by the user. https://bugzilla.gnome.org/show_bug.cgi?id=787922 2017-12-08 00:31:32 +0000 Alvaro Margulis * gst/udp/gstmultiudpsink.c: multiudpsink: fix bind address leak https://bugzilla.gnome.org/show_bug.cgi?id=790986 2017-12-07 11:15:19 +0000 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: Revert "flacparse: fix header rewriting being ignored" This caused broken metadata and also looks a bit dodgy. Revert until we can figure out a solution that works for all cases and doesn't break anything. This reverts commit adeee44b07a173b9ab4253216caba8f66dd43abb. https://bugzilla.gnome.org/show_bug.cgi?id=727802 https://bugzilla.gnome.org/show_bug.cgi?id=785558 2017-12-05 15:14:04 +0100 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Handle drivers that only round up height Commit 1f31715c9861 ("v4l2videodec: use visible size, not coded size, for downstream negotiation filter") added support for removing the padding obtained as the difference between width/height from G_FMT and visible width/height from G_SELECTION from the probed caps obtained via TRY_FMT. This patch fixes the padding removal for drivers that only round up height, but not width, to the padded frame size. This might happen because horizontal padding can be handled by line stride (bytesperline), but there is no such thing as plane stride in the V4L2 API for single-buffer planar formats. https://bugzilla.gnome.org/show_bug.cgi?id=791271 2017-11-01 08:21:37 -0600 Matt Staples * gst/rtsp/gstrtspsrc.c: rtspsrc: Add a signal to allow outgoing messages to be modified or dropped This feature allows applications to implement extensions to the RTSP protocol, such as those defined in the ONVIF Streaming Specification. https://bugzilla.gnome.org/show_bug.cgi?id=762884 2017-08-25 11:57:26 +0200 Haakon Sporsheim * gst/rtpmanager/rtpsession.c: * tests/check/elements/rtpsession.c: rtpsession: Handle zero length feedback packets https://bugzilla.gnome.org/show_bug.cgi?id=791074 2017-07-10 15:19:34 +0200 Florian Zwoch * gst/isomp4/qtdemux.c: qtdemux: fix debug log for 'hvcC' codec_data https://bugzilla.gnome.org/show_bug.cgi?id=784749 2017-12-01 13:04:41 +0100 Havard Graff * tests/check/elements/rtpsession.c: tests: rtpsession: refactor tests to use GstHarness This patch simplifies the tests (44% less code) and makes them much more readable. The provided SessionHarness also makes it much easier to write new tests for rtpsession. https://bugzilla.gnome.org/show_bug.cgi?id=791070 2017-11-24 10:36:01 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Request at least the full header size when parsing headers Otherwise baseparse will incrementally send us bigger buffers until the full header size is reached, which is not only pointless but also means that baseparse will reallocate and copy into a bigger buffer for every input buffers. In pull mode that's done in 64kb increments, in push mode usually in much smaller increments, causing a lot of overhead for example when parsing high-quality coverart. 2017-11-29 11:29:31 +0100 Florent Thiéry * sys/v4l2/gstv4l2object.c: v4l2object: Fix dmabuf support detection This resulted in improper selection of dmabuf on unsupported drivers. The checked ioctl errno was not correct. https://bugzilla.gnome.org/show_bug.cgi?id=790940 2017-11-27 20:10:51 +1100 Matthew Waters * common: Automatic update of common submodule From 3f4aa96 to e8c7a71 2017-11-27 14:44:58 +1100 Matthew Waters * ext/qt/gstqtglutility.cc: gl/caopengllayer: use public GstGLContext instead of Cocoa-specific one Allows keeping the GstGLCAOpenGLLayer public but not the winsys-specific context/display/window. 2017-11-26 15:13:15 +0000 Tim-Philipp Müller * configure.ac: autotools: stop controlling symbol visibility with -export-symbols-regex Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT. This should result in consistent behaviour for the autotools and Meson builds. 2017-11-24 15:37:44 +0100 Edward Hervey * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Do more checks for seekability When receiving a seek event, check whether we can actually seek based on the information the server provided. Also add more documentation on what the seekable field means 2017-11-25 00:53:42 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Always update reserved-duration-remaining If a reserved-max-duration is set, we should always track and update the reserved-duration-remaining estimate, even if we're not sending periodic moov updates downstream for full robust muxing. 2015-04-07 23:53:19 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmuxsink: Use muxer reserved space properties if present. If the use-robust-muxing property is set, check if the assigned muxer has reserved-max-duration and reserved-duration-remaining properties, and if so set the configured maximum duration to the reserved-max-duration property, and monitor the remaining space to start a new file if the reserved header space is about to run out - even though it never ought to. 2017-11-24 08:00:21 +0100 Edward Hervey * ext/gtk/gtkgstglwidget.c: gtk: Fix possibility of NULL variable It's quite unlikely since it's initialized in instance initialization. CID #1417721 2017-11-24 16:56:03 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * tests/check/elements/splitmux.c: splitmux: Fix file switch-on-caps-change. Switching to a new fragment because the input caps have changed didn't properly end the previous file. Use the normal EOS sequence to ensure that happens. Add a test that it works. 2017-11-24 16:53:40 +1100 Jan Schmidt * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: Update output caps on input caps change If the input changes width/height that should be reflected in the output caps, so make sure they get updated 2017-11-23 22:58:40 +1100 Jan Schmidt * ext/qt/gstqtglutility.cc: Revert "gl: Use GstGLDisplayEGL directly instead of creating a GstGLDisplayVIVFb subclass" This reverts commit 47fd4d391e775c11f529705bb0f457a9d25ba5e7. This patch is incorrect. It doesn't actually compile, and causes a crash because the viv-fb window implementation needs a native EGL handle to pass to fbCreateWindow, but the GstGLDisplayEGL handleis actually an EGLDisplay now (and gets cast to the wrong type) 2017-09-05 15:55:03 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: don't insert SPS/PPS inline for hvc1 output Only for byte-stream or hev1. For hvc1 the SPS/PPS are in the caps as codec_data field and in this case they shouldn't be in the stream data as well. The output caps should be updated with the new codec_data if needed, for hvc1. 2017-09-05 15:47:42 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: rtph265depay: store negotiated output format as enum We keep the boolean byte_stream around since it's nicer for readability and most of the code just cares about byte_stream or not. This is useful for future-proofing the code for when we add support for hev1 output as well. 2017-08-29 17:05:51 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: add support for hvc1 as output format 2017-08-08 18:58:11 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265pay.c: rtph265pay: don't add trailing zeros to VPS/PPS/SPS This would happen if input is byte-stream with four-byte sync markers instead of three-byte ones. The code that scans for sync markers will place the start of the NALU on the third-last byte of the NALU sync marker, which means that any additional zeros may be counted as belonging to the previous NALU instead of being part of the next sync marker. Fix that so we don't send VPS/SPS/PPS with trailing zeros in this case. See https://bugzilla.gnome.org/show_bug.cgi?id=732758 2017-06-16 12:41:49 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: assemble AUs into downstream-allocated memory When merging NALs into AUs, use downstream-provided allocator to allocate memory and copy NALs directly into that memory when assembling them. 2017-06-16 12:30:13 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: rtph265depay: try to negotiate an allocator with downstream 2017-06-16 12:13:32 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: simplify buffer accumulation control flow There is no difference between pushing out a buffer directly with gst_rtp_base_depayload_push() and returning it from the process function. The base class will just call _depayload_push() on the returned buffer as well. So instead of marshalling buffers through three layers and back, just push them from one place in handle_nal() and always return NULL from the process vfunc. This simplifies the code a little. Also rename _push_fragmentation_unit() to _finish_fragmentation_unit() for clarity. Push sounds like it means being pushed out, whereas it might just be pushed into an adapter. This change has the side-effect that multiple NALs in a single STAP (such as SPS/PPS) may no longer be pushed out as a single buffer if we output NALs in byte-stream format (i.e. not aggregate AUs), but that shouldn't really make any difference to anyone. 2017-06-16 11:18:16 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: fix crash with empty sprops-parameters https://bugzilla.gnome.org/show_bug.cgi?id=780040 2017-06-16 12:20:34 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: minor clean-up Declutter caps update code a bit. 2017-08-08 13:10:15 +0100 Tim-Philipp Müller * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: add unit test for rtph264pay codec_data Make sure no trailing zero bytes sneak into our SPS or PPS. https://bugzilla.gnome.org/show_bug.cgi?id=732758 2014-07-05 06:21:48 +0000 Philip Craig * gst/rtp/gstrtph264pay.c: rtph264pay: don't add trailing zeros to PPS/SPS This would happen if input is byte-stream with four-byte sync markers instead of three-byte ones. The code that scans for sync markers will place the start of the NALU on the third-last byte of the NALU sync marker, which means that any additional zeros may be counted as belonging to the previous NALU instead of being part of the next sync marker. Fix that so we don't send SPS/PPS with trailing zeros in this case. https://bugzilla.gnome.org/show_bug.cgi?id=732758 2017-05-20 15:50:22 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/rtph264.c: * tests/files/Makefile.am: * tests/files/h264.rtp: tests: rtph264depay: add test for using downstream memory allocator 2017-06-03 00:58:05 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: assemble AUs into downstream-allocated memory When merging NALs into AUs, use downstream-provided allocator to allocate memory and copy NALs directly into that memory when assembling them. 2017-06-02 21:27:40 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: try to negotiate an allocator with downstream 2017-06-02 20:54:20 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: minor clean-up Declutter caps update code a bit. 2017-11-23 08:00:58 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Run gst-indent 2017-11-23 07:59:07 +0100 Edward Hervey * gst/replaygain/rganalysis.c: rganalysis: Fix left shift of signed values left shifting signed values is undefined. Instead of doing "x << offs" which is undefined, do the equivalent "x * (1 << offs)" which is well defined 2017-11-23 07:57:44 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Check presence of bitrate tags Check whether the tag was present before printing it out CID #1418501 2017-11-21 09:33:49 +0100 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Use the proper maximum value for seekable it's a gfloat, not a gdouble 2017-11-18 02:27:50 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Use new GST_SEQNUM_INVALID constant 2017-11-18 02:01:58 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: splitmuxsrc: Don't return FALSE from event handling. Returning FALSE because we drop an event means that internal sources like qtdemux might throw an error and break the whole pipeline. The only time it can happen is either flushing or shutdown, and those will be handled anyway. 2017-10-22 18:26:12 +0800 Jun Xie * gst/isomp4/qtdemux.c: qtdemux: reset reused QtDemuxStream while parsing a new 'trak' if QtDemuxStream is reused, then we need to reset it. https://bugzilla.gnome.org/show_bug.cgi?id=788759 2017-11-13 10:43:11 +0900 Seungha Yang * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: isomp4: Add official fourcc for VP8 codec fourcc for VP8 codec is "vp08" defined by spec. To follow it, add it to demux and change legacy VP8 fourcc "VP80" to "vp08" in mux. Also, enable sync table in case of VP8 codec. See also https://www.webmproject.org/vp9/mp4/ https://bugzilla.gnome.org/show_bug.cgi?id=790026 2017-11-13 10:38:06 +0900 Seungha Yang * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: isomp4: Add support VP9 codec Add fourcc for VP9 codec and support it by qtdemux and qtmux See also https://www.webmproject.org/vp9/mp4/ https://bugzilla.gnome.org/show_bug.cgi?id=790026 2017-11-13 13:51:20 +0100 Edward Hervey * gst/matroska/matroska-demux.c: matroskademux: Remove bogus error message It's just informational 2017-11-10 15:51:05 +0100 Edward Hervey * gst/rtp/gstrtpmpvpay.c: rtpmpvpay: Don't create empty buffer list If there's nothing to send, just return 2017-03-13 18:14:12 +0900 paul.kim * ext/soup/gstsouphttpsrc.c: souphttpsrc: Remove range header when seek to 0 This fixes the previous range header is remained if seek to 0 is attempted. https://bugzilla.gnome.org/show_bug.cgi?id=779957 2017-11-08 16:34:01 +0100 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Fix seeking back to 0 This is a regression introduced by "03db374 - souphttpsrc: retry request on early termination from the server" The problem was that when seeking back to 0, we would not end up calling add_range_header() which in addition to adding range headers *ALSO* sets the read_position to the requested one. This would result in a wide variety of later failures, like reading again and again instead of stopping properly. 2017-11-07 18:03:53 +0900 Seungha Yang * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: matroskademux: Add parsing Colour element ... and forward colorimetry to downstream. The Colour element describes various color information (similar to 'colr' box in isobmff). Note that, due to the comparatively limited syntax for color information in vpx codecs, the color information in mkv/wemb container level should be used for sophisticated color handling (e.g., HDR video). https://bugzilla.gnome.org/show_bug.cgi?id=790023 2017-10-19 14:02:37 +0200 Jan Alexander Steffens (heftig) * sys/v4l2/gstv4l2deviceprovider.c: v4l2deviceprovider: Ignore touch sensing devices With GST_V4L2_USE_LIBV4L2=1, my laptop's touchpad shows up as a video source device in gst-device-monitor, but attempting to stream from it fails because the device doesn't actually support any video formats. name : Synaptics RMI4 Touch Sensor class : Video/Source caps : video/x-raw, format=(string)I420, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1; video/x-raw, format=(string)YV12, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1; video/x-raw, format=(string)BGR, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1; video/x-raw, format=(string)RGB, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)0, height=(int)0, interlace-mode=(string)progressive, pixel-aspect-ratio=(fraction)1/1; properties: udev-probed = true device.bus_path = /sys/devices/rmi4-00/rmi4-00.fn54/video4linux/v4l-touch0 sysfs.path = /sys/devices/rmi4-00/rmi4-00.fn54/video4linux/v4l-touch0 device.subsystem = video4linux device.product.name = "Synaptics\ RMI4\ Touch\ Sensor" device.capabilities = :capture: device.api = v4l2 device.path = /dev/v4l-touch0 v4l2.device.driver = rmi4_f54 v4l2.device.card = "Synaptics\ RMI4\ Touch\ Sensor" v4l2.device.bus_info = rmi4:rmi4-00.fn54 v4l2.device.version = 265480 (0x00040d08) v4l2.device.capabilities = 2501902337 (0x95200001) v4l2.device.device_caps = 354418689 (0x15200001) gst-launch-1.0 v4l2src device=/dev/v4l-touch0 ! ... v4l2-ctl -d /dev/v4l-touch0 --list-formats reports: ioctl: VIDIOC_ENUM_FMT Index : 0 Type : Video Capture Pixel Format: 'TD16' Name : 16-bit signed deltas Index : 1 Type : Video Capture Pixel Format: 'TD08' Name : 8-bit signed deltas Index : 2 Type : Video Capture Pixel Format: 'TU16' Name : 16-bit unsigned touch data https://bugzilla.gnome.org/show_bug.cgi?id=789197 2017-11-03 13:27:50 -0400 Youness Alaoui * gst/rtp/gstrtpg722pay.c: rtpg722pay: Add encoding-params to the src caps template The G722 payload only accepts G722 audio with channels=1, so it must specify the encoding-params=1 in its src caps, otherwise it causes issues with farstream which thinks it supports 2 channels G722 and when confronted with a remote that has G722/8000/2, it will negotiate it and error out with a not-negotiated when the caps don't intersect at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=789878 2017-10-06 17:36:34 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: v4l2allocator: Add support for data_offset In MPLANE mode, the driver may set data_offset, which represent some padding at the start of the buffer used internally. This portion of the data need to be skipped, though it is included in bytesused. This patch removes frame size sanity check as the method used will no longer work. This check was simply there to help detect broken kernel drivers. It would be re-implement by estimating the plane size, which is not totally trivial and may be too much work for a simple debug check. https://bugzilla.gnome.org/show_bug.cgi?id=733501 2017-07-17 17:09:18 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Add "accept-certificate" signal for manually checking a TLS certificate for validity https://bugzilla.gnome.org/show_bug.cgi?id=785024 2017-10-30 19:15:56 +0900 Sangkyu Park * gst/rtsp/gstrtspsrc.c: rtspsrc: Print RTSP/SDP messages to gstreamer log instead of stdout - 'debug' property is deprecated - All RTSP messages are printed to gstreamer log with 'log' level. https://bugzilla.gnome.org/show_bug.cgi?id=788917 2017-11-01 15:29:58 +0900 Justin Kim * gst/rtpmanager/rtpsession.c: rtpsesson: downgrade message level to debug when detected XR When XR packet is detected, warning message leads to misunderstandings. Until RFC3611 is implemented in gst-plugins-base, the level needs to be downgraded to avoid confusion. https://bugzilla.gnome.org/show_bug.cgi?id=789746 2017-10-24 20:12:29 +0530 Ashish Kumar * gst/isomp4/atomsrecovery.c: gst-plugins-good: atoms_recovery: Handled buffer mapping failure https://bugzilla.gnome.org/show_bug.cgi?id=789413 2017-07-08 22:11:49 -0700 Thiago Santos * gst/isomp4/atomsrecovery.c: * gst/isomp4/atomsrecovery.h: * gst/isomp4/gstqtmoovrecover.c: atomsrecovery: read from mdat only what is on headers It is possible that the mdat has more data than what was stored in the headers file. If we put that to the output the file will have bogus data at the end and some players will complain. https://bugzilla.gnome.org/show_bug.cgi?id=784258 2017-07-05 22:23:21 -0700 Thiago Santos * gst/isomp4/atomsrecovery.c: isomp4: atomsrecovery: handle common and large atom headers Do not assume all files are large files. Check and use the short or extended atom size field only if needed. https://bugzilla.gnome.org/show_bug.cgi?id=784258 2017-10-20 11:08:24 +0200 Andreas Frisch * configure.ac: pngdec: fix build with libpng versions between 1.2 and 1.5.1 (revised) https://bugzilla.gnome.org/show_bug.cgi?id=765927 2017-10-19 18:23:34 +0200 Andreas Frisch * configure.ac: * ext/libpng/gstpngdec.c: pngdec: fix build with libpng versions between 1.2 and 1.5.1 https://bugzilla.gnome.org/show_bug.cgi?id=765927 2017-10-19 16:17:45 +0200 Andreas Frisch * ext/libpng/gstpngdec.c: pngdec: Extract icc profiles and send them downstreams for colormanagement elements https://bugzilla.gnome.org/show_bug.cgi?id=765927 2017-10-16 14:20:47 +0200 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtsp: Add missing Since marker 2017-10-13 12:25:22 +0100 Tim-Philipp Müller * ext/qt/qtplugin.pro: qt: update qmake .pro file Update for renaming of plugin file, and add some missing source files. 2017-06-13 18:51:32 +0200 Mathieu Duponchelle * ext/gdk_pixbuf/gstgdkpixbufdec.c: gstgdkpixbufdec: stop pretending to decode gifs. If you can't decode an animated gif, you can't decode a gif, so stop squatting GST_RANK_SECONDARY for that format, libav does a better job. https://bugzilla.gnome.org/show_bug.cgi?id=784683 2017-09-28 22:51:57 +0200 Philippe Renon * sys/directsound/gstdirectsoundsink.c: directsoundsink: simplify how DirecSoundBuffer is cleared we always want to clear the whole buffer so no need to start from offset even if the offset is always zero. https://bugzilla.gnome.org/show_bug.cgi?id=788847 2017-09-28 22:49:31 +0200 Philippe Renon * sys/directsound/gstdirectsoundsink.c: directsoundsink: fix comment https://bugzilla.gnome.org/show_bug.cgi?id=788847 2017-09-28 22:48:41 +0200 Philippe Renon * sys/directsound/gstdirectsoundsink.c: directsoundsink: don't call set_volume with private scaled volume use get_volume() instead to get unscaled volume https://bugzilla.gnome.org/show_bug.cgi?id=788847 2017-09-28 22:46:23 +0200 Philippe Renon * sys/directsound/gstdirectsoundsink.c: directsoundsink: remove duplicate volume initialization https://bugzilla.gnome.org/show_bug.cgi?id=788847 2017-10-10 18:04:50 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix compiler warning qtdemux.c: In function ‘gst_qtdemux_configure_stream’: qtdemux.c:7764:34: error: suggest parentheses around ‘&&’ within ‘||’ [-Werror=parentheses] if ((stream->n_samples == 1) && (stream->first_duration == 0) ~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 2017-09-22 18:41:52 +0200 Nael Ouedraogo * gst/isomp4/qtdemux.c: qtdemux: fix assert when moof containing one sample Avoid computing frame rate when a stream contain moof with only one sample, to avoid an assert. The moof is considered as still picture. The same is already done for one sample given in the moov. https://bugzilla.gnome.org/show_bug.cgi?id=782217 2017-10-09 14:17:25 +0200 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtspsrc: Avoid potentially dereferencing NULL pointer CID 1418986 2017-10-08 00:07:43 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix debug message on pt mismatch 2017-10-07 21:11:41 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Fix driver capability dectection Use the right set of caps when checking if caps intersect. That makes the check only select the supported devices. 2017-09-20 01:46:15 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc/dec: Don't leak template caps 2017-10-07 21:17:53 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videodec: Protect against null pool in _stop This may happen if the negotiation fails, as we will have never created the pools. 2017-10-07 15:55:24 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: * gst/rtsp/gstrtspsrc.c: rtpbin, rtspsrc: fix compiler warnings about 64-bit integer signednes "warning: this decimal constant is unsigned only in ISO C90" with gcc 4.8.4 (Ubuntu/Linaro 4.8.4-2ubuntu1~14.04.3) 2017-10-07 15:39:18 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2: fix build without libv4l https://bugzilla.gnome.org/show_bug.cgi?id=779466 2017-10-07 14:06:38 +0300 Sebastian Dröge * gst/rtp/gstrtpsbcdepay.c: rtpsbcdepay: Fix potential NULL pointer dereference CID 1418864 2017-10-07 01:21:19 +0300 Sebastian Dröge * gst/audiofx/audioecho.c: audioecho: Micro-optimize Gives 1.28x speedup in surround-delay=false mode 2017-10-06 23:59:43 +0300 Sebastian Dröge * gst/audiofx/audioecho.c: audioecho: Don't do linear interpolation between samples Linear interpolation adds quite some noise, and it's unlikely that anybody will ever need sub-sample accurate delays. Proper resampling before that will lead to better results. 2017-09-29 22:19:42 -0400 Enrico Jorns * sys/v4l2/gstv4l2object.c: v4l2object: auto-detect dmabuf export for V4L2_IO_AUTO on capture side Issue an invalid VIDIOC_EXPBUF ioctl to the driver to check if the driver supports dmabuf export. If the driver does not implement the IOCTL, the error is ENOTTY. Any other error codes mean that the driver implements VIDIOC_EXPBUF. https://bugzilla.gnome.org/show_bug.cgi?id=779466 2017-09-24 14:35:01 -0400 Nicolas Dufresne * gst/flv/gstflvdemux.c: flvdemux: Only set pixel-aspect-ratio if specified If it's not specified, we should let the decoder figure it out. Apparently the code was already in place, all was to make the code conditional. https://bugzilla.gnome.org/show_bug.cgi?id=787795 2017-09-23 15:44:09 -0400 Nicolas Dufresne * gst/flv/gstflvdemux.c: flvdemux: Don't pull passed the EOS When a truncated FLV is provided and processed in pull mode, we may endup trying to pull passed EOS, causing a rather confusing warning as the pull offset is an integer overflow. https://bugzilla.gnome.org/show_bug.cgi?id=787795 2017-09-23 15:41:30 -0400 Nicolas Dufresne * gst/flv/gstflvdemux.c: flvdemux: Ignore invalid H.264 codec data This code basically skip over codec_data with empty payload. In this case, the codec_data variable is the size of the header for the CODEC part of Video Tag. The remaining is supposed to be the H.264 codec data, hence should not be empty. https://bugzilla.gnome.org/show_bug.cgi?id=787795 2017-09-23 15:38:07 -0400 Nicolas Dufresne * gst/flv/gstflvdemux.c: flvdemux: Avoid integer overflow on invalid CTS If the CTS is negative an would lead to a negtive PTS, clip the CTS so the PTS will be 0. https://bugzilla.gnome.org/show_bug.cgi?id=787795 2017-10-05 14:36:28 -0300 Thibault Saunier * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-isomp4.xml: docs: Update for git changes 2017-10-05 14:35:27 -0300 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix build 2017-07-13 14:46:55 -0400 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtspsrc: Handle TCP as lower transport with RTSP 2.0 Meaning that the interleave fields have to be updated as if streams setup was working when using pipelined setup request. Otherwise there is a mismatch between the server channel count and our own. This also makes RTSP 2.0 over HTTP working. https://bugzilla.gnome.org/show_bug.cgi?id=781446 2017-04-20 17:45:39 -0300 Thibault Saunier * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtsp: Start implementing support for RTSP 2.0 - Handle version negotation: Added a `default-version` property so that the user can configure what to use in case the server does not support version negotation (which actually exist) - Handle pipelined requests, which allow avoiding full round trip to setup the RTP streams (request are sent in a raw, and response are handled as they arrive). - Handle the new Media-Properties header - Handle the new Seek-Style header - Handle the new Accept-Ranges header Handling of IPV6 should already be OK. We are still missing (at least) the following features (which do not seem really mandatory as they require a "persistent connection between server and client"): - Server to Client TEARDOWN command (Not so usefull fmpov) - PLAY_NOTIFY (not needed for our server yet) - Support for the new REDIRECT features and probably some more protocol changes might not be handled yet. https://bugzilla.gnome.org/show_bug.cgi?id=781446 2017-05-03 11:19:03 -0300 Thibault Saunier * gst/rtsp/gstrtspsrc.c: rtspsrc: Use a macro to debug RTSP messages Simplifying the code a little. https://bugzilla.gnome.org/show_bug.cgi?id=781446 2017-10-03 16:30:10 -0700 Reynaldo H. Verdejo Pinochet * ext/gdk_pixbuf/gstgdkpixbufsink.c: * gst/level/gstlevel.c: * gst/matroska/matroska-mux.c: * gst/multifile/gstmultifilesink.c: * gst/replaygain/gstrganalysis.c: * gst/spectrum/gstspectrum.c: Use proper GtkDoc notation for NULL/FALSE/TRUE 2017-10-02 12:35:48 -0700 Cassandra Rommel * ext/qt/gstqtglutility.cc: gl: Use GstGLDisplayEGL directly instead of creating a GstGLDisplayVIVFb subclass This simplifies the code a lot without any functional changes apart from not closing the display connection. Closing the display connection is not safe to do as it is shared between all other code in the same process and no reference counting or anything happens at the platform layer. 2017-10-01 16:09:13 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Ignore medias marked as sendonly We're never going to receive anything from them, so don't create pads for them. These medias are destinations where *we* could send something. 2017-09-05 11:41:35 +0300 Sebastian Dröge * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsbcdepay.h: sbcdepay: Add property to ignore input timestamps This then just counts samples and calculates the output timestamps based on that and the very first observed timestamp. The timestamps on the buffers are continued to be used to detect discontinuities that are too big and reset the counter at that point. When receiving data via Bluetooth, many devices put completely wrong values into the RTP timestamp field. For example iOS seems to put a timestamp in milliseconds in there, instead of something based on the current sample offset (RTP clock-rate == sample rate). https://bugzilla.gnome.org/show_bug.cgi?id=787297 2017-09-21 13:59:00 +0530 Ponnam Srinivas * gst/rtp/gstrtph265depay.c: rtph265depay: Fix Memory leak in error case https://bugzilla.gnome.org/show_bug.cgi?id=787937 2017-09-22 16:55:21 +0530 Deepak Srivastava * gst/deinterlace/gstdeinterlace.c: deinterlace: Fixed memory leak in error code path https://bugzilla.gnome.org/show_bug.cgi?id=788041 2017-09-20 09:37:59 +0530 Ponnam Srinivas * ext/libpng/gstpngenc.c: pngenc: fix memory leak in error code path Don't leak row_pointers if frame can't be mapped. https://bugzilla.gnome.org/show_bug.cgi?id=787885 2017-09-19 17:55:58 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Don't leak codec name 2017-08-05 12:23:30 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2bufferpool: Don't stop streaming when pool is flushing The purpose of being able to flush the buffer pool is only to unlock any blocked operation. Doing streamoff/streamon had the side effect of turning off and on the camera. As we do a flush_start / flush_stop sequence when shutting down, that would cause a really quick sequence of streamoff/streamon/streamoff/close which was causing some cameras to stop working. https://bugzilla.gnome.org/show_bug.cgi?id=783945 2017-09-17 16:18:48 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: implement basic chain_list function Doesn't do anything fancy yet, but still avoids lots of unnecessary locking/unlocking that would happen if the default chain_list fallback function in GstPad got invoked. 2017-09-17 12:50:30 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: use new gst_buffer_list_calculate_size() 2017-09-14 13:00:56 +0200 Patrick Radizi * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtpbin: add option for sanity checking timestamp offset Timestamp offsets needs to be checked to detect unrealistic values caused for example by NTP clocks not in sync. The new parameter max-ts-offset lets the user decide an upper offset limit. There are two different cases for checking the offset based on if ntp-sync is used or not: 1) ntp-sync enabled Only negative offsest are allowed since a positive offset would mean that the sender and receiver clocks are not in sync. Default vaule of max-ts-offset = 0 (disabled) 2) ntp-sync disabled Both positive and negative offsets are allowed. Default vaule of max-ts-offset = 3000000000 The reason for different default values is to be backwards compatible. https://bugzilla.gnome.org/show_bug.cgi?id=785733 2017-09-14 11:20:17 +0200 Patrick Radizi * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpsource.c: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtpbin: add option for increasing ts_offset gradually Instant large changes to ts_offset may cause timestamps to move backwards and also cause visible effects in media playback. The new option max-ts-offset-adjustment lets the application control the rate to apply changes to ts_offset. https://bugzilla.gnome.org/show_bug.cgi?id=784002 2017-09-06 07:59:56 +0000 Jochen Henneberg * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qmlglsink: Expose itemInitialized as property Instead of just signalling when ready exposing the state as a property allows us to bind at any time if player is loaded async. 2017-09-13 16:05:08 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Ensure all caps a fixated The code relied on the list compare function to fixate the caps but if the caps only has one structure, the compare function will never get called. Capture device for which there is only one structure in the caps would then get some assertion and later fail badly. Instead, fixate before inserting into the list and split the reading and the fixation of the structures. 2017-09-13 11:52:09 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't leak the par value 2017-09-13 11:38:44 -0400 Nicolas Dufresne * tests/examples/v4l2/v4l2src-renegotiate.c: v4l2-renegotiate: Don't leak the option context 2017-09-13 11:33:33 -0400 Nicolas Dufresne * tests/examples/v4l2/v4l2src-renegotiate.c: v4l2src-renegotiate: Don't leak pipeline desc string 2017-09-13 11:32:09 -0400 Nicolas Dufresne * tests/examples/v4l2/v4l2src-renegotiate.c: v4l2-renegotiate: Change --enable-dmabuf into --io-mode= This gives allow testing dmabuf importation but also exportation buy letting user pick anything from the io-mode property on v4l2src. 2017-09-11 20:24:27 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: search_cluster should find preceding cluster before target ... since failing this constraint takes search_pos by surprise which might then end up in an infinite loop. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=787538 2017-09-07 14:33:57 +0300 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxsend.c: rtprtx{send,receive}: improve the debug messages * use INFO/DEBUG/LOG/TRACE equaly and meaningfully; previously rtprtxsend:LOG and rtprtxreceive:LOG would generate a totally different amount of log traffic and sometimes it was impossible to see the information you wanted without useless spam being printed around * improve the wording, give a reasonable and self-explanatory amount of information * print SSRCs in hex * avoid G_FOO_FORMAT for readability (we are just printing integers) 2017-09-07 09:39:13 +0100 Tim-Philipp Müller * ext/qt/gstplugin.cc: * ext/qt/qtplugin.pro: qt: fix build with qmake Move the package defines for GST_PLUGIN_DEFINE from the command line into the source file to avoid quoting issues (-DPACKAGE_NAME="foo" means the quotes won't actually make it to the compiler and then it no longer gets a string constant). 2017-09-05 16:20:44 -0400 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: Request minimum buffer even if need_pool is FALSE When tee is used, it will not request a pool, but still it wants to know how many buffers are required. https://bugzilla.gnome.org/show_bug.cgi?id=730758 2017-09-05 16:20:44 -0400 Nicolas Dufresne * ext/qt/gstqtsink.cc: Request minimum buffer even if need_pool is FALSE When tee is used, it will not request a pool, but still it wants to know how many buffers are required. https://bugzilla.gnome.org/show_bug.cgi?id=730758 2017-09-05 15:30:40 +0100 Ian Jamison * sys/v4l2/gstv4l2object.c: v4l2object: Handle BT2020 for colorspace and transfer This was not fully handled in switches and ub gst_v4l2_object_get_colorspace(); https://bugzilla.gnome.org/show_bug.cgi?id=787313 2017-09-05 15:29:24 +0100 Ian Jamison * sys/v4l2/gstv4l2object.c: v4l2object: Fix colorimetry transfer lookup for 4K video https://bugzilla.gnome.org/show_bug.cgi?id=787160 2017-09-06 11:25:53 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Check if caps have changed after try_fmt try_fmt will update the caps colorimetry and interlace-mode. Before this call, those field are missing. The caps equality check was always failing when a spurious reconfigure event was received. 2017-09-06 23:55:38 +1000 Jan Schmidt * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Allow MPEG layer 1/2, AC3 and Opus in qtmux qtmux is supposed to be the muxer that allows all formats, with others (mp4mux and friends) being profile-restricted. 2017-09-05 12:56:44 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: fix keyunit detection https://bugzilla.gnome.org/show_bug.cgi?id=787254 2017-09-05 15:42:17 +0300 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Fix decoding of streams that don't signal exactly twice the height ... and also progressive streams. 2017-09-05 13:28:16 +0300 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Handle interlaced MJPEG streams These come with two JPEG images per buffer of half height than signalled in the container. Changes based on Tim-Philipp Müller's 0.10 branch: https://cgit.freedesktop.org/~tpm/gst-plugins-good/log/?h=jpegdec-interlaced https://bugzilla.gnome.org/show_bug.cgi?id=568555 2017-09-01 15:00:12 +1000 Matthew Waters * ext/gtk/gstgtkglsink.c: * ext/gtk/gtkgstglwidget.c: gtkglsink: expose the created display and context correctly 1. Propagate the GstGLDisplay we create 2. Add the created GstGLContext to the propagated GstGLDisplay Otherwise with multi-branch GL pipelines involving gtkglsink, things will fall apart and errors will be genarated somewhere. 2017-09-04 17:06:39 +0200 Edward Hervey * gst/audioparsers/gstdcaparse.c: dcaparse: Really fix "usage before unmap" Previous patch would try to unref a buffer that was pushed downstream. Instead only unref when/if needed and keep usage of the cleanup: goto block 2017-09-03 15:23:10 +0530 Arun Raghavan * gst/audioparsers/gstdcaparse.c: dcaparse: Don't unmap buffer before accessing data from it The previous patch added a check for a substream header after gst_buffer_unmap(), which is incorrect. 2017-06-24 18:47:14 +0200 Matej Knopp * gst/audioparsers/gstdcaparse.c: dcaparse: preserve DTS HD substream 2017-09-01 15:56:04 +0200 Edward Hervey * ext/qt/gstqtgl.h: qt: Only include qtgui-config.h on qt >= 5.9.0 The file does not exist in previous versions 2017-08-31 14:40:44 +1000 Matthew Waters * ext/qt/gstqtgl.h: qt: the defines for QT_OPENGL_ES_2 have moved Update the includes to account for that 2017-04-26 13:50:41 +0200 Jochen Henneberg * ext/qt/qtwindow.cc: qt: ensure GL_DRAW_FRAMEBUFFER 2017-08-14 18:18:07 +0530 Arun Raghavan * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Fix some tabs that crept in somehow 2017-08-29 19:13:58 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Also log local and SR RTP running times when doing ntp-sync=true 2017-08-24 17:06:38 +1000 Matthew Waters * gst/rtpmanager/gstrtpbin.c: rtpbin: also create session when creating the send_rtcp_src_%u pad If one requests the send_rtcp_src_%u pad before a recv_rtcp_sink_%u pad, the session/pad would never be created and NULL was returned. Switching the request order would work. https://bugzilla.gnome.org/show_bug.cgi?id=786718 2017-08-26 12:59:35 +0100 Tim-Philipp Müller * tests/files/Makefile.am: * tests/files/cbr_stream.mp3: * tests/files/stream.mp2: * tests/files/vbr_stream.mp3: tests: mpg123audiodec: add files needed by unit tests 2017-08-26 10:10:19 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/gst-plugins-good.supp: * tests/check/pipelines/.gitignore: * tests/check/pipelines/lame.c: * tests/check/pipelines/twolame.c: tests: add basic unit test for twolame as well 2017-08-26 09:59:22 +0100 Tim-Philipp Müller * tests/check/pipelines/lame.c: tests: lame: fix build 2017-08-26 09:52:33 +0100 Tim-Philipp Müller * tests/examples/v4l2/.gitignore: tests: ignore another binary 2017-08-26 09:41:13 +0100 Tim-Philipp Müller * REQUIREMENTS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-twolame.xml: * ext/Makefile.am: * ext/meson.build: * ext/twolame/meson.build: * po/POTFILES.in: twolame: hook up to build system https://bugzilla.gnome.org/show_bug.cgi?id=774252 2017-08-26 09:21:44 +0100 Tim-Philipp Müller Moving twolame mp2 encoder plugin from -ugly https://bugzilla.gnome.org/show_bug.cgi?id=774252 2017-08-26 09:03:08 +0100 Tim-Philipp Müller * REQUIREMENTS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-lame.xml: * ext/Makefile.am: * ext/lame/Makefile.am: * ext/lame/meson.build: * ext/meson.build: * po/POTFILES.in: * tests/check/Makefile.am: * tests/check/gst-plugins-good.supp: * tests/check/meson.build: lame: hook up to build system https://bugzilla.gnome.org/show_bug.cgi?id=774252 2017-08-25 21:13:58 +0100 Tim-Philipp Müller Moving lame mp3 encoder plugin from -ugly https://bugzilla.gnome.org/show_bug.cgi?id=774252 2017-08-22 12:39:43 +0100 Julien Isorce * ext/qt/gstqsgtexture.cc: * ext/qt/gstqtglutility.cc: * ext/qt/gstqtsink.cc: * ext/qt/qtwindow.cc: qt: fix broken build due to commit 2fd84a6c for gstgl https://bugzilla.gnome.org/show_bug.cgi?id=784779 2017-07-07 16:15:12 +0100 Julien Isorce * ext/gtk/Makefile.am: * ext/gtk/gstgtkglsink.c: * ext/gtk/gtkgstglwidget.c: * tests/examples/gtk/glliveshader.c: gl: do not include GL headers in public gstgl headers Except for gst/gl/gstglfuncs.h It is up to the client app to include these headers. It is coherent with the fact that gstreamer-gl.pc does not require any egl.pc/gles.pc. I.e. it is the responsability of the app to search these headers within its build setup. For example gstreamer-vaapi includes explicitly EGL/egl.h and search for it in its configure.ac. For example with this patch, if an app includes the headers gst/gl/egl/gstglcontext_egl.h gst/gl/egl/gstgldisplay_egl.h gst/gl/egl/gstglmemoryegl.h it will *no longer* automatically include EGL/egl.h and GLES2/gl2.h. Which is good because the app might want to use the gstgl api only without the need to bother about gl headers. Also added a test: cd tests/check && make libs/gstglheaders.check https://bugzilla.gnome.org/show_bug.cgi?id=784779 2017-08-20 20:41:19 -0300 Thibault Saunier * tests/check/meson.build: meson: Link mpeg123audiodec test against gstfft Fixing build error: /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:150: undefined reference to `gst_fft_s32_new' /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:151: undefined reference to `gst_fft_s32_window' /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:151: undefined reference to `gst_fft_s32_fft' /run/build/gst-plugins-good/_flatpak_build/../tests/check/elements/mpg123audiodec.c:147: undefined reference to `gst_fft_s32_free' 2017-08-20 17:15:33 +0100 Tim-Philipp Müller * tests/check/pipelines/tagschecking.c: tests: tagschecking: remove gst-check-xmp-* temp files when done Also fix temp file creation a bit. 2017-08-20 15:49:12 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-video4linux2.xml: docs: update for changes in git 2017-08-20 15:48:24 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-mpg123.xml: mpg123: add to docs 2017-08-20 13:56:19 +0100 Tim-Philipp Müller * REQUIREMENTS: * configure.ac: * ext/Makefile.am: * ext/meson.build: * ext/mpg123/meson.build: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/meson.build: mpg123: hook up to build system https://bugzilla.gnome.org/show_bug.cgi?id=774252 2017-08-20 13:48:48 +0100 Tim-Philipp Müller Moving mpg123 plugin from -ugly 2017-08-17 12:23:25 +0100 Tim-Philipp Müller * README: * common: Automatic update of common submodule From 48a5d85 to 3f4aa96 2017-08-14 15:28:22 +0800 Sky Juan * gst/audioparsers/gstac3parse.c: ac3parse: fix not-linked handling causing glitches when selecting stream Fix chain function not handling not-linked from baseparse. When an input data is separated into 2 buffers, the second buffer would not be pushed into the adapter if baseparse returns not-linked for first buffer. This caused glitches when switching streams and selecting a stream that was previously unselected. https://bugzilla.gnome.org/show_bug.cgi?id=786268 2017-08-16 13:57:50 +0200 Jan Alexander Steffens (heftig) * gst/goom2k1/filters.c: * gst/goom2k1/filters.h: * gst/goom2k1/goom_core.c: goom2k1: Convert source files to UTF-8 Causes problems with the new gtk-doc 1.26 otherwise, but is a good idea in any case. https://bugzilla.gnome.org/show_bug.cgi?id=786364 2017-08-14 03:08:41 -0500 Eduard Sinelnikov * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: Add support for growing WAV files With some fixes by me. 2017-08-14 17:39:15 +0530 Arun Raghavan * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Fix compile error 2017-05-21 16:01:14 +0200 Carlos Rafael Giani * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qmlglsink: Add itemInitialized signal to QML item This is useful for autoplay for example. With autoplay, it is necessary to wait until the scene graph is fully set up. This signal is emitted once the QML item node is ready. So, inside a connected slot, the pipeline's state can be set to PLAYING to automatically start playback as soon as the QML script is loaded. https://bugzilla.gnome.org/show_bug.cgi?id=786246 2017-08-14 10:36:56 +0000 Jochen Henneberg * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: fix if buffer size exceeds MTU The plugin queued buffer data if not all buffer data fit into a single RTP packet. Now RTP packets are pushed as long as enough data is available. 2017-07-27 17:21:48 +0300 George Kiagiadakis * ext/vpx/gstvpxenc.c: vpxenc: discard frames that have been dropped by libvpx This fixes a memory leak. When dropframe-threshold has been set, libvpx may output less frames than the input ones, which causes some GstVideoCodecFrames to queue up in GstVideoEncoder's internal frame queue with no chance of ever being all released. And because the frames keep references to the input buffers, the input buffer pool keeps allocating new buffers and memory usage grows very fast. For example the following pipeline's memory usage grows at a rate of about 1GB per minute! videotestsrc ! capsfilter caps=video/x-raw,width=1920,height=1080,framerate=30/1,format=I420 ! \ vp8enc target-bitrate=1000000 end-usage=cbr dropframe-threshold=95 ! fakesink https://bugzilla.gnome.org/show_bug.cgi?id=783086 2017-08-08 13:11:58 +0200 Mathieu Duponchelle * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpstats: fix unsigned integer comparisons. Callers of the API (rtpsource, rtpjitterbuffer) pass clock_rate as a signed integer, and the comparison "<= 0" is used against it, leading me to think the intention was to have the field be typed as gint32, not guint32. This led to situations where we could call scale_int with a MAX_UINT32 (-1) guint32 as the denom, thus raising an assertion. https://bugzilla.gnome.org/show_bug.cgi?id=785991 2017-08-10 14:44:35 +0100 Tim-Philipp Müller * ext/taglib/meson.build: taglib: use -fvisibility=hidden with this C++ plugin in meson too Also pass args as cpp_args. 2017-03-22 15:25:17 +0100 Michael Olbrich * gst/isomp4/qtdemux.c: qtdemux: allow larger files For really long files such as contiguous recordings of a whole day, the 50MB limit is not sufficient. https://bugzilla.gnome.org/show_bug.cgi?id=781458 2017-08-10 16:08:06 +0300 Sebastian Dröge * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Fix offsets for reading lpcm specific fields We were reading at the completely wrong positions, 16 bytes later in the data. Also add support for high-aligned samples. 2017-08-10 14:01:09 +0100 Tim-Philipp Müller * meson.build: meson: don't export symbols by default Only plugin entry points should be exported. Currently plugins might export more symbols with the meson build, as we don't have the exports regexp there that we pass to libtool. 2017-08-10 15:14:31 +0530 Deepak Srivastava * gst/wavparse/gstwavparse.c: wavparse: Fix memory leak in wavparse element Fixing of leaking the text field of the GstWavParseNote and GstWavParseLabl structure. https://bugzilla.gnome.org/show_bug.cgi?id=785429 2017-08-08 10:37:12 +0000 Cyril Lashkevich * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't mark jpeg frames as deltas JPEG formats are encoded, but they never have keyframe flag. But in fact they are keyframes https://bugzilla.gnome.org/show_bug.cgi?id=785990 2017-08-06 13:06:45 +0100 Philippe Normand * sys/osxvideo/Makefile.am: osxvideo: rename library according to the plugin name https://bugzilla.gnome.org/show_bug.cgi?id=785880 2017-08-02 17:16:21 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't drop buffer ref on qbuf This function no longer take ownership of the buffer. CID 1414800 2017-08-02 17:13:55 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Enable VP9 format This was missing, preventing the encoder and decoder to work properly. This also adds support for camera that would produce VP9 (if that exists). 2017-08-02 12:28:38 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2h263enc.h: * sys/v4l2/gstv4l2h264enc.h: * sys/v4l2/gstv4l2mpeg4enc.h: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2transform.h: * sys/v4l2/gstv4l2videodec.h: * sys/v4l2/gstv4l2videoenc.h: * sys/v4l2/gstv4l2vp8enc.h: * sys/v4l2/gstv4l2vp9enc.h: v4l2: Remove spurious CATEGORY_EXTERN These have been copy pasted all over the place and are not used anymore. All object have it's own category now. This fixes build warning since the VP9 decoder had vp8 category declared. 2017-08-02 10:39:46 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2mpeg4enc.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2videoenc.h: * sys/v4l2/gstv4l2vp8enc.c: * sys/v4l2/gstv4l2vp9enc.c: v4l2videoenc: Move the profile/level negotation in the base class This removes duplicated code across different codec. 2017-08-02 09:36:08 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2h263enc.c: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2mpeg4enc.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2videoenc.h: * sys/v4l2/gstv4l2vp8enc.c: * sys/v4l2/gstv4l2vp9enc.c: v4l2videoenc: Turn gst_v4l2_is_video_enc into a helper This reduces the amount of code needed in each codec class. 2017-08-01 16:01:11 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2vp8enc.c: * sys/v4l2/gstv4l2vp8enc.h: * sys/v4l2/gstv4l2vp9enc.c: * sys/v4l2/gstv4l2vp9enc.h: * sys/v4l2/meson.build: v4l2: Add VP8/9 encoder support 2017-07-31 11:56:05 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Use mmap64 to match libv4l2 signature https://bugzilla.gnome.org/show_bug.cgi?id=785628 2017-08-01 09:22:43 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Copy flags and timestamp when importing Whenever we import from downstream pool (userptr or dmabuf-import), we should copy over the flags and timestamp, otherwise downstream will not get proper synchronization or will not be able to notice frames that has corruption in it. https://bugzilla.gnome.org/show_bug.cgi?id=785680 2017-07-31 16:09:30 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2h263enc.c: * sys/v4l2/gstv4l2h263enc.h: * sys/v4l2/meson.build: v4l2: Add H263 Encoder support 2017-07-27 13:51:25 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: v4l2: Add missing no-inst header 2017-07-26 15:18:01 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2mpeg4enc.c: * sys/v4l2/gstv4l2mpeg4enc.h: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2videoenc.h: * sys/v4l2/meson.build: v4l2: Add interface for MPEG4 encoding 2017-07-27 10:51:07 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2h264enc.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2transform.h: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2videoenc.h: v4l2: Ignore register issue and keep probing Don't stop registering the other dynamic plugins if one registration fails. 2017-07-27 14:21:34 +0300 Sebastian Dröge * gst/law/mulaw-decode.c: mulawdec: Unmap input buffer if failing to map the output buffer 2017-07-27 09:22:25 +0530 Satya Prakash Gupta * gst/law/alaw-decode.c: alawdec: Fix Memory leak in error case https://bugzilla.gnome.org/show_bug.cgi?id=785435 2017-07-26 20:36:15 -0400 Nicolas Dufresne * sys/v4l2/ext/v4l2-common.h: * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: v4l2: Update external files with latest This is copied from the linux kernel with only some include changes so it works outside the kernel headers. 2017-07-18 10:41:40 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: For audio tracks, take the default duration from the first buffer ... if we don't have any better idea from the caps. This allows writing SimpleBlocks for a majority of audio streams where the duration of frames is usually fixed. And as a side effect, allows VLC to play streams with Opus as it only works with SimpleBlocks currently: https://trac.videolan.org/vlc/ticket/18545 https://bugzilla.gnome.org/show_bug.cgi?id=784969 2017-07-24 16:45:40 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.h: v4l2: Fix compilation without libv4l2 2017-07-24 16:13:56 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: v4l2: Keep ref to element in allocator/pool Removes the FIXME/Question in the buffer pool and add a ref to the element in the GstAllocator too. This ref is strictly required to keep the GstV4l2Object structure around. 2017-07-24 14:27:05 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Removed unused members 2017-07-24 14:19:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/v4l2_calls.c: v4l2: Add run-time environment to enable libv4l2 The library has started preventing a lot of interesting use cases, like CREATE_BUFS, DMABuf, usage of TRY_FMT. As the libv4l2 is totally inactive and not maintained, we decided to disable it. As a convenience we added a run-time environment that let you enable it for testing. GST_V4L2_USE_LIBV4L2=1 This of course only works if you have enabled libv4l2 at build time. 2017-07-17 10:04:02 +0200 Nicola Murino * ext/jpeg/gstjpegenc.c: jpegenc: declare quality property changeable in PLAYING state https://bugzilla.gnome.org/show_bug.cgi?id=785012 2017-07-21 23:34:59 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix colorimetry validation While not documented, gst_video_colorimetry_matches() only accepts well known names. Looking at the code and unit test, this seems to be on purpose, so fixing by parsing the string and compating the colorimetry structures. 2017-07-21 15:40:24 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2encoder: Fix negotiation error handling The subclass negotiated function will call set_format, if that fails the pool will not be created. We ended up with an assertion. GStreamer-CRITICAL **: gst_buffer_pool_set_active: assertion 'GST_IS_BUFFER_POOL (pool)' failed 2017-07-19 22:25:49 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Speedup camera startup by skipping try_fmt In this commit, we enabled skip_try_fmt_probes quirk in order to speed up the start which is known to be disastrously slow with certain USB cameras. This has the side effect that we needed to rewrite the entire negotiation process in a way that we iterate over the possible caps until we find one that works. The new negotiation method consist of extracting a preferred structure from the peer caps and using this to fixate and sort the caps. To reflect the old behaviour, we sort all resolution strictly bigger to the preferred one with the closes one first. The rest is appended, keeping the same order. We then normalize the caps in case there was some list of interlace-mode or colorimetry left. We finally iterate over all fixed caps and try it. 99% of the time, the first or the second one should work, whit the result of a single S_FMT being issues. From there, it will be relatively easy to introduce new negotiation algorithm. The current algorithm is made for optimal image quality with a scaling sink that sets it's window resolution as preference. This the case if for: v4l2src ! videoconvert ! videoscale ! ximagesink Other strategy would be needed to optimize for non-scaling sink like ximagesink or kmssink when the driver does not scale. https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-19 22:09:38 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Introduce quirk to skip slow probes skip_try_fmt_probes quirk is set, V4L2 object will not probe for interlace-mode and colorimetry to avoid relying on try_fmt. This quirk will be used by v4l2src to avoid desastrous startup time with slow USB webcams. When this quirk is enabled, caller will have to iterate over the negotiated caps as it may contains unsupported formats. If the peer didn't choose a specific interlace-mode, or colorimetry, the value chosen by the driver is set into the caps. For this reason, when this mode is enabled, gst_v4l2_object_set_format() will require writable caps. https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-19 22:07:32 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: always set the GstV4l2Error on error Some of the error case were conditional to using try_fmt or not. This is slightly unexpected, always set the error so the caller can decide. https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-19 22:05:49 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Minor style fix and useful trace https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-19 22:03:29 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix try/s_fmt errors According to the spec,TRY_FMT cannot return EBUSY, though it can return EINVAL if it was not possible to update the format to something supported. https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-19 22:01:26 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Validate colorimetry in S/TRY_FMT This is in preparation for removing slow TRY_FMT probes for colorimetry. As we won't have tried that colorimetry we cannot assume the driver will accept it. https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-19 21:56:14 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Validate field in S/TRY_FMT This is in preparation from removing the slow TRY_FMT probes for interlacing. As we won't have tried that interlace-mode already we need to validate that the driver isn't refusing it. https://bugzilla.gnome.org/show_bug.cgi?id=785156 2017-07-21 19:01:19 +0100 Tim-Philipp Müller * tests/icles/test-accurate-seek.c: tests: icles: fix build Can't do additions/subtractions on void* pointers. 2017-07-21 11:04:17 -0400 Thibault Saunier * tests/icles/test-accurate-seek.c: tests:icles: Fix previous patch by implementing our memmem Using the string version of it will fail on '\0'. 2017-07-21 10:17:00 -0400 Thibault Saunier * tests/icles/test-accurate-seek.c: tests:icles: Do not use memmem GNU extension function As it is not avalaible on windows/msvc and we can use pure GLib for that 2017-07-20 17:21:05 -0400 Nicolas Dufresne * sys/directsound/Makefile.am: directsound: Fix .c file name in Makefile This was broken by accident, bad search and replace. 2017-07-20 11:02:10 -0400 Nicolas Dufresne * Makefile.am: * sys/waveform/Makefile.am: waveform: Fix DLL name to match plugin name https://bugzilla.gnome.org/show_bug.cgi?id=785168 2017-07-20 10:38:32 -0400 Nicolas Dufresne * Makefile.am: * sys/directsound/Makefile.am: * sys/directsound/meson.build: directsound: Fix DLL name to match plugin name https://bugzilla.gnome.org/show_bug.cgi?id=785168 2017-07-19 12:38:03 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: preferably send open-ended segment rather than repeated segment events 2017-07-19 11:27:32 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: fix seeking in fragmented file without mfra random access info ... which no longer worked due to unconditionally clearing sample info and ending up in inconsistent state. Let's tread a bit more carefully and also allow for the old seek handling that resorts to scanning if no mfra info is available. 2017-07-19 10:42:46 +0200 Nicolas Dechesne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: add some useful debug messages Add a couple of useful debug traces , they happened to be useful to debug/investigate a 4K video playback issue with v4l2, so let's make these changes more permanent. Signed-off-by: Nicolas Dechesne https://bugzilla.gnome.org/show_bug.cgi?id=785109 2017-07-18 11:28:37 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Fix 4K colorimetry Since 1.6, the transfer function for BT2020 has been changed from BT709 to BT2020_12. It's the same function, but with more precision. As a side effect, the V4L2 colorpsace didn't match GStreamer colorspace. When GStreamer ended up making a guess, it would not match anything supported by V4L2 anymore. This this by using BT2020_12 for BT2020 colorspace and BT2020 transfer function in replacement of BT709 whenever a 4K resolution is detected. 2017-07-14 16:21:38 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Only check CROPCAP for par once The pixel aspect ratio is documented to not change unless the TV Standard is changed. So this mean that this will be uniform across all possible format and resolutions. https://bugzilla.gnome.org/show_bug.cgi?id=784674 2017-07-18 10:01:13 +0300 Sebastian Dröge * tests/check/elements/matroskamux.c: Revert "matroskamux: adjust unit test to modified behaviour" This reverts commit 8fe478c8a7746cd2c63f20d23e97e26e1a0e6192. We're back to previous behaviour 2017-07-18 00:26:11 +0200 Nicola Murino * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: add properties to control cluster duration https://bugzilla.gnome.org/show_bug.cgi?id=784971 2017-07-17 20:47:26 -0400 Nicolas Dufresne * sys/v4l2/v4l2_calls.c: v4l2: UVC driver is named uvcvideo these days The quirk to avoid probing interlacing didn't work anymore as the driver is now name uvcvideo. This should slightly speed up camera startup. 2017-07-12 21:02:39 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Remove unused defines 2017-07-12 20:53:51 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: Make gst_v4l2_get_capabilities static It's not used outside of v4l2_calls.c 2017-07-12 20:49:47 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: v4l2: Merge v4l2_calls.h into gstv4l2object.h First step of a larger cleanup, all function from v4l2_calls are in fact methods on GstV4l2Object. This split makes the code really confusing. This also remove no longer unused macros. 2017-07-15 14:57:49 +0100 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.c: mpg123audiodec: fix caps leak The pad template takes its own ref, so we should unref the caps. https://bugzilla.gnome.org/show_bug.cgi?id=784982 2017-07-15 12:48:19 +0100 Tim-Philipp Müller * po/meson.build: meson: po: use glib preset and read language list from LINGUAS Supported since meson 0.37, so we can use it now. 2017-07-14 12:12:56 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Trace unknown fourcc as text This makes it easier to find out what is not supported. 2017-07-14 11:54:57 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videoenc.c: v4l2: Don't probe for unneeded format For v4l2videodec/enc, we generate elements per formats, and in this case we can speed up the start up by only probing the format we care about. 2017-07-13 12:32:00 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Implement stable element names Before that, each m2m node would be wrapped as a single, multi-format decoder element. As a unique name was needed, we where using the device name, which changes between re-boots. This led to unpredictable element names. In this patch, we generate an element per codec, using v4l2dec name. If there is multiple decoder for the same format, the following elements will be named v4l2dec. https://bugzilla.gnome.org/show_bug.cgi?id=784908 2017-07-13 14:50:44 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Post an element message with the HTTP headers on the bus too Instead of just sending a sticky event with them downstream. This allows getting the HTTP headers easily in the application, and especially also on errors. 2017-07-13 12:47:02 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix parsing of RLE depth Regression introduced by 86b427dc70562f891a551ffc9f96cefe1cafcddd https://bugzilla.gnome.org/show_bug.cgi?id=784812 2017-07-12 15:29:32 +1000 Jan Schmidt * ext/qt/gstqtsink.cc: * ext/qt/gstqtsink.h: * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qt: Use a proxy object for access to the QML widget QML can destroy the video widget at any time, leaving us with a dangling pointer. Use a lock and a proxy object to cope with that, and block in the widget destructor if there are ongoing calls into the widget. 2017-07-10 18:57:11 +0200 Philippe Renon * ext/shout2/gstshout2.h: shout2: use gint and guint in place of int and uint this fixes a compilation error with gcc 7.1.0 on mys2 where uint is not defined https://bugzilla.gnome.org/show_bug.cgi?id=784758 2017-07-07 21:15:57 +0900 Yasushi SHOJI * gst/rtp/gstrtpgsmpay.c: rtpgsmpay: fix accidental garbage data before actual payload Do not allocate payload size outbuf if appending payload buffer. The commit 137672ff1824948bda4b1b1967de8c24a0055b67 attached payload to the output buffer but forgot to remove payload allocation. That effectively doubled payload size and add zero'ed or random bytes. Makes the following pipeline work again: gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink https://bugzilla.gnome.org/show_bug.cgi?id=784616 2017-07-01 18:57:47 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: segment seek position is expressed in buffer time ... so it need not be corrected again for stream start 2017-07-09 10:54:27 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: provide average bitrate tag 2017-07-07 23:49:44 -0700 Reynaldo H. Verdejo Pinochet * tests/examples/v4l2/v4l2src-renegotiate.c: examples: v4l2: fix wrong initializations brought by 4e8ad583022671c5 https://bugzilla.gnome.org/show_bug.cgi?id=682770 2015-02-27 13:03:42 -0300 Nicolas Dufresne * tests/examples/v4l2/Makefile.am: * tests/examples/v4l2/meson.build: * tests/examples/v4l2/v4l2src-renegotiate.c: examples: v4l2: add example for v4l2src renegotiation Based on work from Thiago Santos https://bugzilla.gnome.org/show_bug.cgi?id=682770 2017-07-07 11:58:10 +0100 Tim-Philipp Müller * meson.build: meson: find python3 via python3 module https://bugzilla.gnome.org/show_bug.cgi?id=783198 2017-07-05 14:44:41 +0100 Tim-Philipp Müller * tests/check/Makefile.am: tests: rtpbin: fix build in uninstalled setup 2017-07-04 17:42:25 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * tests/check/Makefile.am: * tests/check/elements/rtpbin.c: rtpsession: Send EOS if all internal sources sent bye The ones which are not internal should not matter, and we should wait for all sources to have sent their BYEs. And add unit test https://bugzilla.gnome.org/show_bug.cgi?id=773218 2017-07-04 12:24:41 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Only send EOS if all sources have been marked bye Now that multiple sender RTPSource can share the same RTPSession, we must not send an EOS unless they're all marked bye. 2017-07-04 11:49:29 -0400 Thibault Saunier * ext/libcaca/gstcacasink.c: caca: Do not include, unused, sys/time.h Which moreover makes building on windows (mingw/msvc) fail: https://ci.appveyor.com/project/thiblahute/gst-build-ge9m5 2017-07-03 11:47:13 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: Add memory and boudary checks This element was not checking if mapping the RTP buffer and the payload worked, and was not checking if the RTX payload was large enough. https://bugzilla.gnome.org/show_bug.cgi?id=784484 2017-07-04 14:58:00 +0900 Seungha Yang * ext/soup/gstsouphttpsrc.c: souphttpsrc: Unset limit on the number of connection if soup session sharing is used Soup allows only up to two connections per host in a session, if we use default value. When session sharing is used, however, more connections might be required in a session. (e.g., multi-audio adaptive streaming case) https://bugzilla.gnome.org/show_bug.cgi?id=784495 2017-07-03 20:27:29 +0100 Tim-Philipp Müller * gst/imagefreeze/gstimagefreeze.c: imagefreeze: fix use-after-free on seek event Get seqnum before unreffing the seek event. https://bugzilla.gnome.org/show_bug.cgi?id=784486 2017-07-01 18:59:14 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: robustify time tracking for sparse subtitle stream 2017-07-01 18:59:07 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: correctly track chunk size of subtitle stream ... thereby ensuring correct chunk offset tracking for all streams. 2017-06-27 15:59:18 +0100 Julien Isorce * gst/rtpmanager/rtpstats.h: rtpstats: fix assertion 'denom > 0' failed gst_util_uint64_scale_int takes a gint as denom parameter whereas ctx->clock_rate is a guint32. It happens when gst_rtp_packet_rate_ctx_reset set clock_rate to -1. So just define clock_rate as gint like it is done in rtpsource.h https://bugzilla.gnome.org/show_bug.cgi?id=784250 2017-06-28 14:05:27 -0500 Matt Fischer * sys/v4l2/gstv4l2bufferpool.c: v4l2: Block recursive calls to resurect_buffer When resurrecting a buffer, the subsequent free call can result in the group-released handler being called again, which causes a recursive loop. This patch blocks the signal handler during the time that it executes, ensuring that the loop will not occur. https://bugzilla.gnome.org/show_bug.cgi?id=759292 2017-06-20 16:39:36 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: Avoid deprecated ssl-ca-file property SoupSession's ssl-ca-file property is deprecated. Use the recommended tls-database property. This is a bit more complex as it requires creating a GTlsFileDatabase object for an absolute (!) path to the CA certificates file. https://bugzilla.gnome.org/show_bug.cgi?id=784005 2017-06-20 16:37:55 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: Avoid deprecated server ssl properties The ssl-cert-file and ssl-key-file properties are deprecated. Use the soup_server_set_ssl_cert_file function to load the files. https://bugzilla.gnome.org/show_bug.cgi?id=784005 2017-06-20 16:34:41 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: Make ssl_cert/key_file static Just a bit of cleanup. https://bugzilla.gnome.org/show_bug.cgi?id=784005 2017-06-20 16:28:35 +0200 Jan Alexander Steffens (heftig) * tests/files/test-cert.pem: tests: souphttpsrc: Update test-cert.pem Recent GnuTLS disregards the Common Name and only looks at the Subject Alternative Name extension. Since our test-cert has no SAN extension, validation fails. Generate a new certificate with SAN. In addition to 127.0.0.1, for good measure make it valid for localhost and ::1, too. https://bugzilla.gnome.org/show_bug.cgi?id=784005 2017-06-29 15:22:39 -0400 Nicolas Dufresne * ext/soup/gstsouphttpsrc.c: souphttpsrc: Allow any type of proxy Currently we only allowed HTTP proxy. Don't filter for the scheme, just check if it looks like an URI. Soup will warn if the URI is invalid or if proxy protocol is not supported. This enables using SOCKS 4/5 which is directly implemented into GIO. https://bugzilla.gnome.org/show_bug.cgi?id=783012 2017-05-24 15:07:51 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: increase by one the number of allocated buffers Increasing this number fix a buffer starvation problem I'm hitting with a "v4l2src ! kmssink" pipeline. kmssink requests 2 buffer as it keeps a reference on the last rendered one. So we were allocating 3 buffers for the pipeline. Once the first 2 buffers have been pushed we ended up with: - one buffer queued in v4l2 - one being pushed - one kept as last rendered If this 3rd buffer is released after that v4l2 used the first one to capture we end up with a buffer starvation problem as no buffer is currently queued in v4l2 for capture. Fixing this by adding one extra buffer to the pipeline so when one buffer is being pushed downstream the other can already be queued to capture the next frame. We were already adding 3 buffers if downstream didn't reply to the allocation query. I reduced this number to 2 to compensate the extra buffer which is now always added. https://bugzilla.gnome.org/show_bug.cgi?id=783049 2017-06-29 18:59:58 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Create send/recv mutexes once, not on every connect() Also fixes a crash caused by freeing an uninitialized mutex in an error case. https://bugzilla.gnome.org//show_bug.cgi?id=784282 2017-06-27 18:20:17 -0500 Matt Fischer * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix memory leak with dmabuf This patch fixes a memory leak that is caused if the dmabuf file descriptor dup fails. Previously, _cleanup_failed_alloc() would not unref the memory because mems_allocated had not yet been incremented. https://bugzilla.gnome.org/show_bug.cgi?id=784302 2017-06-28 19:46:04 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux_types.c: qtdemux: specify '_swr' atom as a container atom ... so it is parsed as an mp4 style metadata atom as written by muxer 2017-06-27 20:14:57 +0200 Mark Nauwelaerts * gst/isomp4/atoms.c: qtmux: initialize mdhd language code as undefined 2017-06-22 15:34:42 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: Add a faststart-min-packets property When set this property will allow the jitterbuffer to start delivering packets as soon as N most recent packets have consecutive seqnum. A faststart-min-packets of zero disables this feature. This heuristic is also used in rtpsource which implements the probation mechanism and a similar heuristic is used to handle long gaps. https://bugzilla.gnome.org/show_bug.cgi?id=769536 2017-06-23 16:18:57 -0400 Thibault Saunier * meson.build: meson: Allow using glib as a subproject 2017-06-26 11:09:48 +0100 Tim-Philipp Müller * tests/examples/audiofx/meson.build: * tests/examples/cairo/meson.build: * tests/examples/equalizer/meson.build: * tests/examples/jack/meson.build: * tests/examples/level/meson.build: * tests/examples/meson.build: * tests/examples/rtp/meson.build: * tests/examples/shapewipe/meson.build: * tests/examples/spectrum/meson.build: * tests/examples/v4l2/meson.build: * tests/meson.build: meson: build examples https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-26 09:47:55 +0100 Tim-Philipp Müller * meson.build: meson: fix with-package-name option https://bugzilla.gnome.org/show_bug.cgi?id=784082 2017-06-26 09:38:46 +0100 Tim-Philipp Müller * tests/icles/meson.build: meson: tests: icles: simplify build file 2017-06-26 00:22:05 +0100 Tim-Philipp Müller * tests/icles/meson.build: * tests/meson.build: meson: build tests/icles/ https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-19 21:13:42 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: correctly calculate overall first_ts to ensure stream sync ... by minding and compensating for the dts_adjustment that may have been introduced in the PTS timeline. 2017-06-10 15:14:41 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: track highest known cluster position and time ... to use as a fallback initial duration estimate and to provide for interpolation when scanning for position. 2017-06-10 13:46:20 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: improve and simplify searching for cluster and position ... avoiding inefficiency proportional to file size 2017-06-08 16:55:29 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: increase chunk size when scanning for cluster 2017-06-08 16:39:06 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: maintain variable state when searching for position ... so skipping to next cluster happens efficiently 2017-06-24 00:21:00 +0100 Tim-Philipp Müller * ext/meson.build: * ext/raw1394/meson.build: meson: build raw1394 plugin https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-23 23:50:00 +0100 Tim-Philipp Müller * ext/aalib/meson.build: * ext/meson.build: meson: build aalib plugin https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-23 23:38:27 +0100 Tim-Philipp Müller * ext/libcaca/meson.build: * ext/meson.build: meson: build caca plugin https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-23 20:01:59 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update for git master 2017-06-23 19:52:04 +0100 Tim-Philipp Müller * README: * configure.ac: * meson.build: * po/POTFILES.in: * sys/Makefile.am: * sys/meson.build: * sys/sunaudio/Makefile.am: * sys/sunaudio/gstsunaudio.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixeroptions.c: * sys/sunaudio/gstsunaudiomixeroptions.h: * sys/sunaudio/gstsunaudiomixertrack.c: * sys/sunaudio/gstsunaudiomixertrack.h: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: * sys/sunaudio/gstsunaudiosrc.c: * sys/sunaudio/gstsunaudiosrc.h: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/sunaudio.c: * tests/check/meson.build: sys: remove sunaudio plugin Even though hooked up to the build system, it's clear that no one has ever built or used this with GStreamer 1.x. It wants to link against libgstinterfaces, which no longer exists. And uses 0.10-style raw audio caps. And the last meaningful change was done in 2009. Let's just remove it. 2017-06-23 19:35:28 +0100 Tim-Philipp Müller * sys/meson.build: * sys/oss4/meson.build: meson: build oss4 plugin https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-23 19:23:52 +0100 Tim-Philipp Müller * sys/meson.build: * sys/oss/meson.build: meson: build oss plugin https://bugzilla.gnome.org/show_bug.cgi?id=784134 2017-06-22 11:38:56 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Actually use the receive lock when receiving, not the send lock 2017-06-22 01:01:40 +1000 Jan Schmidt * tests/examples/qt/qmlsink/CMakeLists.txt: qmlsink example: Add CMakeLists.txt Make it possible to build using cmake instead of qmake 2017-06-22 01:01:40 +1000 Jan Schmidt * ext/qt/qtitem.cc: qt: Remove misleading reference to GTK in qtitem.cc 2017-06-15 11:46:54 -0400 Thibault Saunier * ext/flac/gstflactag.c: flactag: Fix warning with the newly added GstStateChange values https://bugzilla.gnome.org/show_bug.cgi?id=783798 2017-06-15 19:09:26 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: rtspsrc: do not checksum the stream id https://bugzilla.gnome.org/show_bug.cgi?id=783307 2017-06-15 23:31:24 +0100 Tim-Philipp Müller * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: qtmux: add support for muxing PNG Demuxer already supported it. 2017-06-15 10:40:51 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Use a mutex for protecting against concurrent send/receives We currently send data to the RTSP connection from multiple threads: whenever a command is to be handled and whenever RTCP is generated. This can cause data corruption or worse if both happen at the same time. As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive() calls with a mutex. While this means that we hold a mutex during the IO operation, this is not actually a problem as the IO operation can be interrupted (gst_rtsp_connection_flush()) at any time and is blocking by itself anyway. 2017-06-15 11:50:44 +0300 Sebastian Dröge * gst/isomp4/atoms.c: qtmux: Un-merge the last two stsc entries after serializing The last entry will most likely get new samples added to it in "robust" muxing mode, changing the samples_per_chunk and thus making it wrong to keep the last two entries merged. It will run into an assertion later when adding a new sample to the chunk. Thanks to gdiener@cardinalpeak.com for the analysis of the bug and proposal for a solution. 2017-06-14 00:09:25 +0300 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Actually clip to upstream size instead of size of the data chunk There might be other chunks after the data chunk, so clipping the chunk size with the data size can lead to a negative number and all following calculations go wrong and cause crashes or worse. This was introduced in 3ac119bbe2c360e28c087cf3852ea769d611b120. https://bugzilla.gnome.org/show_bug.cgi?id=783760 2017-06-13 17:40:19 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmux: Drop allocation queries They can cause us to deadlock, while we're waiting for a new frame and upstream is waiting for the allocation query to be answered before sending a frame https://bugzilla.gnome.org/show_bug.cgi?id=783753 2017-06-01 02:03:27 +0200 Mathieu Duponchelle * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: uniquify stream ids https://bugzilla.gnome.org/show_bug.cgi?id=783307 2017-06-07 12:47:59 -0400 Thibault Saunier * tests/check/meson.build: meson: Do not use path separator in test names Avoiding warnings like: WARNING: Target "elements/audioamplify" has a path separator in its name. 2017-06-06 11:29:29 -0400 Nicolas Dufresne * tests/examples/v4l2/camctrl.c: Fix v4l2 example 2017-06-05 16:55:13 +0900 Jimmy Ohn * gst/isomp4/qtdemux.c: qtdemux: remove not needed code remove not needed code about res variable. https://bugzilla.gnome.org/show_bug.cgi?id=783422 2017-06-02 14:01:17 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Make sure min_buffers is valid When upstream does no use the v4l2videoenc pool, we need to activate that internal pool. Though, we relied the driver to provide a minimum required buffer, which Qualcomm Venus driver don't currently provide. https://bugzilla.gnome.org/show_bug.cgi?id=783361 2017-06-02 11:30:15 +0100 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: rtph265depay: fix caps leak 2017-05-26 16:30:06 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: simplify buffer accumulation control flow There is no difference between pushing out a buffer directly with gst_rtp_base_depayload_push() and returning it from the process function. The base class will just call _depayload_push() on the returned buffer as well. So instead of marshalling buffers through three layers and back, just push them from one place in handle_nal() and always return NULL from the process vfunc. This simplifies the code a little. Also rename _push_fragmentation_unit() to _finish_fragmentation_unit() for clarity. Push sounds like it means being pushed out, whereas it might just be pushed into an adapter. This change has the side-effect that multiple NALs in a single STAP (such as SPS/PPS) may no longer be pushed out as a single buffer if we output NALs in byte-stream format (i.e. not aggregate AUs), but that shouldn't really make any difference to anyone. 2017-05-30 22:23:10 +0200 Juan Navarro * gst/rtpmanager/rtpsession.c: rtpsession: print value of unknown RTCP Payload Type This adds printing the actual value of any unknown RTCP PT to the already existing WARNING log message. https://bugzilla.gnome.org/show_bug.cgi?id=783248 2017-05-26 17:52:19 +0200 Edward Hervey * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Don't leak VideoCodecState CID #1409852 2017-05-26 17:48:01 +0200 Edward Hervey * ext/dv/gstdvdemux.c: dvdemux: Remove un-needed variable check if pad wasn't present by now everything would have broken before CID #1409854 2017-05-25 15:26:37 +0200 Piotr Drąg * po/POTFILES.in: po: update POTFILES https://bugzilla.gnome.org/show_bug.cgi?id=783093 2017-05-25 10:09:04 +0800 Haihua Hu * ext/qt/qtwindow.cc: glframebuffer: check frame buffer status need use specific fbo target https://bugzilla.gnome.org/show_bug.cgi?id=783065 2017-05-24 14:19:27 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videoenc.c: v4l2videoenc: Remove unused function 2017-05-21 15:29:11 +0200 Nicolas Dufresne * sys/v4l2/ext/types-compat.h: v4l2: Don't redefine __bitwise if already set https://bugzilla.gnome.org/show_bug.cgi?id=728438 2017-05-23 14:40:56 -0400 Ayaka * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2h264enc.c: * sys/v4l2/gstv4l2h264enc.h: * sys/v4l2/gstv4l2videoenc.c: * sys/v4l2/gstv4l2videoenc.h: * sys/v4l2/meson.build: v4l2: Add Video Encoder support This implements H264 encoding support using generic V4L2 interface. It is reported to work with Samsung MFC driver, IXM.6 CODA driver and Qualcomm mainline Venus driver. Other platform should be supported as none of this work is platform specific. The implementation consist of a GstV4l2VideoEnc base class, which implements the core streaming functionality. This base class is implemented by GstV4l2H264Enc class that implements the caps negotiation specific to H264 profiles and level. This implementation supports hardware with multiple H264 encoder. Though, to make it simplier to use, the first discovered H264 encoder will be named v4l2h264enc. Other encoder found during discovery will have a unique name like v4l2video0h264enc. This work is the combined work of multiple developpers in the last 3 years. Thanks to all of the contributors: Ayaka Frédéric Sureau Jean-Michel Hautbois Nicolas Dufresne Pablo Anton https://bugzilla.gnome.org/show_bug.cgi?id=728438 2017-05-23 14:36:37 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Remove unused forward declaration https://bugzilla.gnome.org/show_bug.cgi?id=728438 2015-10-05 16:30:46 +0100 Ayaka * sys/v4l2/gstv4l2bufferpool.c: v4l2pool: Fix wrong error message https://bugzilla.gnome.org/show_bug.cgi?id=728438 2015-10-05 16:20:19 +0100 Ayaka * sys/v4l2/gstv4l2object.c: v4l2: increase pre-allocated encoded buffer size As of today, the MFC encoder often need to exceed that 1 MB size for encoded buffer we fixed earlier for decoding. https://bugzilla.gnome.org/show_bug.cgi?id=728438 2017-05-24 16:32:30 +0100 Tim-Philipp Müller * gst/rtp/gstrtpopusdepay.c: rtpopusdepay: minor perf improvements Use the ::process_rtp_packet() vfunc to avoid mapping the RTP buffer twice. gst_rtp_buffer_get_payload_buffer() returns a new sub-buffer which will always be writable, so no need to make it writable. 2017-05-24 16:14:54 +0100 Tim-Philipp Müller * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: rtp: opus: use existing utility funcs for copying/dropping metas We had our own copies of those while the code was in -bad, but now we can use the existing utility functions instead of re-implementing them. 2017-05-24 12:57:10 +0100 Tim-Philipp Müller * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtputils.c: * gst/rtp/gstrtputils.h: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9depay.c: * gst/rtp/gstrtpvp9pay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: cache meta tag quarks and add more utility functions for metas Every g_quark_from_static_string() is a hash table lookup serialised on the global quark lock in GLib. Let's just look up the two quarks we need once and cache them locally for future use. While we're at it, add new utility functions for the two most commonly used tags (audio + video). Make first argument a gpointer so we don't have to cast and make the code ugly. These are used for logging purposes only anyway. 2017-05-24 11:33:05 +0530 vijay * gst/audioparsers/gstaacparse.c: aacparse : Fix, Caps were not set while reusing aacparse While reusing aacparse caps were not set.This fix enables aacparse to reuse in same pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=783027 2017-05-21 17:45:34 +0100 Tim-Philipp Müller * Makefile.am: * config.h.meson: * meson.build: meson: don't need config.h.meson any longer 2017-05-21 15:26:12 +0200 Carlos Rafael Giani * ext/qt/gstqsgtexture.cc: * ext/qt/gstqsgtexture.h: qmlglsink: Add dummy texture that is shown as placeholder for NULL buffers https://bugzilla.gnome.org/show_bug.cgi?id=782917 2017-04-24 16:55:22 +0300 George Kiagiadakis * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2send: use non-blocking I/O and a configurable network operations timeout This allows timing out on network errors much earlier (currently it takes ~15min to timeout) and we can still unlock and change state in the meantime. https://bugzilla.gnome.org/show_bug.cgi?id=571722 2017-05-21 10:37:19 +0100 Tim-Philipp Müller * ext/taglib/meson.build: * meson.build: meson: make C++ compiler optional It's only needed for the taglib plugin which is optional. 2017-05-21 10:33:43 +0100 Tim-Philipp Müller * gst/multifile/multifile.vproj: multifile: remove some cruft 2017-05-20 17:09:52 +0200 Josep Torra * sys/osxaudio/gstosxcoreaudio.c: osxaudio: fixes playback of mono streams with no channel-mask field in caps Fixes a negotiation error seen when trying to playback of a .MOV file with a mono AAC audio stream decoded by avcdec_aac that doesn't set channel-mask field but sink was requiring channel-mask=0x3. 2015-09-06 20:49:59 +0530 Ravi Kiran K N * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Push tag event to both pads Tags are pushed to "videosrcpad"/"audiosrcpad" in gst_dvdemux_add_pad() method, however they will be NULL in this method, hence tags are not pushed. Instead, send tag event to "pad" created gst_dvdemux_add_pad(). Signal no-more-pads when both pads are created https://bugzilla.gnome.org/show_bug.cgi?id=743657 2017-05-20 14:53:42 +0100 Tim-Philipp Müller * meson.build: * meson_options.txt: * tests/check/elements/autodetect.c: meson: add options to set package name and origin https://bugzilla.gnome.org/show_bug.cgi?id=782172 2017-05-20 11:40:33 +0100 Luis de Bethencourt * gst/multifile/gstmultifilesink.c: multifilesink: fix property name in example pipeline Since the move from CVS the property name of the documentation example has been filename instead of location. Users trying the gst-launch command as is will get: no property name "filename" in element Fixing it. 2017-05-20 11:13:40 +0200 Josep Torra * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideo: fix macOS 10.12 deprecation warnings Add #defines to allow older versions of macOS to use the new constant names. 2017-05-13 09:05:57 +0200 Edward Hervey * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: isomp4: Safely ignore [skip] atoms Instead of warning about them 2017-05-18 15:23:14 +0300 Simon Himmelbauer * ext/qt/gstqtglutility.cc: qt: Use GST_GL_HAVE_PLATFORM_CGL instead of GST_GL_HAVE_PLATFORM_COCOA The latter is not used/available anymore since years. Also fix a typo in the include path for the Cocoa GL display header. 2017-05-18 15:10:30 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Make session sharing thread-safe on our side https://bugzilla.gnome.org/show_bug.cgi?id=780140 2017-05-18 10:53:48 +0100 Tim-Philipp Müller * gst/audiofx/gststereo.c: stereo: fix typo in plugin description 2017-05-18 10:43:19 +0100 Tim-Philipp Müller * ext/shout2/gstshout2.c: * gst/audiofx/gstscaletempoplugin.c: Fix up package name and origin in some plugins 2017-05-15 19:51:47 +0300 Sebastian Dröge * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: gst: Clear floating flag in constructor of all GstObject subclasses that are not owned by any parent https://bugzilla.gnome.org/show_bug.cgi?id=743062 2017-05-15 14:22:34 +0300 Sebastian Dröge * ext/raw1394/gst1394clock.c: 1394: Sink the clock reference in the constructor This is now needed as GstClock does not do that internally anymore, because that broke bindings. https://bugzilla.gnome.org/show_bug.cgi?id=743062 2017-05-17 10:58:05 +0800 Haihua Hu * ext/qt/gstqtglutility.cc: qml: Add EGL platform support for x11 backend Add support for EGL platform when x11 is available. This can work e.g. on imx6 platform. https://bugzilla.gnome.org/show_bug.cgi?id=782718 2017-04-28 23:05:35 -0400 Nicolas Dufresne * ext/pulse/pulseutil.h: pulse: Accept MPEG 1 layer 3 version 2.5 https://bugzilla.gnome.org/show_bug.cgi?id=781929 2017-05-16 13:50:16 -0400 Nicolas Dufresne * configure.ac: * ext/aalib/Makefile.am: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/flac/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/jack/Makefile.am: * ext/jpeg/Makefile.am: * ext/libcaca/Makefile.am: * ext/libpng/Makefile.am: * ext/pulse/Makefile.am: * ext/raw1394/Makefile.am: * ext/shout2/Makefile.am: * ext/soup/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/Makefile.am: * ext/vpx/Makefile.am: * ext/wavpack/Makefile.am: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/audioparsers/Makefile.am: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debugutils/Makefile.am: * gst/deinterlace/Makefile.am: * gst/dtmf/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flv/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/imagefreeze/Makefile.am: * gst/interleave/Makefile.am: * gst/isomp4/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/rtsp/Makefile.am: * gst/shapewipe/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * gst/y4m/Makefile.am: * sys/directsound/Makefile.am: * sys/oss/Makefile.am: * sys/oss4/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxvideo/Makefile.am: * sys/sunaudio/Makefile.am: * sys/v4l2/Makefile.am: * sys/waveform/Makefile.am: * sys/ximage/Makefile.am: Remove plugin specific static build option Static and dynamic plugins now have the same interface. The standard --enable-static/--enable-shared toggle are sufficient. 2017-05-16 14:07:56 -0400 Nicolas Dufresne * ext/twolame/Makefile.am: Remove plugin specific static build option Static and dynamic plugins now have the same interface. The standard --enable-static/--enable-shared toggle are sufficient. 2017-05-16 14:07:56 -0400 Nicolas Dufresne * ext/lame/Makefile.am: Remove plugin specific static build option Static and dynamic plugins now have the same interface. The standard --enable-static/--enable-shared toggle are sufficient. 2017-05-16 14:07:56 -0400 Nicolas Dufresne * ext/mpg123/Makefile.am: Remove plugin specific static build option Static and dynamic plugins now have the same interface. The standard --enable-static/--enable-shared toggle are sufficient. 2017-05-16 14:05:52 -0400 Nicolas Dufresne * ext/gtk/Makefile.am: Remove plugin specific static build option Static and dynamic plugins now have the same interface. The standard --enable-static/--enable-shared toggle are sufficient. 2017-05-16 14:05:52 -0400 Nicolas Dufresne * ext/qt/Makefile.am: Remove plugin specific static build option Static and dynamic plugins now have the same interface. The standard --enable-static/--enable-shared toggle are sufficient. 2017-05-12 17:53:57 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Add alignment-threshold argument If a non-reference stream is behind the reference stream by an amount of time smaller than the alignment threshold (in nsec), it counts as being after it. https://bugzilla.gnome.org/show_bug.cgi?id=782563 2017-05-16 12:56:15 +0300 Vivia Nikolaidou * gst/isomp4/gstqtmux.c: qtmux: Do not check timecode data for mp4 container Timecode trak is only supported for mov right now, not for mp4. That code would otherwise create an invalid trak if the muxed video contained timecode metadata. https://bugzilla.gnome.org/show_bug.cgi?id=782684 2017-05-11 20:01:15 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: When accepting renegotiation, just return TRUE and change nothing We only accept new caps if they are basically the same. We don't want to reset anything as if the caps are new, otherwise various state could get out of sync with the current run. 2017-05-11 19:21:22 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: In prefill mode, only pad buffers with > 0 sized memories as needed Adding a 0-byte memory has not much effect. Also add some debug output. 2017-05-10 15:58:41 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Lateness is in QT timescale, diff in GstClockTime Print the right one in debug output to get meaningful numbers. 2017-05-10 14:31:40 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Error out if a gap edit list has to be written in prefill mode We don't have any space reserved for this in the moov and the pre-finalized moov would have broken A/V synchronization. Error out here now 2017-05-10 11:42:09 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Calculate with reserved moov size instead of last moov size We have some padding added after the initial moov, so a bigger updated moov can be handled to some degree and is expected. Previously we just ignored the padding and errored out in cases when the padding would've just been enough. 2017-05-10 11:12:23 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Error out directly if sending filler data results in a flow error CID 1405994 2017-05-09 16:02:43 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: In prefill mode, handle the case when only the first chunk was ever used 2017-05-09 09:47:10 -0400 Nicolas Dufresne * ext/qt/gstplugin.cc: qmlgl: Make the plugin name match the pugin file name 2017-03-16 15:12:07 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Use a in-memory cookie jar by default in sessions we created This ensures that cookies are stored and used as set by the server, and shared with other souphttpsrc that use the same SoupSession. https://bugzilla.gnome.org/show_bug.cgi?id=780140 2017-03-16 13:58:41 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Implement soup session sharing souphttpsrc now shares its SoupSession with other elements in the pipeline via GstContext if possible (session-wide settings are all the defaults), or if the context was forced by the application. This allows multiple souphttpsrcs to reuse connections, cookies, etc. https://bugzilla.gnome.org/show_bug.cgi?id=780140 2017-03-09 10:15:34 +0200 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Add new prefill recording mode This sets up a moov with the correct sample positions beforehand and only works with constant framerate, I-frame only streams. Currently only support for ProRes and raw audio is implemented but adding new codecs is just a matter of defining appropriate maximum frame sizes. https://bugzilla.gnome.org/show_bug.cgi?id=781447 2017-03-29 14:01:25 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Error out on discontinuities/gaps when muxing raw audio When muxing raw audio, we have no way of storing timestamps but are just storing a continuous stream of audio samples. If the difference between the expected and the real timestamp becomes to big, we should error out instead of silently creating files with wrong A/V sync. https://bugzilla.gnome.org/show_bug.cgi?id=780679 2017-05-09 11:41:25 +0200 Sebastian Dröge * ext/vpx/gstvpxdec.c: vpxdec: Set fb->priv to NULL after freeing just in case https://bugzilla.gnome.org/show_bug.cgi?id=782359 2017-05-08 15:22:00 +0000 Dustin Spicuzza * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: directsoundsink: Use GstClock API instead of Sleep() for waiting It's more accurate and allows cancellation. https://bugzilla.gnome.org/show_bug.cgi?id=773681 2017-05-08 15:05:45 +0000 Tim-Philipp Müller * ext/vpx/gstvp9dec.c: vpx: fix build against older libvpx versions Such as 1.3.0 as on raspbian. 2017-05-03 23:23:10 +0530 Nirbheek Chauhan * sys/directsound/gstdirectsoundsink.c: directsoundsink: Fix corner case causing large CPU usage We were unnecessarily looping/goto-ing repeatedly when we had exactly the amount of data as the free space, and also when the free space was too small. This, as it turns out, is a very common scenario with Directsound on Windows. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=773681 We have to do polling here because the event notification API that Directsound exposes cannot be used with live playback since all events must be registered in advance with the capture buffer, you cannot add/remove them once playback has begun. Directsoundsrc had the same problem. See also: https://bugzilla.gnome.org/show_bug.cgi?id=781249 2017-05-03 23:31:00 +0530 Nirbheek Chauhan * sys/directsound/gstdirectsoundsink.c: directsoundsink: Clean up some debug logging Don't need to print the function name, gstreamer does it for you. https://bugzilla.gnome.org/show_bug.cgi?id=773681 2017-05-06 22:30:20 +0100 Tim-Philipp Müller * gst/matroska/matroska-ids.h: matroskademux: improve index memory usage Re-arrange order of index entry struct members to avoid padding bytes in the middle of the struct, thus potentially reducing the overall size of the struct and reducing memory used by the index. On Linux x86_64 the size goes down from 32 bytes to 24 bytes for each index entry. 2017-05-04 18:59:14 +0300 Sebastian Dröge * configure.ac: * meson.build: Back to development === release 1.12.0 === 2017-05-04 15:38:34 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * meson.build: Release 1.12.0 2017-05-04 15:07:27 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/fur.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2017-05-04 13:47:20 +0300 Sebastian Dröge * po/el.po: po: Update translations 2017-05-02 10:32:30 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Fix crash on mss stream caused by invalid stsd entry access Since mss has no moov, default stsd entry should be created with media-caps. https://bugzilla.gnome.org/show_bug.cgi?id=782042 === release 1.11.91 === 2017-04-27 17:29:58 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * meson.build: Release 1.11.91 2017-04-27 15:58:47 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/fur.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2017-04-27 15:28:02 +0300 Sebastian Dröge * po/LINGUAS: * po/el.po: * po/fur.po: po: Update translations 2017-04-27 12:56:27 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Don't crash in debug output if stream==NULL That case is correctly handled below but not in the debug output. https://bugzilla.gnome.org/show_bug.cgi?id=781270 2017-04-25 17:11:27 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Don't perform seeks with inconsistent seek values If gst_segment_do_seek() fails, we shouldn't try seeking on that resulting segment but just error out. Crashes further down the line otherwise. 2017-04-24 20:27:49 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 60aeef6 to 48a5d85 2017-04-24 17:31:04 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: add test for rtph264depay avc/byte-stream output Make sure avc output doesn't contain SPS/PPS inline, but byte-stream output does. 2017-04-24 17:29:37 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: don't insert SPS/PPS inline for AVC output SPS/PPS are in the caps in this case and shouldn't be in the stream data. 2017-04-21 19:09:14 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Chain up to the parent class' provide_clock() implementation If no clock was provided directly by rtspsrc. This behaviour was removed by f8013487c91a6ffc552a4b25aa1a70f0bd5377f8 and results in rtspsrc not providing the system clock via the rtpjitterbuffer. As a result, if another element like an audio sink, provides a clock, the pipeline would select that (when going to PAUSED/PLAYING again later). Audio clocks usually don't progress in PAUSED, and thus our live source won't be able to use the clock to produce data, making the sink never preroll and everything is stuck. 2017-04-20 11:22:15 +0200 Jürgen Sachs * gst/isomp4/qtdemux.c: qtdemux: reset sample_description_id to default Fixes stream where sample_description_id is specified in the tfhd https://bugzilla.gnome.org/show_bug.cgi?id=778337 2017-04-20 13:16:24 +0100 Sebastian Dröge * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't use an explicit name for requesting audio pads ... unless the muxer uses the same audio pad template name as splitmuxsink. We can't request a pad called "audio_0" on a muxer that wants pads to be "sink_%d". 2017-02-23 09:31:36 +0900 ChangBok Chae * gst/flv/gstflvdemux.c: flvdemux: remove duplicated segment initialization It's also done in gst_flv_demux_cleanup(). https://bugzilla.gnome.org/show_bug.cgi?id=779106 2017-04-20 20:17:35 +1000 Xavier Claessens * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Correctly catch FLUSH events in probes https://bugzilla.gnome.org/show_bug.cgi?id=767498 2017-04-19 12:28:12 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: Revert "rtpbin: pipeline gets an EOS when any rtpsources byes" This reverts commit eeea2a7fe88a17b15318d5b6ae6e190b2f777030. It breaks EOS in some sender pipelines, see https://bugzilla.gnome.org/show_bug.cgi?id=773218#c20 2017-04-14 17:01:49 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Reset adapter in more discontinuity cases In push mode we process as much as possible in the adapter. When we receive a DISCONT buffer which we can't match to an actual sample (based on the existing sample table) and there is still data remaining in the incoming adapter,there is one of two cases happening: 1) We are doing reverse playback, in which case we should flush out all pending data 2) We have leftover data from the previous incoming buffer... which we can't do anything about. For the second case, make sure we flush out the remaining data so that we can start parsing again from scratch. https://bugzilla.gnome.org/show_bug.cgi?id=781319 2017-04-14 10:56:41 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Use GST_ELEMENT_ERROR_WITH_DETAILS Allows the application to know the exact status code that was returned by the server in a programmatic fashion. https://bugzilla.gnome.org/show_bug.cgi?id=781304 2017-04-16 18:47:56 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Fix leak on QtDemuxStreamStsdEntry Fix unit test failure https://bugzilla.gnome.org/show_bug.cgi?id=781362 2017-04-14 13:38:53 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Fix timescale of timecode tracks They should have ideally the same timescale of the video track, which we can't guarantee here as in theory timecode configuration and video framerate could be different. However we should set a correct timescale based on the framerate given in the timecode configuration, and not just use the framerate numerator. 2017-04-13 13:25:06 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Properly reset demuxer when all streams are EOS Make sure offset and neededbytes are properly resetted when all streams are EOS in push-mode. Avoids cases when some data might still be pushed by upstream (because it didn't yet see the resulting GST_FLOW_EOS yet) and qtdemux gets completely lost. https://bugzilla.gnome.org/show_bug.cgi?id=781266 2017-04-13 08:00:30 +0200 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Make more usage of error macro And make sure we actually use the provided soup_msg argument in the macro 2017-03-08 15:01:13 -0300 Thibault Saunier * gst/audiofx/gststereo.c: docs: Port all docstring to gtk-doc markdown 2017-03-08 15:01:13 -0300 Thibault Saunier * ext/gtk/gstgtkbasesink.c: * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtksink.c: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: docs: Port all docstring to gtk-doc markdown 2017-04-12 18:46:53 +0530 Nirbheek Chauhan * ext/meson.build: meson: Print message when disabling taglib on MSVC 2017-04-12 13:26:59 +0200 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Don't forget to update pad->last_buf buf is the current pad->last_buf value. If ever it gets copied/unreffed, we need to make sure to write back the new pointer to the last_buf variable. Fixes using wrong pointer values in the case of decrasing DTS value 2017-04-12 11:33:05 +0200 Edward Hervey * tests/check/elements/.gitignore: tests: Add vp9enc to gitignore 2017-04-11 13:41:48 +0200 Jürgen Sachs * gst/isomp4/qtdemux.c: qtdemux: fix: sample description index override in tfhd not evaluated https://bugzilla.gnome.org/show_bug.cgi?id=778337 2017-04-12 11:03:24 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Add out-of-bound check Make sure we don't read invalid memory 2016-04-27 12:17:37 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: move parsing of tkhd out of stsd entry loop It needs only to be read once. 2016-04-07 12:23:35 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: check for a different stsd entry before pushing a sample Before pushing a sample, check if there was a change in the current stsd entry. This patch also assumes that the first stsd entry is used as default for the first sample. It might cause an uneeded caps renegotiation when this isn't the case. 2016-04-06 12:55:18 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: parse all stsd entries stsd can have multiple format entries, parse them all. This is required to play DVB DASH profile that uses multiple entries to identify the different available bitrates/options on dash streams The stream format-specific data is not stored into QtDemuxStreamStsdEntry 2016-04-05 14:34:00 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: rework stsd sample entries access Instead of using the stsd as a base pointer, use the actual stsd entry as the stsd can have multiple entries. This is rarely used for file playback but is a possible profile with in DVB DASH specs. This still doesn't support stsd with multiple entries but makes it easier to do so. 2016-04-05 18:00:10 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: get stsd child by index instead of type There might be multiple children with the same type 2017-04-07 16:33:18 +0300 George Kiagiadakis * tests/check/elements/rtprtx.c: tests/check/rtprtx: add checks for rtprtxqueue's max-size-{time,packets} properties https://bugzilla.gnome.org/show_bug.cgi?id=780867 2017-04-04 17:33:31 +0300 George Kiagiadakis * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtprtxqueue: implement handling of the max-size-time property https://bugzilla.gnome.org/show_bug.cgi?id=780867 2017-04-10 23:49:06 +0100 Tim-Philipp Müller * autogen.sh: * common: Automatic update of common submodule From 39ac2f5 to 60aeef6 2017-04-10 08:56:00 +0000 Todor Tomov * sys/v4l2/gstv4l2bufferpool.c: v4l2object: Copy timestamp when importing buffers This is needed for V4L2_OUTPUT interface, and is harmless of V4L2_CAPTURE interfaces. This will fix timestamp in cases like: v4l2src io-mode=dmabuf ! v4l2videoNenc output-io-mode=dmabuf-import ! ... Same apply for userptr. https://bugzilla.gnome.org/show_bug.cgi?id=781119 2017-04-10 15:55:30 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Fix last_dts tracking for raw audio and similar formats Accumulate the durations directly and don't scale yet another time by the number of samples. 2017-04-07 10:48:50 +0100 Vincent Penquerc'h * tests/check/elements/splitmux.c: tests: fix leak in splitmux test https://bugzilla.gnome.org/show_bug.cgi?id=781025 2017-04-07 15:29:43 +0800 Lyon Wang * gst/audiofx/gstscaletempo.c: scaletempo: Scale GAP event timestamp and duration like for buffers https://bugzilla.gnome.org/show_bug.cgi?id=781008 2017-02-17 10:01:08 -0300 Thibault Saunier * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: v4l2dec: Fix race when going from PAUSED to READY Running `gst-validate-launcher -t validate.file.playback.change_state_intensive.vorbis_vp8_1_webm` on odroid XU4 (s5p-mfc v4l2 driver) often leads to: ERROR:../subprojects/gst-plugins-good/sys/v4l2/gstv4l2videodec.c:215:gst_v4l2_video_dec_stop: assertion failed: (g_atomic_int_get (&self->processing) == FALSE) This happens when the following race happens: - T0: Main thread - T1: Upstream streaming thread - T2. v4l2dec processing thread) [The decoder is in PAUSED state] T0. The validate scenario runs `Executing (36/40) set-state: state=null repeat=40` T1- The decoder handles a frame T2- A decoded frame is push downstream T2- Downstream returns FLUSHING as it is already flushing changing state T2- The decoder stops its processing thread and sets `->processing = FALSE` T1- The decoder handles another frame T1- `->process` is FALSE so the decoder restarts its streaming thread T0- In v4l2dec-> stop the processing thread is stopped NOTE: At this point the processing thread loop never started. T0- assertion failed: (g_atomic_int_get (&self->processing) == FALSE) Here I am removing the whole ->processing logic to base it all on the GstTask state to avoid duplicating the knowledge. https://bugzilla.gnome.org/show_bug.cgi?id=778830 === release 1.11.90 === 2017-04-07 16:31:56 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * meson.build: Release 1.11.90 2017-04-07 15:18:11 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2017-04-07 15:06:30 +0300 Sebastian Dröge * po/el.po: po: Update translations 2017-04-06 12:01:00 +0200 Edward Hervey * gst/audioparsers/gstaacparse.c: aacparse: streamline and improve AudioSpecificConfig parsing AudioSpecifigConfig is used in a variety of AAC streams but was being parsed differently. Instead, make everyone use the same parsing. * Remove unused 'bits' field (it was always set to 0 if present) * Add proper GAConfig parsing (to know the number of samples per frame if present). Fixes wrong rate/channels configuration in streams coming from qtdemux https://bugzilla.gnome.org/show_bug.cgi?id=780966 2017-04-05 09:46:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Fix 32bit only printf format The previous patch was using %llu for 64bits printf, which is 32bit specific. We also trace the latency in time human readable form now. 2016-03-16 16:22:48 +0100 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: set streamparm for outputs that support it Without a specified framerate from the sink, the decoder frame interval should be set using the framerate of the encoded video stream. Therefore, the v4l2object should be able to change the framerate on the output if the V4L2 device accepts it. This is also necessary for mem2mem encoders so that their bitrate calculation code may work correctly and they may report the correct frame duration on the capture queue. https://bugzilla.gnome.org/show_bug.cgi?id=779466 2016-03-16 16:24:55 +0100 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2videodec: only set latency if the frame duration is valid If the duration of the v4l2object is GST_CLOCK_TIME_NONE, because the sink did not specify a framerate in the caps and the driver accepts the framerate, the decoder element uses GST_CLOCK_TIME_NONE to calculate and set the element latency. While this is a bug of the capture driver, the decoder element should not use the invalid duration to calculate a latency, but print a warning instead. https://bugzilla.gnome.org/show_bug.cgi?id=779466 2016-11-23 12:17:55 -0500 Olivier Crête * sys/v4l2/gstv4l2sink.c: v4l2sink: Block in preroll_wait on unlock The correct behaviour of anything stuck in the ->render() function between ->unlock() and ->unlock_stop() is to call gst_base_sink_wait_preroll() and only return an error if this returns an error, otherwise, it must continue where it left off! https://bugzilla.gnome.org/show_bug.cgi?id=774945 2017-04-05 15:55:20 +1000 Jan Schmidt * ext/vpx/gstvp9dec.c: vp9dec: Add warnings for unsupported frame formats At least output an element warning on the bus when we encounter a frame format GStreamer doesn't currently support. 2017-04-04 17:55:13 +0200 Edward Hervey * gst/audioparsers/gstaacparse.c: aacparse: Handle Parametric Stereo with HE-AAC(v2) According to ISO/IEC:14496-2:2009 , in the case of HE-AACv2 (audioObjecType 29) parametric stereo is used (a single mono track is used and then transformations are applied to it to provide a stereo output). We therefore report two channels in the case where there is one reported in the audioChannelConfiguration. Fixes the various issues where a demuxer would report two channels, but then the parser would say there's only one channel, and then the decoder would output two channels. 2017-04-04 15:22:25 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Simplify buffer refcounting in add_buffer() and remove unneeded NULL checks 2017-04-04 15:08:33 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Select the best pad based on the cached last_buf if any last_buf is the one we're going to write next, not buf. As such we should check timestamps against that one if there is one to select the earliest pad. Also remember the currently selected pad in the very beginning when storing the first last_buf. This both solves some edge cases where not the correct next pad was selected corresponding to the target interleave. 2017-04-04 15:07:40 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Error out immediately if a timecode is to be written but downstream return not-OK 2017-04-03 11:34:49 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Update variables before early exit This is an update of d78d5896272d78df41e696fac929e7dfb3bb3dfa We still exit as early as possible in case of non-ok/non-unlinked combined flow, but we first make sure that we update the internal position variables. This ensures that if upstreams "ignores" the flow return (and carries on pushing), we don't end up processing data with completely bogus variables/positions. 2017-03-24 00:11:13 +1300 Douglas Bagnall * gst/interleave/interleave.c: * gst/interleave/interleave.h: interleave: avoid using uninitialised ordering_map If self->channel_positions == NULL (which seems unlikely), self->default_channels_ordering_map will be used unintialised. We avoid that by keeping track of the channel_mask, which is set when the ordering map is initialised. https://bugzilla.gnome.org/show_bug.cgi?id=780331 2017-03-23 23:56:31 +1300 Douglas Bagnall * gst/interleave/interleave.c: interleave: don't overflow channel map with >64 channels When there are more than 64 channels, we don't want to exceed the bounds of the ordering_map buffer, and in these cases we don't want to rempa at all. Here we avoid doing that. https://bugzilla.gnome.org/show_bug.cgi?id=780331 2017-03-28 14:23:16 -0300 Thibault Saunier * tests/check/meson.build: meson: Use get_pkgconfig_variable instead of calling pkg-config ourself It is avalaible in meson 0.36 which is now are requirement 2017-03-28 14:22:41 -0300 Thibault Saunier * pkgconfig/gstreamer-plugins-good.pc.in: * pkgconfig/meson.build: pkgconfig: Do not ever build an installed .pc file 2017-03-28 11:15:53 -0300 Thibault Saunier * tests/check/meson.build: meson: test: Fix environment object usage 2017-03-28 11:14:47 -0300 Thibault Saunier * meson.build: * pkgconfig/gstreamer-plugins-good.pc.in: * pkgconfig/meson.build: pkgconfig: Generate the pkg-config with meson too 2017-03-27 21:52:00 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: In gap mode, consider the mdat offset when calculating the remaining mdat size The mdat generally does not start at offset 0, we have to include the size of the moof and whatever else was in front of the mdat. 2017-03-27 11:43:31 +0300 Sebastian Dröge * gst/isomp4/atomsrecovery.c: atomsrecovery: Error out when fseek() fails instead of silently ignoring CID 1403262 2017-03-23 22:13:05 +0100 Carlos Rafael Giani * sys/v4l2/gstv4l2object.c: v4l2object: Also add videometa if there is padding to the right and bottom https://bugzilla.gnome.org/show_bug.cgi?id=780478 2017-03-21 12:54:27 +0200 George Kiagiadakis * gst/rtpmanager/gstrtpmux.c: rtpmux: fix output segment and buffer DTS to correspond to the flattened PTS https://bugzilla.gnome.org/show_bug.cgi?id=780347 2017-03-23 17:53:19 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Remove some unused variables 2017-03-23 15:01:16 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Remove a couple of unneeded levels of indentation 2017-03-22 18:18:40 +0000 Enrique Ocaña González * gst/isomp4/qtdemux.c: qtdemux: distinguish TFDT with value 0 from no TFDT at all TFDTs with time 0 are being ignored since commit 1fc3d42f. They're mistaken with the case of not having TFDT, but those two cases must be distinguished in some way. This patch passes an extra boolean flag when the TFDT is present. This is now the condition being evaluated, instead of checking for 0 time. https://bugzilla.gnome.org/show_bug.cgi?id=780410 2017-03-22 19:15:09 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Reset current chunk after writing out timecode If we have multiple tracks with timecodes, or it's not the first track that has timecodes, or not the first buffer, we already started a chunk for media data. We now need to "close" that chunk because we wrote data for the timecode track and a new chunk has to be started for the original track the next time it has data. 2017-03-22 18:52:51 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Do timecode handling per track, not per muxer instance There could be multiple video tracks with timecodes. 2017-03-22 00:38:51 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: qtdemux: matroskademux: Ignore repeated seek events Similar to what was done in adaptivedemux, ignore seek events we've already handled - such as when they are received on every srcpad of files with lots of streams. 2017-03-21 14:55:32 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: dashdemux: Update mdatleft from overall mdatsize and offset when observing a gap Otherwise mdatleft will have a value calculated from the initial mdatsize minus the parts of the stream that we saw, which is not including all the parts of the stream that might've been skipped. 2017-03-20 17:03:32 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * gst/audioparsers/gstmpegaudioparse.c: docs: update two references to the removed 'mad' plugin https://bugzilla.gnome.org/show_bug.cgi?id=776140 2017-03-20 12:03:29 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxqueue.c: rtprtxqueue: add basic documentation and example pipelines Mostly explaining the difference between rtprtxqueue and rtprtxsend. 2017-03-17 20:58:28 -0400 Nicolas Dufresne * sys/v4l2/meson.build: v4l2: Fix meson plugin shared object name It didn't match between AutoMake and Meson, and the Meson name didn't math the plugin name (video4linux2). 2017-03-16 18:20:54 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: fix example pipelines and improve the documentation https://bugzilla.gnome.org/show_bug.cgi?id=771383 2017-03-17 14:10:40 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: fix playback if sample number does not start at 0 This reverts commit 29b807685d3c962bbe8afe351c5dca97d59eb5e0, while fixing the original breaking tests/check/pipelines/flacdec. 2017-03-17 11:30:04 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: Revert "flacparse: fix playback if sample number does not start at 0" This breaks gst-validate on the build server (though not locally), and a unit test, and I can't run unit tests right now for some unrelated reason. This reverts commit 0747b56f8e7f4731d67f8d13a4bdc453dde0fdf7. 2017-03-16 17:44:41 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: print the correct variable in debug statement This debug statement is meant to print the time since the last (early) RTCP transmission, not the last regular RTCP transmission (which also happens to be set a few lines above to current_time, so the debug output is just confusing) 2017-03-16 17:42:27 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: convert LOG message to TRACE This is printed too often (for every chained buffer!) and just clutters the logs. 2017-03-16 14:58:45 +0100 Miguel París Díaz * gst/rtpmanager/rtpsource.c: rtpsource: fix warning message https://bugzilla.gnome.org/show_bug.cgi?id=780105 2017-03-16 13:54:54 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: fix playback if sample number does not start at 0 https://bugzilla.gnome.org/show_bug.cgi?id=777738 2017-03-15 18:58:55 +0100 Miguel París Díaz * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsource: get clock-rate from pt if needed to generate SR https://bugzilla.gnome.org/show_bug.cgi?id=780105 2017-03-16 13:52:48 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Include GStreamer souphttpsrc version in default User-Agent string 2017-03-16 00:41:44 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: fix crash with empty sprops-parameters https://bugzilla.gnome.org/show_bug.cgi?id=780040 2017-03-11 21:20:40 -0800 Thiago Santos * gst/isomp4/atomsrecovery.c: * gst/isomp4/atomsrecovery.h: atomsrecovery: also handle extra atoms after 'mdia' in a 'trak' Take into account the atoms at the end of the 'trak' atom when recovering it. So that its size (already computed and added in the trak size) isn't making offsets wrong. https://bugzilla.gnome.org/show_bug.cgi?id=771478 2017-03-11 12:56:33 -0800 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: avoid fallthrough to moovrecovery failure section Return before that to preserve our successfull results, otherwise no moov recovery information would be written https://bugzilla.gnome.org/show_bug.cgi?id=771478 2017-03-11 12:27:28 -0800 Thiago Santos * gst/isomp4/atomsrecovery.c: atomsrecovery: expect more atom types at the headers Skip more atoms at the header until it finds the 'mdat' to continue the moov recovery https://bugzilla.gnome.org/show_bug.cgi?id=771478 2017-03-14 16:42:25 -0400 Olivier Crête * Makefile.am: * configure.ac: * tests/examples/Makefile.am: * tests/examples/pulse/.gitignore: * tests/examples/pulse/Makefile.am: * tests/examples/pulse/pulse.c: pulse example: Remove That example only tested the property probe interface, which has been removed. The same kind of thing can now be done with the generic gst-device-monitor tool. 2017-03-14 16:38:02 -0400 Olivier Crête * sys/v4l2/gstv4l2object.h: v4l2: Remove unused macro 2017-03-14 16:35:25 -0400 Olivier Crête * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: Remove unused definitions 2017-03-14 10:10:19 +0100 Emeric Grange * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: qtmux: add CineForm support https://bugzilla.gnome.org/show_bug.cgi?id=780024 2017-03-14 15:09:44 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Only create new chunks if we have more than a single stream There's no point in creating multiple chunks otherwise, it only wastes some bytes for storing the chunk offsets. 2017-03-14 10:09:46 +0100 Emeric Grange * gst/isomp4/qtdemux.c: qtdemux: add S16L support https://bugzilla.gnome.org/show_bug.cgi?id=780022 2017-03-14 15:48:08 +1100 Jan Schmidt * tests/check/elements/splitmux.c: splitmux test: Use passed first/last timestamps Don't hard-code the expected timestamp range, use the values the caller is passing in. 2017-03-14 14:15:00 +1100 Matthew Waters * ext/gtk/gtkgstglwidget.c: gl: GL_ARRAY_BUFFER is not a part of VAO state As a result we need to bind it on every draw in order to have the correct state in the GL state machine. 2017-03-13 14:28:47 +1100 Matthew Waters * ext/qt/gstqtsrc.cc: gl/format: use our own GL format enum's instead of gstvideo's They can describe in more detail (such as component sizes) the requested format. 2017-03-12 11:42:25 -0400 Nicolas Dufresne * Makefile.am: * docs/plugins/inspect/plugin-soup.xml: Add old plugin names to cruft list This will help fixing uninstalled setup. Also fix missing path correction in one of the plugin xml. https://bugzilla.gnome.org/show_bug.cgi?id=779344 2016-12-15 12:38:40 +0100 Michael Dutka * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph265depay.c: rtph264depay, rtph265depay: remove stray g_debug() https://bugzilla.gnome.org/show_bug.cgi?id=779858 2017-03-10 11:24:14 +0100 Wim Taymans * gst/isomp4/gstqtmux.c: qtmux: init fourcc Initialize the fourcc to 0 so that we can detect failure later. 2017-03-08 22:50:52 -0500 Nicolas Dufresne * tests/check/Makefile.am: * tests/check/elements/level.c: * tests/check/elements/rglimiter.c: tests: Add missing LDADD for libm in tests using math.h Also, remove the math.h include for the one that just prentend to need it. 2017-03-08 22:15:46 -0500 Nicolas Dufresne * Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: Fix shout2 plugin doc generation In the previous patch, we also renamed shout2send to shout2, so it does not clash with it's feature. Though we forgot to rename it in the doc reference. This patch also add a cruft detection on the xml that made me miss this error. https://bugzilla.gnome.org/show_bug.cgi?id=779344 2017-03-04 11:03:53 -0500 Nicolas Dufresne * ext/gtk/Makefile.am: * ext/gtk/gstplugin.c: Rename plugin filesnames to match plugin names - libgstgtksink.so -> libgstgtk.so - libgstteletextdec.so -> libgstteletex.so - libgstcamerabin2.so -> libgstcamerabin.so - libgstonvif.so -> libgstrtponvif.so (meson only) - sdp -> sdpelem (avoid clash with libgstsdp) - gstsiren -> siren - libgstkmssink.so -> libgstkms.so https://bugzilla.gnome.org/show_bug.cgi?id=779344 2017-03-04 10:52:47 -0500 Nicolas Dufresne * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-shout2.xml: * ext/pulse/Makefile.am: * ext/pulse/meson.build: * ext/shout2/gstshout2.c: * ext/soup/Makefile.am: * ext/soup/meson.build: * sys/oss4/Makefile.am: Fix plugin filenames to match plugin names - libgstpulse.so becomes libgstpulseaudio.so - libgstsouphttpsrc.so becomes libgstsoup.so - libgstoss4audio.so becomes libgstoss4.so https://bugzilla.gnome.org/show_bug.cgi?id=779344 2017-03-08 16:01:02 +0200 Sebastian Dröge * gst/isomp4/atoms.c: qtmux: Free EDTS instead of just clearing it and setting it to NULL 2017-03-08 15:27:32 +0200 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/gstqtmux.c: qtmux: Fix some memory leaks related to timecode tracks 2017-03-04 00:34:44 +1100 Jan Schmidt * tests/check/elements/splitmux.c: splitmux: Add unit test for reverse playback Ensure that reverse playback works and generates the range of timestamps (0-3s) we expect, in monotonically descending order. 2017-02-28 11:50:45 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Fix reverse playback Fix the check for whether the start time of the segment has been reached when playing in reverse. Otherwise, playback stops after reaching the start of any file part, instead of continuing until all parts within the segment have played 2017-02-22 03:01:31 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Don't lose crypto info on a new moof We parse the next moof in advance of having pushed all samples from the previous one in some cases, and we'll still need the crypto info from the previous fragment so keep around any unused crypto info entries when adding new ones 2017-02-27 13:55:58 +0200 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Update modification times when sending the moov https://bugzilla.gnome.org/show_bug.cgi?id=779422 2017-03-01 16:11:47 -0800 Michael Smith * gst/audioparsers/gstsbcparse.h: sbcparse: Fix up values for allocation enumeration. https://bugzilla.gnome.org/show_bug.cgi?id=779389 2017-02-28 13:10:50 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: fix potential leak of old, unassociated, association requests https://bugzilla.gnome.org/show_bug.cgi?id=722560 2017-02-28 15:47:23 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Don't increment -1 / unset indices CID 1398545 2017-02-28 15:20:31 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Protect against NULL pointer dereference for streams without caps CID 1363332 2017-02-28 12:57:02 +0200 Sebastian Dröge * gst/rtp/gstrtph263pay.c: rtph263pay: Free mac on errors CID 1212149 2017-02-28 12:45:24 +0200 Sebastian Dröge * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: Add missing break to for loop 2017-02-28 11:02:54 +0100 Edward Hervey * tests/check/Makefile.am: check: Fix splitmux test CFLAGS Needs to know where the gstapp headers are 2017-02-27 21:02:51 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix compilation with gcc 7 qtdemux.c: In function ‘qtdemux_parse_samples’: qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context] if (stream->samples_per_frame * stream->bytes_per_frame) { ~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~ 2017-02-27 21:01:23 +0200 Sebastian Dröge * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: Fix compilation with gcc 7 gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’: gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size] memset (mp3parse->xing_seek_table_inverse, 0, 256); ^~~~~~ gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_handle_first_frame’: gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size] memset (mp3parse->xing_seek_table_inverse, 0, 256); ^~~~~~ 2017-02-27 19:31:39 +0200 Sebastian Dröge * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: When getting new headers, replace the old version of them This prevents storing an infinite amount of e.g. comment headers if they come without a new initialization header in front of them. There can only be one header of each type. 2017-02-27 19:25:35 +0200 Sebastian Dröge * tests/check/Makefile.am: * tests/check/elements/rtp-payloading.c: rtp-payloading: Add new test for Vorbis renegotiation Check if encoding, payloading, depayloading and decoding works if the stream configuration (and thus the headers) change. 2017-02-27 19:24:07 +0200 Sebastian Dröge * gst/rtp/gstrtpvorbispay.c: vorbispay: Only replace headers when receiving a new config header If we also replace all headers when receiving any possibly following comments header, we would throw away the config header before being able to make use of it. 2017-02-23 12:11:15 +0200 George Kiagiadakis * tests/check/Makefile.am: * tests/check/elements/splitmux.c: tests: splitmux: add unit test for content with sparse streams https://bugzilla.gnome.org/show_bug.cgi?id=761086 2017-02-22 11:23:19 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxpartreader.c: splitmuxpartreader: ignore sparse streams when calculating the end offset of a part A sparse stream's ending timestamp can be considerably smaller than the ending timestamps of the other streams, which can lead to skipping considerable time from the next part. https://bugzilla.gnome.org/show_bug.cgi?id=761086 2017-02-22 11:21:06 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxpartreader.c: splitmuxpartreader: identify sparse streams 2017-02-17 14:37:08 +0200 Sebastian Dröge * ext/qt/gstqtglutility.cc: qml: Add support for Vivante EGL FS windowing system https://bugzilla.gnome.org/show_bug.cgi?id=778825 2017-02-25 21:47:03 -0300 Edgard Lima * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * gst/audioparsers/gstamrparse.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726depay.h: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg726pay.h: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpspeexpay.h: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/gstv4l2vidorient.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: Update Edgard Lima's email https://bugzilla.gnome.org/show_bug.cgi?id=779230 2017-02-08 13:36:00 +0000 Andrew * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: Don't always reset PTS to 0 after a gap In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP timestamps is more than (3 * jbuf->clock_rate) we call rtp_jitter_buffer_reset_skew which resets pts to 0. So components down the pipeline (playes, mixers) just skip frames/samples until pts becomes equal to pts before gap. In version 1.10.2 and before this checking was bypassed for packets with "estimated dts", and gaps were handled correctly. https://bugzilla.gnome.org/show_bug.cgi?id=778341 2017-02-24 15:59:41 +0200 Sebastian Dröge * meson.build: meson: Update version 2017-02-24 15:37:36 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.11.2 === 2017-02-24 15:07:23 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: Release 1.11.2 2017-02-24 12:50:21 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2017-02-24 12:44:58 +0200 Sebastian Dröge * po/el.po: po: Update translations 2017-02-10 20:50:17 +0900 Seungha Yang * ext/soup/gstsouphttpsrc.c: souphttpsrc: Extract redirection uri on libsoup's restarted callback Let libsoup handle redirection automatically. And then, to figure out redirection uri, extract it on "restarted" callback which will be fired before soup_session_send() is returned. https://bugzilla.gnome.org/show_bug.cgi?id=778428 2017-01-02 19:29:04 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Update image size when extrapolating Update the image size according the amount of data we are going to read/write. This workaround bugs in driver where the sizeimage provided by TRY/S_FMT represent the buffer length (maximum size) rather then the expected bytesused (buffer size). https://bugzilla.gnome.org/show_bug.cgi?id=775564 2017-02-17 15:50:32 -0800 Reynaldo H. Verdejo Pinochet * sys/v4l2/gstv4l2object.c: v4l2: fix typo in _acquire_format() error messages Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=778815 2017-02-07 17:27:56 +0100 Guillaume Desmottes * tests/check/elements/matroskamux.c: * tests/check/elements/qtmux.c: tests: matroskamux, qtmux: don't add codec_data buffers to template caps streamheader and codec_data buffers fields are only meant to be in the negotiated caps, not the template caps. Fixes false-positive leaks of those buffers detected by the leaks tracer, as template caps are static, and we decided to not include code in gstreamer core to handle this unusual case of template caps having buffers in them. https://bugzilla.gnome.org/show_bug.cgi?id=768762 2017-02-09 12:46:54 +0000 Jochen Henneberg * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: Update and send out headers when new headers are received The payloader needs to reset and update the vorbis config data which is pushed on the network if it receives new headers, or at least, it may have to do so. Without this, the stream configuration could change without the payloader sending the new configuration to the other side. 2017-02-15 14:48:58 -0500 Olivier Crête * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Change files on incompatible caps https://bugzilla.gnome.org/show_bug.cgi?id=761761 2017-02-15 16:35:01 -0500 Olivier Crête * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Reset ready_for_output on state change https://bugzilla.gnome.org/show_bug.cgi?id=761761 2017-02-15 15:09:06 -0500 Olivier Crête * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Remove unused next_max_out_running_time https://bugzilla.gnome.org/show_bug.cgi?id=761761 2017-02-15 15:07:32 -0500 Olivier Crête * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Remove unused muxed_out_time https://bugzilla.gnome.org/show_bug.cgi?id=761761 2017-02-17 13:07:05 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: Revert "qtdemux: Always snap to the start of the keyframe" This reverts commit 107902ec514bd826aa29d2298107e2c091e1c779. This commit intended to ensure that keyframe seeks land at the start timestamp of a keyframe, rather than in the middle of one, but they cause trouble on files with sparse streams, or with JPEG 'cover art' tracks that have only one or a few JPEG samples with very long durations. That's still desirable for doing seamless cutting of videos, but needs a rethink for implementation. https://bugzilla.gnome.org/show_bug.cgi?id=778690 2017-02-17 01:22:11 +1100 Jan Schmidt * gst/audiofx/audioecho.c: * gst/audiofx/audioecho.h: audiofx/echo: added surround-delay and surround-mask Add a new boolean surround-delay property that makes audioecho just apply a delay to certain channels to create a surround effect, rather than an echo on all channels. This is useful when upmixing from stereo - for example. Add a surround-mask property to control which channels are considered surround sound channels when adding a delay with surround-delay = true Original patch from Jochen Henneberg 2017-02-15 00:13:30 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Use IP_MULTICAST_ALL for filtering IPv4 packets if available This goes around the inefficient control message based filtering and does all the filtering kernel-side. Unfortunately this is Linux-only and there is no IPv6 variant of it (yet). 2017-02-14 19:53:30 +0000 Tim-Philipp Müller * Makefile.am: meson: dist meson build files Ship meson build files in tarballs, so people who use tarballs in their builds can start playing with meson already. 2017-02-10 10:53:05 +0100 Søren Juul * gst/icydemux/gsticydemux.c: * tests/check/elements/icydemux.c: icydemux: reset tags on empty value Some radio streams uses StreamTitle='' to reset the title after a track stopped playing, e.g. while the host talks between tracks or during news segments. This change forces an empty tag object to be distributed if StreamTitle or StreamUrl is received with empty value, thus allowing downstream elements to get notified about this. https://bugzilla.gnome.org/show_bug.cgi?id=778437 2017-02-13 11:17:25 +0100 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Properly notify missing elements If the srtp elements are not present, post a message on the bus informing about the missing plugins. 2017-02-10 10:32:57 -0300 Juan Pablo Ugarte * sys/v4l2/gstv4l2object.c: v4l2object: mark singleton caps as "may be leaked" objects. Set MAY_BE_LEAKED flag on static pads returned by gst_v4l2_object_get_*_caps() functions. Made functions thread safe by using g_once_init[enter|leave] funtions. https://bugzilla.gnome.org/show_bug.cgi?id=778453 2017-02-09 14:18:30 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Remove now unused done label 2017-02-09 12:55:32 +0100 Nick Kallen * gst/imagefreeze/gstimagefreeze.c: imagefreeze: do not cache caps Upstream elements like videoflip can transform caps, such as changing width and height. When an imagefreeze downstream receives an ACCEPT_CAPS query it will NOW return all caps that it can accept. https://bugzilla.gnome.org/show_bug.cgi?id=778389 2017-02-09 11:29:43 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Add a comment about how atom_trak_set_elst_entry() works 2014-08-22 09:55:43 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux_dump.c: qtdemux: demote some log messages to TRACE level Don't spam debug log with uninteresting stuff. 2017-02-08 17:24:26 +0200 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Clear edit lists every time we recalculate them We recalculate them, so any old information has to be forgotten. Otherwise we write invalid edit lists when writing headers multiple times. https://bugzilla.gnome.org/show_bug.cgi?id=778330 2017-02-07 13:10:18 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: splitmuxsrc: Allow for buffers before the segment when measuring Used signed calculations when measuring the max_ts of an input fragment, so as to calculate the correct duration and offset when buffers have timestamps preceding their segment 2017-02-02 12:55:25 +0100 Miguel París Díaz * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsession: relate received FIRs and PLIs to source This is needed in order to: - Avoid ignoring requests for different media sources. - Add SSRC field in the GstForceKeyUnit event. https://bugzilla.gnome.org/show_bug.cgi?id=778013 2017-01-30 20:20:08 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: sanity check number of segments in edit list Fixes crash with fuzzed file. https://bugzilla.gnome.org/show_bug.cgi?id=777940 2017-01-02 22:16:39 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Skip seeking query if upstream format is time Don't need to querying byte-format seeking for time-format upstream case https://bugzilla.gnome.org/show_bug.cgi?id=776715 2016-12-01 12:47:08 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Use upstream's StreamFlags if there are When multiple demuxer's are used, upstream might want to indicate default streams using GST_STREAM_FLAG_{SELECT, UNSELECT} https://bugzilla.gnome.org/show_bug.cgi?id=775440 2017-01-27 16:14:16 +0200 Vivia Nikolaidou * gst/isomp4/atoms.c: qtmux: Timecode track fixes for STSD entry The n_frames field (frames per second) should follow the nominal frame rate for drop-frame timecodes. Also, the trak's timescale (and duration, accordingly) should follow the STSD entry's timescale and frame duration (fps_n and fps_d accordingly), not the other way around. https://bugzilla.gnome.org/show_bug.cgi?id=777832 2017-01-19 11:08:11 +0100 Arnaud Vrac * ext/soup/gstsouphttpsrc.c: souphttpsrc: retry request on early termination from the server Fix a regression introduced by commit 183695c61a54f1 (refactor to use Soup's sync API). The code previously attempted to reconnect when the server closed the connection early, for example when the stream was put in pause for some time. Reintroduce this feature by checking if EOS is received before the expected content size is downloaded. In this case, do the request starting at the previous read position. https://bugzilla.gnome.org/show_bug.cgi?id=776720 2017-01-10 09:40:56 -0700 Matt Staples * gst/rtsp/gstrtspsrc.c: rtspsrc: find_stream_by_channel should ignore unconfigured streams https://bugzilla.gnome.org/show_bug.cgi?id=777101 2017-01-25 18:43:00 +0000 Brendan Shanks * gst/isomp4/gstqtmux.c: qtmux: Fix debug typo and remove misleading warning https://bugzilla.gnome.org/show_bug.cgi?id=777362 2017-01-25 20:56:24 +0200 Sebastian Dröge * tests/examples/rtp/client-PCMA.c: rtp: Remove unused variable in example client-PCMA.c:84:22: warning: unused variable 'isrc' [-Wunused-variable] GObject *session, *isrc, *osrc; ^ 2017-01-25 19:21:03 +0200 Sebastian Dröge * ext/qt/Makefile.am: qt: The code requires at least C++11 ... and clang requires this to be specified on the commandline while gcc nowadays defaults to C++11 or even newer. 2017-01-09 11:32:35 +0530 Rahul Bedarkar * gst/wavparse/gstwavparse.c: wavparse: check for not NULL before clearing adapter In case wavparse receives a manually injected FLUSH_STOP event while operating in pull mode we get criticals because we'd try to clear a NULL adapter. https://bugzilla.gnome.org/show_bug.cgi?id=777123 2017-01-24 19:23:44 -0300 Thibault Saunier * tests/check/meson.build: meson: Properly use ':' for defining keywords 2017-01-17 16:41:58 +0100 Jean-Christophe Trotin * sys/v4l2/gstv4l2allocator.c: v4l2allocator: reference memory before the buffer is queued In gst_v4l2_allocator_qbuf(), the memory is referenced after the buffer is queued. Once queued (VIDIOC_QBUF), the buffer might be handled by the V4L2 driver (e.g. decoded) and dequeued (gst_v4l2_allocator_dqbuf), through a different thread, before the memory is referenced (gst_memory_ref). In this case, in gst_v4l2_allocator_dqbuf(), the memory is unreferenced (gst_memory_unref) before having been referenced: the memory refcount reaches 0, and the memory is freed. So, to avoid this crossing case, in gst_v4l2_allocator_qbuf(), the memory shall be referenced before the buffer is queued. https://bugzilla.gnome.org/show_bug.cgi?id=777399 2017-01-24 17:59:59 +0200 Sebastian Dröge * gst/isomp4/atoms.c: qtmux: Only write 4 byte zero padding to the Video Sample Description in MOV For MP4 this is not defined, and it actually breaks things for MSE in Chrome if we do this. For MOV this is required by some broken software but the official specification says it's optional: https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html https://bugzilla.gnome.org/show_bug.cgi?id=777540 2017-01-02 13:42:04 +0100 Santiago Carot-Nemesio * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.h: rtpstats: Keep number of nacks sent/received per source Currently, the nack packets sent or received are kept at session level, which makes it impossible to distinguish how many of these packages were sent/received per ssrc when several sources are in the same session. This patch is aligned with the https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats https://bugzilla.gnome.org/show_bug.cgi?id=776714 2016-12-08 15:59:33 +0100 Jonas Holmberg * gst/rtp/gstrtph265pay.c: rtph265pay: Fix handling of config-interval Insert VPS/SPS/PPS before the first NAL unit containing an I-frame in an access unit only. If an access unit consists of several such NAL units (tiles) VPS/SPS/PPS should only be inserted before the first of them so that parameters are only updated between frames. Do not insert VPS/SPS/PPS before P-frames when config-interval is -1. https://bugzilla.gnome.org/show_bug.cgi?id=775817 2017-01-19 12:29:44 +0100 Arnaud Vrac * ext/soup/gstsouphttpsrc.c: souphttpsrc: report a useful error message when soup_session_send fails This helps to understand cases where libsoup doesn't set the message status code after running soup_session_send. https://bugzilla.gnome.org/show_bug.cgi?id=777222 2017-01-19 11:05:00 +0100 Arnaud Vrac * ext/soup/gstsouphttpsrc.c: souphttpsrc: properly check that seek range was respected This check must be done only when we are sure the request was successfully sent. soup_session_send() might fail without setting the status code. In this case status code is 0 so we would only catch the error after the seek range check. In this case we would report an error saying that the seek range was not respected, instead of reporting the underlying error that triggered the soup_session_send() failure. https://bugzilla.gnome.org/attachment.cgi?bugid=777222 2017-01-09 21:04:51 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: gdkpixbufoverlay: add a positioning coefficient pair ... so as to allow one clearly defined (absolute) positioning mode that can cater for a variety of absolute but also relative positioning with respect to edge or center. 2017-01-21 20:48:22 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: update composition in _before_transform ... since we need to determine passthrough mode for buffer preparation before calling into _transform_ip. 2017-01-07 20:11:13 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: handle setting NULL gdkpixbuf ... which is a clearer way to clear any current overlay, other than fiddling with alpha or positioning properties to make it virtually go away. 2017-01-20 17:16:10 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Stop reading a ncdt sub-tag if it goes behind the surrounding tag https://bugzilla.gnome.org/show_bug.cgi?id=777532 2017-01-20 07:58:26 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Fix various out of bounds reads when parsing ncdt tags https://bugzilla.gnome.org/show_bug.cgi?id=777500 2017-01-19 13:46:58 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Increment current stts index whenever we finished one stts entry Otherwise we could read more chunks than there are available, doing an out of bounds read and potentially crash. https://bugzilla.gnome.org/show_bug.cgi?id=777469 2017-01-19 13:25:53 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: Revert "qtdemux: Increment current stts index in all code paths after reading one chunk" This reverts commit 99d5d7570d0b53dad3bc8eb653b1320ee422aace. It broke playback of various valid files. 2017-01-19 07:52:33 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Increment current stts index in all code paths after reading one chunk Otherwise we could read more chunks than there are available, doing an out of bounds read and potentially crash. https://bugzilla.gnome.org/show_bug.cgi?id=777469 2017-01-19 08:37:37 +0100 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Initialize return variable In the normal use-case we would end up with ret being unitialized causing havoc. https://bugzilla.gnome.org/show_bug.cgi?id=777222 2017-01-13 12:27:40 +0000 David Warman * gst/isomp4/qtdemux.c: qtdemux: avoid XMP tag parsing fatal error. qtdemux_handle_xmp_taglist() requires a writable taglist, but qtdemux->tag_list can become non-writable, specifically after sending global tags (qtdemux.c:958), which adds a second reference. Ensure the list is made writable before calling (make_writable will copy the list if necessary). https://bugzilla.gnome.org/show_bug.cgi?id=766177 2016-05-31 13:17:45 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: rework taglist handling Keep taglist around during element existance to avoid having to create it at different places before usage. Makes code simpler to handle. https://bugzilla.gnome.org/show_bug.cgi?id=766177 2017-01-16 11:58:02 +0100 Arnaud Vrac * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: make flow return values handling clearer The flow return values was stored in the element before because the result had to be set from callbacks. This is not the case anymore, we can return the flow result directly from functions, making the code easier to understand. https://bugzilla.gnome.org/show_bug.cgi?id=777222 2017-01-13 16:40:43 +0100 Arnaud Vrac * ext/soup/gstsouphttpsrc.c: souphttpsrc: properly track redirections The current code configures libsoup to handle redirections transparently, without informing the caller, thus preventing the element to record the redirect code and location uri. Fix this by always setting the SOUP_MESSAGE_NO_REDIRECT, preventing libsoup from handling the redirection. When we receive a redirection request and libsoup can safely handle it, return a custom error which triggers a retry with the new URI. https://bugzilla.gnome.org/show_bug.cgi?id=777222 2017-01-17 10:53:39 +0100 Aurélien Zanelli * gst/isomp4/gstqtmuxmap.c: qtmux: add 4444 and 4444xq variants to video/x-prores pad template caps They are handled since commit 7b565475bf551c53b8eed46f7086f3b372f1f6c4 (qt: Add support for ProRes 4444 XQ). https://bugzilla.gnome.org/show_bug.cgi?id=777377 2017-01-17 10:48:57 +1100 Jan Schmidt * gst/matroska/ebml-read.c: matroska: Quiet a WARN when parsing push mode This warning was noisy when returning EOS, which is just used to indicate more data is needed from upstream. 2017-01-16 14:50:22 +0100 Georg Lippitsch * gst/isomp4/gstqtmux.c: qtmux: Don't write Sync Sample Atom for ProRes https://bugzilla.gnome.org/show_bug.cgi?id=777331 2015-01-28 08:58:26 +0100 Enrico Jorns * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2_calls.h: v4l2: Remove usage and definition of LOG_CAPS macro Unlike former definitions of LOG_CAPS, the current implementation simply expands to GST_DEBUG_OBJECT. The LOG_CAPS macro is rarely used and most uses duplicate already existing GST_DEBUG_OBJECT lines. Therefore, the caps are often printed twice which unnecessarily clutters the debug log. Replace LOG_CAPS calls with GST_DEBUG_OBJECT, remove LOG_CAPS calls, and delete the definition of LOG_CAPS. https://bugzilla.gnome.org/show_bug.cgi?id=776899 2017-01-16 15:40:43 +0100 Jean-Christophe Trotin * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: remove duplicated line of code https://bugzilla.gnome.org/show_bug.cgi?id=777330 2017-01-16 15:17:15 +0100 Jean-Christophe Trotin * sys/v4l2/gstv4l2allocator.c: v4l2allocator: fix memory type in allocator probe The buffer memory type provided to the VIDIOC_CREATE_BUFS ioctl shall be set with the value ("memory") given as input parameter of the gst_v4l2_allocator_probe() function. https://bugzilla.gnome.org/show_bug.cgi?id=777327 2017-01-14 15:27:19 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: fix other icon counter check It's never going to be 0 if we first increment and then check. 2017-01-14 15:16:53 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: boldly assume that first 'covr' image is the front cover 2017-01-14 15:09:07 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: extract cover art images into GST_TAG_IMAGE not PREVIEW_IMAGE These are usually much bigger than icon size and required by iTunes to be certain fairly large sizes. In qtmux it is also the IMAGE tags which we write out as 'covr' atoms. 2017-01-14 15:05:36 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: also set PICTURE tag width and height if available 2017-01-14 14:58:52 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: fix encoder init error with some GST_TAG_PREVIEW_IMAGEs The encoder fails to initialise when we try to set GST_TAG_PREVIEW_IMAGEs sent to use by qtdemux from iTunes-generated m4a files. We should not just blindly translate the PREVIEW tag to file icon image types, but check if the specific conditions required are met (i.e. image type 1 must be a 32x32 PNG icon, and what we're getting is 500x500). https://bugzilla.gnome.org/show_bug.cgi?id=776962 2017-01-13 12:39:00 +0000 Tim-Philipp Müller * meson.build: meson: bump version 2017-01-11 10:32:23 -0300 Juan Pablo Ugarte * tests/examples/gtk/glliveshader.c: gl/examples/gtk: fixed compilation on systems without GL_GEOMETRY_SHADER https://bugzilla.gnome.org/show_bug.cgi?id=777143 2017-01-12 21:35:25 +1100 Matthew Waters * ext/qt/gstqtsink.cc: * ext/qt/gstqtsrc.cc: gl/utils: also take care of the local GL context in query functions Simplifies a deduplicates a lot of code in elements retrieving/setting the local OpenGL context. 2017-01-12 21:35:25 +1100 Matthew Waters * ext/gtk/gstgtkglsink.c: gl/utils: also take care of the local GL context in query functions Simplifies a deduplicates a lot of code in elements retrieving/setting the local OpenGL context. 2016-12-22 17:40:40 +0200 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Add option for timecode-based split If this option is given, it will calculate the next split point based on timecode difference. https://bugzilla.gnome.org/show_bug.cgi?id=774209 2017-01-13 00:01:06 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Don't reset request pad numbering across uses When reset, don't restart request pad numberings, as request pads can survive across state changes. Only restart at 0 if all request pads are handed back first. https://bugzilla.gnome.org/show_bug.cgi?id=777174 2017-01-11 18:52:28 +0100 Mathieu Duponchelle * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtxqueue: Expose basic statistics as properties. Statistics about the total number of retransmission requests and the actual number of retransmitted packets can be helpful at application-level. https://bugzilla.gnome.org/show_bug.cgi?id=777182 2017-01-12 17:45:35 +0100 Aurélien Zanelli * gst/isomp4/gstqtmux.c: qtmux: simplify video/x-h264 caps handling 'stream-format' and 'alignment' are defined in pad template caps so there is no need to check them again here. Also remove bitrate parsing from caps as bitrate in caps doesn't make sense but from tags, which is actually the case. https://bugzilla.gnome.org/show_bug.cgi?id=777181 2016-12-08 17:02:22 +0100 Aurélien Zanelli * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: add basic HEVC/H.265 muxing support https://bugzilla.gnome.org/show_bug.cgi?id=736752 2017-01-11 18:29:05 +0100 Georg Lippitsch * gst/isomp4/gstqtmux.c: qtmux: Calculate clean aperture size Calculate clean aperture dimensions by first guessing display aspect ratio based on pixel aspect ratio and frame size. https://bugzilla.gnome.org/show_bug.cgi?id=777100 2017-01-10 18:19:55 +0200 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux_types.c: qtmux: Write tapt atom for MOV files if PAR not 1/1 Needed for QuickTime 7 to properly play files. Also write the clap atom for MOV files always, not only when ProRes is used as a video codec. It's mandatory for MOV. https://bugzilla.gnome.org/show_bug.cgi?id=777100 2017-01-12 16:32:45 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.11.1 === 2017-01-12 15:31:02 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: Release 1.11.1 2017-01-12 14:38:55 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2017-01-12 14:36:22 +0200 Sebastian Dröge * po/el.po: * po/hr.po: * po/id.po: * po/zh_CN.po: po: Update translations 2017-01-11 17:53:32 -0800 Andre McCurdy * gst/isomp4/qtdemux.c: qtdemux: free seqh after calling qtdemux_parse_svq3_stsd_data() The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to be freed by the caller after use. https://bugzilla.gnome.org/show_bug.cgi?id=777157 Signed-off-by: Andre McCurdy 2017-01-10 16:01:35 +0100 Edward Hervey * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: isomp4: Don't spam debug log with knonw/padding atoms Only output WARNING messages for atoms we don't know how to handle instead of for padding/known atoms we don't need to do any processing on https://bugzilla.gnome.org/show_bug.cgi?id=777095 2017-01-10 16:54:48 +0800 Haihua Hu * ext/qt/qtwindow.cc: * ext/qt/qtwindow.h: qmlglsrc: use glBlitFramebuffer to copy texture for GLES3.0 If support glBlitFrameBuffer, use it for texture copy instead of glCopyTexImage2D https://bugzilla.gnome.org/show_bug.cgi?id=777078 2017-01-09 19:05:10 +0000 Tim-Philipp Müller * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: * sys/v4l2/gstv4l2bufferpool.c: Fix indentation 2017-01-09 19:04:04 +0000 Tim-Philipp Müller * tests/check/elements/rtpjitterbuffer.c: tests: rtpjitterbuffer: fix compiler warning due to c99-ism rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode 2016-11-11 14:31:03 +1100 Matthew Waters * gst/autodetect/gstautodetect.c: autodetect: bring the element state down after success Otherwise some messages that are emitted by the element on NULL->READY will not reach the application. https://bugzilla.gnome.org/show_bug.cgi?id=764947 2017-01-08 01:13:32 +1100 Jan Schmidt * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Write tfdt atom into fragmented files. The DASH spec requires that tfdt atoms be present, so write one out. ISO/IEC 23009-1:2014 6.3.4.2 https://bugzilla.gnome.org/show_bug.cgi?id=708221 2017-01-07 23:55:42 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Don't reset output timestamps when no tfdt If a fragmented stream doesn't have a tfdt, don't reset the output timestamps at each fragment boundary by erroneously using the default value of 0. Introduced by commit 69fc48 https://bugzilla.gnome.org/show_bug.cgi?id=754230 2016-12-16 16:51:48 -0300 Thibault Saunier * ext/vpx/meson.build: * gst/equalizer/meson.build: * gst/isomp4/meson.build: * meson.build: meson: Install presets files 2017-01-03 10:12:30 +0530 Garima Gaur * gst/avi/gstavidemux.c: avidemux: fix some caps leaks https://bugzilla.gnome.org//show_bug.cgi?id=776789 2016-12-22 17:34:08 +0200 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Return a bin with a "location" property as a sink Splitmuxsink might be called with a custom bin as a sink. If it has a "location" property, it can be used. 2016-11-18 22:42:18 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmux: Rewrite buffer collection and scheduling Majorly change the way that splitmuxsink collects incoming data and sends it to the output, so that it makes all decisions about when / where to split files on the input side. Use separate queues for each stream, so they can be grown individually and kept as small as possible. This removes raciness I observed where sometimes some data would end up put in a different output file over multiple runs with the same input. Also fixes hangs with input queues getting full and causing muxing to stall out. 2016-11-17 23:40:27 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * tests/check/elements/splitmux.c: splitmuxsink: Add format-location-full signal Add a new signal for formatting the filename, which receives a GstSample containing the first buffer from the reference stream that will be muxed into that file. Useful for creating filenames that are based on the running time or other attributes of the buffer. To make it work, opening of files and setting filenames is now deferred until there is some data to write to it, which also requires some changes to how async state changes and gap events are handled. 2016-12-31 01:54:01 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Always snap to the start of the keyframe When performing a key-unit seek, always snap to the start ts of the keyframe buffer we landed on so that the keyframe is entirely within the resulting outgoing segment. That seems the most sensible result, since the user requested snapping to the keyframe position. 2016-12-31 01:48:04 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Omit cslg_shift when snapping seeks Segments times and seek requests are stored and handled in raw 'PTS' time, without the cslg_shift - which only applies to outgoing samples. Omit the cslg_shift portion when extracting PTS to compare for internal seek snaps. If the cslg_shift is included, then keyframe+snap-before seeks generate a segment start/stop time that already includes the cslg_shift, and it's then added a 2nd time, causing the first buffer(s) to have timestamps that are out of segment. 2016-12-30 22:31:38 +1100 Jan Schmidt * gst/isomp4/atoms.c: qtmux: Remove bogus check in atom_stsc_add_new_entry() Remove an old check from atom_stsc_add_new_entry() that extends the last entry in the STSC if the samples per chunk matches, as the new interleave merging logic requires that the final entry by updateable. There's already code below which simply merges the final entry into the previous one when needed, so rely on that instead. Fixes asserts like: ERROR:atoms.c:2940:atom_stsc_update_entry: assertion failed: (atom_array_index (&stsc->entries, len - 1).first_chunk == first_chunk) 2016-04-24 21:38:51 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Fix key_time in gst_qtdemux_adjust_seek() time in segment should be PTS based (not DTS). https://bugzilla.gnome.org/show_bug.cgi?id=765498 2016-12-28 22:49:27 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Pass seek flags when activating. Pass all seek flags when activating a part based on a seek, so that SNAP flags are preserved. 2016-11-26 01:13:19 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: splitmux: Fix a small race in the splitmuxsrc Make sure the state of the parser is set to collecting streams before chaining up to the parent change_state() method, to close a small window that can cause playback to never commence. 2017-01-02 15:06:33 +0100 Edward Hervey * tests/check/elements/amrparse.c: check: Remove dead code 2016-12-31 09:52:25 +0000 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: refactor max_files handling a bit Use GQueue instead of a GSList so we don't have to traverse the whole list to append something every time. And it also keeps track of the number of items in it for us. Add a function to add filenames to the list of old files and use it in more places, so that memory doesn't build up in other modes either if no max_files limit is specified. https://bugzilla.gnome.org/show_bug.cgi?id=766991 2016-05-29 17:21:47 +0100 Ursula Maplehurst * gst/multifile/gstmultifilesink.c: multifilesink: don't leak memory when no max-files limit is set Technically we weren't leaking the memory, just storing it internally and never using it until the element is freed. But we'd still use more and more memory over time, so this is not good over longer periods of time. Only keep track of files if there's actually a limit set, so that we will prune the list from time to time. https://bugzilla.gnome.org/show_bug.cgi?id=766991 2016-12-29 12:39:20 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: adjust segment stop for KEY_UNIT negative rate seeking 2016-12-29 12:25:35 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: implement pull mode SNAP flag seeking 2016-12-29 11:26:33 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: tweak KEY_UNIT SNAP seek handling Previously, seeking to position y where y is (strictly) within a keyframe would seek to that keyframe both with SNAP_BEFORE and SNAP_AFTER, where the latter is now adjusted to really snap to the next keyframe. 2016-12-28 13:23:11 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: correctly perform pull mode KEY_UNIT seeking Rather amazingly (and equally unnoticed), keyunit seeking resulted in segments where start != time (which is bogus for simple avi timeline). So, properly adjust the segment (start) rather than fiddling with segment time (only). 2016-12-28 13:04:54 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: restore considering of pull mode KEY_UNIT seeking ... by using the original seek event's flags rather than the corresponding segment flags, which do not have such counterpart flags (and do no longer have them covertly sneaking in nowadays). 2015-05-08 12:44:01 +0200 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: only drop actual streamheader buffers with xiph codecs With Xiph codecs the stream header buffers are both in the caps and are usually also at the beginning of each input stream, but it's perfectly possible that the input stream does not have the stream header buffers inline in the data. Matroskamux would drop the first N buffers assuming they're stream headers, but this meant it would drop actual payload data when the stream didn't contain the stream headers inline. Fix this by only dropping leading buffers if they're flagged as stream headers. This fixes issues with streams that are being tapped into after streaming has started. https://bugzilla.gnome.org/show_bug.cgi?id=749098 2016-12-21 17:43:58 +0100 Nicola Murino * tests/check/elements/matroskamux.c: matroskamux: adjust unit test to modified behaviour Now matroskamux mark all packets of audio-only streams as keyframes so in test_block_group after pushing the test audio data 4 buffers are produced and not more 2. The last buffer is the original data and must match with what pushed. The remaining ones are matroskamux headers https://bugzilla.gnome.org/show_bug.cgi?id=754696 2016-05-30 01:15:31 +0200 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: mark all packets of audio-only streams as keyframes This helps with streaming audio-only streams via multifdsink, tcpserversink and such. https://bugzilla.gnome.org/show_bug.cgi?id=754696 2015-03-28 18:15:36 +0100 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: add G722 audio support https://bugzilla.gnome.org/show_bug.cgi?id=746574 2016-12-13 11:11:07 +0900 Wonchul Lee * gst/udp/gstudpsrc.c: updsrc: Add to join multiple multicast interfaces https://bugzilla.gnome.org/show_bug.cgi?id=776030 2015-03-25 13:51:30 +0000 Tim-Philipp Müller * gst/rtp/gstrtpklvdepay.c: rtpklvdepay: add the SPARSE flag to the outgoing stream-start event 2016-12-17 13:42:34 +0000 Tim-Philipp Müller * ext/qt/gstqtsink.cc: * ext/qt/gstqtsrc.cc: qt: improve element and property descriptions a bit 2016-12-14 14:37:45 -0800 Reynaldo H. Verdejo Pinochet * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: rtpmanager: place content before Since-version API marker Avoids confusing the parser 2016-12-14 14:16:53 -0800 Reynaldo H. Verdejo Pinochet * ext/shout2/gstshout2.c: shout2: fix 404 in package origin 2016-12-14 21:45:15 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Check if we have enough data available when parsing edit lists Also consume the data entry by entry to get complicated indexing out of the code. https://bugzilla.gnome.org/show_bug.cgi?id=776107 2016-12-14 19:15:03 +0100 Víctor Manuel Jáquez Leal * sys/v4l2/gstv4l2object.c: v4l2object: Don't check size in a non-list value After commit 1ea9735a I see these error while using the webcam integrated in my laptop: GStreamer-CRITICAL **: gst_value_list_get_size: assertion 'GST_VALUE_HOLDS_LIST (value)' failed The issue is gst_v4l2src_value_simplify() was doing its job of generating a single value, rather than the original list. That why, when getting the list size, a critical warning was raised. This patch takes advantage of the compiler optimizations to verify first if the list was simplified, thus use it directly, otherwise, if it is a list, verify its size. https://bugzilla.gnome.org/show_bug.cgi?id=776106 2016-12-14 10:39:12 +0100 Havard Graff * tests/check/elements/rtpjitterbuffer.c: tests/jitterbuffer: Major refactoring and cleanups * Changed PCMU->TEST for common macros * Changed verify-functions (lost & rtx) into macros. * Remove option to add marker-bit for test-buffers (not used anywhere) * Add new push_test_buffer function that makes sure there are correlation between dts and the time on the clock. (classic test-mistake) * Established a generic starting-point for tests with the construct_deterministic_initial_state function and use it where applicable, which removes lots of "boilerplate" everywhere. * Add basic lost-event test * Remove as much "magic constants" as possible. * Remove 3 tests that no longer are testing anything that others don't, and was completely unmaintainable. * Remove unnecessary use of the testclock * Verify each test is testing what it actually says it does (and modify where it doesn't) In general, make the tests much smaller, better, more maintainable and readable. https://bugzilla.gnome.org/show_bug.cgi?id=774409 2016-12-14 09:54:11 +0000 Tim-Philipp Müller * .gitignore: * Makefile.am: * configure.ac: * gst-plugins-good.spec.in: Remove generated .spec file Likely extremely bitrotten, and we should not ship this anyway. 2016-12-14 10:15:10 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Check that the XiTh size is big enough https://bugzilla.gnome.org/show_bug.cgi?id=775794 2016-12-09 20:27:53 +0900 Heekyoung Seo * gst/isomp4/qtdemux.c: qtdemux: Check node length of video sample description Add check for node length of video sample description and its fields and for the XiTh atom. Also unify the code a bit. https://bugzilla.gnome.org/show_bug.cgi?id=775794 2016-12-08 18:50:52 +0900 Heekyoung Seo * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Enable xvid/mp2 codec support Add support for xvid video and mp2 audio, add m2v1 fourcc. https://bugzilla.gnome.org/show_bug.cgi?id=775794 2016-12-13 22:32:46 +0200 Sebastian Dröge * gst/rtp/gstrtpvp9depay.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/rtprtx.c: * tests/check/elements/vp9enc.c: gst: Don't declare variables inside the for loop header This is a C99 feature. 2016-12-11 13:27:27 +0200 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: Ensure to reinit buffers whenever they were not allocated yet That is, whenever we go through start/stop we have to ensure that on the next opportunity the buffers are reallocated again. Otherwise the buffers might be NULL because the element was reused with the same configuration as before (i.e. set_caps() wouldn't have reinited the buffers). https://bugzilla.gnome.org/show_bug.cgi?id=775898 2016-12-10 12:52:18 +0000 Tim-Philipp Müller * docs/design/Makefile.am: * docs/design/design-rtpauxiliary.txt: * docs/design/design-rtpcollision.txt: * docs/design/design-rtpretransmission.txt: docs: design: remove, moved to gst-docs 2016-12-09 17:17:35 -0300 Thibault Saunier * meson.build: meson: Support building without Gst debug 2016-12-09 17:55:39 +0200 Sebastian Dröge * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: flxdec: Only send SEGMENT events after CAPS I.e., don't just forward the event but delay it if we don't have caps on the srcpad yet. 2016-12-09 17:49:40 +0200 Sebastian Dröge * gst/flx/gstflxdec.c: flxdec: Unref and unmap buffers in all code paths as needed https://bugzilla.gnome.org/show_bug.cgi?id=775888 2016-12-08 12:37:25 +0300 Sergey Borovkov * ext/qt/gstqtglutility.cc: qml: Fix egl being deinitialized on display cleanup Use the with_egl_display() variant in order to not destroy the EGLDisplay on destruction. https://bugzilla.gnome.org/show_bug.cgi?id=775793 2016-12-06 17:42:31 +0530 Arun Raghavan * sys/v4l2/gstv4l2object.c: v4l2object: Don't set empty interlace-mode list If for some reason we fail to probe formats (all try_fmt calls fail, for example), this is not a critical error, but we end up with an empty list of interlace modes. This causes all subsequent negotiation to fail. This patch fixes interlace-mode setting to be skipped if we failed to detect any. https://bugzilla.gnome.org/show_bug.cgi?id=775702 2016-12-07 17:22:22 +0530 Garima Gaur * gst/monoscope/gstmonoscope.c: monoscope: Unref allocation query after finished with it https://bugzilla.gnome.org/show_bug.cgi?id=775752 2016-12-07 22:55:46 +1100 Matthew Waters * ext/qt/qtitem.cc: qml/item: also unref the display on destruction Leaking objects (and a thread!) is never a good idea. https://bugzilla.gnome.org/show_bug.cgi?id=775746 2016-12-07 22:58:29 +1100 Matthew Waters * tests/examples/qt/qmlsink/main.cpp: tests/examples/qmlsink: scope QApplication/Engine So they are destroyed before gst_deinit() is run and the leaks tracer doesn't show false-positives. https://bugzilla.gnome.org/show_bug.cgi?id=775746 2016-12-06 07:48:47 +0200 Sebastian Dröge * gst/flx/gstflxdec.c: flxdec: Allocate 0-initialized memory for the decoded frame Otherwise we might leak arbitrary information from the uninitialized memory if not every pixel is written. https://scarybeastsecurity.blogspot.gr/2016/12/1days-0days-pocs-more-gstreamer-flic.html 2016-12-05 07:57:19 -0700 Matt Staples * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix session cleanup when handling redirect on PLAY Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by removing code from gst_rtspsrc_send that changed the state varable upon encountering a redirect. Better to let the redirect handlers in gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own state-dependent cleanup. https://bugzilla.gnome.org/show_bug.cgi?id=775543 2016-09-07 16:10:27 +0300 Aleix Conchillo Flaque * gst/rtsp/gstrtspsrc.c: rtspsrc: always send teardown request Allow CMD_CLOSE to cancel all commands not only CMD_PAUSE and ignore CMD_WAIT while closing. https://bugzilla.gnome.org/show_bug.cgi?id=748360 2016-12-03 08:19:27 +0100 Edward Hervey * README: * common: Automatic update of common submodule From f980fd9 to 39ac2f5 2016-12-01 17:08:09 +0100 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: Don't leak duplicate items When providing items with a seqnum, there is a (very small) probability that an element with the same seqnum already exists. Don't forget to free that item if it wasn't inserted. And avoid returning undefined values when dealing with duplicate items 2016-12-01 11:23:02 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Sanitize unknown codec caps We might have non-printable characters in the unknown fourcc, replace them with '_', in the same way we do it for unknown tags. 2016-12-01 20:04:28 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Free vprp chunk also if it existed but we made no use of it https://bugzilla.gnome.org/show_bug.cgi?id=775479 2016-12-01 17:38:33 +0200 Sebastian Dröge * gst/matroska/matroska-read-common.c: matroskademux: Fix memory leak when parsing attachments gst_tag_image_data_to_image_sample() does not take ownership of the passed memory, so don't set it to NULL to allow us to free it later. https://bugzilla.gnome.org/show_bug.cgi?id=775472 2016-12-01 14:56:18 +0200 Sebastian Dröge * gst/matroska/matroska-read-common.c: matroskademux: Unify zlib/bzip2 decompress loops with the ones from qtdemux Especially, simplify the code a bit. 2016-12-01 14:41:48 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Increase inflate buffer in bigger steps 1024 bytes is quite small, let's do 4096 bytes (or one page). Also remove redundant if, we're always in that case when getting here. 2016-12-01 14:30:49 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Ensure that size of the pasp atom is as much as we need https://bugzilla.gnome.org/show_bug.cgi?id=775455 2016-12-01 14:30:10 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Free compressed moov node and it's corresponding decompressed data https://bugzilla.gnome.org/show_bug.cgi?id=775455 2016-12-01 14:29:21 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Check size of compressed MOOV header against available data And actually read the size of the cmvd atom from the right position. https://bugzilla.gnome.org/show_bug.cgi?id=775455 2016-12-01 14:27:55 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix zlib inflate loop Handle errors cleanly, deallocate all memory and return the actual size of the inflated data. https://bugzilla.gnome.org/show_bug.cgi?id=775455 2016-12-01 13:38:16 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Make sure we have enough data in the codec_data to be able to parse it Also error out cleanly if mapping the buffer failed. https://bugzilla.gnome.org/show_bug.cgi?id=775450 2016-12-01 13:32:22 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix out of bounds read in tag parsing code We can't simply assume that the length of the tag value as given inside the stream is correct but should also check against the amount of data we have actually available. https://bugzilla.gnome.org/show_bug.cgi?id=775451 2016-12-01 15:06:06 +0530 Garima Gaur * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpsbcdepay.c: rtp: Fix some memory leaks in usage of gst_pad_get_current_caps() https://bugzilla.gnome.org/show_bug.cgi?id=775071 2016-11-30 17:56:02 +0200 Vivia Nikolaidou * gst/isomp4/qtdemux.c: qtdemux: Read interlacing information from 'fiel' atom Read interlacing and TFF/BFF information from the 'fiel' atom and pass it into the caps https://bugzilla.gnome.org/show_bug.cgi?id=775414 2016-11-29 13:55:40 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix compiler warning qtdemux.c: In function ‘qtdemux_parse_trak’: qtdemux.c:10184:38: error: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 9 has type ‘gint {aka const int}’ [-Werror=format=] GST_DEBUG_OBJECT (qtdemux, "Found jpeg: len %u, need %lu", len, ^ 2016-11-28 13:45:24 -0800 Scott D Phillips * gst/isomp4/qtdemux.c: qtdemux: Change off_t type to gint off_t is a signed integer type provided by sys/types.h on posix systems. Replace with gint for building on non-posix systems (like windows). https://bugzilla.gnome.org/show_bug.cgi?id=775287 2016-11-22 21:00:25 -0800 Scott D Phillips * meson.build: meson: add libm to has_function checks The functions from math.h may be implemented in libm. https://bugzilla.gnome.org/show_bug.cgi?id=774876 2016-10-27 23:02:37 +0530 Nirbheek Chauhan * ext/meson.build: Revert "meson: dv plugin now works on MSVC" This reverts commit 05a89613feff70cff416367f5aa807a1d5c68b63. Let's not put in stuff that needs unreleased Meson. This can go in for the next cycle. 2016-11-28 13:51:41 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Ensure that tags are valid UTF-8 before adding them to the taglist https://bugzilla.gnome.org/show_bug.cgi?id=775219 2016-11-28 12:22:49 +0200 Sebastian Dröge * gst/multipart/multipartdemux.c: multipartdemux: Post an error message on the bus if we got EOS without having added any pads 2016-11-28 12:00:09 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Handle non-UTF8 headers and error reasons more gracefully Especially don't put them into GstStructures in one way or another, just ignore them or error out cleanly depending on the importance of their content. 2016-11-28 09:30:25 +0200 Sebastian Dröge * gst/rtp/gstrtpvrawpay.c: vrawpay: Error out cleanly if mapping the video frame fails Instead of later dereferencing NULL and crashing. 2016-11-27 11:14:13 +0100 Edward Hervey * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Update statistics before pushing If an element queries the number of retransmission buffers pushed *while* the push is still taking place (and before the object lock is taken just after) it would end up with the wrong statistic being reported. Increment it just before the push, avoids races when getting statistics https://bugzilla.gnome.org/show_bug.cgi?id=768723 2016-11-26 11:20:51 +0000 Tim-Philipp Müller * .gitmodules: common: use https protocol for common submodule https://bugzilla.gnome.org/show_bug.cgi?id=775110 2016-07-28 18:51:24 +0200 Philipp Zabel * sys/v4l2/gstv4l2bufferpool.c: gstv4l2bufferpool: lock flush_stop against regular qbuf These can be called from different threads and both manipulate the pool->buffers array. Lock them properly and let flush_stop move the array contents into a temporary array on the stack to avoid having to call release_buffer under the object lock. https://bugzilla.gnome.org/show_bug.cgi?id=775015 2016-11-24 14:25:22 +0100 Philipp Zabel * sys/v4l2/gstv4l2bufferpool.c: gstv4l2bufferpool: remove critical error message when process is called on an inactive pool If the pool is inactive, it is guaranteed to also be flushing, so the following check will return GST_FLOW_FLUSHING anyway. This can happen if a v4l2src is blocking on DQBUF in create and is sent an EOS event on another thread. In that case the pool is set to flushing/inactive without locking, the v4l2src is unblocked, and may call pool_process with a valid buffer on the already inactive pool. https://bugzilla.gnome.org/show_bug.cgi?id=775014 2016-11-24 14:41:52 +0100 Philipp Zabel * sys/v4l2/gstv4l2src.c: v4l2src: release buffer if create fails gst_base_src_get_range does not expect a buffer to be returned in the error case, so we are leaking a reference here if create fails. https://bugzilla.gnome.org/show_bug.cgi?id=775014 2016-11-23 18:34:04 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Handle create_session() returning NULL in bundle code CID 1394492. 2016-11-22 16:42:55 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Make sure to only change DTS of writable buffers And trivial cleanup https://bugzilla.gnome.org/show_bug.cgi?id=774840 2016-11-22 16:42:26 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Error out much earlier if we don't have a valid PTS https://bugzilla.gnome.org/show_bug.cgi?id=774840 2016-11-22 16:18:41 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Only use buffer durations if they are actually valid https://bugzilla.gnome.org/show_bug.cgi?id=774840 2016-11-22 15:59:19 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Revert commits that set DTS and duration on buffers unconditionally 39f7e52266fde3b3c035e22cbcbb2bb1fa207b17 was setting the buffer duration to 0 if is not valid, under the assumption that this is "the last" buffer and no others are coming next. This is wrong, last_buf is the previous buffer and not the very last one. 4e3c13c87c258c9c95e2217d32ab314d12b5fffc was setting DTS to 0 if there was none. This will set DTS to 0 for all e.g. audio streams, completely messing up calculations if streams don't start at 0. https://bugzilla.gnome.org/show_bug.cgi?id=774840 2016-11-22 15:58:37 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Only write "gap" edit list if there is a non-zero gap https://bugzilla.gnome.org/show_bug.cgi?id=774840 2016-11-23 07:09:06 +1100 Matthew Waters * gst/flx/flx_color.c: * gst/flx/flx_fmt.h: * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: flxdec: rewrite logic based on GstByteReader/Writer Solves overreading/writing the given arrays and will error out if the streams asks to do that. Also does more error checking that the stream is valid and won't overrun any allocated arrays. Also mitigate integer overflow errors calculating allocation sizes. https://bugzilla.gnome.org/show_bug.cgi?id=774859 2016-11-23 11:20:49 +0200 Sebastian Dröge * gst/flx/gstflxdec.c: flxdec: Don't unref() parent in the chain function We don't own the reference here, it is owned by the caller and given to us for the scope of this function. Leftover mistake from 0.10 porting. https://bugzilla.gnome.org/show_bug.cgi?id=774897 2016-11-22 20:33:29 +0200 Sebastian Dröge * ext/vpx/gstvpxdec.c: vpxdec: libvpx's release buffer is sometimes called with fb->priv==NULL Don't assert on this but just ignore these cases. 2016-11-22 20:24:59 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Fix cluster searching if we search multiple times in one chunk After finding a cluster id in the byte reader, we skip ahead the reader position by one further byte to be able to continue searching from there inside the same chunk if the cluster candidate was a false positive. We have to accomodate for that additional byte when resuming the search, otherwise all following pulls are off-by-one for every resume and we run into an assertion. 2016-11-22 20:01:20 +0200 Sebastian Dröge * gst/matroska/matroska-ids.c: matroska: Add size checks to the parsing of FLAC headers 2016-11-22 23:46:00 +1100 Matthew Waters * gst/flx/gstflxdec.c: flxdec: fix some warnings comparing unsigned < 0 bf43f44fcfada5ec4a3ce60cb374340486fe9fac was comparing an unsigned expression to be < 0 which was always false. gstflxdec.c: In function ‘flx_decode_brun’: gstflxdec.c:322:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] if ((glong) row - count < 0) { ^ gstflxdec.c:332:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] if ((glong) row - count < 0) { ^ https://bugzilla.gnome.org/show_bug.cgi?id=774834 2016-11-21 16:17:31 +0200 Vivia Nikolaidou * gst/isomp4/gstqtmuxmap.c: qtmux: Enable up to 16 unpositioned raw audio channels https://bugzilla.gnome.org/show_bug.cgi?id=774789 2016-11-22 19:05:00 +1100 Matthew Waters * gst/flx/gstflxdec.c: flxdec: add some write bounds checking Without checking the bounds of the frame we are writing into, we can write off the end of the destination buffer. https://scarybeastsecurity.blogspot.dk/2016/11/0day-exploit-advancing-exploitation.html https://bugzilla.gnome.org/show_bug.cgi?id=774834 2016-11-21 15:25:23 +0000 David Evans * gst/isomp4/qtdemux.c: qtdemux: Be sure not to read off end of FLAC dfLa box https://bugzilla.gnome.org/show_bug.cgi?id=773712 2016-11-21 11:48:58 +0100 Nicola Murino * gst/matroska/matroska-demux.c: matroskademux: add support for skipping invalid data in push mode https://bugzilla.gnome.org/show_bug.cgi?id=774566 2016-11-21 11:48:29 +0100 Nicola Murino * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroskaparse: add support for skipping invalid data https://bugzilla.gnome.org/show_bug.cgi?id=774566 2016-11-18 17:00:59 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Move to new helper function to parse authentication responses https://bugzilla.gnome.org/show_bug.cgi?id=774416 2016-11-20 14:12:16 +0100 christophecvr * gst/isomp4/qtdemux.c: qtdemux: Fix wrong compiler warning with gcc 6.2 | ../../../git/gst/isomp4/qtdemux.c: In function 'qtdemux_parse_tree': | ../../../git/gst/isomp4/qtdemux.c:10224:24: error: 'size' may be used uninitialized in this function [-Werror=maybe-uninitialized] | offset += size; | ^~ | ../../../git/gst/isomp4/qtdemux.c:10197:25: note: 'size' was declared here | guint32 size, tag; | ^~~~ https://bugzilla.gnome.org/show_bug.cgi?id=774747 2016-11-20 16:15:07 +0000 Tim-Philipp Müller * Makefile.am: * configure.ac: * win32/MANIFEST: * win32/common/config.h: win32: remove copies of generated headers 2016-11-20 13:14:08 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Ensure that raw video have properly aligned buffers That is, aligned to to 32 bytes for video. Fixes crashes if the raw buffers are passed to SIMD processing functions. https://bugzilla.gnome.org/show_bug.cgi?id=774428 2016-11-20 13:08:27 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Ensure that raw audio and video have properly aligned buffers That is, aligned to the basic type for audio and to 32 bytes for video. Fixes crashes if the raw buffers are passed to SIMD processing functions. https://bugzilla.gnome.org/show_bug.cgi?id=774428 2016-11-14 14:44:11 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Always write edit lists for the tracks to give a more accurate duration Always write an edit list for the whole track. In general this is not necessary except for the case of having a gap or DTS adjustment but it allows to give the whole track's duration in the usually more accurate media timescale. https://bugzilla.gnome.org/show_bug.cgi?id=774403 2016-11-18 22:45:45 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Remove useless return variable qtdemux_expose_streams() returns flow error immediately, if there is an error. So, the variable for the flow return is not needed. https://bugzilla.gnome.org/show_bug.cgi?id=774674 2016-11-17 13:59:48 +0000 David Evans * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_types.c: qtdemux: Add support for FLAC encapsulated in ISOBMFF As defined by https://git.xiph.org/?p=flac.git;a=blob_plain;f=doc/isoflac.txt https://bugzilla.gnome.org/show_bug.cgi?id=773712 2016-11-17 19:59:53 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpmux.c: rtpmux: Mark pad as needing reconfiguration again if it failed And return FLUSHING instead of NOT_NEGOTIATED on flushing pads. https://bugzilla.gnome.org/show_bug.cgi?id=774623 2016-11-17 19:59:26 +0200 Sebastian Dröge * gst/monoscope/gstmonoscope.c: monoscope: Mark pad as needing reconfiguration again if it failed And return FLUSHING instead of NOT_NEGOTIATED on flushing pads. https://bugzilla.gnome.org/show_bug.cgi?id=774623 2016-11-17 19:58:52 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Mark pad as needing reconfiguration again if reconfiguration failed And consider negotiation failures on flushing pads as FLUSHING, not as NOT_NEGOTIATED. https://bugzilla.gnome.org/show_bug.cgi?id=774623 2016-11-17 19:56:23 +0200 Sebastian Dröge * ext/dv/gstdvdec.c: dvdec: Fix handling of negotiation failures Return NOT_NEGOTIATED if sending the caps event fails, or FLUSHING if the pad was flushing at that point. https://bugzilla.gnome.org/show_bug.cgi?id=774623 2016-11-17 17:16:26 -0800 Scott D Phillips * meson.build: meson: add_global_arguments -> add_project_arguments https://bugzilla.gnome.org/show_bug.cgi?id=774656 2016-11-16 10:53:51 +0530 Vinod Kesti * gst/multifile/gstsplitmuxsink.c: splitmuxsink: pad request fails for flvmux splitmuxsink requests pad from element using pad template like "video_%u", "audio_%u" and "sink_%d". This is true for most of the muxers. But splitmuxsink not able to request pad to flvmux as flvmux has "audio" and "video" as pad templates. fix: splitmuxsink should fallback to "audio" and "video" when template not found. https://bugzilla.gnome.org/show_bug.cgi?id=774507 2016-11-17 10:24:28 +0200 Sebastian Dröge * gst/matroska/matroska-parse.c: matroskaparse: Add remaining relevant parts from a3a55305 to the parser https://bugzilla.gnome.org/show_bug.cgi?id=774566 2016-11-16 22:39:01 +0100 Nicola Murino * gst/matroska/matroska-parse.c: matroskaparse: ignore parsing errors at the end of the file This is the same change as a3a55305 for the parser. https://bugzilla.gnome.org/show_bug.cgi?id=774566 2016-11-16 08:56:34 +0100 Philippe Normand * docs/plugins/gst-plugins-good-plugins.signals: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtpbundle.c: * tests/check/meson.build: * tests/examples/rtp/.gitignore: * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-rtpbundle.c: * tests/examples/rtp/server-rtpbundle.c: rtpbin: receive bundle support A new signal named on-bundled-ssrc is provided and can be used by the application to redirect a stream to a different GstRtpSession or to keep the RTX stream grouped within the GstRtpSession of the same media type. https://bugzilla.gnome.org/show_bug.cgi?id=772740 2016-11-15 16:52:39 +0530 Vinod Kesti * gst/audioparsers/gstaacparse.c: aacparse: assertion while converting ADTS stream to RAW aacparse resizes input buffer while converting ADTS stream to RAW, During buffer resize buffer write permission is not checked. This throws gst_buffer_is_writable assertion and leads to AV sync issue some times. It is corrected by making buffer writeable using gst_buffer_make_writable https://bugzilla.gnome.org/show_bug.cgi?id=774129 2016-11-15 21:17:51 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Don't modify upstream TIME segment TIME segment implies that stream/running time is being handled by upstream. So, we shouldn't override it without any clue. This patch is for fixing seek in DASH streaming. https://bugzilla.gnome.org/show_bug.cgi?id=774196 2016-11-14 22:33:27 +0530 Arun Raghavan * config.h.meson: meson: Add define for v4l2-probe config option 2016-11-14 17:37:51 +0200 Sebastian Dröge * gst/interleave/deinterleave.c: deinterleave: Reset caps accumulator to ANY when resyncing the adapter, not EMPTY The accumulator is filled by intersecting with all the pad caps, as such it must be initialized with ANY (like it is before the iteration is started) and not to EMPTY. Fixes the CAPS query always returning EMPTY caps when resyncing happened during the query, e.g. because pads were added/removed. 2016-11-14 12:13:14 +0100 Petr Kulhavy * gst/udp/gstudpsrc.c: udpsrc: remove redundant saddr unref The g_object_unref (saddr) before receiving message seems to be redundant as it is done just before jumping to retry Though not directly related, part of https://bugzilla.gnome.org/show_bug.cgi?id=772841 2016-11-12 23:34:23 +0100 Petr Kulhavy * gst/udp/gstudpsrc.c: udpsrc: receive control messages only in multicast Control messages are used only in multicast mode - to detect if the destination address is not ours and possibly drop the packet. However in non-multicast modes the messages are still allocated and freed even if not used. Therefore request control messages from g_socket_receive_message() only in multicast mode. https://bugzilla.gnome.org/show_bug.cgi?id=772841 2016-11-11 10:45:01 -0800 Scott D Phillips * gst/matroska/matroska-mux.c: Use intermediate guint when handling GstVideoMultiviewFlags The underlying integer type of the enum GstVideoMultiviewFlags is implementation defined and may not have the same size as guint. https://bugzilla.gnome.org/show_bug.cgi?id=774293 2016-11-11 10:44:18 -0800 Scott D Phillips * gst/multifile/gstsplitfilesrc.c: splitfilesrc: update uri_get_type to match the prototype in GstURIHandlerInterface https://bugzilla.gnome.org/show_bug.cgi?id=774293 2016-10-26 22:37:34 -0700 Scott D Phillips * meson.build: meson: don't add_global_arguments when being built as a subproject https://bugzilla.gnome.org/show_bug.cgi?id=773568 2016-10-21 15:49:36 +0100 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: fix header rewriting being ignored https://bugzilla.gnome.org/show_bug.cgi?id=727802 2016-11-09 06:25:27 +0000 Sean DuBois * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Add metadatacreator property Allow users to set metadatacreator value in the meta packet https://bugzilla.gnome.org/show_bug.cgi?id=774131 2016-11-01 19:56:36 +0200 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Use first buffer TS as mux start time Do not use last buffer TS + buffer duration because buffer duration might be inaccurate, especially for frame rates like 30fps where a rounding error is observed. https://bugzilla.gnome.org/show_bug.cgi?id=773785 2016-11-07 14:47:22 +0800 Haihua Hu * ext/qt/gstqtsrc.cc: * ext/qt/gstqtsrc.h: * ext/qt/qtwindow.cc: * ext/qt/qtwindow.h: qmlglsrc: some enhancements for qmlglsrc 1. Need set use-default-fbo to qquickwindow during set property to support change render target on the fly. 2. Calculate qmlglsrc refresh frame rate in qtglwindow https://bugzilla.gnome.org/show_bug.cgi?id=774035 2016-11-03 15:03:59 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix timer-reuse bug When doing rtx, the jitterbuffer will always add an rtx-timer for the next sequence number. In the case of the packet corresponding to that sequence number arriving, that same timer will be reused, and simply moved on to wait for the following sequence number etc. Once an rtx-timer expires (after all retries), it will be rescheduled as a lost-timer instead for the same sequence number. Now, if this particular sequence-number now arrives (after the timer has become a lost-timer), the reuse mechanism *should* now set a new rtx-timer for the next sequence number, but the bug is that it does not change the timer-type, and hence schedules a lost-timer for that following sequence number, with the result that you will have a very early lost-event for a packet that might still arrive, and you will never be able to send any rtx for this packet. Found by Erlend Graff - erlend@pexip.com https://bugzilla.gnome.org/show_bug.cgi?id=773891 2016-10-09 15:59:05 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix lost-event using dts instead of pts The lost-event was using a different time-domain (dts) than the outgoing buffers (pts). Given certain network-conditions these two would become sufficiently different and the lost-event contained timestamp/duration that was really wrong. As an example GstAudioDecoder could produce a stream that jumps back and forth in time after receiving a lost-event. The previous behavior calculated the pts (based on the rtptime) inside the rtp_jitter_buffer_insert function, but now this functionality has been refactored into a new function rtp_jitter_buffer_calculate_pts that is called much earlier in the _chain function to make pts available to various calculations that wrongly used dts previously (like the lost-event). There are however two calculations where using dts is the right thing to do: calculating the receive-jitter and the rtx-round-trip-time, where the arrival time of the buffer from the network is the right metric (and is what dts in fact is today). The patch also adds two tests regarding B-frames or the “rtptime-going-backwards”-scenario, as there were some concerns that this patch might break this behavior (which the tests shows it does not). 2016-11-03 16:33:53 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix bug in reschedule_timer The new timeout is always going to be (timeout + delay), however, the old behavior compared the current timeout to just (timeout), basically being (delay) off. This would happen if rtx-delay == rtx-retry-timeout, with the result that a second rtx attempt for any buffers would be scheduled immediately instead of after rtx-delay ms. Simply calculate (new_timeout = timeout + delay) and then use that instead. https://bugzilla.gnome.org/show_bug.cgi?id=773905 2016-11-03 13:27:51 +0000 Tim-Philipp Müller * tests/check/elements/wavparse.c: * tests/files/Makefile.am: * tests/files/audiotestsrc.wav: tests: wavparse: add test for processing an actual .wav file https://bugzilla.gnome.org/show_bug.cgi?id=773861 2016-11-03 12:34:51 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Don't set caps to NULL after setting them on the srcpad We would like to check later on EOS if we found a known stream type or not, to possibly post an error message. https://bugzilla.gnome.org/show_bug.cgi?id=773861 2016-10-05 12:19:12 +1100 Matthew Waters * ext/gtk/gstgtkglsink.c: gl: GST_GL_TYPE -> GST_TYPE_GL Some deprecated symbols are kept for backwards compatibility 2016-10-05 12:19:12 +1100 Matthew Waters * ext/qt/gstqtsink.cc: * ext/qt/gstqtsrc.cc: gl: GST_GL_TYPE -> GST_TYPE_GL Some deprecated symbols are kept for backwards compatibility 2016-11-02 14:33:28 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Don't deref NULL pads in debug output That tends to crash. 2016-11-02 11:46:07 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: isomp4: Don't use gst_video_colorimetry_to_string_full() The API was reverted. Just use the plain gst_video_colorimetry_to_string() function. 2016-11-02 11:00:13 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix GObject warnings on shutdown. Commit 83e718 added a pad template to splitmux request pads, which means that GstElement now releases the pads on dispose, but after having removed all elements in the bin and unlinked them. Make sure we can handle cleanup in that case without throwing assertions. https://bugzilla.gnome.org/show_bug.cgi?id=773784 2016-11-02 02:25:51 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: splitmuxsrc: Store seek seqnum and send it on EOS / segment events. GES relies on the EOS event having the seqnum of the seek that caused it. 2016-11-02 02:25:00 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Forward a not-linked error on the bus Handle not-linked as for other fatal errors and post it onto the bus so the app knows 2016-11-01 21:00:15 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix compiler warning qtdemux.c: In function ‘qtdemux_parse_tree’: qtdemux.c:10139:16: error: ‘color_table_id’ may be used uninitialized in this function [-Werror=maybe-uninitialized] if (color_table_id != 0) { ^ qtdemux.c:10121:19: note: ‘color_table_id’ was declared here guint16 color_table_id; ^~~~~~~~~~~~~~ 2016-10-20 17:40:59 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Use a default interleave of 250ms for all codecs https://bugzilla.gnome.org/show_bug.cgi?id=773217 2016-10-19 14:33:33 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Use a default interleave when ProRes is used The ProRes guidelines suggest an interleave of 0.5s is common, but specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should be used per chunk. It might also make sense to use similar numbers in general. https://bugzilla.gnome.org/show_bug.cgi?id=773217 2016-10-19 14:25:28 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Allow configuring the interleave size in bytes/time Previously we were switching from one chunk to another on every single buffer. This wastes some space in the headers and, depending on the software, might depend in more reads (e.g. if the software is reading multiple samples in one go if they're in the same chunk). The ProRes guidelines suggest an interleave of 0.5s is common, but specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should be used per chunk. This will be handled in a follow-up commit. https://bugzilla.gnome.org/show_bug.cgi?id=773217 2016-09-30 18:22:27 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Set compressor name, horizontal/vertical resolution and depth for ProRes This is also required by some software to handle ProRes files. https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-09-30 18:05:38 +0300 Sebastian Dröge * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: qt: Add support for ProRes 4444 XQ And also 4444 in the muxer. https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-09-30 17:58:37 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux_types.c: qtmux: Write 'clap' atom for ProRes It's required for ProRes to work with other software. It is also in the MP4 standard, but inventing values here seems a bit tricky for the general case and it does not really give any extra information. https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-09-30 09:55:58 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Read colorimetry information from colr atom if available https://bugzilla.gnome.org/show_bug.cgi?id=772181 2016-09-29 21:56:18 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: qtmux: Always write colr atom with the colorimetry information https://bugzilla.gnome.org/show_bug.cgi?id=772181 2016-09-29 18:16:18 +0300 Sebastian Dröge * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: Fix writing of the 'fiel' extension atom This was also wrong for JPEG2000. Also write it for all MOV files and JPEG2000, not only for ProRes. https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-09-29 17:40:23 +0300 Sebastian Dröge * gst/isomp4/atoms.c: qtmux: Write 4 bytes of zeroes at the end of the sample description extensions This is working around some broken software. https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-09-28 20:55:24 +0300 Sebastian Dröge * gst/isomp4/atoms.c: atoms: 'pasp' atom is also part of MP4, write it always https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-07-11 19:30:12 +0300 Vivia Nikolaidou * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: qtmux: Write additional atoms for prores video These required atoms are: colorimetry, field information, spatial/temporal quality, and vendor. https://bugzilla.gnome.org/show_bug.cgi?id=769048 2014-06-16 17:20:32 +0200 Stian Selnes * gst/rtp/gstrtph263depay.c: rtph263depay: Don't drop mode b packets with picture start code Some buggy payloaders, e.g. rtph263pay, may use mode B for packets that starts with a picture (or GOB) start code although it's not allowed. Let's be nice and not drop these packets/frames. https://bugzilla.gnome.org/show_bug.cgi?id=773516 2016-06-22 13:59:35 +0200 Havard Graff * gst/rtp/gstrtph263ppay.c: * tests/check/elements/rtph263.c: rtph263ppay: Fix caps leak Fix leaking caps when downstream has not-fixed caps. https://bugzilla.gnome.org/show_bug.cgi?id=773515 2016-10-26 16:42:19 +0200 Stian Selnes * gst/rtp/gstrtph263pay.c: rtph263pay: Fix indentation https://bugzilla.gnome.org/show_bug.cgi?id=773514 2016-10-18 11:35:58 +0200 Stian Selnes * gst/rtp/gstrtph263pay.c: rtph263pay: Use GST_TRACE_OBJECT for logging bitstream parsing Bump the bitstream parsing to TRACE log level so it doesn't flood the output when trying to read the more useful DEBUG and LOG messages. Also use GST_DEBUG_OBJECT instead of GST_DEBUG in various places https://bugzilla.gnome.org/show_bug.cgi?id=773514 2016-10-18 11:09:10 +0200 Stian Selnes * gst/rtp/gstrtph263pay.c: rtph263pay: Fix leak for B-fragments Altough commits 6a16be7, 64f9d08 and 0c7e3a8 fixed some issues they introduced others. This patch fixes the leak of one macroblock for every B fragment. Macroblock structures must not be freed immediately after finding the boundaries as they are stored and used later. However the inital dummy structure (used for finding the first boundary) must be freed. CID #1212156 https://bugzilla.gnome.org/show_bug.cgi?id=773512 2016-10-20 13:14:13 +0200 Alejandro G. Castro * gst/rtpmanager/rtpsession.c: rtpbin: avoid generating errors when rtcp messages are empty and check the queue is not empty Add a check to verify all the output buffers were empty for the session in a timout and log an error. https://bugzilla.gnome.org/show_bug.cgi?id=773269 2016-10-26 13:21:29 +0200 Alejandro G. Castro * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpbin: pipeline gets an EOS when any rtpsources byes Instead of sending EOS when a source byes we have to wait for all the sources to be gone, which means they already sent BYE and were removed from the session. We now handle the EOS in the rtcp loop checking the amount of sources in the session. https://bugzilla.gnome.org/show_bug.cgi?id=773218 2016-10-21 17:31:00 +0000 Matt Staples * gst/rtsp/gstrtspsrc.c: rtspsrc: Also handle redirect on PLAY https://bugzilla.gnome.org/show_bug.cgi?id=772610 2016-08-30 10:24:43 +0200 Petr Kulhavy * gst/rtsp/gstrtspsrc.c: rtspsrc: allow missing control attribute in case of a single stream Improve RFC2326 - chapter C.3 compatibility: In case just a single stream is specified in SDP and the control attribute is missing do not drop the stream but rather assume "a=control:*" https://bugzilla.gnome.org/show_bug.cgi?id=770568 2016-10-08 18:11:17 +0200 William Manley * sys/v4l2/gstv4l2allocator.c: v4l2: Warn, don't assert if v4l gives us a buffer with a too large size I've seen problems where the `bytesused` field of `v4l2_buffer` would be a silly number causing the later call to: gst_memory_resize (group->mem[i], 0, group->planes[i].bytesused); to result in this error to be printed: (pulsevideo:11): GStreamer-CRITICAL **: gst_memory_resize: assertion 'size + mem->offset + offset <= mem->maxsize' failed besides causing who-knows what other problems. We make the assumption that this buffer has still been dequeued correctly so just clamp to a valid size so downstream elements won't end up in undefined behaviour. The invalid `v4l2_buffer` I saw from my capture device was: buffer = { index = 0, type = 1, bytesused = 534748928, // <- Invalid flags = 8260, // V4L2_BUF_FLAG_TIMESTAMP_MONOTONIC | V4L2_BUF_FLAG_ERROR | V4L2_BUF_FLAG_DONE field = 01330, // <- Invalid timestamp = { tv_sec = 0, tv_usec = 0 }, timecode = { type = 0, flags = 0, frames = 0 '\000', seconds = 0 '\000', minutes = 0 '\000', hours = 0 '\000', userbits = "\000\000\000" }, sequence = 0, memory = 2, m = { offset = 3537219584, userptr = 140706665836544, // Could be nonsense, not sure planes = 0x7ff8d2d5b000, fd = -757747712 }, length = 2764800, reserved2 = 0, reserved = 0 } This is from gdb with my own annotations added. This was with gst-plugins-good 1.8.1, a Magewell XI100DUSB-HDMI video capture device and kernel 3.13 using a dodgy HDMI cable which is great at breaking HDMI capture devices. I'm using io-mode=userptr and have built gst-plugins-good without libv4l. https://bugzilla.gnome.org/show_bug.cgi?id=769765 2016-10-20 20:41:07 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Use a better default value for the movie header timescale Take the maximum video timescale, or if no video track is present the previous value of 1800. https://bugzilla.gnome.org/show_bug.cgi?id=769041 2016-10-20 20:07:19 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Be more clever with the default video track timescale Use the number of milliframes per second for integral and drop-frame framerates, as suggested by the QT file format specification and other places. We already did that for integral framerates before, but not for drop-frame framerates. This now keeps precision better. For all other framerates, check if it's close to a well-known framerate and use that instead. https://bugzilla.gnome.org/show_bug.cgi?id=769041 2016-10-10 13:00:01 +0100 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: extract interlaced information from jpeg video This information is hidden in a small chunk of data. Format found at https://developer.apple.com/standards/qtff-2001.pdf, page 92, "Video Sample Description", under table 3.1. https://bugzilla.gnome.org/show_bug.cgi?id=767771 2016-10-26 12:46:28 +0530 Jagadish * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: Fixing x and y offset computation While computing the x and y offsets, it's the video resolution and resized overlay resolution to be used instead of actual overlay image resoltuion. Due to this, the overlay image used to get wrongly overlayed in undesired location https://bugzilla.gnome.org/show_bug.cgi?id=757292 2016-11-01 18:09:00 +0000 Tim-Philipp Müller * meson.build: meson: update version 2016-10-24 16:56:31 +0000 Enrique Ocaña González * gst/isomp4/qtdemux.c: qtdemux: Use the tfdt decode time on byte streams when it's significantly different than the time in the last sample We consider there's a sifnificant difference when it's larger than on second or than half the duration of the last processed fragment in case the latter is larger. https://bugzilla.gnome.org/show_bug.cgi?id=754230 === release 1.11.0 === 2016-11-01 18:53:15 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.10.0 === 2016-11-01 17:57:44 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.10.0 2016-11-01 17:47:31 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2016-11-01 17:41:51 +0200 Sebastian Dröge * po/el.po: po: Update translations 2016-10-27 12:01:55 +0200 Tobias Schneider * sys/v4l2/gstv4l2object.c: v4l2object: fix extra-controls leak Gst struct v4l2object->extra_controls is created if user sets appropriate option but it is not freed on destruction of v4l2object. https://bugzilla.gnome.org/show_bug.cgi?id=773580 2016-10-31 18:00:07 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: Revert "souphttpsrc: reduce reading latency by using non-blocking read" This reverts commit 8816764112408766889c8b680a3af51115df4bf5. It causes issues with the timeouts, and causes connections to be closed without actual reason. Needs further investigation. https://bugzilla.gnome.org/show_bug.cgi?id=773509 2016-10-31 09:00:49 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Don't try to add srcpad if we don't know valid caps yet Otherwise we'll run into an assertion on specially crafted files. https://bugzilla.gnome.org/show_bug.cgi?id=773643 2016-10-27 09:11:26 +0530 Nirbheek Chauhan * gst/audiofx/gststereo.c: Explicitly define float constants as float With MSVC, this gives the following warning: warning C4305: 'function': truncation from 'double' to 'gfloat' Apparently, MSVC does not figure out what type to use for constants based on the assignment. This warning is very spammy, so let's try to fix it. 2016-10-27 11:23:51 +0530 Nirbheek Chauhan * meson.build: meson: Remove uselessly duplicated dep checks These checks are done inside the meson.build files for each plugin. 2016-10-27 11:22:59 +0530 Nirbheek Chauhan * ext/meson.build: meson: dv plugin now works on MSVC Needs a Meson patch to filter out the useless -lpthread https://github.com/mesonbuild/meson/pull/962 2016-10-27 14:03:48 +0200 Branko Subasic * gst/matroska/matroska-mux.c: matroskamux: allow resolutions above 4096 Modify the caps string to allow width and height greater than 4096. There is no need to restrict it since the matroska format allows the width and height values to be up to eight bytes long. https://bugzilla.gnome.org/show_bug.cgi?id=773582 2016-10-23 17:23:10 -0700 Scott D Phillips * gst/udp/gstudpsrc.c: udpsrc: Check for G_PLATFORM_WIN32 for presence of ipi_spec_dest G_OS_WIN32 is only set when not building with cygwin, but ipi_spec_dest is missing both with and without cygwin. https://bugzilla.gnome.org/show_bug.cgi?id=773114 2016-10-26 08:51:40 +0200 Michael Olbrich * ext/soup/gstsouphttpsrc.c: souphttpsrc: reset read_position when reading fails souphttpsrc maintains two variables for the position: * 'request_position' is where we want to be * 'read_position' is where we are During Normal operations both are updated in sync when data arrives. A seek changes 'request_position' but not 'read_position'. When the two positions get out of sync, then a new request is send and the 'Range' header is adjusted to the current 'request_position'. Without this patch, if reading fails, then the source is destroyed. This triggers a new request, but the range remains unchanged. As a result, the old range is used and old data will be read. Changing the 'read_position' to -1 makes it explicitly different from 'request_position' and as a result the 'Range' header is updated correctly. https://bugzilla.gnome.org/show_bug.cgi?id=773509 2016-10-25 08:54:34 -0700 Scott D Phillips * meson.build: meson: Don't depend on gstreamer-check-1.0 on windows https://bugzilla.gnome.org/show_bug.cgi?id=773114 2016-10-25 15:24:20 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: reset connection info to non-flushing when closing This solves a hanging mainloop in following scenario: * connect to source * network/server drops * pipeline set to NULL (and connection to flushing as part) * pipeline set to PAUSED/PLAYING (connection to non-flushing, but not recorded) * [connecting still not possible] * pipeline set to NULL => mainloop hangs (since no actual flushing is done) 2016-10-26 14:32:48 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Only allow one video request pad The pacing of the overall muxing is controlled by the video GOPs arriving, so we can only handle 1 video stream, and the request pad is named accordingly. Ignore a request for a 2nd video pad if there's already an active one. 2016-10-26 11:59:32 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Take ownership of floating refs sink the floating ref when handed a muxer or sink to use so we clearly take ownership. 2016-10-25 14:51:52 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Set child elements to NULL when removing. Make sure that elements are in the NULL state when removing. Fixes critical warnings when errors occur early on in starting up. 2016-10-25 14:50:53 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Set pad template on request sink pads Ensure that the ghost pad returned as a request pad has the template that was requested 2016-10-25 10:50:47 +0530 Nirbheek Chauhan * meson.build: * tests/check/meson.build: Revert "meson: move gstreamer-check-1.0 dependency to tests/check" This reverts commit 46632694662b96fddb848a1f2091a215b28a2d35. Does not actually work. See: https://bugzilla.gnome.org/show_bug.cgi?id=773114#c31 2016-06-08 11:24:37 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: flvmux: Assume PTS is DTS when PTS is missing This fixes issue for encoders that only sets the DTS. We assume that there was no re-ordering when that happens. https://bugzilla.gnome.org/show_bug.cgi?id=762207 2016-10-24 00:34:15 +0100 Tim-Philipp Müller * tests/check/meson.build: meson: fix build outside of gst-all 2016-10-21 00:42:54 -0700 Scott D Phillips * sys/directsound/meson.build: meson: directsound: Add ole32 library dependency https://bugzilla.gnome.org/show_bug.cgi?id=773114 2016-10-21 00:42:18 -0700 Scott D Phillips * meson.build: * tests/check/meson.build: meson: move gstreamer-check-1.0 dependency to tests/check https://bugzilla.gnome.org/show_bug.cgi?id=773114 2016-10-20 22:08:14 +0100 Tim-Philipp Müller * tests/check/elements/videomixer.c: tests: videomixer: disable racy flush_start_flush_stop test It's been broken for years, and it's unlikely it will ever be fixed for collectpads/videomixer now that there's compositor which works fine. So let's disable it, since all it does is that it creates noise that distracts from other failures. Also see the corresponding adder bug as it failed in the same way: https://bugzilla.gnome.org/show_bug.cgi?id=708891 2016-10-09 16:56:10 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/souphttpsrc.c: tests: Fix souphttpsrc tests without CK_FORK=no It seems that the forked processes all attempt to handle the listening socket from the server, and only one has to shutdown the socket to break the server completely. Create a new server inside each test to avoid this. https://bugzilla.gnome.org/show_bug.cgi?id=772656 2016-10-09 15:23:51 +0200 Jan Alexander Steffens (heftig) * tests/check/elements/level.c: tests: Fix level test in CK_FORK=no mode The tests accumulate buffers in GstCheck's buffers list, and the list is not (consistently) reset between tests. Do that and remove the now conflicting unrefs for outbuffers. https://bugzilla.gnome.org/show_bug.cgi?id=772644 2016-10-07 13:04:27 +0530 Gaurav Gupta * sys/waveform/gstwaveformsink.c: waveformsink: Fix Memory leak using GST_PTR_FORMAT https://bugzilla.gnome.org/show_bug.cgi?id=772497 2016-10-18 12:23:42 +0530 Nirbheek Chauhan * gst/monoscope/meson.build: meson: Add missing gstaudio dep to monoscope In file included from ../subprojects/gst-plugins-good/gst/monoscope/gstmonoscope.c:42:0: ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory #include ^ compilation terminated. https://ci.gstreamer.net/job/GStreamer-master-meson/271/console 2016-10-16 12:40:22 +0200 Sergey Borovkov * ext/qt/qtwindow.cc: qt: Fix failing build on RPI https://bugzilla.gnome.org/show_bug.cgi?id=773026 2016-10-16 02:18:22 +0530 Nirbheek Chauhan * gst/multifile/meson.build: meson: Add missing pbutils dependency to multifile Found via the Jenkins CI: FAILED: subprojects/gst-plugins-good/gst/multifile/gstmultifile@sha/gstsplitmuxsink.c.o [...] In file included from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.h:24:0, from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.c:59: ../subprojects/gst-plugins-base/gst-libs/gst/pbutils/pbutils.h:30:43: fatal error: gst/pbutils/pbutils-enumtypes.h: No such file or directory #include ^ compilation terminated. https://ci.gstreamer.net/job/GStreamer-master-meson/263/console 2016-10-15 22:11:08 +0530 Nirbheek Chauhan * meson.build: meson: Don't set c_std to gnu99 Use the default for each compiler on every platform instead. This improves our compatibility with compilers that don't have gnu99 as a c_std. 2016-10-04 18:04:11 -0300 Thibault Saunier * meson.build: * tests/check/getpluginsdir: * tests/check/meson.build: meson: Make use of new environment object and set plugin path to builddir Workaround source_root being the root directory of all projects in the subproject case and remove now unneeded getpluginsdir Bump meson requirement to 0.35 2016-10-06 11:15:54 +0530 Gaurav Gupta * tests/examples/rtp/client-rtpaux.c: tests: Fix memory leak in test rtpaux test https://bugzilla.gnome.org/show_bug.cgi?id=772496 2016-10-03 11:27:54 +0530 Nirbheek Chauhan * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Forward latency queries to upstream Without this, latency queries to imagefreeze will fail. 2016-09-30 11:35:39 -0300 Thibault Saunier * hooks/pre-commit.hook: * meson.build: * tests/check/getpluginsdir: meson: Setup pre commit hook and fix getpluginsdir for standalone case 2016-09-29 04:55:14 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Handle stop point from segment If the seek stop point (or start, during reverse play) was within the segment we just finished, go EOS immediately instead of proceeding through all other parts and sending 0 length seeks to them. https://bugzilla.gnome.org/show_bug.cgi?id=772138 2016-09-29 03:21:26 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Drop lock shutting down pads Avoid a sporadic deadlock on shutdown by dropping the splitmux lock around pad shutdown https://bugzilla.gnome.org/show_bug.cgi?id=772138 2016-09-29 02:47:36 +1000 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: splitmuxsrc: Fix extra unref handling queries https://bugzilla.gnome.org/show_bug.cgi?id=772138 2016-09-29 04:50:25 +1000 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Avoid stall when parts get out of sync When one part moves ahead of the others - due to excessive downstream queueing, or really small input files - then we can end up activating parts more than once. That can lead to effects like shutting down pad tasks prematurely. https://bugzilla.gnome.org/show_bug.cgi?id=772138 2016-09-30 11:41:19 +0100 Tim-Philipp Müller * meson.build: meson: update version === release 1.9.90 === 2016-09-30 13:02:19 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.9.90 2016-09-30 12:17:26 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2016-09-30 11:43:54 +0300 Sebastian Dröge * po/el.po: po: Update translations 2016-09-30 13:22:32 +0530 Arun Raghavan * tests/check/pipelines/tagschecking.c: tests: Fix tagschecking failure due to missing PTS qtmux now needs the PTS (commit a993883b7), so let's make sure we produce one with our buffers. https://bugzilla.gnome.org/show_bug.cgi?id=772228 2016-09-28 23:03:58 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE Just error out if there is no valid PTS. https://bugzilla.gnome.org/show_bug.cgi?id=772143 2016-09-29 17:37:28 +0300 Sebastian Dröge * gst/isomp4/qtdemux_types.c: qtdemux: Add JPEG2000 ihdr atom to the list of known ones Otherwise qtdemux is always going to complain about it being unknown. 2016-09-29 10:19:56 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Always write the default frame duration for VP8/9 too The WebM spec allows this now, and it allows us to guess a framerate. See https://bugzilla.gnome.org/show_bug.cgi?id=772141 and also https://bugzilla.gnome.org/show_bug.cgi?id=654379 2016-09-27 15:26:19 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph265depay.c: rtph26[45]depay: Don't handle NALs inside STAP units twice They've already been handled before pushing them into the adapter. 2016-09-27 12:39:12 +0100 Tim-Philipp Müller * tests/check/meson.build: meson: tests: fix vp8 availability checks Those variables are not defined if vp8 was not found. 2016-09-27 10:23:38 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: Revert "multifilesink: streamline the file-switch code a bit" This reverts commit f1ceaab02f3f557e23b77b14771a575788f92bb4. This broke atomic file writes in "buffer" mode. It did make sure that any streamheaders are prepended to each file in buffer mode as well, but that's not really needed in practice, whereas atomic file writes are, so let's restore the status quo ante for now since this was primarily a code cleanup anyway, and if anyone needs to streamheaders in buffer mode too they can make a patch to implement that differently. Re-implementing the atomic writes in the element also seems way too much work. https://bugzilla.gnome.org/show_bug.cgi?id=766990 2016-09-27 10:22:57 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: Revert "multifilesink: close file on write error with next-file mode is set to buffer" This reverts commit 84e441d2685cf223d348a95be0c5ba693bbf6624. This will no longer be needed once we revert f1ceaab02. 2016-09-26 13:22:29 -0300 Thibault Saunier * tests/check/meson.build: meson: Add gst-plugins-base plugins directories to be used by tests 2016-09-26 14:30:00 +0100 Tim-Philipp Müller * ext/vpx/meson.build: * meson.build: * tests/check/getpluginsdir: * tests/check/meson.build: meson: add unit tests Only works properly in an installed setup currently, most likely won't work with a subprojects setup yet. 2016-09-24 09:36:24 +0100 Tim-Philipp Müller * meson.build: * po/meson.build: meson: hook up translations 2016-09-08 17:30:41 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Don't negotiate to less than two segments GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make sure that if our buffer parameters are such that the maxlength is not at least 2x fragsize, we still request the ringbuffer to keep that much space so it continues to work. https://bugzilla.gnome.org/show_bug.cgi?id=770446 2016-09-24 23:22:01 +0530 Arun Raghavan * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Fix timestamping We were just picking the timestamp of the last buffer pushed into our adapter before we had enough data to push out. This fixes things to figure out how large each frame is and what duration it covers, so we can set both the timestamp and duration correctly. Also adds some DISCONT handling. 2016-07-12 18:14:52 +0200 Georg Lippitsch * gst/isomp4/gstqtmux.c: qtmux: Fix fourcc for ProRes Proxy This is apco, according to https://wiki.multimedia.cx/index.php?title=Apple_ProRes https://bugzilla.gnome.org/show_bug.cgi?id=769048 2016-09-18 20:55:31 +0100 Tim-Philipp Müller * ext/vpx/meson.build: meson: fix build with vpx 1.3.x vpx >= 1.4.0 is optional 2016-09-15 18:19:35 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Use new bin suppressed flags API for managing the element flags 2016-09-15 09:52:31 +0100 Tim-Philipp Müller * ext/jack/gstjackaudioclient.c: * gst/rtp/dboolhuff.c: * gst/rtpmanager/rtpsession.c: * gst/videofilter/gstvideoflip.c: ext, gst: fix indentation 2016-09-15 09:52:17 +0100 Tim-Philipp Müller * tests/check/elements/flvmux.c: * tests/check/elements/rtph263.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/rtpsession.c: * tests/check/elements/rtpvp9.c: tests: fix indentation 2016-08-11 11:04:22 -0600 Thomas Bluemel * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue. Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one definitely lost packets is encountered. https://bugzilla.gnome.org/show_bug.cgi?id=769757 2016-08-11 23:07:44 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: improved rtx-rtt averaging The basic idea is this: 1. For *larger* rtx-rtt, weigh a new measurement as before 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less 3. For very large measurements, consider them "outliers" and count them a lot less The idea being that reducing the rtx-rtt is much more harmful then increasing it, since we don't want to be underestimating the rtt of the network, and when using this number to estimate the latency you need for you jitterbuffer, you would rather want it to be a bit larger then a bit smaller, potentially losing rtx-packets. The "outlier-detector" is there to prevent a single skewed measurement to affect the outcome too much. On wireless networks, these are surprisingly common. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-08-05 12:51:59 +0200 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Detect whether to assume equidistant spacing when loss Assuming equidistant packet spacing when that's not true leads to more loss than necessary in the case of reordering and jitter. Typically this is true for video where one frame often consists of multiple packets with the same rtp timestamp. In this case it's better to assume that the missing packets have the same timestamp as the last received packet, so that the scheduled lost timer does not time out too early causing the packets to be considered lost even though they may arrive in time. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-07-27 10:39:50 +0200 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Don't request rtx if 'now' is past retry period There is no need to schedule another EXPECTED timer if we're already past the retry period. Under normal operation this won't happen, but if there are more timers than the jitterbuffer is able to process in real-time, scheduling more timers will just make the situation worse. Instead, consider this packet as lost and move on. This scenario can occur with high loss rate, low rtt and high configured latency. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-07-26 18:01:48 +0200 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix lost duration when gap after lost timer This patch fixes an issue with the estimated gap duration when there is a gap immediately after a lost timer has been processed. Previously there was a discrepancy beteen the gap in seqnum and gap in dts which would cause wrong calculated duration. The issue would only be seen with retranmission enabled since when it's disabled lost timers are only created when a packet is received and the actual gap length and last dts is known. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-07-19 01:11:58 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Expose rtx-deadline as a property The default -1 gives the old behavior. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-08-11 12:02:19 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Improved expected-timer handling when gap > 0 https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-08-11 11:51:50 +0200 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Major improvements for RTX stats Stats should also be collected for unsuccessful packets. rtx-rtt is very important for determining the necessary configured latency on the jitterbuffer. It's especially important to be able to increase the latency when retransmitted packets arrive too late and are considered lost. This patch includes these late packets in the calculation of the various rtx stats, making them more correct and useful. Also in the case where the original packet arrives after a NACK is sent, the received RTX packet should update the stats since it provides useful information about RTT. The RTT is only updated if and only if all requested retranmissions are received. That way the RTT is guaranteed to make sense. If not we don't know which request the packet is a response to and the RTT may be bogus. A consequence of this patch is that RTT is not updated for a request when one of the RTX packets for that seqnum is lost, but that since measured RTT will be more accurate. The implementation store the RTX information from the timed out timers and use this when the retransmitted packet arrives. For performance these timers are stored separately from the "normal" timers in order to not impact performance (see attached performance test). https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-08-11 11:02:44 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Add and expose more stats and increase testing of it Add num-pushed and num-lost. Expose num-late, num-duplicates and avg-jitter. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-07-07 10:20:02 +0200 Stian Selnes * gst/rtpmanager/gstrtprtxreceive.c: rtxreceive: Set buffer flag for retransmitted packets https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-07-09 23:47:41 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Option to disable rtx-delay-reorder When disabled we can save some iterations over timers. There is probably an argument for rtx-delay-reorder to exist, but for normal operations, handling jitter (reordering) is something a jitterbuffer should do, and this variable feels like functionality that is not "in-sync" with what the jitterbuffer is trying to achieve. Example: You have 50ms jitter on your network, and are receiving audio packets with 10ms durations. An audio packet should not be considered late until its rtx-timeout has expired (and hence a rtx-event is sent), but with rtx-delay-reorder, events will be sent pretty much all the time due to the jitter on the network. Point being: The jitterbuffer should adapt its size to the measured network jitter, and then rtx-delay-reorder needs to adapt as well, or simply get out of the way and let the other (better) rtx-mechanisms do their job. Also change find_timer to only use seqnum as an argument, since there will only ever be one timer per seqnum at any given time. In the one case where the type matters, the caller simply checks the type. https://bugzilla.gnome.org/show_bug.cgi?id=769768 2016-09-14 09:58:41 -0400 Olivier Crête * gst/rtp/gstrtph263pay.c: rtph263pay: Fix double free from coverity CID #1372887 2016-09-14 09:58:37 -0400 Olivier Crête * gst/rtp/gstrtph263pay.c: rtph263pay: Indent as per gst-indent 2016-09-14 11:30:41 +0200 Sebastian Dröge * configure.ac: configure: Depend on gstreamer 1.9.2.1 2016-09-14 10:17:02 +0900 Wonchul Lee * gst/autodetect/gstautodetect.c: autodetect: Use gst_bin_set_suppressed_flags() API https://bugzilla.gnome.org/show_bug.cgi?id=771395 2016-09-09 15:36:12 +0200 Thomas Scheuermann * ext/jack/gstjackaudioclient.c: jack: Fix pipeline hang when jack changes sample rate or buffer size If jackd changes the buffer size or sample rate, jackaudiosink hangs and can't be stopped. This also happens if jack is configured as slave and a gstreamer pipeline is started on the slave machine while the jack master isn't running yet. If the the jack master is started it changes the buffer size / sample rate and jackaudiosink can't be stopped. This fix calls jack_shutdown_cb when jack_sample_rate_cb or jack_buffer_size_cb is called. https://bugzilla.gnome.org/show_bug.cgi?id=771272 2016-09-12 20:08:36 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix field ordering for reverse playback And actually calculate the field duration instead of a frame duration so that we can properly timestamp output frames in fields=all mode. This is probably still broken for reverse playback in telecine mode. 2016-09-12 09:02:00 +0000 Thomas Klausner * gst/udp/gstudpsrc.c: udpsrc: Fix compilation on NetBSD https://bugzilla.gnome.org/show_bug.cgi?id=771278 2016-09-10 20:51:10 +1000 Jan Schmidt * autogen.sh: * common: Automatic update of common submodule From b18d820 to f980fd9 2016-09-09 14:02:25 +0200 Xabier Rodriguez Calvar * gst/isomp4/qtdemux.c: qtdemux: offset is irrelevant when no crypto info Cause later it will try to use the crypto info array to get an index and attach on of the positions as buffer's crypto info. https://bugzilla.gnome.org/show_bug.cgi?id=770951 2016-09-10 09:53:57 +1000 Jan Schmidt * autogen.sh: * common: Automatic update of common submodule From f49c55e to b18d820 2016-09-09 16:36:03 +1000 Matthew Waters * ext/gtk/meson.build: meson: add build files for the gtk plugin 2016-09-07 15:33:30 -0400 Nicolas Dufresne * sys/osxaudio/Makefile.am: osxaudio: Distribute device provider files Those where missing the the dev release tarballs for 1.9.2 which prevented building from tarball on OSX platform 2016-09-06 09:49:39 +0200 Xabier Rodriguez Calvar * gst/isomp4/qtdemux.c: qtdemux: Fix crash with no cenc aux offset https://bugzilla.gnome.org/show_bug.cgi?id=770951 2016-09-06 13:13:39 +0800 Haihua Hu * ext/qt/gstqsgtexture.cc: qmlglsink: check qt_context_ first in GstQSGTexture::bind() When start qmlglsink app, it will set NULL buffer to GstQSGTexture in which case that qt_context_ will be a random value and cause gst_gl_context_activate() fail. https://bugzilla.gnome.org/show_bug.cgi?id=770925 2016-09-05 09:39:33 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: parse a bit more of the humongous LOAS data https://bugzilla.gnome.org/show_bug.cgi?id=769278 2016-09-05 09:39:08 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: make it clear when a potential LOAS frame is not one https://bugzilla.gnome.org/show_bug.cgi?id=769278 2016-09-05 09:38:26 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: add a few comments to anchor parsing to the spec https://bugzilla.gnome.org/show_bug.cgi?id=769278 2016-09-05 09:37:02 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: improve channel/rate handling Keep track of the last parsed channels/rate fields so they can be used even if the element was not yet configured. https://bugzilla.gnome.org/show_bug.cgi?id=769278 2016-09-05 09:35:53 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: fix varlength number reading as per spec https://bugzilla.gnome.org/show_bug.cgi?id=769278 2016-09-05 09:35:02 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: strip uneeded static arrays slack https://bugzilla.gnome.org/show_bug.cgi?id=769278 2016-07-18 19:18:58 -0400 Olivier Crête * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4adepay.h: rtpmp4adepay: Only declare a stream to be framed once a marker bit has been seen This may cause a few packets to be processed by the parser, but it's better than never pushing out buffers from a slightly broken stream where no marker bits are set. 2016-09-06 14:25:42 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Fix timestamping in reverse playback mode This is only supported right now if after a demuxer that supports reverse playback, e.g. with DV container inside AVI container. 2016-09-05 12:23:54 -0300 Thibault Saunier * meson.build: meson: Bump version to 1.9.2 2015-06-26 20:13:17 +0200 Mathieu Duponchelle * gst/isomp4/GstQTMux.prs: * gst/isomp4/Makefile.am: * gst/isomp4/gstqtmux.c: qtmux: Implement the preset interface. + And provide a "youtube" preset, which based on https://support.google.com/youtube/answer/1722171 sets faststart to True. https://bugzilla.gnome.org/show_bug.cgi?id=751559 2016-09-01 12:27:35 +0300 Sebastian Dröge * configure.ac: Back to development === release 1.9.2 === 2016-09-01 12:27:15 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.9.2 2016-09-01 11:23:33 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: Update translations 2016-09-01 10:59:51 +0300 Sebastian Dröge * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS We use gdk_cairo_create() which is deprecated since 3.22. 2016-08-31 05:50:44 +1000 Jan Schmidt * sys/osxvideo/Makefile.am: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/osxvideosink.h: osxvideo: Remove QuickTime references. QuickTime.h is no longer available on OS X 10.12 (Sierra), and both the header and the framework seem unnecessary for compilation - at least as of 10.11 (El Capitan). https://bugzilla.gnome.org/show_bug.cgi?id=770526 2016-08-19 11:11:03 -0700 Thibault Saunier * ext/dv/gstdvdemux.c: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * gst/avi/gstavidemux.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/multifile/gstsplitmuxsrc.c: * gst/rtsp/gstrtspsrc.c: * gst/wavparse/gstwavparse.c: Use the new API to post flow ERROR messages on the bus https://bugzilla.gnome.org/show_bug.cgi?id=770158 2016-08-26 21:32:07 +0200 Josep Torra * tests/check/elements/.gitignore: gitignore: ignore qtdemux, rtph261 and rtpvp9 tests 2016-08-26 21:22:16 +0200 Josep Torra * tests/check/Makefile.am: tests: use GST_NET_LIBS instead of hardcoded -lgstnet Fixes build in OSX when running 'make check' in gst-uninstalled. 2016-08-26 21:14:47 +0200 Josep Torra * tests/check/elements/rtp-payloading.c: tests: remove a wrong 'const' specifier Fixes "error: duplicate 'const' declaration specifier" 2016-08-26 21:11:59 +0200 Josep Torra * configure.ac: * tests/check/Makefile.am: build: silence error about pthread for 'make check' in osx Fixes "clang: error: argument unused during compilation: '-pthread'" 2016-08-26 20:31:10 +0300 Sebastian Dröge * tests/check/Makefile.am: vp9enc: Fix build of unit test by letting it link to libgstvideo 2016-08-26 12:06:35 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems" This broke API, so we need a better solution! This reverts commit c7579d31a6e9d788e94b83258309063d0aae481e. 2016-06-08 15:06:28 +0200 Stian Selnes * gst/rtp/gstrtpvp9depay.c: * tests/check/Makefile.am: * tests/check/elements/rtpvp9.c: rtpvp9depay: Support flexible mode 2016-06-06 17:03:36 +0200 Stian Selnes * ext/vpx/gstvp9enc.c: * tests/check/Makefile.am: * tests/check/elements/vp9enc.c: vp9enc: Fix leak of vpx_image_t 2016-05-06 13:33:22 +0200 Stian Selnes * gst/rtp/gstrtph263pdepay.c: * tests/check/elements/rtph263.c: rtph263pdepay: Don't try to push empty frame If the result of depayloading is an empty frame, just drop it. This is likely the result of a buggy payloader. 2016-05-06 16:06:53 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: fix PROP_TIMESTAMP_OFFSET range problems It could not set the offset for the full guint32 range. 2016-05-06 09:44:42 +0200 Havard Graff * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: introduce max-streams property To be able to cap the number of allowed streams for one session. This is useful for preventing DoS attacks, where a sender can change SSRC for every buffer, effectively bringing rtpbin to a halt. https://bugzilla.gnome.org/show_bug.cgi?id=770292 2016-03-31 00:10:49 +0200 Havard Graff * gst/rtpmanager/rtpsource.c: rtpsource: reordered packets are very normal, and should not be a warning 2016-02-05 14:19:25 +0100 Havard Graff * gst/rtpmanager/rtpsession.c: rtpsession: degrade g_warning to GST_ERROR So we don't blow up while investigating 2016-02-04 14:16:40 +0100 Stian Selnes * gst/rtp/gstrtph263pdepay.c: * tests/check/elements/rtph263.c: rtph263pdepay: Fix picture header for non-writable payload Under certain conditions gst_rtp_buffer_get_payload() returns a copy of the payload. In this case the payload modifications will not affect the rtp buffer. So instead of modifying the payload buffer directly we should modify the buffer that actually gets pushed on the adapter. 2015-11-19 11:50:47 +0100 Stian Selnes * gst/rtp/gstrtph261depay.c: * tests/check/Makefile.am: * tests/check/elements/rtph261.c: rtph261depay: Fix check of valid payload length Packets with no H.261 payload should be dropped to avoid invalid write/reads. 2015-11-09 10:06:21 +0100 Stian Selnes * gst/rtp/gstrtph263pay.c: * tests/check/elements/rtph263.c: rtph263pay: Fix double free, invalid reads and leak 2014-06-30 15:43:58 +0200 Stian Selnes * gst/rtpmanager/rtpsession.c: rtpsession: sanity check RTT before ignoring PLI/FIR 2014-06-30 15:07:45 +0200 Stian Selnes * gst/rtpmanager/rtpsession.c: rtpsession: handle sdes messages with non-utf8 more gracefully 2014-06-17 08:52:50 +0200 Stian Selnes * gst/rtp/gstrtph263pay.c: rtph263pay: change log level on bitstream parsing messages 2016-07-07 11:13:18 +0200 Mikhail Fludkov * tests/check/elements/rtprtx.c: tests/rtprtx: refactor the tests to use gstharness The functionality of all the tests was kept exactly the same. Some tests were renamed: test_push_forward_seq -> test_rtxsend_rtxreceive test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss test_rtxreceive_data_reconstruction was testing that retransmitted buffer produced by rtxsend was correctly transformed to the original buffer by rtxreceive. Now we are checking for this in all the tests where both rtxsend & rtxreceive are involved. That's why the test was removed. 2016-08-25 15:52:36 +0200 Jonas Holmberg * gst/rtp/gstrtph265pay.c: rtph265pay: Set RTP marker bit Set the RTP marker bit on the last RTP packet of an H.265 access unit. https://bugzilla.gnome.org/show_bug.cgi?id=770394 2016-07-26 19:39:58 +0200 Xabier Rodriguez Calvar * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: videoflip: added GstVideoDirection interface It implements now this interface with its video-direction property. Values are changed to GstVideoOrientationMethod but they have the same value than the originals. https://bugzilla.gnome.org/show_bug.cgi?id=768687 2015-11-06 10:39:16 +0100 Havard Graff * gst/rtpmanager/gstrtpsession.c: gstrtpsession: refactor duplicate code into a function Less code, easier to read, more consistent. https://bugzilla.gnome.org/show_bug.cgi?id=770293 2016-08-23 17:06:44 +0100 Vincent Penquerc'h * gst/rtpmanager/gstrtpbin.c: rtpbin: fix typo in max-misorder-time property name 2016-08-22 00:05:52 +0100 Tim-Philipp Müller * gst/multifile/gstsplitmuxsink.c: splitmuxsink: fix printf format compiler warning in debug message On 32-bit x86: gstsplitmuxsink.c:966:31: warning: format ‘%u’ expects argument of type ‘unsigned int’, but argument 9 has type ‘guint64 {aka long long unsigned int}’ 2016-08-12 21:25:34 +0530 Nirbheek Chauhan * ext/twolame/meson.build: Add support for Meson as alternative/parallel build system https://github.com/mesonbuild/meson With contributions from: Tim-Philipp Müller Jussi Pakkanen (original port) Highlights of the features provided are: * Faster builds on Linux (~40-50% faster) * The ability to build with MSVC on Windows * Generate Visual Studio project files * Generate XCode project files * Much faster builds on Windows (on-par with Linux) * Seriously fast configure and building on embedded ... and many more. For more details see: http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html Building with Meson should work on both Linux and Windows, but may need a few more tweaks on other operating systems. 2016-08-12 21:25:34 +0530 Nirbheek Chauhan * ext/lame/meson.build: Add support for Meson as alternative/parallel build system https://github.com/mesonbuild/meson With contributions from: Tim-Philipp Müller Jussi Pakkanen (original port) Highlights of the features provided are: * Faster builds on Linux (~40-50% faster) * The ability to build with MSVC on Windows * Generate Visual Studio project files * Generate XCode project files * Much faster builds on Windows (on-par with Linux) * Seriously fast configure and building on embedded ... and many more. For more details see: http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html Building with Meson should work on both Linux and Windows, but may need a few more tweaks on other operating systems. 2016-08-12 21:25:34 +0530 Nirbheek Chauhan * ext/mpg123/meson.build: Add support for Meson as alternative/parallel build system https://github.com/mesonbuild/meson With contributions from: Tim-Philipp Müller Jussi Pakkanen (original port) Highlights of the features provided are: * Faster builds on Linux (~40-50% faster) * The ability to build with MSVC on Windows * Generate Visual Studio project files * Generate XCode project files * Much faster builds on Windows (on-par with Linux) * Seriously fast configure and building on embedded ... and many more. For more details see: http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html Building with Meson should work on both Linux and Windows, but may need a few more tweaks on other operating systems. 2016-08-12 21:12:30 +0530 Nirbheek Chauhan * .gitignore: * config.h.meson: * ext/cairo/meson.build: * ext/dv/meson.build: * ext/flac/meson.build: * ext/gdk_pixbuf/meson.build: * ext/jack/meson.build: * ext/jpeg/meson.build: * ext/libpng/meson.build: * ext/meson.build: * ext/pulse/meson.build: * ext/shout2/meson.build: * ext/soup/meson.build: * ext/speex/meson.build: * ext/taglib/meson.build: * ext/vpx/meson.build: * ext/wavpack/meson.build: * gst/alpha/meson.build: * gst/apetag/meson.build: * gst/audiofx/meson.build: * gst/audioparsers/meson.build: * gst/auparse/meson.build: * gst/autodetect/meson.build: * gst/avi/meson.build: * gst/cutter/meson.build: * gst/debugutils/meson.build: * gst/deinterlace/meson.build: * gst/dtmf/meson.build: * gst/effectv/meson.build: * gst/equalizer/meson.build: * gst/flv/meson.build: * gst/flx/meson.build: * gst/goom/meson.build: * gst/goom2k1/meson.build: * gst/icydemux/meson.build: * gst/id3demux/meson.build: * gst/imagefreeze/meson.build: * gst/interleave/meson.build: * gst/isomp4/meson.build: * gst/law/meson.build: * gst/level/meson.build: * gst/matroska/meson.build: * gst/meson.build: * gst/monoscope/meson.build: * gst/multifile/meson.build: * gst/multipart/meson.build: * gst/replaygain/meson.build: * gst/rtp/meson.build: * gst/rtpmanager/meson.build: * gst/rtsp/meson.build: * gst/shapewipe/meson.build: * gst/smpte/meson.build: * gst/spectrum/meson.build: * gst/udp/meson.build: * gst/videobox/meson.build: * gst/videocrop/meson.build: * gst/videofilter/meson.build: * gst/videomixer/meson.build: * gst/wavenc/meson.build: * gst/wavparse/meson.build: * gst/y4m/meson.build: * meson.build: * meson_options.txt: * sys/directsound/meson.build: * sys/meson.build: * sys/v4l2/meson.build: * sys/ximage/meson.build: * tests/check/meson.build: * tests/meson.build: Add support for Meson as alternative/parallel build system https://github.com/mesonbuild/meson With contributions from: Tim-Philipp Müller Jussi Pakkanen (original port) Highlights of the features provided are: * Faster builds on Linux (~40-50% faster) * The ability to build with MSVC on Windows * Generate Visual Studio project files * Generate XCode project files * Much faster builds on Windows (on-par with Linux) * Seriously fast configure and building on embedded ... and many more. For more details see: http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html Building with Meson should work on both Linux and Windows, but may need a few more tweaks on other operating systems. 2016-08-20 16:59:30 +0800 Jie Jiang * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: Fixed splitmuxsink 32-bit overflow bug Extend the byte tracking counters to 64-bit on all platforms, instead of using gsize, which overflows after 4GB. https://bugzilla.gnome.org/show_bug.cgi?id=770019 2016-08-19 17:18:16 +0300 Vivia Nikolaidou * gst/isomp4/atoms.c: isomp4: Fix coverity warning If atom_copy_data fails to write anything, return 0 CID #1371458 2016-04-09 07:51:03 +0530 Nirbheek Chauhan * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/v4l2-utils.c: v4l2: consistently check #ifdef HAVE_GUDEV instead of #if Both work with autotools but they definitely don't mean the same thing, cause problems with other build systems, and are bad form. Existence should always be checked with #ifdef or #if defined. 2016-04-19 10:53:05 +0530 Nirbheek Chauhan * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: directsound: port away from old DirectX API D3DX has been deprecated for the last 4 years and latest versions of Windows no longer ship headers for it. This is fine as long as you're building with Cerbero's Wine-based DirectX headers, but sucks if you want to build against the actual Windows SDK. We were just using it to get error strings anyway, so just use the generic error string API. 2016-08-18 12:02:01 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: Revert "flacparse: Add maximum bitrate tag" This reverts commit c703ab69f526092bb26cce41ca691a896c8383d8. https://bugzilla.gnome.org/show_bug.cgi?id=769392 2016-08-18 09:57:51 +0300 Sebastian Dröge * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations Need to set max-misorder-time and max-dropout-time to 0 so the jitterbuffer does not base them on packet rate calculations. If it does, out gap is big enough to be considered a new stream and we wait for a few consecutive packets just to be sure https://bugzilla.gnome.org/show_bug.cgi?id=751311 2016-08-09 12:55:59 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Add option to split at exactly max-size-time Will try to request a keyframe from the encoder to be sent at the target running time. https://bugzilla.gnome.org/show_bug.cgi?id=769664 2016-08-09 20:16:16 +0300 Vivia Nikolaidou * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Allow time and bytes to reach their respective thresholds https://bugzilla.gnome.org/show_bug.cgi?id=769664 2016-08-17 09:49:04 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Allow mimetypes with properties as long as they're application/sdp Some servers add properties like charset, e.g. application/sdp; charset=utf8 Ideally we should also parse the charset and do conversion of all messages, but that's for a later time. 2016-06-24 16:32:37 +0300 Vivia Nikolaidou * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Added support for writing timecode track https://bugzilla.gnome.org/show_bug.cgi?id=767950 2016-08-16 00:40:53 +1000 Jan Schmidt * ext/qt/gstqtglutility.cc: qt: Use wglShareLists() workaround unconditionally. Sometimes wglCreateContextAttribsARB() exists, but isn't functional (some Intel drivers), so it's easiest to do the workaround unconditionally. 2016-08-08 13:41:14 +1000 Jan Schmidt * ext/qt/gstqtglutility.cc: qt: Move debug statement to after the category init Don't output debug to an uninitialised debug category. 2016-08-11 16:32:21 -0600 Thomas Bluemel * gst/udp/gstmultiudpsink.c: multiudpsink: Initialize bytes_sent field. This fixes endpoints not receiving any data intermittently. https://bugzilla.gnome.org/show_bug.cgi?id=769773 2016-08-10 11:45:13 -0600 Thomas Bluemel * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpstats.c: rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations. https://bugzilla.gnome.org/show_bug.cgi?id=751311 2016-08-10 11:26:17 -0600 Thomas Bluemel * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Don't warn for duplicate packets This is a normal scenario and should not be a warning. This can happen frequently when re-transmits of lost packets are enabled. https://bugzilla.gnome.org/show_bug.cgi?id=762208 2016-08-08 13:49:19 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmux: Fix typo converting to running time. Use the correct collected timestamp. 2016-08-08 02:53:48 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: Revert "splitmuxsink: Use GstBin async-handling instead of our own." This reverts commit fa008f271a52f82dededc28bd81b020ca7939b47. async-handling in GstBin causes the pipeline to spin at 100% CPU as the top-level pipeline tries to change that state to PLAYING constantly. This is a workaround for a core problem, essentially, but an improvement in this case for now. 2016-08-08 00:56:38 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmux: Recheck state after unlocking mutex. After dropping the splitmux lock, re-check the state, don't just fall through and sleep unconditionally, as we may have already missed the wakeup. https://bugzilla.gnome.org/show_bug.cgi?id=769514 2016-08-03 03:32:07 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Don't stop and error on EOS flow return Don't immediately halt on EOS flow return from downstream due to out of segment. Let the demuxer handle it and send EOS. 2016-08-04 00:36:28 -0300 Thiago Santos * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: avoid unref of null buffer The current 'l' pointer will be NULL when the loop is interrupted with a 'break' statement. Need to have it advance to the next list item before interrupting. 2016-07-27 09:28:23 +0800 Haihua Hu * tests/examples/qt/qmlsink/.gitignore: * tests/examples/qt/qmlsink/main.cpp: * tests/examples/qt/qmlsink/main.qml: * tests/examples/qt/qmlsink/play.pro: * tests/examples/qt/qmlsink/qml.qrc: * tests/examples/qt/qmlsrc/.gitignore: * tests/examples/qt/qmlsrc/grabqml.pro: * tests/examples/qt/qmlsrc/main.cpp: * tests/examples/qt/qmlsrc/main.qml: * tests/examples/qt/qmlsrc/qml.qrc: qmlglsrc: Add qmlglsrc unit test example https://bugzilla.gnome.org/show_bug.cgi?id=768160 2016-07-27 08:16:47 +0800 Haihua Hu * ext/qt/Makefile.am: * ext/qt/gstplugin.cc: * ext/qt/gstqtglutility.cc: * ext/qt/gstqtglutility.h: * ext/qt/gstqtsrc.cc: * ext/qt/gstqtsrc.h: * ext/qt/qtitem.cc: * ext/qt/qtwindow.cc: * ext/qt/qtwindow.h: qt: implement qmlglsrc for qml view grab [Matthew Waters]: gst-indent sources https://bugzilla.gnome.org/show_bug.cgi?id=768160 2016-08-02 14:01:14 +0200 Carlos Rafael Giani * gst/wavparse/Makefile.am: * gst/wavparse/gstwavparse.c: wavparse: Add tags for container format and bitrate for uncompressed PCM The PCM bitrate is added to help downstream elements (like uridecodebin) figure out a proper network buffer size https://bugzilla.gnome.org/show_bug.cgi?id=769390 2016-08-01 18:52:26 +0200 Carlos Rafael Giani * gst/audioparsers/gstflacparse.c: flacparse: Add maximum bitrate tag https://bugzilla.gnome.org/show_bug.cgi?id=769392 2016-07-28 17:58:16 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: When receiving a DISCONT buffer that does not point to a sample, remember the offset And don't just reset everything. This makes sure that we can continue to handle data in the following scenario: moov: discont moof: discont mdat: continuous Previously this would fail because the offset would be the accumulated offset from moov and moof at the mdat position, while the buffer offset might be something completely different. 2016-07-25 13:34:02 +0300 Sebastian Dröge * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpilbcpay.c: rtp: Filter with the filter caps in the payloader's getcaps 2016-03-03 11:35:06 +0000 Vincent Penquerc'h * ext/soup/gstsouphttpsrc.c: souphttpsrc: include http-status-code in error message details https://bugzilla.gnome.org/show_bug.cgi?id=763038 2016-07-25 18:20:03 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix debug statement signedness. The ts variable is a GstClockTime, don't print it as a GstClockTimeDiff. 2016-07-22 17:00:14 +0300 Sebastian Dröge * tests/examples/qt/qml/main.cpp: qml: Don't forget to unref the actual sink element after setting it on glsinkbin 2016-07-22 16:57:45 +0300 Sebastian Dröge * tests/examples/qt/qml/main.cpp: qml: Use glsinkbin instead of glupload directly 2016-07-17 22:41:02 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Handle negative running time Use signed clock times for running time everywhere so that we handle negative running times without going haywire, similar to what queue and multiqueue do these days. 2016-07-18 00:12:55 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Drop lock when sending dummy event When pushing the dummy event into the multiqueue, drop the splitmux lock or else we might deadlock. 2016-06-30 01:56:41 +1000 Jan Schmidt * gst/rtp/gstrtph264pay.c: rtph264pay: Intersect with filter caps in getcaps function. Always intersect with the filter caps in the getcaps function to make sure we return a subset of what was requested. Other payloaders also have this problem and need fixing in future commits. 2016-07-12 17:30:56 +0200 Guillaume Desmottes * tests/check/elements/qtdemux.c: tests: qtdemux: fix element and pad leak https://bugzilla.gnome.org/show_bug.cgi?id=768739 2016-07-12 16:45:36 +0200 Guillaume Desmottes * tests/check/elements/audiofirfilter.c: * tests/check/elements/audioiirfilter.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/videobox.c: * tests/check/pipelines/effectv.c: tests: fix bus leaks gst_bus_add_signal_watch() takes a ref on the bus which should be released using gst_bus_remove_signal_watch(). https://bugzilla.gnome.org/show_bug.cgi?id=768739 2016-07-14 03:07:11 +0800 Ting-Wei Lan * configure.ac: configure: Call AG_GST_PKG_CONFIG_PATH to set GST_PKG_CONFIG_PATH GST_PKG_CONFIG_PATH is used in docs/plugins directory, so AG_GST_PKG_CONFIG_PATH must be called to set it. https://bugzilla.gnome.org/show_bug.cgi?id=768787 2016-07-12 07:39:58 +0200 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't drop final bytes of a range request At the end of a range request, we don't want to return GST_FLOW_EOS otherwise the last bytes we just read will be dropped by basesrc. Instead just return GST_FLOW_OK (which was set just before) and let basesrc handle the fact we are at the end of the segment. 2016-07-11 18:30:18 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2deviceprovider.c: v4l2provider: Fix device type detection The type detection would lead to assertion as it would try to create a device without having found any type for it. It also didn't detect MPLANE devices properly. 2016-07-11 18:29:01 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't assert when used by the monitor The monitor sets the object->element object as a GstObject. This works for debug traces, but will assert for ELEMENT_ERROR. This was the only case where that could happen. Add a check for that. 2016-07-11 17:38:00 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Indent very long line 2016-07-12 00:42:02 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: At the end of a range request, read another time to finalize the request If we're at the end of a range request, read again to let libsoup finalize the request. This allows to reuse the connection again later, otherwise we would have to cancel the message and close the connection. 2016-07-11 21:13:47 +0200 Stefan Sauer * common: Automatic update of common submodule From f363b32 to f49c55e 2016-07-11 19:57:18 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Fix keep-alive handling We have to get rid of the message on EOS when the complete stream is read to remember that we successfully finished handling this specific message. Otherwise we will cancel it later and close the connection instead of reusing it at a later time. It might also make sense to reuse connections if a non-200 response is received. As long as there was no connection error, the HTTP connection should be re-usable. 2016-07-11 12:05:06 -0400 Nicolas Dufresne * configure.ac: Also enable V4L2 probe on aarch64 (aka ARM 64bit) 2016-07-11 11:59:19 -0400 Olivier Crête * tests/examples/rtp/client-PCMA.c: rtp example: Fix leak Also stop fetching the internal source as this functionality has been broken. 2016-07-08 14:58:37 -0400 Nicolas Dufresne * configure.ac: Enable v4l2 probe on Linux/ARM Most of those have V4L2 drivers these days enabling it make sure that it this code is enabled in major distribution, hence that HW accelerated decoder/encoder can be used on platforms that support it. The probes are slightly increasing the first init of gstreamer library, though the result is cached in the registry for later use. 2016-07-11 09:46:49 +0200 Jonas Holmberg * gst/rtp/gstrtph265pay.c: * tests/check/elements/rtp-payloading.c: rtph265pay: Accept array_completeness=1 When parsing NAL unit type in codec_data, check the 6bits of NAL_unit_type only and do not require the array_completeness bit to be 0, since the default and mandatory value of array_completeness is 1 for hvc1. https://bugzilla.gnome.org/show_bug.cgi?id=768653 2016-07-10 21:35:06 -0400 Nicolas Dufresne * sys/v4l2/v4l2_calls.c: v4l2: Also copy device_caps in gst_v4l2_dup This fixes regression where M2M error out saying they have no output format (the V4L2 CAPTURE side). https://bugzilla.gnome.org/show_bug.cgi?id=768195 2016-07-10 21:30:27 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Use correct in6_pktinfo struct instead of in_pktinfo Fixes the build on FreeBSD, which does not have the latter. https://bugzilla.gnome.org/show_bug.cgi?id=768623 2016-07-08 17:28:19 +0000 Luis de Bethencourt * sys/v4l2/v4l2_calls.c: v4l2: fix multiplanar capture After switching to using V4L2_CAP_DEVICE_CAPS we lost support for multiplanar device types. After some research, it looks like vcap.capabilities treated the multiplanar flag of output and capture devices equally, but not the new device_caps. https://bugzilla.gnome.org/show_bug.cgi?id=768195 2016-07-08 14:56:30 +0200 Mats Lindestam * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: multipartmux: Use PTS and DTS instead of timestamp And pass-through both of them. Based on a patch by Göran Jönsson https://bugzilla.gnome.org/show_bug.cgi?id=767900 2016-06-30 14:40:40 +0200 Thomas Scheuermann * ext/jack/gstjackaudioclient.c: jack: don't wait for callbacks if the jack server shut down Otherwise we'll wait forever. https://bugzilla.gnome.org/show_bug.cgi?id=747275 2016-06-23 15:30:19 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Let upstream events go through upstream There's no real reason to avoid sending QOS/NAVIGATION events upstrea. Some elements might want to have that information. 2016-06-23 15:22:56 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Let upstream events go through upstream There's no real reason to avoid sending QOS/NAVIGATION events upstrea. Some elements might want to have that information. 2016-06-23 15:17:36 +0200 Edward Hervey * ext/dv/gstdvdemux.c: dvdemux: Let upstream events go through upstream There's no real reason to avoid sending QOS/NAVIGATION events upstrea. Some elements might want to have that information. Also remove downstream-only CAPS event handling and minimize code 2016-07-07 23:53:54 +0100 Luis de Bethencourt * sys/v4l2/gstv4l2.c: v4l2: fix v4l2 probe build error A typo in gst_v4l2_probe_and_register() caused a build error when building with --enable-v4l2-probe. Fixing it. gstv4l2.c: In function 'gst_v4l2_probe_and_register': gstv4l2.c:150:25: error: 'struct v4l2_capability' has no member named 'capabilitites' device_caps = vcap.capabilitites; 2016-07-01 22:53:33 -0700 Reynaldo H. Verdejo Pinochet * sys/v4l2/gstv4l2src.c: v4l2src: use gst_caps_intersect_full in negotiate() Instead of reimplementing the GST_CAPS_INTERSECT_FIRST interection mode. https://bugzilla.gnome.org/show_bug.cgi?id=768195 2016-07-02 01:56:07 -0700 Reynaldo H. Verdejo Pinochet * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2_calls.c: v4l2: use opened device caps instead of physical device ones The same physical device can export multiple devices. In this case, the capabilities field now contains a union of all caps available from all exported V4L2 devices alongside a V4L2_CAP_DEVICE_CAPS flag that should be used to decide what capabilities to consider. In our case, we need the ones from the exported device we are using. https://bugzilla.gnome.org/show_bug.cgi?id=768195 2016-07-07 18:24:59 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Remove suspicious checks for pads being active and linked We should add all pads, no matter if they are linked or active or not at this point. Skipping some that are not will cause different behaviour than with other muxers. 2016-07-07 18:23:07 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Error out if we start writing data with some pads not having a codec id yet This can only happen if a) upstream somehow gets around the CAPS event failing or b) there never being any CAPS event. The following code assumes that all pads have a codec-id. https://bugzilla.gnome.org/show_bug.cgi?id=768509 2016-07-07 18:14:43 +0300 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Consistently use gst_matroska_mux_set_codec_id() for setting the codec id 2016-07-04 09:50:11 +0200 Jonas Holmberg * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: * tests/check/elements/rtp-payloading.c: rtph265pay/depay: Sync against RFC 7798 Handle sprop-vps, sprop-sps and sprop-pps in caps instead of sprop-parameter-sets. rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It handles profile-id, tier-flag and level-id in caps query. https://bugzilla.gnome.org/show_bug.cgi?id=753760 2016-07-06 09:25:00 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Push nominal bitrate tags Add per-stream tag lists, which are used to send nominal bitrate tags. When remuxing FLV => FLV, this now passes through the upstream bitrate. https://bugzilla.gnome.org/show_bug.cgi?id=768440 2016-07-06 09:24:49 +0200 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Refactor metadata tag handling The FLV header cannot be trusted to indicate video or audio presence, as the comments already mention. Don't delay pushing tags waiting for streams that might never appear. Tags are now pushed immediately after they change: - After parsing an onMetaData script object - After negotiating caps on a pad https://bugzilla.gnome.org/show_bug.cgi?id=768440 2016-07-06 12:44:10 +0100 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: fix AAC codec_data values As seen in the parent switch for object_type_id, the 4 possible values are 0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values. Looks like it was a typo making them decimal instead of hexadecimal. CID 1363328 2016-07-06 13:51:03 +0300 Sebastian Dröge * configure.ac: Back to development === release 1.9.1 === 2016-07-06 13:06:44 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.9.1 2016-07-06 11:46:26 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2016-07-06 11:22:53 +0300 Steven Hoving * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix error messages to first convert to doubles before division 2016-07-06 10:18:30 +0300 Sebastian Dröge * po/da.po: * po/hr.po: * po/pt_BR.po: * po/sk.po: po: Update translations 2016-07-05 21:11:35 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else There's a small window for a race condition otherwise. 2016-07-04 17:45:40 +0200 Sebastian Dröge * tests/check/elements/qtmux.c: qtmux: Use complete AAC caps with codec_data in the tests 2016-07-04 16:58:38 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Reject raw AAC if no codec_data is found in the caps If necessary, a demuxer will have to invent something here but this is only a problem with non-conformant files anyway. 2016-07-04 16:55:32 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Invent AAC codec_data if none is present Without, raw AAC can't be handled and we have some information available in the decoder that most likely allows us to decode the stream in one way or another. This is the same code already used by matroskademux for the same reasons, and ffmpeg/vlc play such files just fine too by guesswork. 2016-07-04 14:54:13 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Reject raw AAC caps without codec_data The resulting file is not going to be playable without guesswork and raw caps should always have codec_data. 2016-07-01 19:22:32 +0100 Tim-Philipp Müller * ext/qt/Makefile.am: qt: fix build some more when QPA is not available Compiler would complain about include directory that didn't exist because QPA_INCLUDE_PATH gets subst-ed regardless (and if it didn't we'd have just an empty -I argument). https://bugzilla.gnome.org/show_bug.cgi?id=767553 2016-05-10 15:48:49 +0200 Edward Hervey qtdemux: Handle upstream GAP in push-mode/time segment This is to handle cases where upstream handles the fragmented streaming in TIME segments and sends us data with gaps within fragments. This would happen when dealing with trick-modes. When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples, it must obey the following rules: * The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET * The buffers containing the first sample after a gap: * MUST start at the beginning of a sample, * MUST have the DISCONT flag set, * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment. https://bugzilla.gnome.org/show_bug.cgi?id=767354 2016-07-01 11:54:57 +0100 Tim-Philipp Müller * sys/v4l2/v4l2-utils.c: v4l2: fix potential double-free of error debug string gst_v4l2_clear_error() doesn't work like g_clear_error(), it doesn't NULLify the pointer, so set freed debug string to NULL so it doesn't get freed again if gst_v4l2_clear_error() is called twice on the error. CID 1362901 2016-07-01 10:05:00 +0000 Brad Lackey * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't disable UDP protocols on redirecting https://bugzilla.gnome.org/show_bug.cgi?id=768232 2016-07-01 17:28:17 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Push caps only when it was updated Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps event per moof without consideration of duplication. https://bugzilla.gnome.org/show_bug.cgi?id=768268 2016-06-30 15:01:46 +0200 Jonas Holmberg * gst/rtp/gstrtph265depay.c: rtph265depay: fix invalid memory access 10 bytes was allocated for stream_format but size of "byte-stream" is more. Use g_strdup() instead. https://bugzilla.gnome.org/show_bug.cgi?id=753760 2016-06-29 23:31:20 +0200 Sebastian Dröge * ext/shout2/gstshout2.c: shout2: Use a non-timer GstPoll Otherwise set_flushing() will have undefined semantics and nowadays causes a g_critical() to warn about that. 2016-06-19 02:08:25 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: dynamically adjust blocksize Update the blocksize depending on how much is obtained from a read of the input stream. This avoids doing too many reads in small chunks when larger amounts of data are available and also prevents using a very large memory area to read a small chunk of data. https://bugzilla.gnome.org/show_bug.cgi?id=767833 2016-06-28 16:44:50 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo 2016-06-28 15:15:14 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS 2016-06-28 15:08:04 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Move #includes around to a) work around broken glibc header and b) Windows 2016-06-28 14:25:03 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Fix compilation on Windows and *BSD/OSX 2016-06-23 20:21:59 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Filter out multicast packets that are not for our multicast address https://bugzilla.gnome.org/show_bug.cgi?id=767980 2016-06-28 10:57:27 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state If we consider the RTSP state, what can happen is that it is PLAYING but the element already asynchronously tried to PAUSE and it just did not happen yet. We would then override this setting to PAUSED (while the element actually is in PAUSED) and set the RTSP state to PLAYING again. This would then cause us to produce packets while the sinks are all PAUSED, piling up thousands of packets in the rtpjitterbuffer and other elements and finally failing. 2016-06-27 18:15:08 +0800 Haihua Hu * ext/qt/qtitem.cc: qmlglsink: Fix build error when don't have QPA installed. Check header file existance and wrap the header file include in the necessary #ifdef to avoid build error. https://bugzilla.gnome.org/show_bug.cgi?id=767553 2016-06-27 09:20:35 +0300 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Add comment about H263/MPEG4P2 being non-standard for FLV They are however supported by ffmpeg and apparently used out there. https://bugzilla.gnome.org/show_bug.cgi?id=768006 2016-06-24 14:48:53 +0300 Vivia Nikolaidou * gst/flv/gstflvdemux.c: flvdemux: Add support for H263 and MPEG4 part2 https://bugzilla.gnome.org/show_bug.cgi?id=768006 2016-06-16 15:13:02 +1000 Matthew Waters * ext/qt/qtitem.cc: * ext/qt/qtplugin.pro: qmlglsink: add win32 support The current state of c++ ABI's on Window's and Gst's/Qt's conflicting mingw builds means that we cannot use mingw for building the qt plugin. Instead, a qmake .pro file is provided that is expected to be used with the msvc binaries provided by Qt like so: (with the PATH environment variable containing the path to the qt biniaries and PKG_CONFIG_PATH containing the path to GStreamer modules) cd /path/to/sources/gst-plugins-bad/ext/qt qmake -tp vc Then open the resulting VS project and build the library. Then cp debug/libgstqtsink.dll /path/to/prefix/lib/gstreamer-1.0/libgstqtsink.cll https://bugzilla.gnome.org/show_bug.cgi?id=761260 2016-06-21 17:10:56 -0400 Nicolas Dufresne * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: Update plugins doc This is partly automated using "make update" in docs/plugins, but also required manual merge. Additionally, missing plugins and elements have been added. 2016-06-21 17:51:38 +0100 Tim-Philipp Müller * tests/check/elements/splitmux.c: tests: splitmux: skip tests if theora or ogg plugins are not available https://bugzilla.gnome.org/show_bug.cgi?id=767861 2016-06-21 11:46:13 -0400 Nicolas Dufresne * common: Automatic update of common submodule From ac2f647 to f363b32 2016-06-21 07:40:42 -0400 Aaron Boxer * gst/rtp/gstrtpj2kpay.c: gstrtpj2kpay: use tile bit and tile number to determine if there are multiple tiles in packet Now we don't have to rely on a special value for the tile number. https://bugzilla.gnome.org/show_bug.cgi?id=767817 2016-06-21 09:34:56 +0100 Tim-Philipp Müller * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: fix compiler warning on OS/X gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535 https://bugzilla.gnome.org/show_bug.cgi?id=767817 2016-06-21 09:34:37 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-rtp.xml: docs: update 2016-05-16 17:31:58 +0200 Guillaume Desmottes * tests/check/elements/capssetter.c: * tests/check/elements/icydemux.c: * tests/check/elements/jpegenc.c: * tests/check/elements/level.c: * tests/check/elements/multifile.c: * tests/check/elements/qtmux.c: * tests/check/elements/rtprtx.c: * tests/check/elements/udpsrc.c: fix buffer leaks in tests Need to call gst_check_drop_buffers() to release the buffers exchanged during the test. https://bugzilla.gnome.org/show_bug.cgi?id=766561 2016-05-17 12:52:43 +0300 Guillaume Desmottes * tests/check/elements/interleave.c: interleave: fix message leaks in test Flush the bus when cleaning up so pending messages are destroyed. https://bugzilla.gnome.org/show_bug.cgi?id=766561 2016-05-17 12:58:06 +0300 Guillaume Desmottes * tests/check/elements/videomixer.c: videomixer: fix event leaks in test https://bugzilla.gnome.org/show_bug.cgi?id=766561 2016-05-13 15:12:22 +0200 Guillaume Desmottes * tests/check/elements/deinterleave.c: deinterleave: fix leaks - Flush the bus so messages aren't leaked - Fix pad leak https://bugzilla.gnome.org/show_bug.cgi?id=766561 2016-06-17 15:29:16 +0300 Sebastian Dröge * gst/rtp/gstrtph264pay.c: rtph264pay: Deprecated sprop-parameter-set property This is supposed to be either in the codec_data (avc stream format) or inside the stream, and we extract it from there. It should not be set from a property as it's stream specific. https://bugzilla.gnome.org/show_bug.cgi?id=767789 2016-06-17 12:16:32 -0700 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: make all srtp encoder properties explicit The Session Data Protocol doesn't allow specifying a cipher for the SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher "aes-128-icm" is the default for SRTP and SRTCP, but if we want to have an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead. https://bugzilla.gnome.org/show_bug.cgi?id=767799 2016-06-17 19:59:13 +0100 Tim-Philipp Müller * ext/soup/gstsoup.c: soup: work around frequent deadlocks in GLib type initialisation .. by registering the types from the plugin init function. This seems to help, but we'll see if it's enough (might need similar things elsewhere). https://bugzilla.gnome.org/show_bug.cgi?id=693911 https://bugzilla.gnome.org/show_bug.cgi?id=674885 2016-06-17 16:08:08 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: The prores variant is stored in the variant field, not format And the caps in the sink pad template already used variant (only). 2016-06-17 13:00:48 +0200 Jonas Holmberg * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: rtph265pay: Remove sprop-parameter-sets property There is no valid use case when this property is needed since the values must be in either codec_data or buffer data. https://bugzilla.gnome.org/show_bug.cgi?id=753760 2016-06-10 16:17:26 +0200 Jonas Holmberg * docs/plugins/scanobj-build.stamp: * gst/rtp/gstrtph265pay.c: rtph265pay: Read NALU type the same way everywhere Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the same way as in other places. https://bugzilla.gnome.org/show_bug.cgi?id=753760 2016-06-17 13:58:33 +0200 Aurélien Zanelli * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: fix RTPJitterBufferMode documentation Documentation lacks '@' before each enum values and there was an extra line after symbol section which confuses GTK-Doc parser. https://bugzilla.gnome.org/show_bug.cgi?id=767788 2016-05-23 10:18:48 +0200 Miguel París Díaz * gst/rtpmanager/rtpsession.c: rtpsession: take the lock when changing stats https://bugzilla.gnome.org/show_bug.cgi?id=766025 2016-04-14 18:14:32 +0300 Sergey Borovkov * ext/qt/qtitem.cc: qml: Enable qmlglsink for eglfs https://bugzilla.gnome.org/show_bug.cgi?id=763044 2016-06-16 00:44:48 +1000 Matthew Waters * ext/qt/qtitem.cc: qmlglsink: propagate GL context creation failure upwards Otherwise an application cannot know if the qmlglsink will be displaying frames incorrectly/at all. 2016-06-16 00:44:16 +1000 Matthew Waters * ext/qt/qtitem.cc: qmlglsink: also allow wayland-egl as a platform name 2016-06-12 15:35:28 +0800 Haihua Hu * ext/qt/Makefile.am: * ext/qt/qtitem.cc: qmlglsink: Add Wayland support Don't use gstgldisplay to get wayland display. Should use QPA on wayland to get wayland display for QT. https://bugzilla.gnome.org/show_bug.cgi?id=767553 2016-06-15 11:19:43 +0200 Jürgen Slowack * gst/rtp/gstrtph265pay.c: rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection Fixes sps/pps/vps insertion via the config-interval property. https://bugzilla.gnome.org//show_bug.cgi?id=767680 2016-06-11 12:16:03 +0300 Sebastian Dröge * tests/check/pipelines/simple-launch-lines.c: simple-launch-lines: Use correct JPEG2000 caps 2016-06-10 13:43:09 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: fix indentation 2016-06-10 13:42:01 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: fix date parsing when there are trailing spaces Fixes parsing of "Thu May 11 15:57:46 2006 ". https://bugzilla.gnome.org/show_bug.cgi?id=767496 2016-05-13 15:08:24 -0400 Aaron Boxer * gst/rtp/gstrtpj2kcommon.h: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: gstrtpj2k: set sampling field required by RFC This field is now required in the sink caps. https://bugzilla.gnome.org/show_bug.cgi?id=766236 2016-06-09 09:30:48 +0900 Seungha Yang * gst/flv/gstflvdemux.c: flvdemux: Fix unref assertion failure Fix unref assertion failure https://bugzilla.gnome.org/show_bug.cgi?id=767424 2016-05-14 14:46:17 +0200 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Work with non-TIME segments With non-time segments, it now assumes that the arrival time of packets is not relevant and that only the RTP timestamp matter and it produces an output segment start at running time 0. https://bugzilla.gnome.org/show_bug.cgi?id=766438 2016-06-07 20:53:34 -0400 Nicolas Dufresne * ext/libpng/gstpngdec.c: pngdec: Wait for segment event before checking it The heuristic to choose between packetise or not was changed to use the segment format. The problem is that this change is reading the segment during the caps event handling. The segment event will only be sent after. That prevented the decoder to go in packetize mode, and avoid useless parsing. https://bugzilla.gnome.org/show_bug.cgi?id=736252 2016-06-06 17:00:22 -0400 Nicolas Dufresne * ext/jpeg/gstjpegdec.c: jpegdec: Wait for segment event before checking it The heuristic to choose between packetise or not was change to use the segment format. The problem is that this change is reading the segment during the caps event handling. The segment event will only be sent after. That prevented the decoder to go in packetize mode, and avoid useless parsing. https://bugzilla.gnome.org/show_bug.cgi?id=736252 2016-06-07 16:42:09 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Keep part of the input buffer Instead of completely getting rid of the input buffer, copy the metadata, the flags and the timestamp into an empty buffer. This way the decoder base class can copy that information again to the output buffer. https://bugzilla.gnome.org/show_bug.cgi?id=758424 2016-06-07 16:41:58 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Coding style fixes 2016-06-07 16:09:23 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Coding style fixes 2016-06-07 16:04:52 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Add an error return to _try/_set_format This way one can easily ignore errors. Previously, error were always posted ont he bus. https://bugzilla.gnome.org/show_bug.cgi?id=766172 2016-06-07 16:01:55 -0400 Nicolas Dufresne * sys/v4l2/v4l2-utils.c: * sys/v4l2/v4l2-utils.h: v4l2-util: Introduce GstV4l2Error This is to allow returning an error that can easily be sent as message to the application if the element needs it. Using this also allow ignoring errors. https://bugzilla.gnome.org/show_bug.cgi?id=766172 2016-06-07 12:41:19 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Avoid decide allocation on active pool v4l2src will renegotiate only if the format have changed. As of now, it's not possible to change the allocationw without resetting the camera. To avoid unwanted side effect, simply keep the old allocation if no renegotiation is taking place. This fixes assertion and possible failures in USERPTR or DMABUF import mode (when using downstream pools). https://bugzilla.gnome.org/show_bug.cgi?id=754042 2016-04-28 13:44:49 +0200 Edward Hervey * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Show state name in debugging Makes it easier to trace what's going on 2016-05-10 15:45:42 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Remove useless variable That variable is only needed for a debug statement, move it there 2016-05-10 15:10:36 +0200 Edward Hervey * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Add/Fix comments on the various structure variables No variables were added/removed. This was just a good excuse to: * Comment what most variables are used for (and when) * Order them in such a way as to show first the common variables used in all cases, followed by those only used in push-mode 2016-05-10 15:07:40 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Remove unused structure Let's just remove it, been commented for 7+ years :) 2015-09-02 11:48:29 +0200 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2videodec: use decoder stop command instead of queueing empty buffers Only if the decoder stop command fails, keep queueing empty buffers to signal end of stream as before. https://bugzilla.gnome.org/show_bug.cgi?id=733864 2014-12-12 14:31:36 +0100 Peter Seiderer * sys/v4l2/gstv4l2videodec.c: v4l2videodec: add gst_v4l2_decoder_cmd helper https://bugzilla.gnome.org/show_bug.cgi?id=733864 2016-06-01 20:28:39 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Forward segments directly if we are operating in PUSH mode on fragmented streams We shouldn't go through segment activation as we will only have a limited understanding of how the whole stream timeline looks like from the moof. We only know about the current fragment, while upstream knows about the whole stream. This fixes seeking in DASH streams, both for seeks after the current moof and for seeks into the current moof. The former would fail because the moof ends and we can't activate any segment, the latter would cause a segment that stops at the moof end, and no further fragments would be played because we end up being EOS. https://bugzilla.gnome.org/show_bug.cgi?id=767071 2016-06-06 17:54:10 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Use looser caps for upstream When we fixate for upstream, try to not introduce new fields when not needed. This was imported from videoconvert element. 2015-01-28 12:07:58 +0100 Enrico Jorns * sys/v4l2/gstv4l2transform.c: gstv4l2transform: format fixation for preferring passthrough * If outgoing format is unfixated, try to set it to input format. * Call gst_caps_fixate () at end of fixation routine https://bugzilla.gnome.org/show_bug.cgi?id=766719 2016-05-20 12:49:53 +0200 Philipp Zabel * sys/v4l2/gstv4l2transform.c: v4l2transform: allow to change pixel aspect ratio Scalers may change width and height independently, allow to change pixel aspect ratio. https://bugzilla.gnome.org/show_bug.cgi?id=766712 2016-05-20 12:32:25 +0200 Philipp Zabel * sys/v4l2/gstv4l2transform.c: v4l2transform: fix scaling in case of fixed pixel aspect ratio To change pixel aspect ratio from DAR to PAR, the necessary scaling factor is DAR/PAR, not DAR*PAR. For good measure, add debug output similar to the fixed-width and fixed-height cases. https://bugzilla.gnome.org/show_bug.cgi?id=766711 2016-05-13 16:39:25 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: fill colorimetry in gst_v4l2_object_acquire_format Instead of relying on the default colorimetry chosen by gst_video_info_set_format(), set info.colorimetry from the values returned by G_FMT. This allows decoders to propagate their input colorimetry downstream. https://bugzilla.gnome.org/show_bug.cgi?id=766383 2016-05-18 10:17:12 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: refactor gst_v4l2_object_get_colorspace to take a v4l2_format parameter Move the extraction of colorimetry parameters from struct v4l2_format and the setting of the identity matrix for RGB formats into the function to avoid code duplication. https://bugzilla.gnome.org/show_bug.cgi?id=766383 2016-05-13 14:58:41 +0200 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2videodec: use visible size, not coded size, for downstream negotiation filter gst_v4l2_probe_caps() returns the coded size, not the visible size. Subtract the known padding from probed caps with the coded size before using them as filter for caps negotiation with downstream elements. https://bugzilla.gnome.org/show_bug.cgi?id=766382 2016-05-13 14:45:02 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2object: use G_SELECTION instead of G_CROP in gst_v4l2_object_acquire_format The gst_v4l2_object_acquire_format() function is used by v4l2videodec to obtain the currently set capture format. Since G_FMT returns the coded size, the visible size needs to be obtained from the compose rectangle in order to negotiate it with downstream elements. The G_CROP call hasn't worked on mem2mem capture queues for a long time. Instead use the G_SELECTION call to obtain the compose rectangle and only fall back to G_CROP for ancient kernels. https://bugzilla.gnome.org/show_bug.cgi?id=766381 2016-01-27 09:57:38 +0100 Andreas Naumann * sys/v4l2/gstv4l2sink.c: v4l2sink: Use V4L2_BUF_TYPE_VIDEO_OUTPUT_OVERLAY if driver advertises it. On modern kernels, the G/S_FMT ioctls will always fail using V4L2_BUF_TYPE_VIDEO_OVERLAY with VFL_DIR_TX (e.g. real overlay out drivers) since this is not the intented use (rather rx, according to v4l2 API doc). Probably this is why the Video Output Overlay interface was created, so if the driver advertises it we might as well use. For old kernels (pre 2012) the old way might still work so keeping this for compatibility. https://bugzilla.gnome.org/show_bug.cgi?id=761165 2016-06-06 18:52:01 +0100 Kieran Bingham * sys/v4l2/gstv4l2object.c: v4l2object: Use non-deprecated V4L2 type for RGB15 Support for the updated V4L2_PIX_FMT_XRGB555 was added in commit 2538fee2fd8fdb74b05f0a511281bc4707e7cc44 however, when setting the format for use in v4l2 ioctls, the old deprecated format is still used. Convert this to the new accepted format type, as the preferred format. https://bugzilla.gnome.org/show_bug.cgi?id=767300 2016-05-04 14:50:32 +0200 Michael Olbrich * gst/matroska/matroska-demux.c: matroskademux: preserve seek flags Without this some flags get lost in streaming mode. https://bugzilla.gnome.org/show_bug.cgi?id=767194 2016-06-06 10:47:52 +0300 Sebastian Dröge * ext/soup/Makefile.am: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: Revert "WIP revert soup" This reverts commit fdac3a7a231f3848665636cf8122f96103b46e3b. Was not supposed to be pushed but a local workaround for https://bugzilla.gnome.org/show_bug.cgi?id=693911#c13 2016-06-03 13:09:35 +0200 Miguel París Díaz * gst/rtpmanager/rtpsource.c: rtpsource: complete warn log with SSRC https://bugzilla.gnome.org/show_bug.cgi?id=767195 2016-05-31 15:29:13 +0300 Sebastian Dröge * ext/soup/Makefile.am: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: WIP revert soup 2016-06-03 13:18:31 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Unref seek event in any case It would be leaked if no seek handler was currently set. 2016-06-03 10:49:17 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Properly set event/message sequence numbers based on the previous seek See https://bugzilla.gnome.org/show_bug.cgi?id=765935 https://bugzilla.gnome.org/show_bug.cgi?id=767157 2016-06-03 10:36:32 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Remember if upstream had a time segment and if not properly create time segments Previously the segment.time was wrong, and the position was not updated correctly, resulting in seeks in PUSH mode with upstream providing a BYTES segment to not work at all. https://bugzilla.gnome.org/show_bug.cgi?id=767157 2016-06-03 09:54:53 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Implement SEEKING query so we can actually seek if upstream can't seek in TIME https://bugzilla.gnome.org/show_bug.cgi?id=767157 2016-06-02 14:19:15 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Recalculate the frame offsets at the beginning of each BYTE segment and whenever upstream gives us a timestamp This fixes seeking in DV streams where upstream operates in PUSH mode with a TIME segment (e.g. avidemux). Without this, we would generate wrong durations and timestamps after a seek. https://bugzilla.gnome.org/show_bug.cgi?id=767157 2016-06-02 13:53:44 +0300 Sebastian Dröge * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Pass-through buffer DISCONT flags https://bugzilla.gnome.org/show_bug.cgi?id=767157 2016-06-02 16:16:45 -0400 Olivier Crête * gst/rtp/gstrtpvp9depay.c: rtpvp9depay: Don't assert on flexible mode packets Instead just post a warning on the bus for now. 2016-06-02 15:03:17 +0200 Guillaume Desmottes * tests/check/elements/rtpbin.c: tests: rtpbin: fix caps leak https://bugzilla.gnome.org/show_bug.cgi?id=767156 2016-06-02 15:00:01 +0200 Guillaume Desmottes * tests/check/elements/amrparse.c: tests: amrparse: clean up test - use GST_CHECK_MAIN() to reduce boilerplate - unref the input caps using a teardown function to prevent leaks https://bugzilla.gnome.org/show_bug.cgi?id=767156 2016-05-20 15:22:35 +0200 Edward Hervey * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Ensure DISCONT flag is properly propagated The output of deinterlace at startup, or when receiving a new DISCONT buffer, should have the DISCONT flag set on the first buffer. 2016-05-31 21:34:04 +0200 Josep Torra * sys/v4l2/gstv4l2bufferpool.c: v4l2src: check for valid size on raw video buffers Discard buffers that doesn't contain enough data when dealing with raw video inputs. https://bugzilla.gnome.org/show_bug.cgi?id=767086 2016-05-31 17:10:36 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Use the demuxer segment instead of a new one for MSS streams Upstream might have told us something about the to be expected segment, so let's use that information instead of coming up with a [0,-1] segment. https://bugzilla.gnome.org/show_bug.cgi?id=767071 2016-05-31 17:04:32 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Only activate segments and send SEGMENT events if we have streams But in that case also remove the pending newsegment event, otherwise we would later send a possibly outdated event. https://bugzilla.gnome.org/show_bug.cgi?id=767071 2016-05-31 16:53:50 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: In PULL mode, nothing is ever going to send us a SEGMENT event https://bugzilla.gnome.org/show_bug.cgi?id=767071 2016-05-31 16:38:34 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Don't override TIME segments from upstream that we just saw The point of d8fb7a9c96b108814beeaa0e63f818d4648c7fe9 was to not have any spurious segments stored for later if we do BYTES->TIME conversion, but overriding any TIME segments from upstream does not make any sense. See https://bugzilla.gnome.org/show_bug.cgi?id=763165 https://bugzilla.gnome.org/show_bug.cgi?id=767071 2015-07-16 09:48:46 +0530 Prashant Gotarne * gst/multifile/gstmultifilesrc.c: multifilesrc: set position as offset from start-index query position in GST_FORMAT_BUFFER returns offset from start-index rather than index. https://bugzilla.gnome.org/show_bug.cgi?id=752462 2016-05-27 12:49:32 +0100 Tim-Philipp Müller * tests/check/pipelines/simple-launch-lines.c: * tests/files/Makefile.am: * tests/files/gradient.j2k: tests: add unit test for JPEG-2000 rtp payloader leak https://bugzilla.gnome.org/show_bug.cgi?id=766870 2016-05-25 17:11:13 +0200 Pierre Lamot * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: Fix buffer memory leak Input buffer memory was not unmapped https://bugzilla.gnome.org/show_bug.cgi?id=766870 2016-05-18 12:12:15 +0300 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2object: fix caps leak gst_v4l2_object_probe_caps() was taking an extra ref on the returned caps for no reason. https://bugzilla.gnome.org/show_bug.cgi?id=766610 2016-05-22 20:14:18 +0100 Tim-Philipp Müller * gst/videocrop/gstvideocrop.c: videocrop mark crop properties as mutable in playing state 2016-05-20 16:47:35 +0300 Guillaume Desmottes * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix buffer leak when flushing When early returning in gst_soup_http_src_read_buffer() because the element is FLUSHING, we need to unmap and unref the buffer which was just created. https://bugzilla.gnome.org/show_bug.cgi?id=766718 2016-05-20 11:15:44 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Set seek event seqnum on all SEGMENT events Some were forgotten. See https://bugzilla.gnome.org/show_bug.cgi?id=765935 2016-05-20 11:12:44 +0300 Sebastian Dröge * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Pass through seek event seqnums in all SEGMENT/EOS events and SEGMENT_DONE messages/events See https://bugzilla.gnome.org/show_bug.cgi?id=765935 2016-05-20 10:56:52 +0300 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Set seek event seqnum in EOS and SEGMENT_DONE messages/events Also actually store the seqnum in pull mode seeks. See https://bugzilla.gnome.org/show_bug.cgi?id=765935 2016-05-17 13:40:38 +0300 Guillaume Desmottes * gst/deinterlace/gstdeinterlace.c: deinterlace: fix caps leak The caps returned by gst_pad_get_current_caps() was never unreffed when not early returning. Fix a leak with the elements/deinterlace test. https://bugzilla.gnome.org/show_bug.cgi?id=766558 2016-01-25 16:25:51 +0100 Mikhail Fludkov * gst/rtpmanager/rtpsession.c: * tests/check/Makefile.am: * tests/check/elements/rtpsession.c: rtpsession: don't act on suspicious BYE RTCP Some endpoints (like Tandberg E20) can send BYE packet containing our internal SSRC. I this case we would detect SSRC collision and get rid of the source at some point. But because we are still sending packets with that SSRC the source will be recreated immediately. This brand new internal source will not have some variables incorrectly set in its state. For example 'seqnum-base` and `clock-rate` values will be -1. The fix is not to act on BYE RTCP if it contains internal or unknown SSRC. https://bugzilla.gnome.org/show_bug.cgi?id=762219 2015-11-15 14:54:28 +0100 Mikhail Fludkov * tests/check/elements/rtpsession.c: rtpsession: Add test for locking of the stats signal Keeping the lock while emitting the stats signal introduces potential deadlock in those situations when the signal callback wants the access to rtpsession's properties which also requre the lock. https://bugzilla.gnome.org/show_bug.cgi?id=762216 2016-05-19 15:36:57 +0900 Seungha Yang * gst/matroska/matroska-demux.c: matroskademux: don't hold object lock whilst pushing out headers matroskademux would take the GST_OBJECT_LOCK in - gst_matroska_demux_push_codec_data_all() - gst_matroska_demux_query() Some parse element such as FLAC checks upstream seekability, and there is some use cases that matroska-demux is linked to a parse element (e.g.,FLAC format) without intermediate elements (e.g., queue). In this case, matroska-demux never returns from _push_codec_data_all() because the parser can return only after it receives the response to the upstream query, but that's not going to happen because it's deadlocked. Elements must not hold the object lock whilst pushing out events or data. https://bugzilla.gnome.org/show_bug.cgi?id=766645 2016-05-19 12:43:01 +0300 Sebastian Dröge * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Set sent_buffers and streamheader_buffers to NULL after freeing Otherwise we might use an already freed list later and crash or worse. 2016-05-18 18:32:57 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: fix Since version for new "loop" property 2016-05-16 16:18:37 +0200 Guillaume Desmottes * gst/rtsp/gstrtpdec.c: rtpdec: fix clock leak gst_system_clock_obtain() returns a new ref. https://bugzilla.gnome.org/show_bug.cgi?id=766521 2016-05-17 05:33:35 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: add doc blurb with since marker for new "loop" property 2015-11-13 15:52:35 +0100 Dimitrios Katsaros * gst/avi/gstavimux.c: avimux: add support for png https://bugzilla.gnome.org/show_bug.cgi?id=758059 2016-05-15 22:07:14 +1000 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: splitmuxsrc: Connect to demux signals before activating Fix a race in splitmuxsrc by properly connecting to the demuxer signals we're interested in *before* setting it running. 2016-05-15 13:31:37 +0300 Sebastian Dröge * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: Update for git master 2016-05-15 12:16:23 +0200 Olivier Crête * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: rtpmp4gpay: Don't produce timestamps based on byte count The GST_BUFFER_OFFSET of output buffers returned to GstRtpBasePayload should reflect the number of "samples" in the unit of the RTP clock in this buffer. If this is not true, then it shouldn't be set. https://bugzilla.gnome.org/show_bug.cgi?id=761943 2016-05-15 12:24:03 +0200 Edward Hervey * gst/matroska/matroska-mux.c: matroska-mux: Fix strcmp usage Just use g_strcmp0 which can handle NULL entries 2016-03-04 10:14:00 +0100 Carlos Rafael Giani * ext/soup/gstsouphttpsrc.c: souphttpsrc: Use audio/x-unaligned-raw instead of audio/x-raw for L16 data Directly setting audio/x-raw caps leads to problems when the delivered data blocks do not align properly at sample boundaries (for example, a data block with 391 bytes). So, instead, set audio/x-unaligned-raw to let a parser be autoplugged. https://bugzilla.gnome.org/show_bug.cgi?id=689460 2016-05-12 11:52:09 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Parsing elst box based on version segment_duration and media_time should be parsed based on version of elst box. Specification defines that an elst box with version 1 has uint64 and int64 values for segment_duration and media_time, respectively. https://bugzilla.gnome.org/show_bug.cgi?id=766301 2016-05-14 12:57:41 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: souphttpsrc: check if request was cancelled when sending message It might be that the request was aborted by the application and we can return immediatelly 2016-05-14 12:43:54 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: souphttpsrc: proxy resolver is on by default Remove from the session creation parameters 2016-05-14 12:15:48 -0300 Thiago Santos * ext/soup/Makefile.am: soup: update build to warn about newer deprecated functions We already depend on 2.48 2016-05-14 11:09:33 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: reduce reading latency by using non-blocking read Non-blocking read will return the amount of data available without blocking to wait for the full requested size. The downside is that now it souphttpsrc needs to have a waiting mechanism in case there is no data available yet to avoid busy looping arond the inputstream. 2016-05-15 12:30:50 +0300 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Take the lock already when reading the other stats, not just for the hash table https://bugzilla.gnome.org/show_bug.cgi?id=766025 2016-05-14 17:04:57 +0100 Tim-Philipp Müller * gst/matroska/ebml-read.c: matroska: use math-compat.h for NAN define 2016-05-14 23:39:22 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Use GstBin async-handling instead of our own. Set the async-handling property on GstBin to let it manage async-handling instead of the local handling from the previous commit. Works because of #174a5e in core 2016-05-13 10:17:33 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: refactor to use Soup's sync API Replace the async API with the sync API to remove all the extra mainloop and context handling. Currently it blocks reading until 'blocksize' bytes are available but that can be improved by using: https://developer.gnome.org/gio/unstable/GPollableInputStream.html#g-pollable-input-stream-read-nonblocking https://bugzilla.gnome.org/show_bug.cgi?id=693911 2016-05-14 04:50:36 -0300 Thiago Santos * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: replace deprecated API Avoid using soup_server_run_async and old get_port() APIs, replace with me soup_server_listen and get the port through the URIs list returned from the server. 2016-05-14 12:34:10 +0200 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Upgrade debug message to error It causes the entire pipeline to fail, it should be easier to find. 2016-05-14 18:32:52 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Hide internal async state changes. When switching fragments, hide the async-start/async-done messages from the parent bin, as otherwise we sometimes (very rarely) hang in PAUSED instead of returning / continuing to PLAYING state. 2016-05-13 21:20:28 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Remove stray carriage-return from debug 2016-05-13 16:43:21 +0300 Sebastian Dröge * gst/rtp/Makefile.am: rtp: Ship gstrtpj2kcommon.h file to fix distcheck 2015-04-30 14:43:04 +0200 Jesper Larsen * gst/avi/gstavimux.c: avimux: Do not write index and header if idx is NULL Fixes criticals with e.g. videotestsrc num-buffers=1 ! identity drop-probability=1.0 ! avimux ! fakesink https://bugzilla.gnome.org/show_bug.cgi?id=748700 2016-05-12 08:43:39 -0400 Aaron Boxer * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: manage T tile invalidation bit correctly, update tile id in header correctly. 1. according to RFC, T bit is only set when either the RTP packet only contains the J2K main header, or the packet contains tile parts from multiple tiles. This is now being managed correctly in the code. The second scenario cannot happen with our payloader, since tile headers are always placed in their own RTP packet, and so a packet cannot contain tile parts from multiple tiles. However, I have added code to track if multiple tile parts are included in a single RTP packet, in case in the future we want to put header and data in same packet. 2. Old code would set the tile id to zero for all J2K packets. This is now set correctly to the appropriate tile id. https://bugzilla.gnome.org/show_bug.cgi?id=745187 2016-05-12 08:41:51 -0400 Aaron Boxer * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: manage fragmented headers correctly J2K main header framentation across multiple RTP packets is now handled correctly https://bugzilla.gnome.org/show_bug.cgi?id=745187 2016-05-11 15:04:26 -0400 Aaron Boxer * gst/rtp/gstrtpj2kcommon.h: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: rtpj2k: move common code to shared header, code clean up https://bugzilla.gnome.org/show_bug.cgi?id=745187 2016-05-11 15:01:32 -0400 Aaron Boxer * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: rtpj2k: update documentation https://bugzilla.gnome.org/show_bug.cgi?id=745187 2016-05-12 14:43:43 +0200 Patricia Muscalu * gst/auparse/gstauparse.c: * gst/auparse/gstauparse.h: auparse: Fix sticky event misordering warning Make sure that src pad has caps before sending segment event. https://bugzilla.gnome.org/show_bug.cgi?id=766359 2016-05-11 09:28:13 +0300 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Don't notify about stats property changes while taking the session lock The signal handlers might want to actually get the value of the stats property, which would take the session lock again and deadlock. This was introduced by 2e960e70750a0cb7e1117d0c09d08597866a29ee. https://bugzilla.gnome.org/show_bug.cgi?id=766025 2016-05-03 13:59:54 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: improve edts segment handling after seeks in push mode Properly handle edts segments for push-based operation seeking. We only support edts that a single segment that has media at the end, being preceeded by any number of gap segments. This also allows the qt segment rate to be respected after seeks https://bugzilla.gnome.org/show_bug.cgi?id=765669 2016-05-03 10:41:06 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: properly activate segment with rate != 1.0 Also use the qt rate to identify the position within a qt segment to properly translate playback time to qt media time https://bugzilla.gnome.org/show_bug.cgi?id=765669 2016-05-03 11:45:01 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix stall when receiving already lost packet When a packet arrives that has already been considered lost as part of a large gap the "lost timer" for this will be cancelled. If the remaining packets of this large gap never arrives, there will be missing entries in the queue and the loop function will keep waiting for these packets to arrive and never push another packet, effectively stalling the pipeline. The proposed fix conciders parts of a large gap definitely lost (since they are calculated from latency) and ignores the late arrivals. In practice the issue is rare since large gaps are scheduled immediately, and for the stall to happen the late arrival needs to be processed before this times out. https://bugzilla.gnome.org/show_bug.cgi?id=765933 2016-05-05 14:18:21 +0200 Miguel París Díaz * gst/rtpmanager/rtpsession.c: rtpsession: Take session lock when creating stats The access to the session hash table must happen while the session lock is taken, otherwise another thread might modify the hash table while we're creating the stats. https://bugzilla.gnome.org/show_bug.cgi?id=766025 2016-05-03 21:17:01 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: update segment when new duration is found Otherwise the old segment will have a shorter stop time and would cause the stream to end too early. 2016-05-04 11:37:29 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: dismember activate_segment into 2 parts One that updates and push a new segment, the other will move the stream to the new segment starting position 2016-05-04 09:30:27 +0300 Sebastian Dröge * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: dv: Use correct pixel-aspect-ratio values The previous ones resulted in odd display aspect ratios and were different from the ones used by e.g. ffmpeg. The new ones now result in display aspect ratios of 4:3 and 16:9. https://bugzilla.gnome.org/show_bug.cgi?id=765946 2015-11-09 17:55:09 +0100 George Kiagiadakis * tests/check/elements/splitmux.c: tests: add splitmuxsrc test for new "format-location" signal https://bugzilla.gnome.org/show_bug.cgi?id=753625 2015-11-09 17:51:12 +0100 George Kiagiadakis * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: add a format-location signal that allows bypassing the location property This signal allows a user to directly return a sorted list of files to be joined, so that they don't have to follow the filename pattern that the "location" property expects. https://bugzilla.gnome.org/show_bug.cgi?id=753625 2016-05-04 11:15:20 -0400 Xavier Claessens * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix deadlock case when source reaches EOS https://bugzilla.gnome.org/show_bug.cgi?id=765072 2016-05-03 22:59:27 -0700 Stefan Sauer * gst/wavparse/gstwavparse.c: wavparse: simplify and correct header scanning The wav spec tells that 'fmt' (and 'bext' if present) must come before 'data'. There is no requirement for 'fmt' to be first. We already had a list of chunks to skip, but it is easier to just skip any chunk while seeking for 'fmt'. This fixes reading files generated by ProTools. 2016-04-30 22:15:13 +0900 Hyunjun Ko * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudiodeviceprovider.c: * sys/osxaudio/gstosxaudiodeviceprovider.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: osxaudio: Support audio device provider on osx https://bugzilla.gnome.org/show_bug.cgi?id=753265 2016-05-01 15:09:27 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: set audio header rate according to calculated bps in stop_file ... now that set_fields is no longer called there by e538608b3f90539003de21c1db238f3c9b946e30 2016-04-29 15:04:11 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Store the segment sequence number in the EOS events and SEGMENT_DONE events/message Also instead of storing it per stream, store it globally in the demuxer. It's the same for each stream anyway. https://bugzilla.gnome.org/show_bug.cgi?id=765806 2016-04-11 10:54:38 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Always bind to ANY when address is a multicast address and not only on Windows For IPv6 addresses, binding to a multicast group does not work on Linux either. Always bind to ANY and then later join the multicast group. https://bugzilla.gnome.org/show_bug.cgi?id=764679 2016-04-26 17:01:49 +0800 Song Bing * sys/ximage/ximageutil.c: ximageutil: shouldn't implement transform if don't support it shouldn't implement transform if don't support it. Or gst_buffer_copy_into() will print ERROR log. https://bugzilla.gnome.org/show_bug.cgi?id=765583 2016-04-28 16:24:52 +0300 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Allow MPEG-1 Layer 1 and 2 in addition to 3 in MP4 Via the MPEG-4 Part 3 spec we can support the other layers too. Also correct the samples per frame calculation for MP3 if it's MPEG-2 or MPEG-2.5. https://bugzilla.gnome.org/show_bug.cgi?id=765725 2016-04-27 20:46:34 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Update caps for TCP whenever they change We only changed them for UDP so far, which caused the wrong seqnum-base and other information to be passed to rtpjitterbuffer/etc when seeking. This usually wasn't that much of a problem as the code there is robust enough, but every now and then it causes us to drop up to 32756 packets before we continue doing anything meaningful. https://bugzilla.gnome.org/show_bug.cgi?id=765689 2016-04-27 20:33:38 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Ensure to not take caps with the wrong pt for getting the clock-rate Especially the caps on the pad might be out of date, and the new caps would be provided for the current pt via the request-pt-map signal. https://bugzilla.gnome.org/show_bug.cgi?id=765689 2016-04-27 18:27:17 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't propagate spurious state change returns from internal elements further We handle them inside rtspsrc and override them in all other cases anyway, so do the same for "internal" state changes like PAUSED->PAUSED and PLAYING->PLAYING. This keeps unexpected NO_PREROLL to confuse state changes in GstBin. See also https://bugzilla.gnome.org/show_bug.cgi?id=760532 https://bugzilla.gnome.org/show_bug.cgi?id=765689 2016-04-27 14:09:03 +0300 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Don't override maximum audio chunk size with the scale again just before writing it set_fields() should only be called in the beginning, otherwise we will never remember the maximum audio chunk size and write a wrong block align... which then causes wrong timestamps and other problems. 2016-04-27 13:53:00 +0300 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Actually store the largest audio chunk size for the VBR case of MP2/MP3 3ea338ce271e1f6a96d2ed49d4472b091f6f8b7e changed avimux to do that, but it never actually kept track of the max audio chunk for MP3 and MP2. These are knowing the hdr.scale only after parsing the frames instead of at setcaps time. 2016-04-25 15:03:14 +0200 Mats Lindestam * gst/udp/gstmultiudpsink.c: multiudpsink: Allow setting "socket-v6" without setting "socket" too https://bugzilla.gnome.org/show_bug.cgi?id=764897 2016-04-22 15:02:16 +0100 Mario Sanchez Prada * ext/vpx/gstvpxenc.c: vpxenc: Properly handle frames with too low duration When a frame's duration is too low, calling gst_util_uint64_scale() to scale its value can result into it being truncated to zero, which will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error when trying to encode. To prevent this from happening, we simply ignore the duration when encoding if it becomes zero after scaling, logging a warning message. https://bugzilla.gnome.org/show_bug.cgi?id=765391 2016-04-22 15:48:08 +0100 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix description of linear interlacing method 2016-04-21 14:08:19 -0300 Thibault Saunier * gst/flv/gstflvmux.c: flv: Handle the case where we do not get any CollectData in handle_buffer https://bugzilla.gnome.org/show_bug.cgi?id=765320 2016-04-11 22:41:20 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Do not use unreliable framerate timescale/1 is unreliable value for framerate. Due to downstream element usually use framerate generated by qtdemux, let it be omitted until the framerate can be reliably calculated. https://bugzilla.gnome.org/show_bug.cgi?id=764733 2016-04-21 12:53:33 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: Revert "qtdemux: expose streams with first moof for fragmented format" This reverts commit d8bb6687ea251570c331038279a43d448167d6ad. https://bugzilla.gnome.org/show_bug.cgi?id=764733 2016-02-09 17:17:09 +0000 Alex Ashley * gst/isomp4/qtdemux.c: qtdemux: support seeking of CENC encrypted streams When playing a stream that has been protected by DASH CENC, playback will fail if a seek is performed. Qtdemux produces the error "stream is protected using cenc, but no cenc protection system information has been found" and playback stops. The problem is that gst_qtdemux_reset() gets called as part of the FLUSH during a seek. This function frees the protection_system_ids array. When gst_qtdemux_configure_protected_caps() is called after the seek has completed, the protection_system_ids array is empty and qtdemux is unable to create the correct output caps for the protected stream. This commit changes it to only free the protection_system_ids on hard resets. https://bugzilla.gnome.org/show_bug.cgi?id=761787 2016-04-18 14:33:10 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: add "retrieve-sender-address" property This allows disabling of sender address retrieval, which might be useful in certain scenarios, like when the socket is connected, or the sender address is not of interest (e.g. when receiving an MPEG-TS stream). Disabling sender address retrieval in those cases can have minor performance advantages. https://bugzilla.gnome.org/show_bug.cgi?id=563323 2015-11-26 13:15:06 +0100 Dimitrios Katsaros * sys/v4l2/v4l2_calls.c: v4l2: Change warning handling to break infinite message loop v4l2src can cause an "infinite message loop" when a base control exposed as a property is not provided by the device. In these cases, if in the warning message handling for the bus, the GST_DEBUG_BIN_TO_DOT_FILE* category of functions are used, the src lookup causes a new warning to be posted on the bus, causing a loop. This patch changes the warning for these controls so they are not posted on the bus. https://bugzilla.gnome.org/show_bug.cgi?id=758703 2016-04-15 10:44:02 -0400 Xavier Claessens * gst/multifile/gstsplitmuxsink.c: spitmuxsink: Avoid creating small file at EOS When EOS is reached, the current file get closed and the last GOP in the mq was written in a new file. https://bugzilla.gnome.org/show_bug.cgi?id=765072 2016-04-15 19:55:03 +0100 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.h: mpg123: fix build with msvc Fix syntax errors when compiling against cerbero-provided libmpg123 headers. We do the same as the libmpg123 internal visual studio build here. mpg123.h(1378): error C2143: syntax error: missing ')' before '(' mpg123.h(1378): error C2081: 'ssize_t': name in formal parameter list illegal mpg123.h(1378): error C2143: syntax error: missing ')' before '*' mpg123.h(1378): error C2091: function returns function mpg123.h(1378): error C2143: syntax error: missing '{' before '*' mpg123.h(1378): error C2059: syntax error: ')' mpg123.h(1379): error C2143: syntax error: missing ')' before '*' mpg123.h(1379): error C2365: 'off_t': redefinition; previous definition was 'typedef' ... 2016-04-15 19:59:15 +0300 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: S16 uses S32 temporary buffers, float/double their own type Make sure to allocate not only a S16 buffer for S16 but a twice as big one to hold S32. https://bugzilla.gnome.org/show_bug.cgi?id=765116 2016-04-16 02:17:26 +1000 Jan Schmidt * ext/pulse/pulsesink.c: Revert "pulsesink: uncork if needed upon commit" This reverts commit 0dd46accf6d282ff07065852bd91c85c78af3394. With some audiosinks, starting the ringbuffer on the first commit causes audio glitches at startup by starting to output segments from the ringbuffer before it has been filled / fully prerolled. This doesn't usually happen with pulsesink because we map the pulseaudio ringbuffer directly, but we should keep things consistent with other sinks with regards to startup latency, plus it gives more headway to avoid glitching, should the initial 2nd segment take more than 10ms to generate. https://bugzilla.gnome.org/show_bug.cgi?id=657076 2016-04-15 00:46:56 -0700 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add srtp rollover counters from mikey crypto sessions The server can send multiple crypto sessions, one for each SSRC with its own rollover counter. We parse this information and pass it to the SRTP decoder via the "request-key" signal. https://bugzilla.gnome.org/show_bug.cgi?id=730540 2016-04-15 14:35:07 +0000 Jan Schmidt * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Fix debug output when resyncing Don't output the pointer value of the time() function as a timestamp by using the correct variable. Fixes build on Raspberry Pi 3. 2016-04-15 11:36:36 +0300 Sebastian Dröge * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: If no proxy is set by properties, use the default libsoup proxy resolver That is, use whatever system settings there might exist. This is the same behaviour we use in the HTTP source. 2016-04-14 10:01:28 +0100 Julien Isorce * README: * common: Automatic update of common submodule From 6f2d209 to ac2f647 2016-04-13 18:45:07 +0100 Damian Ziobro * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Add max_files_number property https://bugzilla.gnome.org/show_bug.cgi?id=744612 2016-04-13 10:57:03 -0700 Reynaldo H. Verdejo Pinochet * gst/videomixer/videomixer2.c: videomixer: drop reference to videomixer 2 Fix a small grammar mistake on "overlayed" while at it. 2016-04-13 09:57:16 +0300 Sebastian Dröge * sys/ximage/ximageutil.c: ximage: Initialize all fields in the meta explicitly The meta is not allocated with all fields initialized to zeroes. https://bugzilla.gnome.org/show_bug.cgi?id=764902 2016-04-12 09:41:00 +0000 Paolo Pettinato * gst/rtpmanager/gstrtpmux.c: rtpmux: Forward sticky events on buffer lists too, not only on buffers https://bugzilla.gnome.org/show_bug.cgi?id=764933 2016-04-12 15:01:28 +0300 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Drain the field history if the caps are changing Otherwise we will use fields from the old caps with everything set up for the new caps, causing crashes and worse. Also don't do anything if the same caps are set twice. 2016-04-12 15:00:31 +0300 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Instead of confusing crashes later, just error out immediately if mapping a video frame fails This probably still crashes but at least we get some hint about what goes wrong instead of random behaviour later. 2016-04-12 11:38:51 +0100 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: check stream is available in PIFF parser qtdemux->streams is an array, it will never evaluate to true when comparing to NULL. Instead we want to check the number of streams to make sure the stream is available. https://bugzilla.gnome.org/show_bug.cgi?id=753614 CID 1358389 2016-04-12 11:37:36 +0100 Luis de Bethencourt * gst/isomp4/qtdemux.c: Revert "qtdemux: redundant check in PIFF parser" This reverts commit 41e10524f3babdd92aac8c8c9d5b9cdf184c2d4e. 2016-04-12 11:05:50 +0100 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: redundant check in PIFF parser qtdemux->streams is an array of size GST_QTDEMUX_MAX_STREAMS, it will never evaluate to true when comparing to NULL. https://bugzilla.gnome.org/show_bug.cgi?id=753614 CID 1358389 2016-04-12 11:56:08 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: avoid leaking GValues unset the GValue if we don't use it any more to avoid leaks. 2016-04-12 10:15:39 +0300 Sebastian Dröge * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation The head of the queue is the oldest packet (as in lowest seqnum), the tail is the newest packet. To calculate the fill level, we should calculate tail-head while considering wraparounds. Not the other way around. Other code is already doing this in the correct order. https://bugzilla.gnome.org/show_bug.cgi?id=764889 2016-04-11 10:44:56 +0300 Sebastian Dröge * gst/rtpmanager/Makefile.am: rtpmanager: It's GST_LIBS, not GST_LIBS_LIBS 2016-04-11 08:33:17 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: Fix parsing segment duration of empty edit list box For empty edit list, segment-duration in edit list box should not be used for segment event. https://bugzilla.gnome.org/show_bug.cgi?id=764870 2016-04-08 13:05:57 +0200 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: make timecodescale configurable In some use cases the default timecodescale will produce blocks with the same timestamp https://bugzilla.gnome.org/show_bug.cgi?id=764769 2016-04-07 13:01:52 +0200 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: jiterbuffer: Move assertion to the right location We shouldn't have "late" lost timers at that point 2016-03-02 14:25:24 +0100 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Speed up lost timeout handling When downstream blocks, "lost" timers are created to notify the outgoing thread that packets are lost. The problem is that for high packet-rate streams, we might end up with a big list of lost timeouts (had a use-case with ~1000...). The problem isn't so much the amount of lost timeouts to handle, but rather the way they were handled. All timers would first be iterated, then the one selected would be handled ... to re-iterate the list again. All of this is being done while the jbuf lock is taken, which in some use-cases would return in holding that lock for 10s... blocking any buffers from being accepted in input... which would then arrive late ... which would create plenty of lost timers ... which would cause the same issue. In order to avoid that situation, handle the lost timers immediately when iterating the list of pending timers. This modifies the complexity from a quadratic to a linear complexity. https://bugzilla.gnome.org/show_bug.cgi?id=762988 2016-03-02 14:23:01 +0100 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Don't create lost events if we don't need them When "do-lost" is set to FALSE we don't use/send the lost events. In that case, don't create them to start with :) https://bugzilla.gnome.org/show_bug.cgi?id=762988 2016-03-02 13:57:07 +0100 Edward Hervey * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Add tracing of lock usage Helps with debugging lock usage https://bugzilla.gnome.org/show_bug.cgi?id=762988 2016-02-10 19:56:59 +0530 Nirbheek Chauhan * sys/v4l2/gstv4l2deviceprovider.c: v4l2: Don't leak v4l2 objects and props on probe errors 2016-04-04 17:42:03 +0100 Tim-Philipp Müller * tests/check/elements/rtp-payloading.c: tests: add unit test for jpeg depayloader packet loss handling Make sure it always outputs something that looks like a valid JPEG frame, ie. starts with an SOI marker and ends with an EOI marker. 2016-03-15 03:25:26 +0530 Nirbheek Chauhan * gst/rtp/gstrtpjpegdepay.c: rtpjpegdepay: Don't send invalid frames downstream after packet loss or a DISCONT After clearing the adapter due to a DISCONT, as might happen when some packet(s) have been lost, the depayloader was pushing data into the adapter (which had no header due to the clear), creating a headerless frame out of it, and sending it downstream. The downstream decoder would then usually ignore it; unless there were lots of DISCONTs from the jitterbuffer in which case the decoder would reach its max_errors limit and throw an element error. Now we just discard that data. It is probaby not worth trying to salvage this data because non-progressive jpeg does not degrade gracefully and makes the video unwatchable even with low packet loss such as 3-5%. 2016-01-05 16:15:16 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtpjitterbuffer: Add RFC7273 media clock handling https://bugzilla.gnome.org/show_bug.cgi?id=762259 2015-07-10 09:44:15 +0200 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: PIFF box detection and parsing support The PIFF data is stored in a custom UUID box which is parsed and the crypto_info of the element is updated accordingly. This allows downstream decryptors to process and decrypt the protected content. https://bugzilla.gnome.org/show_bug.cgi?id=753614 2016-04-01 12:15:05 +0100 Luis de Bethencourt * gst/rtp/gstrtpvorbisdepay.c: rtpvorbisdepay: remove dead code payload_buffer hasn't been assigned a value before the jumps to switch_failed or packet_short. So the value must be NULL. No need to unmap and unref. CID #1316476 2016-03-31 14:57:20 +0100 Luis de Bethencourt * gst/rtp/gstrtph263pay.c: rtph263pay: fix leak Free memory of current macroblock once it isn't needed so it isn't leaked by the call of the gst_rtp_h263_pay_B_mbfinder function. if (!(mac = gst_rtp_h263_pay_B_mbfinder (context, gob, mac, mb))) { CID 1212156 2016-03-31 02:15:04 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmux: Handle a hang draining out at EOS Make sure that all data is drained out when the reference pad goes EOS. Fixes a problem where data that arrives on other pads after the reference pad finishes can stall forever and never pass EOS. https://bugzilla.gnome.org/show_bug.cgi?id=763711 2016-03-18 15:45:01 -0400 Xavier Claessens * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Fix occasional deadlock when ending file with subtitle Deadlock occurs when splitting files if one stream received no buffer during the first GOP of the next file. That can happen in that scenario for example: 1) The first GOP of video is collected, it has a duration of 10s. max_in_running_time is set to 10s. 2) Other streams catchup and we receive the first subtitle buffer at ts=0 and has a duration of 1min. 3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to 1min. That buffer is blocked in handle_mq_input() because max_in_running_time is still 10s. 4) Since all in_running_time are now > 10s, max_out_running_time is now set to 10s. That first GOP gets recorded into the file. The muxer pop buffers out of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the GstDataQueue is empty. 5) A 2nd GOP of video is collected and has a duration of 10s as well. max_in_running_time is now 20s. Since subtitle's in_running_time is already 1min, that GOP is already complete. 6) But let's say we overran the max file size, we thus set state to SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of previous GOP) arrives in handle_mq_output(), EOS event is sent downstream instead. But since the subtitle queue is empty, that's never going to happen. Pipeline is now deadlocked. To fix this situation we have to: - Send a dummy event through the queue to wakeup output thread. - Update out_running_time to at least max_out_running_time so it sends EOS. - Respect time order, so we set out_running_tim=max_in_running_time because that's bigger than previous buffer and smaller than next. https://bugzilla.gnome.org/show_bug.cgi?id=763711 2015-11-17 18:17:35 +0100 Stian Selnes * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * tests/check/elements/rtpsession.c: rtpsession: Add new signal 'on-app-rtcp' Similar to the 'on-feedback-rtcp' signal, but emitted for RTCP APP packets. https://bugzilla.gnome.org/show_bug.cgi?id=762217 2016-03-24 15:57:11 +0900 Minjae Kim * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: rtpmanager: Set to initial value for 'ntpns' in get_current_times() Initialize "ntpns" variable to -1 as the OE compiler for some reason doesn't realize that the variable is set in all code paths. https://bugzilla.gnome.org/show_bug.cgi?id=764119 2016-03-27 14:29:58 +0530 Nirbheek Chauhan * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtksink.c: * ext/gtk/gtkgstbasewidget.c: gtk: Fix logging in base widget and fix desc of GL sink Set a default category for gtkgstbasewidget lest the logging go to the 'default' category where it can't be found easily 2016-01-31 11:08:38 +1100 Sebastian Dröge * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Allow different quantization tables for components 2 and 3 RFC 2435 mentions in section 4.1 that U/V use table number 1, but this seems just like an example. Some encoders are not following that and there seems to be no reason to reject their streams. https://bugzilla.gnome.org/show_bug.cgi?id=761345 2016-03-25 17:49:14 +1100 Matthew Waters * ext/gtk/gtkgstglwidget.c: gtk/gl: don't assert when gdk doesn't provide a GL context Allows the application to check whether gtkglsink is supported by setting the element to READY. https://bugzilla.gnome.org/show_bug.cgi?id=764148 2016-03-24 19:23:12 -0400 Nicolas Dufresne * ext/vpx/gstvpxdec.c: vpxdec: Use threads on multi-core systems This is a redo of commit b848c1b6ffd1e508228820a013f94fb445e4777f. The code was lost when the elements where ported to use a baseclass. https://bugzilla.gnome.org/show_bug.cgi?id=764169 2016-02-29 23:40:03 -0300 Thiago Santos * gst/multifile/gstsplitmuxsink.c: * tests/check/elements/splitmux.c: splitmuxsink: only try to create internal sink if it doesn't exist This allows splitmuxsink to be reused after being put to NULL. Test included https://bugzilla.gnome.org/show_bug.cgi?id=762893 2015-10-01 13:41:23 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: v4l2object: probe all colorspace supported by device A device can support more than one colorspace for a given image dimension and pixel format. So we have to probe all the supported colorspace and not only rely on the default one. Otherwise we could end up with negotiation failure if the caps colorimetry field don't match the v4l2 device default one even if the v4l2 could support such colorimetry. This patch enable probing if colorspace for both capture and output device. It really makes sense for output device since the colorspace shall be set by the application and a little less for capture device which, at the moment, shall provide the colorspace; ie: the v4l2 specification seems to not take into account the fact that a capture device could do colorspace conversion. As a side effet, probing takes some times and so sligthly delay v4l2 initialization. Note that this patch only probe colorspace and not all colorspace, matrix, transfer and range combination to avoid taking too much time, especially with low-speed devices as full probing do 1782 ioctl. https://bugzilla.gnome.org/show_bug.cgi?id=755937 2016-03-24 16:21:56 +0100 Edward Hervey * tests/check/elements/flvdemux.c: check: Fix indentation 2016-03-24 16:20:39 +0100 Edward Hervey * tests/check/elements/flvdemux.c: tests: Remove unused variables 2016-03-10 08:44:57 +0900 Vineeth TM * ext/gtk/gstgtkbasesink.c: gtkbasesink: post message to application for unhandled keyboard/mouse events https://bugzilla.gnome.org/show_bug.cgi?id=763403 2016-03-04 15:50:26 +0900 Vineeth TM * ext/qt/gstqtsink.cc: bad: use new gst_element_class_add_static_pad_template() https://bugzilla.gnome.org/show_bug.cgi?id=763081 2016-03-04 15:50:26 +0900 Vineeth TM * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtksink.c: bad: use new gst_element_class_add_static_pad_template() https://bugzilla.gnome.org/show_bug.cgi?id=763081 2016-03-16 20:26:16 +0200 Sebastian Dröge * gst/interleave/deinterleave.c: deinterleave: Return the current caps on the srcpads on caps queries It's not like we could accept any other caps here. The caps are decided by the upstream caps event. Also keep the filter order intact when filtering the results against the filter caps. https://bugzilla.gnome.org/show_bug.cgi?id=763326 2016-03-04 16:14:44 +0900 Vineeth TM * ext/twolame/gsttwolamemp2enc.c: ugly: use new gst_element_class_add_static_pad_template() https://bugzilla.gnome.org/show_bug.cgi?id=763082 2016-03-04 16:14:44 +0900 Vineeth TM * ext/lame/gstlamemp3enc.c: ugly: use new gst_element_class_add_static_pad_template() https://bugzilla.gnome.org/show_bug.cgi?id=763082 2016-03-24 15:14:23 +0900 Jimmy Ohn * gst/isomp4/qtdemux.c: qtdemux: Fix qtdemux memory leak in src_convert function If we don't find the index of the sample correctly in src_convert function, we have to unref about the qtdemux before returning value. So, I have modify it about instead pass qtdemux as a parameter into src_convert function. https://bugzilla.gnome.org/show_bug.cgi?id=763973 2016-03-22 13:15:20 +0900 Jimmy Ohn * gst/isomp4/qtdemux.c: qtdemux: Add check condition for fail case in get_duration function Currently, get_duration function always return the TRUE even though it can't be set duration correctly. So, we need to add the else condition about the fail case. Also, we already set the GST_CLOCK_TIME_NONE in this function. So I have modify it which is related code in some function. https://bugzilla.gnome.org/show_bug.cgi?id=763968 2016-03-21 10:11:23 +0900 Jimmy Ohn * gst/isomp4/qtdemux.c: qtdemux: Modify data type of duration in handle_src_query function Data type of duration need to modify from guint64 to GstClockTime for consistency in handle_src_query function. https://bugzilla.gnome.org/show_bug.cgi?id=763965 2016-03-18 14:40:58 +0200 Vivia Nikolaidou * tests/check/elements/deinterlace.c: deinterlace: Added unit tests for field=auto https://bugzilla.gnome.org/show_bug.cgi?id=763869 2016-03-17 21:21:02 +0200 Vivia Nikolaidou * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Added "auto" fields mode The "auto" fields mode will detect the upstream and downstream framerates and will decide to deinterlace all or only top fields. https://bugzilla.gnome.org/show_bug.cgi?id=763869 2016-03-16 20:17:55 +0100 Havard Graff * gst/flv/gstflvdemux.c: * tests/check/elements/flvdemux.c: flvdemux: don't emit pad-added until caps are ready In other words, gst_pad_get_current_caps should never return NULL in a pad-added callback from the demuxer. Added tests for the two special cases with AAC and H.264 where this would happen every time. https://bugzilla.gnome.org/show_bug.cgi?id=763780 2016-03-04 10:30:12 +0900 Vineeth TM * ext/aalib/gstaasink.c: * ext/cairo/gstcairooverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/gstscaletempo.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/deinterlace/gstdeinterlace.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpL24pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpklvpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpstreamdepay.c: * gst/rtp/gstrtpstreampay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp9depay.c: * gst/rtp/gstrtpvp9pay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideomedian.c: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: * tests/check/elements/autodetect.c: * tests/check/elements/qtmux.c: good: use new gst_element_class_add_static_pad_template() https://bugzilla.gnome.org/show_bug.cgi?id=763076 2016-03-04 09:42:44 +0100 David Buchmann * tests/check/elements/flvmux.c: flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE https://bugzilla.gnome.org/show_bug.cgi?id=762207 2015-11-04 14:51:19 +0900 Jihae Yi * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid potentially overflowing expression https://bugzilla.gnome.org/show_bug.cgi?id=757569 2016-03-22 10:43:45 +0900 Jimmy Ohn * gst/isomp4/qtdemux.c: qtdemux: Add the function to get channels and sample rate for AAC Add aac_get_channels and sample_rate function to get these value for AAC. https://bugzilla.gnome.org/show_bug.cgi?id=749110 2016-03-24 13:33:02 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.8.0 === 2016-03-24 12:27:33 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.8.0 2016-03-24 12:02:59 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2016-03-16 20:18:41 +0200 Sebastian Dröge * gst/interleave/deinterleave.c: deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time Doing queries while holding the object lock is a bit dangerous, and in this case causes deadlocks. https://bugzilla.gnome.org/show_bug.cgi?id=763326 2016-03-17 20:53:27 +0200 Vivia Nikolaidou * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix typo to not change the input caps but our filtered caps Changing the input caps and not using them anymore afterwards is useless, and it breaks negotiation in pipelines like: gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" ! deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" ! fakesink === release 1.7.91 === 2016-03-15 12:04:39 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.7.91 2016-03-15 11:53:37 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2016-03-15 11:41:22 +0200 Sebastian Dröge * po/hu.po: * po/sr.po: po: Update translations 2016-03-15 03:26:14 +0530 Nirbheek Chauhan * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpsource.c: rtpmanager: Some comment and documentation clarifications/fixes 2016-03-13 10:33:13 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: Revert "flacparse: push tags in pre_push_frame" This reverts commit 4065fcb80a49924b70f0c8fc159dec0ff47943a1. flacparse should not push tags by itself, the base class is going to do that while properly merging in upstream tags. It just didn't because of a bug in the base class, which was hidden by this commit. https://bugzilla.gnome.org/show_bug.cgi?id=763553 2016-02-25 05:17:51 +0530 Nirbheek Chauhan * gst/rtp/dboolhuff.c: * gst/rtp/dboolhuff.h: * gst/rtp/gstrtpsbcpay.c: win32: Don't use __attribute__ on MSVC Use MSVC-equivalents for alignment and packing compiler directives when building on MSVC 2016-02-25 05:16:42 +0530 Nirbheek Chauhan * gst/matroska/ebml-read.c: win32: Don't try to include xmath.h on newer Visual Studio 2016-02-25 05:16:09 +0530 Nirbheek Chauhan * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/monoscope/gstmonoscope.c: gst Factor out endian-order RGB formats MSVC seems to ignore preprocessor conditionals inside static pad template macros. 2016-03-08 17:37:17 +0100 Thomas Roos * sys/directsound/gstdirectsoundsink.c: dirctsoundsink: Setting volume should not unmute https://bugzilla.gnome.org/show_bug.cgi?id=755106 2016-03-08 13:57:24 +0100 Thomas Roos * sys/directsound/gstdirectsoundsink.c: dirctsoundsink: Fix volume reset on unmute https://bugzilla.gnome.org/show_bug.cgi?id=755106 2016-03-08 13:03:55 +0100 Alban Bedel * sys/v4l2/gstv4l2object.c: v4l2object: fix capture with bayer formats other than bggr gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8 for all bayer formats. This is obviously broken if the device use another ordering. Fix this by properly reading the format parameter. https://bugzilla.gnome.org/show_bug.cgi?id=763318 2016-03-07 10:28:06 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: reset pending segment if we are already pushing one When upstream is running in bytes in push-mode, qtdemux will convert seeks from time to bytes and send it upstream. Upstream element will perform a byte seek and send a byte segment to qtdemux that will convert it to time and push it downstream. There is, however, the pending_segment variable that stores a new segment event to be pushed before the next data. When handling seeks as mentioned above this variable was being ignored and, if it contained some segment event, it would override the one resulting from the seek. This would restore a previous segment and would cause the seek segment to be discarded downstream. This patch fixes this issue by unrefing any pending segment as the seek from upstream should contain the latest one that should be used, as requested by the application. https://bugzilla.gnome.org/show_bug.cgi?id=763165 2016-03-07 10:27:41 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: run gst-indent Otherwise commits will fail with our indent check hook 2016-03-04 15:09:45 +0100 Josep Torra * sys/v4l2/gstv4l2object.c: v4l2: fix colorimetry for NV12 Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior. At #v4l it was described that documentation might be wrong and that we should trust this macro instead. https://bugzilla.gnome.org/show_bug.cgi?id=762529 2016-03-05 11:38:46 +0200 Sebastian Dröge * tests/examples/gtk/Makefile.am: gtk: examples: #define GST_USE_UNSTABLE_API and link with X11_LIBS X11_LIBS is needed for XInitThreads() and without the #define we get warnings about the GL API being still unstable. 2016-03-04 14:07:19 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Fix multicast group joining with provided sockets on Windows On Windows the socket will be bound to ANY instead of the multicast group, as binding to a multicast group does not work. Which would mean that we override src->addr to become ANY and won't automatically join a multicast group anymore on Windows. On Linux we would automatically join a multicast group, keep it consistent. https://bugzilla.gnome.org/show_bug.cgi?id=763093 2016-03-01 18:22:37 +0300 Sergey Borovkov * ext/qt/qtitem.cc: qml: Fix leak of the OpenGL contexts [Matthew Waters]: add NULL checks before unreffing https://bugzilla.gnome.org/show_bug.cgi?id=762999 2016-03-02 13:13:24 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases" This reverts commit a7fb7b53592d87f7983544debb74d364fc3257ad. The mutex is taken by the caller, we should keep it locked when returning so the caller can unlock it again. 2016-03-01 15:01:22 +0000 Luis de Bethencourt * gst/audioparsers/gstflacparse.c: flacparse: push tags in pre_push_frame Push a tag event before pre-roll if we have tags. https://bugzilla.gnome.org/show_bug.cgi?id=762660 === release 1.7.90 === 2016-03-01 18:15:43 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.7.90 2016-03-01 17:03:59 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/ca.po: * po/da.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/or.po: * po/pt_BR.po: * po/ro.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/tr.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2016-03-01 16:53:27 +0200 Sebastian Dröge * po/bg.po: * po/cs.po: * po/de.po: * po/fr.po: * po/nl.po: * po/pl.po: * po/ru.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: po: Update translations 2016-03-01 14:14:02 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases 2016-02-29 10:10:24 +0000 Luis de Bethencourt * gst/matroska/matroska-demux.c: matroska-demux: remove impossible condition It is impossible for a guint to have a negative value, no need to check for this. Introduced in commit 6861d11c49ea0f30d2432cf4ebf6108bc89897f1 CID 1354509 2016-02-28 10:12:36 +0100 Petr Viktorin * gst/alpha/gstalpha.c: alpha: Fix sample pipeline Use the zorder pad property to make sure the semitransparent video is on top of the background. https://bugzilla.gnome.org/show_bug.cgi?id=762809 2016-02-28 13:42:28 +0000 Tim-Philipp Müller * gst/replaygain/gstrgvolume.c: * tests/check/elements/rgvolume.c: rgvolume: make tag list writable before modifying it Making the event itself writable is not enough, it won't make the actual taglist in the event writable as well. Instead, just make a copy of the taglist and then create a new tag event from that if required, replacing the old one. Before we would inadvertently modify taglists upstream elements might still be holding on to. Add unit test for this as well. https://bugzilla.gnome.org/show_bug.cgi?id=762793 2016-02-28 13:01:34 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Properly error out if binding the UDP sockets fails udpsrc is not returning us a socket in that case. 2016-02-27 20:33:32 +0200 Sebastian Dröge * gst/goom/gstgoom.c: goom: Use goom_set_resolution() instead of recreating the goom instance when the resolution changes https://bugzilla.gnome.org/show_bug.cgi?id=762765 2016-02-27 20:32:45 +0200 Sebastian Dröge * gst/goom/gstgoom.c: Revert "goom: Initialize the goom struct only once we know width/height and recreate it if those change" This reverts commit cc6e102643c1bae928316dca9f34db028fb9a67e. 2016-02-27 20:31:15 +0200 Sebastian Dröge * gst/goom/gstgoom.c: goom: Initialize the goom struct only once we know width/height and recreate it if those change Fixes crash when the width and/or height is changing. https://bugzilla.gnome.org/show_bug.cgi?id=762765 2016-02-26 12:41:07 +0200 Sebastian Dröge * common: Automatic update of common submodule From b64f03f to 6f2d209 2016-02-25 22:54:18 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-rtp.xml: docs: add rtpopusdepay and rtpopuspay to documentation 2016-02-17 15:15:11 +0000 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopusdepay.h: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: rtp: opus: move Opus RTP payloader/depayloader from -bad to -good https://bugzilla.gnome.org/show_bug.cgi?id=756282 2016-02-17 15:10:00 +0000 Tim-Philipp Müller Merge branch 'plugin-move-rtp-opus' Move Opus RTP depayloader/payloader from -bad to -good. https://bugzilla.gnome.org/show_bug.cgi?id=756282 2016-02-25 11:33:13 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: cenc aux info parsing from mdat support in PULL mode This is already supported for PUSH mode but was failing in PULL mode. The aux info is sometimes stored in the mdat before the first sample, so the loop task needs to pull data stored at that location and perform the aux info cenc parsing. https://bugzilla.gnome.org/show_bug.cgi?id=761700 https://bugzilla.gnome.org/show_bug.cgi?id=762516 2016-02-24 11:28:09 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: prevent buffer flow if any stream failed to be exposed In some cases the stream configuration can fail, for instance if the stream is protected and no decryptor was found. For those situations the demuxer shouldn't emit any data on the corresponding source pad of the stream and bail out. https://bugzilla.gnome.org/show_bug.cgi?id=762516 2016-02-24 09:12:03 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: don't push encrypted buffer without cenc metadata When the cenc metadata is stored outside of the moof box and the stream is exposed it is possible that the cenc metadata hasn't been processed yet while the first buffer is being pushed. When this happens the buffer can't possibly be decrypted downstream so don't push it. https://bugzilla.gnome.org/show_bug.cgi?id=762516 2016-02-23 23:10:20 +1100 Matthew Waters * ext/qt/gstqtsink.cc: * ext/qt/qtitem.cc: qt: use a static_cast instead of dynamic one The dynamic_cast is a little but of overkill as the app will still crash if it fails in the later g_assert. Allows compilation with -fno-rtti https://bugzilla.gnome.org/show_bug.cgi?id=762526 2015-10-21 16:21:45 +0200 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: read saio aux_info_type as a FOURCC https://bugzilla.gnome.org/show_bug.cgi?id=756897 2016-02-23 18:27:47 +0200 Sebastian Dröge * ext/dv/gstdvdec.c: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * gst/deinterlace/gstdeinterlace.c: * gst/smpte/gstsmpte.c: gst: Handle gst_pad_get_current_caps() returning NULL gracefully 2016-02-23 18:12:54 +0200 Dave Craig * gst/rtp/gstrtph265depay.c: rtph265depay: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps() Remove calls to gst_pad_has_current_caps() which then go on to call gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just use gst_pad_get_current_caps() and check for NULL. https://bugzilla.gnome.org/show_bug.cgi?id=759539 2015-12-16 12:40:39 +0000 Dave Craig * ext/flac/gstflacenc.c: * gst/flv/gstflvmux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/rtp/gstrtph264depay.c: * gst/shapewipe/gstshapewipe.c: * gst/videocrop/gstaspectratiocrop.c: gst: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps() Remove calls to gst_pad_has_current_caps() which then go on to call gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just use gst_pad_get_current_caps() and check for NULL. https://bugzilla.gnome.org/show_bug.cgi?id=759539 2015-12-16 10:54:17 +0000 Dave Craig * gst/audioparsers/gstaacparse.c: aacparse: Handle gst_pad_get_current_caps() returning NULL gracefully This can happen when the pipeline is currently shutting down. https://bugzilla.gnome.org/show_bug.cgi?id=759539 2016-02-23 15:57:18 +0100 Linus Svensson * gst/matroska/matroska-demux.c: matroska-demux: Don't handle seek until ready https://bugzilla.gnome.org/show_bug.cgi?id=762542 2016-02-23 15:55:13 +0100 Linus Svensson * gst/matroska/matroska-demux.c: matroska-demux: Unref seek event https://bugzilla.gnome.org/show_bug.cgi?id=762542 2016-02-22 11:01:40 +0100 Aurélien Zanelli * gst/multifile/gstmultifilesink.c: multifilesink: close file on write error with next-file mode is set to buffer If we have an error during fwrite call, file stays open and thus next incoming buffer will trigger an assert when trying to opening a new file. This happens if we do not restart element, file is closed at stop, and if application handles the returned GST_FLOW_ERROR to keep bin alive. https://bugzilla.gnome.org/show_bug.cgi?id=762434 2016-02-19 23:44:42 +0100 Matej Knopp * gst/matroska/matroska-mux.c: matroskamux: don't output empty tags/tag elements Such files will not play on Android, because of bug in libwebm matroska parsing, which is still present in 6.0.1 https://bugzilla.gnome.org/show_bug.cgi?id=762349 2016-02-04 15:59:04 +0000 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroska-demux: make up an OpusHead block if possible when missing https://bugzilla.gnome.org/show_bug.cgi?id=761489 2016-02-04 10:43:15 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroska-mux: make up an OpusHead block if possible when missing This block is needed in the Matroska file, but data coming from RTP may not have one. https://bugzilla.gnome.org/show_bug.cgi?id=761489 2016-02-22 13:53:21 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: make stream-id more readable and order-friendly ... as streams are so ordered by id by e.g. decodebin (and as typically already honoured by other demuxers). 2016-02-22 13:25:51 +0100 Mark Nauwelaerts * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroska: remove confusing duplicate track uid field 2016-02-22 14:03:02 +0000 Luis de Bethencourt * gst/rtp/gstrtpvp9pay.c: rtpvp9pay: add missing break VP9_PAY_PICTURE_ID_7BITS and VP9_PAY_PICTURE_ID_15BITS are mutually exclusive options of the picture-id-mode. We can break after the first case. 1 or 2 bytes need to be added to the header length depending on the PictureID size. https://tools.ietf.org/html/draft-uberti-payload-vp9-00#section-4.2 CID 1353479 2016-01-24 17:40:37 +0300 Sergey Borovkov * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qmlglsink: Schedule onSceneGrpahInitialized to execute on render thread onSceneGraphInitialized() is called from non render thread currently when scene graph is already initialized. https://bugzilla.gnome.org/show_bug.cgi?id=761003 2016-02-22 09:09:01 +0900 Vineeth TM * gst/avi/gstavidemux.c: avidemux: Fix buffer memory leak buffer being mapped is not being unmapped in some cases https://bugzilla.gnome.org/show_bug.cgi?id=762420 2015-11-04 10:19:03 +0100 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: rtpmanager: Don't warn for duplicate/reordered packets This is a normal scenario and should not be a warning. https://bugzilla.gnome.org/show_bug.cgi?id=762208 2016-02-21 09:47:43 +0000 Tim-Philipp Müller * gst/alpha/alpha.vcproj: * gst/auparse/auparse.vcproj: * gst/avi/avi.vcproj: * gst/cutter/cutter.vcproj: * gst/debugutils/debug.vcproj: * gst/debugutils/navigationtest.vcproj: * gst/effectv/effectv.vcproj: * gst/flx/flxdec.vcproj: * gst/goom/goom.vcproj: * gst/goom2k1/goom.vcproj: * gst/interleave/interleave.vcproj: * gst/isomp4/qtdemux.vcproj: * gst/law/alaw.vcproj: * gst/law/mulaw.vcproj: * gst/matroska/matroska.vcproj: * gst/multipart/multipart.vcproj: * gst/rtp/rtp.vcproj: * gst/smpte/smpte.vcproj: * gst/spectrum/spectrum.vcproj: * gst/udp/udp.vcproj: * gst/videobox/videobox.vcproj: * gst/videocrop/videocrop.vcproj: * gst/videofilter/gamma.vcproj: * gst/videofilter/videobalance.vcproj: * gst/videofilter/videofilter.vcproj: * gst/videofilter/videoflip.vcproj: * gst/videomixer/videomixer.vcproj: * gst/wavenc/wavenc.vcproj: * gst/wavparse/wavparse.vcproj: * gst/y4m/y4menc.vcproj: * win32/MANIFEST: * win32/vs6/autogen.dsp: * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstalaw.dsp: * win32/vs6/libgstalpha.dsp: * win32/vs6/libgstalphacolor.dsp: * win32/vs6/libgstapetag.dsp: * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstauparse.dsp: * win32/vs6/libgstautodetect.dsp: * win32/vs6/libgstavi.dsp: * win32/vs6/libgstcutter.dsp: * win32/vs6/libgstdirectsound.dsp: * win32/vs6/libgsteffectv.dsp: * win32/vs6/libgstflx.dsp: * win32/vs6/libgstgoom.dsp: * win32/vs6/libgsticydemux.dsp: * win32/vs6/libgstid3demux.dsp: * win32/vs6/libgstinterleave.dsp: * win32/vs6/libgstjpeg.dsp: * win32/vs6/libgstlevel.dsp: * win32/vs6/libgstmatroska.dsp: * win32/vs6/libgstmedian.dsp: * win32/vs6/libgstmonoscope.dsp: * win32/vs6/libgstmulaw.dsp: * win32/vs6/libgstmultipart.dsp: * win32/vs6/libgstpng.dsp: * win32/vs6/libgstqtdemux.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstsmpte.dsp: * win32/vs6/libgstspeex.dsp: * win32/vs6/libgstudp.dsp: * win32/vs6/libgstvideobalance.dsp: * win32/vs6/libgstvideobox.dsp: * win32/vs6/libgstvideocrop.dsp: * win32/vs6/libgstvideoflip.dsp: * win32/vs6/libgstvideomixer.dsp: * win32/vs6/libgstwaveform.dsp: * win32/vs6/libgstwavenc.dsp: * win32/vs6/libgstwavparse.dsp: * win32/vs7/libgstdirectsound.vcproj: * win32/vs8/gst-plugins-good.sln: * win32/vs8/libgst1394.vcproj: * win32/vs8/libgstaasink.vcproj: * win32/vs8/libgstalaw.vcproj: * win32/vs8/libgstalpha.vcproj: * win32/vs8/libgstalphacolor.vcproj: * win32/vs8/libgstannodex.vcproj: * win32/vs8/libgstapetag.vcproj: * win32/vs8/libgstaudiofx.vcproj: * win32/vs8/libgstauparse.vcproj: * win32/vs8/libgstautodetect.vcproj: * win32/vs8/libgstavi.vcproj: * win32/vs8/libgstcacasink.vcproj: * win32/vs8/libgstcdio.vcproj: * win32/vs8/libgstcutter.vcproj: * win32/vs8/libgstdirectsound.vcproj: * win32/vs8/libgstdv.vcproj: * win32/vs8/libgsteffectv.vcproj: * win32/vs8/libgstflac.vcproj: * win32/vs8/libgstflxdec.vcproj: * win32/vs8/libgstgoom.vcproj: * win32/vs8/libgsticydemux.vcproj: * win32/vs8/libgstid3demux.vcproj: * win32/vs8/libgstjpeg.vcproj: * win32/vs8/libgstladspa.vcproj: * win32/vs8/libgstlevel.vcproj: * win32/vs8/libgstmatroska.vcproj: * win32/vs8/libgstmng.vcproj: * win32/vs8/libgstmonoscope.vcproj: * win32/vs8/libgstmulaw.vcproj: * win32/vs8/libgstmultipart.vcproj: * win32/vs8/libgstpng.vcproj: * win32/vs8/libgstrtp.vcproj: * win32/vs8/libgstrtsp.vcproj: * win32/vs8/libgstshout2.vcproj: * win32/vs8/libgstsmpte.vcproj: * win32/vs8/libgstspeex.vcproj: * win32/vs8/libgsttaglib.vcproj: * win32/vs8/libgstudp.vcproj: * win32/vs8/libgstvideobalance.vcproj: * win32/vs8/libgstvideobox.vcproj: * win32/vs8/libgstvideoflip.vcproj: * win32/vs8/libgstvideomixer.vcproj: * win32/vs8/libgstwavenc.vcproj: * win32/vs8/libgstwavparse.vcproj: win32: remove outdated build cruft This hasn't been touched for generations, doesn't work, and is just causing confusion. We also don't want to maintain these files manually. 2016-02-20 11:51:56 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2bufferpool.c: v4l2: don't use undeclared core debug category symbols 2016-02-06 14:39:05 +0100 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: workaround for files with wrong color_table_id value Instead of erroring out, just use the default color table. https://bugzilla.gnome.org/show_bug.cgi?id=761637 2016-02-19 15:02:04 +0000 Tim-Philipp Müller * gst/flv/gstflvmux.c: * gst/rtp/gstrtpvp9depay.c: flvmux, rtpvp9depay: fix indentation 2016-02-19 15:03:04 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2videodec.c: v4l2src: fix indentation 2015-12-04 00:46:34 +1100 Havard Graff * gst/flv/gstflvmux.c: flvmux: plug leak(s) in error-scenario https://bugzilla.gnome.org/show_bug.cgi?id=762210 2015-12-04 00:46:12 +1100 Havard Graff * gst/flv/gstflvdemux.c: flvdemux: fix eos event leak https://bugzilla.gnome.org/show_bug.cgi?id=762209 2016-02-19 14:41:07 +0000 Tim-Philipp Müller * tests/check/elements/flvdemux.c: * tests/check/elements/flvmux.c: * tests/check/elements/rtph263.c: * tests/check/elements/rtpjitterbuffer.c: tests: fix indentation 2016-02-18 16:09:29 +0100 Havard Graff * tests/check/elements/rtpjitterbuffer.c: tests: rtpjitterbuffer: port testharness to GstHarness and cleanup/improve Probably found a bug as well, in that there are some timestamps in there that are looking very wrong. (marked with FIXME) https://bugzilla.gnome.org/show_bug.cgi?id=762267 2016-02-18 10:27:19 +0100 Havard Graff * tests/check/elements/rtpjitterbuffer.c: tests: rtpjitterbuffer: test cleanups/improvements Use fail_unless and friends instead of g_assert Factor seq-num checking out to separate function Check more return-values from push and crank and others https://bugzilla.gnome.org/show_bug.cgi?id=762254 2015-12-03 11:07:05 +0100 Stian Selnes * tests/check/elements/rtpjitterbuffer.c: tests: rtpjitterbuffer: fix leaks in unit test https://bugzilla.gnome.org/show_bug.cgi?id=762214 2016-02-19 12:38:28 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.7.2 === 2016-02-19 11:49:55 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.7.2 2016-02-19 10:31:48 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: Update translations 2016-02-18 18:33:13 +0100 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: plug leaks in cenc aux info parsing 2016-02-18 13:43:07 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: fix spurious souphttpsrc test timouts Set GSETTINGS_BACKEND=memory, apparently there's something about fork() and the dconf backend (or whatever else that drags in or activates) that messes up locking and causes timeouts due to deadlocks in g_mutex_lock(), since everything works fine with CK_FORK=no as well. 2016-02-18 11:10:14 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Unmap wavpack header buffer after creating it Otherwise it will be mapped writable all the time and we can't read from it anywhere. https://bugzilla.gnome.org/show_bug.cgi?id=762239 2015-12-08 18:49:40 +0100 Stian Selnes * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Add test for big seqnum gap handling Make sure that the packets queued when detecting a big gap are pushed after reset (5 consective seqnums) and not dropped. https://bugzilla.gnome.org/show_bug.cgi?id=762211 2016-02-17 15:03:13 +0000 Tim-Philipp Müller * gst/rtp/gstrtputils.h: rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions 2016-02-09 13:17:00 +0000 Alex Ashley * gst/isomp4/qtdemux.c: qtdemux: only transform protected caps once Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 (https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the behaviour of qtdemux to call gst_qtdemux_configure_stream() for every new moof. When playing a protected stream, gst_qtdemux_configure_stream() calls gst_qtdemux_configure_protected_caps(). The gst_qtdemux_configure_protected_caps() function takes the original media format, puts this in a field called "original-media-type" and then changes the caps to "application/x-cenc". The gst_qtdemux_configure_protected_caps() did not handle the case of being called multiple times, causing it to incorrectly set the caps. The second call was causing the caps to be set to: application/x-cenc, original-media-type"application/x-cenc" This commit makes gst_qtdemux_configure_protected_caps() check that the caps have already been transformed, so that it only gets changed once. https://bugzilla.gnome.org/show_bug.cgi?id=761769 2015-11-03 14:50:53 +0200 Sebastian Dröge * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: opus: Add proper support for multichannel audio https://bugzilla.gnome.org/show_bug.cgi?id=757152 2015-06-30 13:51:33 +0200 Sebastian Dröge * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: opus: Copy metadata in the (de)payloader, but only the relevant ones The payloader didn't copy anything so far, the depayloader copied every possible meta. Let's make it consistent and just copy all metas without tags or with only the audio tag. https://bugzilla.gnome.org/show_bug.cgi?id=751774 2015-05-04 11:23:16 +0200 Sebastian Dröge * gst/rtp/gstrtpopusdepay.c: opusdepay: Set multistream=FALSE on the Opus caps The RTP Opus mapping only allows mono/stereo, and not multistream Opus streams. 2015-03-24 13:57:54 -0400 Olivier Crête * gst/rtp/gstrtpopuspay.c: rtpopuspay: Forward stereo preferences from caps upstream https://bugzilla.gnome.org/show_bug.cgi?id=746617 2015-03-24 13:56:21 -0400 Olivier Crête * gst/rtp/gstrtpopuspay.c: rtpopuspay: Set the number of channels to 2 as per RFC draft https://bugzilla.gnome.org/show_bug.cgi?id=746617 2015-03-23 12:24:55 +0100 Sebastian Dröge * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields https://bugzilla.gnome.org/show_bug.cgi?id=746617 2015-02-19 14:30:10 +0000 Vincent Penquerc'h * gst/rtp/gstrtpopuspay.c: rtpopuspay: default encoding name to OPUS https://bugzilla.gnome.org/show_bug.cgi?id=737810 2015-02-19 14:05:06 +0000 Vincent Penquerc'h * gst/rtp/gstrtpopuspay.c: rtpopuspay: make caps writable before truncating them https://bugzilla.gnome.org/show_bug.cgi?id=737810 2015-02-05 10:27:51 +0000 Vincent Penquerc'h * gst/rtp/gstrtpopuspay.c: rtpopuspay: negotiate the encoding name Chrome uses a different encoding name that gstreamer. https://bugzilla.gnome.org/show_bug.cgi?id=737810 2014-11-01 10:10:27 -0400 Nicolas Dufresne * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: rtpopus: Use OPUS encoding name Both Firefox and Chrome uses OPUS as the encoding in their SDP. Adding this now defacto standard name remove the need for special case in SDP parsing code. https://bugzilla.gnome.org/show_bug.cgi?id=737810 2013-01-31 12:30:49 +0100 Wim Taymans * gst/rtp/gstrtpopuspay.c: opuspay: fix timestamps Copy timestamps to payloaded buffer. Avoid input buffer memory leak. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929 2012-11-03 20:38:00 +0000 Tim-Philipp Müller * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopusdepay.h: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-10-22 12:08:41 +0200 Wim Taymans * gst/rtp/gstrtpopuspay.c: opuspay: remove pointless caps serialization Remove the caps serialization in the rtp caps. the spec nor the receiver does anything with it. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547 2012-10-17 17:34:26 +0100 Tim-Philipp Müller * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there. 2012-09-20 18:41:24 -0400 Olivier Crête * gst/rtp/gstrtpopuspay.c: rtpopuspay: Allocate the rtp buffer correctly Use the right functions to allocate the rtp buffer 2012-09-14 17:08:49 +0200 Mark Nauwelaerts * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopuspay.c: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-03-07 17:14:29 +0100 Mark Nauwelaerts * gst/rtp/gstrtpopuspay.c: opus: port to updated 0.11 2011-12-30 11:41:17 +0100 Edward Hervey * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopusdepay.h: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: Merge remote-tracking branch 'origin/master' into 0.11-premerge Conflicts: docs/libs/Makefile.am ext/kate/gstkatetiger.c ext/opus/gstopusdec.c ext/xvid/gstxvidenc.c gst-libs/gst/basecamerabinsrc/Makefile.am gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h gst-libs/gst/video/gstbasevideocodec.c gst-libs/gst/video/gstbasevideocodec.h gst-libs/gst/video/gstbasevideodecoder.c gst-libs/gst/video/gstbasevideoencoder.c gst/asfmux/gstasfmux.c gst/audiovisualizers/gstwavescope.c gst/camerabin2/gstcamerabin2.c gst/debugutils/gstcompare.c gst/frei0r/gstfrei0rmixer.c gst/mpegpsmux/mpegpsmux.c gst/mpegtsmux/mpegtsmux.c gst/mxf/mxfmux.c gst/videomeasure/gstvideomeasure_ssim.c gst/videoparsers/gsth264parse.c gst/videoparsers/gstmpeg4videoparse.c 2011-12-09 17:25:41 +0000 Vincent Penquerc'h * gst/rtp/gstrtpopuspay.c: opusenc: add upstream negotiation for multistream ability This will help elements that cannot deal with multistream, such as the RTP payloader. The caps now do not include a "streams" field anymore, but a "multistream" boolean, since we have no real use for knowing the exact amount of streams. https://bugzilla.gnome.org/show_bug.cgi?id=665078 2011-12-07 15:13:11 -0200 Danilo Cesar Lemes de Paula * gst/rtp/gstrtpopusdepay.c: * gst/rtp/gstrtpopusdepay.h: * gst/rtp/gstrtpopuspay.c: * gst/rtp/gstrtpopuspay.h: Adding opus RTP payloader/depayloader element Adding OPUS RTP module based on the current draft: http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt https://bugzilla.gnome.org/show_bug.cgi?id=664817 2016-02-17 13:26:02 +0000 Luis de Bethencourt * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtputils.c: * gst/rtp/gstrtputils.h: rtp: h264/h265: avoid duplication of read_golomb() There is no need to have two identical implementations of the read_golomb function. https://bugzilla.gnome.org/show_bug.cgi?id=761606 2016-02-17 14:37:44 +0100 Ognyan Tonchev * gst/matroska/matroska-demux.c: matroskademux: Simple implementation of TRICKMODE_KEY_UNITS When the trickmode key-units flag is set on the segment, simply skip any sample on a video stream that isn't a keyframe https://bugzilla.gnome.org/show_bug.cgi?id=762185 2015-08-21 14:15:18 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: send GAP events for lagging audio and video streams too Send GAP events for non-subtitle streams too if they lag too much behind, but use a higher threshold than for subtitles. This helps with fixing prerolling with a file where one of the audio streams only has data starting from 19s onwards. It's not a complete fix yet, it also requires changes elsewhere, such as in baseparse, to make sure caps are propagated. https://bugzilla.gnome.org/show_bug.cgi?id=614460 https://bugzilla.gnome.org/show_bug.cgi?id=753899 2015-12-23 19:54:13 +0100 Stian Selnes * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpvp9depay.c: * gst/rtp/gstrtpvp9depay.h: * gst/rtp/gstrtpvp9pay.c: * gst/rtp/gstrtpvp9pay.h: rtpvp9pay: rtpvp9depay: Initial implementation of draft 01 Quick and dirty implementation of an RTP payloader and depayloader for VP9. In particalur it assumes no spatial or temporal layering, non-flexible mode, and some other bits and pieces. https://bugzilla.gnome.org/show_bug.cgi?id=754773 2016-02-16 09:02:30 +0900 Vineeth TM * gst/avi/gstavidemux.c: avidemux: Fix string memory leak codec_name is not being freed in all conditions leading to memory leak https://bugzilla.gnome.org/show_bug.cgi?id=762117 2015-12-10 12:15:52 +0100 Miguel París Díaz * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: add "get-session" signal This gets the GstRTPSession element, as compared to the RTPSession object that is returned by get-internal-session. https://bugzilla.gnome.org/show_bug.cgi?id=759293 2015-12-14 11:09:46 +0900 Vineeth TM * ext/mpg123/gstmpg123audiodec.c: plugins-bad: Fix example pipelines rename gst-launch --> gst-launch-1.0 replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**) fix caps in examples https://bugzilla.gnome.org/show_bug.cgi?id=759432 2015-08-17 11:50:28 +0100 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.c: mpg123: still reset pending audio info on hard flush Follow-up to previous commit. https://bugzilla.gnome.org/show_bug.cgi?id=752431 2015-07-15 10:44:02 -0600 Jason Litzinger * ext/mpg123/gstmpg123audiodec.c: mpg123: fix handling of sample rate change during playback If the sample rate of the media changes, the resulting flush will clear the has_next_audioinfo flag, and the caps won't be sent downstream. https://bugzilla.gnome.org/show_bug.cgi?id=752431 2015-08-15 12:58:40 -0300 Thiago Santos * ext/mpg123/gstmpg123audiodec.c: audiodecoders: use default pad accept-caps handling Avoids useless check of downstream caps when handling an accept-caps query Elements: dtsdec, faad, gsmdec, mpg123audiodec, opusdec, sbcdec, adpcmdec, sirendec 2015-04-26 18:04:16 +0100 Tim-Philipp Müller * ext/mpg123/Makefile.am: Remove obsolete Android build cruft This is not needed any longer. 2015-01-11 01:08:08 +0000 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.c: mpg123: fix compiler warning and simplify checks in set_caps https://bugzilla.gnome.org/show_bug.cgi?id=740195 2015-01-03 13:06:45 +0100 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: mpg123: rework set_format code so mpg123audiodec works with decodebin/playbin The old code was using gst_caps_normalize() and was generally overly complex. Simplify by picking sample rate and number of channels from upstream and the sample format from the allowed caps. If the format caps is a list of strins, just pick the first one. And if the srcpad isn't linked yet, use the default format (S16). https://bugzilla.gnome.org/show_bug.cgi?id=740195 2014-09-10 17:24:39 +0100 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.c: Fix up one-element lists in template caps 2014-03-05 00:51:04 +0000 Tim-Philipp Müller * tests/check/elements/mpg123audiodec.c: tests: fix mpg123audiodec test for big-endian architectures 2014-02-04 17:22:27 +0100 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: mpg123: improved error report and checks Signed-off-by: Carlos Rafael Giani 2013-12-05 12:04:39 +0100 Sebastian Dröge * ext/mpg123/gstmpg123audiodec.c: mpg123audiodec: Require caps to be set before any data processing 2013-07-26 17:25:42 +0200 Edward Hervey * ext/mpg123/gstmpg123audiodec.c: mpg123: Remove dead assignment harder ? :) 2013-05-15 11:25:07 +0200 Sebastian Dröge * tests/check/elements/mpg123audiodec.c: mpg123audiodec: Fix event handling in unit test 2012-10-24 12:16:39 +0200 Sebastian Dröge * ext/mpg123/Makefile.am: gst: Add better support for static plugins 2013-04-15 00:22:39 -0700 David Schleef * ext/mpg123/gstmpg123audiodec.c: mpg123: Add conditional on API version for new enum 2016-02-16 19:59:13 +1100 Matthew Waters * ext/gtk/gstgtkbasesink.c: * ext/gtk/gstgtkbasesink.h: gtk(gl)sink: remove the signal handlers on finalize It's possible that the sink element will be freed before the widget is destroyed. When the widget was eventually destroyed, it was attempting to access member variables of the freed sink struct which resulted in undefined behaviour. Fix by disconnecting our signal on finalize. https://bugzilla.gnome.org/show_bug.cgi?id=762098 2016-02-16 00:19:00 +0000 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: rtp: h265: hook up move RTP H.265 payloader/depayloader to build https://bugzilla.gnome.org/show_bug.cgi?id=761606 2016-02-16 00:14:27 +0000 Tim-Philipp Müller * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: * gst/rtp/gstrtph265pay.c: rtp: h265: use common meta utility functions https://bugzilla.gnome.org/show_bug.cgi?id=761606 2016-02-05 18:18:31 +0000 Tim-Philipp Müller * gst/rtp/gstrtph265depay.h: * gst/rtp/gstrtph265pay.h: * gst/rtp/gstrtph265types.h: rtp: h265: remove codecparser dependency from h265 payloader/depayloader Looks like it just uses the NAL enums and nothing else from the codecparsers, and that's the only reason it had to be moved from -good to -bad when it was originally added. We can probably keep those NAL enums up to date enough, so let's remove the codecparser dependency so it can be moved back into -good. https://bugzilla.gnome.org/show_bug.cgi?id=761606 2016-02-16 00:24:58 +0000 Tim-Philipp Müller Merge branch 'plugin-move-rtp-h265' Move RTP H.265 payloader/depayloader from -bad to -good. https://bugzilla.gnome.org/show_bug.cgi?id=761606 2016-02-05 15:34:51 +0000 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: gstrtph265depay: keep consistency with rtph264depay Use gst_rtp_drop_meta() and the same function prototype for gst_rtp_copy_meta() to keep consistency with the RTP elements in gst-plugins-good 2016-02-05 13:56:34 +0000 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: fix termination of access unit Only consider the access unit complete when the next-occurring VCL NAL unit has the first bit after its NAL unit header equal to 1. 2016-01-15 16:10:02 +0000 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: fix unneeded sub-buffer creation We create a sub-buffer just to copy over its metas and then throw it away immediately, just use the original input buffer directly. 2016-01-15 15:56:59 +0000 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtph265pay: add "send VPS/SPS/PPS with every key frame" mode It's not enough to have timeout or event based VPS/SPS/PPS information sent in RTP packets. There are some scenarios when key frames may appear more frequently than once a second, in which case the minimum timeout for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough. It might also be desirable in general to make sure the VPS/SPS/PPS is available with every keyframe (packet loss aside), so receivers can actually pick up decoding immediately from the first keyframe if VPS/SPS/PPS is not signaled out of band. This commit adds the possibility to send VPS/SPS/PPS with every key frame. This mode can be enabled by setting "config-interval" property to -1. In this case the payloader will add VPS, SPS and PPS before every key (IDR) frame. https://bugzilla.gnome.org/show_bug.cgi?id=757892 2016-01-15 15:19:41 +0000 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: rtph265pay: change config-interval property type from uint to int This way we can use -1 as special value, which is nicer than MAXUINT. https://bugzilla.gnome.org/show_bug.cgi?id=757892 2015-08-15 16:22:20 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: make sure we call handle_nal for each NAL Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure we correctly extract the SPS and PPS. https://bugzilla.gnome.org/show_bug.cgi?id=730999 2015-08-15 14:45:34 +0100 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtph265pay: Copy metadata in the payloader, but only the relevant ones The payloader didn't copy anything so far, the depayloader copied every possible meta. Let's make it consistent and just copy all metas without tags or with only the video tag. https://bugzilla.gnome.org/show_bug.cgi?id=751774 2015-08-15 11:41:40 +0100 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING() https://bugzilla.gnome.org/show_bug.cgi?id=753228 2015-08-15 11:30:36 +0100 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtph265pay: fix potential crash when shutting down A race condition in the state change function may cause buffers to be unreffed while they are still used by the streaming thread in gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the parent class first in the state change function to make sure streaming has stopped and only then free those buffers. https://bugzilla.gnome.org/show_bug.cgi?id=741381 2015-08-14 15:08:08 +0100 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtph265pay: fix buffer leak when using SPS/PPS Fixes a buffer leak that would occur if the pipeline was shutdown while a SPS/PPS header was being created. https://bugzilla.gnome.org/show_bug.cgi?id=741271 2015-08-14 11:49:51 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: rtph265depay: copy metadata in the depayloader, but only the relevant ones The payloader didn't copy anything so far, the depayloader copied every possible meta. Let's make it consistent and just copy all metas without tags or with only the video tag. https://bugzilla.gnome.org/show_bug.cgi?id=751774 2015-08-12 17:54:52 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: checking if depay has sps/pps nals before insertion Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430 https://bugzilla.gnome.org/show_bug.cgi?id=753228 2015-08-12 17:22:42 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: only update the srcpad caps if something else than the codec_data changed h264parse and gstrtph264depay do the same, let's keep the behaviour consistent. As we now include the codec_data inside the stream, this causes less caps renegotiation. https://bugzilla.gnome.org/show_bug.cgi?id=753228 2015-08-12 16:43:48 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: PPS replaces old PPS if it has the same id https://bugzilla.gnome.org/show_bug.cgi?id=753228 2015-08-12 16:11:00 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: Insert SPS/PPS NALs into the stream rtph264depay does the same and this fixes decoding of some streams with 32 SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere. This looks like a mistake in the part of the spect about the codec_data. 2015-08-12 15:49:50 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: implement process_rtp_packet() vfunc For more optimised RTP packet handling: means we don't need to map the input buffer again but can just re-use the mapping the base class has already done. Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235 https://bugzilla.gnome.org/show_bug.cgi?id=753228 2015-08-12 15:14:50 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP() Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code. 2015-08-12 14:59:53 +0100 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtph265depay: prevent trying to get 0 bytes from adapter This causes an assertion and would lead to getting a NULL instead of a buffer. Without proper checking this would easily lead to a segfault. Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199 2015-07-29 17:29:28 +0100 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtp: remove dead assignment Value set to ret will be overwritten at least once at the end of the while loop, removing assignment. 2015-04-24 16:48:23 +0100 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: remove unused enum items PROP_LAST This were probably added to the enums due to cargo cult programming and are unused. 2015-03-06 14:54:41 +0000 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtp: donl_present variable unused donl_present is not implemented, yet the value is set and checked a few times. Cleaning this. CID #1249687 2015-01-08 15:36:04 +0000 Luis de Bethencourt * gst/rtp/gstrtph265pay.c: rtp: value truncated too short creates dead code type is truncated to 0-31 with "& 0x1f", but right after that it is checks if the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will never be True if the value is maximum 31 after the truncation. The intention of the code was to truncate to 0-63. 2015-01-08 15:27:44 +0000 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtp: fix nal unit type check After further investigation the previous commit is wrong. The code intended to check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c does. Type 40 would not be complete. 2015-01-08 13:47:09 +0000 Luis de Bethencourt * gst/rtp/gstrtph265depay.c: rtp: fix dead code and check for impossible values nal_type is the index for a GstH265NalUnitType enum. There are two types of dead code here: First, after checking if nal_type is >= 39 there are two OR conditionals that check if the value is in ranges higher than that number, so if nal_type >= 39 falls in the True branch those other conditions aren't checked and if it falls in the False branch and they are checked, they will always also be False. They are redundant. Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41 should never be True. Removing this redundant checks. CID 1249684 2014-10-16 10:34:01 +0200 Thijs Vermeir * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: rtp: add h265 RTP payloader + depayloader 2016-02-15 11:51:46 +0900 Vineeth TM * tests/check/elements/rtpmux.c: tests: rtpmux: Fix element memory leak https://bugzilla.gnome.org/show_bug.cgi?id=762057 2016-02-12 20:57:29 +0100 Stefan Sauer * gst/monoscope/monoscope.c: monoscope: rework the scaling code The running average was wrong and the resulting scaling factor was only held in place using the CLAMP. In addtion we are now convering quickly to volume changes. FInally now with this change, we can change the resolution defines and everythign adjusts. 2016-01-28 17:00:55 +0100 Stefan Sauer * gst/monoscope/convolve.c: * gst/monoscope/monoscope.c: * gst/monoscope/monoscope.h: monoscope: use constants in the drawing code Make all the drawing ops be based on the constants. This way we can change the fixed size at least at compile time. 2016-01-28 09:51:17 +0100 Stefan Sauer * gst/monoscope/gstmonoscope.c: monoscope: replace hardcoded values by constants This at least establishes the relationship. 2016-01-28 09:43:12 +0100 Stefan Sauer * gst/monoscope/convolve.c: * gst/monoscope/convolve.h: * gst/monoscope/monoscope.c: * gst/monoscope/monoscope.h: monoscpe: make the convolver use dynamic memory Replace all #defines with members and initialize the convolver with a parameter. 2016-01-28 08:56:44 +0100 Stefan Sauer * gst/monoscope/README: monoscope: update README We can already create multiple instances. 2016-01-28 08:53:35 +0100 Stefan Sauer * gst/monoscope/convolve.c: * gst/monoscope/monoscope.c: monoscope: code cleanup Use constants more often. Cleanup comments and add more to explain how things work. 2016-02-09 12:14:04 +1100 Matthew Waters * ext/gtk/gtkgstglwidget.c: glsyncmeta: separate out gpu/cpu waits. CPU waits are more expensive and are only required if the CPU is ever going to access the data. GPU waits perform inter-context synchronisation and are cheaper as they don't require CPU intervention. 2016-02-08 23:41:32 +0000 Luis de Bethencourt * gst/deinterlace/gstdeinterlace.c: deinterlace: remove check for impossible condition Commit bd27a1f30b4458f2edee53c76dd07fb35904b61d added a few error handling memory management checks. These check srccaps to see if it needs to be unreferenced before returning, in the case of invalid_caps this goto jump always happens before srccaps is set, so it will always be NULL in this error label. CID #1352035 2016-02-08 12:48:46 +0100 Piotr Drąg * po/POTFILES.in: po: update POTFILES https://bugzilla.gnome.org/show_bug.cgi?id=761705 2016-02-08 15:31:55 +0000 Luis de Bethencourt * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix spelling of reenqueueing To match commit 7d7074cef0272cd5155098bfc2bda6849dd89267. I love the idea of aiming for the maximum number of consecutive vowels. 2016-02-08 10:17:49 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix spelling of queueing Didn't know which one to choose between queuing and queueing, so I picked the one with the biggest amount of vowels in a row ;-P (both are acceptable apparently) 2016-02-07 15:02:35 -0500 Nicolas Dufresne * ext/jpeg/gstjpegdec.c: jpegdec: Don't pass the same data over and over We already pass the entire frame to the decoder. If the decoder ask for more data, don't pass the same data again as this leads to infinit loop. Instead, simply fail the fill function to signal the problem with that frame. It will then be skipped properly. https://bugzilla.gnome.org/show_bug.cgi?id=761670 2016-02-08 00:10:33 +0000 Tim-Philipp Müller * gst/matroska/lzo.c: matroska: get rid of _stdint.h include 2016-02-05 20:00:57 -0300 Thiago Santos * tests/check/Makefile.am: tests: extend the AM_TESTS_ENVIRONMENT from check.mak To get the CK_DEFAULT_TIMEOUT defined for all tests https://bugzilla.gnome.org/show_bug.cgi?id=761472 2016-02-05 18:04:31 -0300 Thiago Santos * autogen.sh: * common: Automatic update of common submodule From 86e4663 to b64f03f 2016-01-24 15:47:12 +0100 Holger Kaelberer * tests/examples/qt/qml/main.qml: tests: fix warning in qml example https://bugzilla.gnome.org/show_bug.cgi?id=756082 2016-01-30 18:43:30 +0100 Sebastian Dröge * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers For APP/JPG markers the size is following and we have to skip that. This is not really a problem unless the marker contains e.g. a preview JPEG or something else that we might interprete as another marker. 2016-01-26 22:37:30 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: fix framerate calculation for fragmented format qtdemux calculates framerate using duration and the number of sample. In case of fragmented mp4 format, however, the number of sample can be figure out after parsing every moof box. Because qtdemux does not parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect framerate calculation. This patch will triger gst_qtdemux_configure_stream() for every new moof. Then, framerate will be calculated by using duration and n_samples of the moof. https://bugzilla.gnome.org/show_bug.cgi?id=760774 2016-01-28 22:36:23 +0900 Seungha Yang * gst/isomp4/qtdemux.c: qtdemux: handling zero segment-duration edit list Based on document ISO_IEC_14496-12, edit list box can have segment duration as zero. It does not imply that media_start equals to media_stop. But, it just indicates a sample which should be presented at the first. This patch derives segment duration using media_time and duration of file. And set derived duration to segment-duration. https://bugzilla.gnome.org/show_bug.cgi?id=760781 2016-01-28 21:36:54 +0900 Seungha Yang * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: expose streams with first moof for fragmented format In case of push mode, qtdemux expose streams after got moov box. We can not guarantee that a moov box has sample data such as sample duration and the number of sample in stbl box for fragmented format case. So, if a moov has no sample data, streams will not be exposed until get the first moof. https://bugzilla.gnome.org/show_bug.cgi?id=760779 2016-01-27 18:48:17 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS 2016-01-27 18:44:23 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps Prevents double-negotiation during startup and in some other cases. 2016-01-27 16:43:22 +0100 Sebastian Dröge * tests/check/elements/deinterlace.c: deinterlace: Add negotiation unit tests for all 4 modes These now check the output caps based on the input caps and a following capsfilter and make sure the caps are exactly as expected. https://bugzilla.gnome.org/show_bug.cgi?id=760995 https://bugzilla.gnome.org/show_bug.cgi?id=720388 2016-01-26 17:39:20 +0100 Vivia Nikolaidou * gst/deinterlace/gstdeinterlace.c: deinterlace: Do passthrough in auto mode if downstream only supports interlaced If the following conditions are met: 1) upstream and downstream caps are compatible 2) upstream is interlaced 3) downstream doesn't support progressive mode then deinterlace will just do passthrough instead of failing to link. This is done with the following scenario in mind: videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. ! queue ! deinterlace name=dein_desktop ! autovideosink In this case, dein_src will do the deinterlacing. However, videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. ! queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue ! "video/x-raw,interlace-mode=interleaved" ! fakesink In this case, caps auto-negotiation will make dein_file and dein_desktop do the deinterlacing, while dein_src will be passthrough. https://bugzilla.gnome.org/show_bug.cgi?id=760995 2016-01-26 18:05:51 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Add mode=auto-strict In this mode we will passthrough all progressive caps but interlaced caps must be caps where we actually support deinterlacing. This is the only difference between auto and auto-strict, auto would passthrough all unsupported interlaced caps. https://bugzilla.gnome.org/show_bug.cgi?id=720388 2016-01-26 17:50:30 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Implement reconfiguration a bit better And e.g. consider reconfiguration caused by RECONFIGURE events too. https://bugzilla.gnome.org/show_bug.cgi?id=720388 2016-01-26 11:57:09 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Rewrite caps negotiation Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind of caps were last set, and e.g. if we last had interlaced caps or not. That's just broken. Also previously the handling of non-sysmem caps features was rather random and unusuable. Now the behaviour is the following, depending on the mode property: 1) mode=disabled Completely do passthrough of everything 2) mode=interlaced Only accept formats we can actually deinterlace, and accept interlaced and progressive content and always run the deinterlacer and output progressive content 3) mode=auto (i.e. playbin) Accept all progressive formats as passthrough, accept all formats that we can deinterlace ourselves (which we do then), but also accept everything else for which we then just passthrough. In auto mode, deinterlacing is best effort: If we can, we deinterlace, if we can't we just output interlaced content. https://bugzilla.gnome.org/show_bug.cgi?id=720388 https://bugzilla.gnome.org/show_bug.cgi?id=760553 2016-01-26 11:34:40 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Remove unused, obsolete bufferalloc code 2016-01-26 18:50:38 +0100 Matej Knopp * gst/matroska/matroska-mux.c: matroskamux: use A_AAC instead of A_AAC/MPEGx/y Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete https://bugzilla.gnome.org/show_bug.cgi?id=761144 2016-01-25 17:21:24 +0100 Víctor Manuel Jáquez Leal * gst/isomp4/qtdemux.c: * gst/rtp/gstrtph261pay.c: gst: Fix unintialized variable warnings While cross-compiling with Linaro GCC 5.1-2015.08, it complained about a couple unitialized variables. This patch initializes them to zero. https://bugzilla.gnome.org/show_bug.cgi?id=761094 2016-01-25 16:29:46 +1100 Matthew Waters * ext/qt/gstqtsink.cc: qt: specify that we currently only take 2D textures Fixes black screen video playback on android without a caps filter. 2016-01-25 15:03:23 +0100 George Kiagiadakis * gst/multifile/gstsplitmuxpartreader.c: splitmuxsrc: print potentially negative offset with a sign 2016-01-21 17:41:55 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Re-add colorimetry field for RGB formats This time, check if it's an RGB format and sets the transformation matrix to identity. The rest of the colorimetry information is meaningfull and shall be kept. https://bugzilla.gnome.org/show_bug.cgi?id=759624 2016-01-22 10:03:50 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: fix sRGB colorspace definition V4l2 can also use the sRGB colorspace for YUV formats and thus needs a default matrix. 2016-01-21 15:29:46 +0000 Tim-Philipp Müller * gst/debugutils/gsttaginject.c: taginject: fix sample pipeline in docs https://bugzilla.gnome.org/show_bug.cgi?id=679571 2016-01-21 10:49:44 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: Add adobe colorspace support Use the new primaries and transfer function for Adobe RGB. Explicitly list the colorimetry instead of using the default GStreamer ones. The defaults for BT2020, for example, do not match. Explicitly set the matrix of SRGB to RGB. 2016-01-20 13:41:33 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Ensure that we always have valid frame user data before using it Otherwise we're going to dereference NULL pointers. 2016-01-20 10:02:48 +0200 Sebastian Dröge * ext/vpx/gstvpxdec.c: vpxdec: Unref frame in all code paths of handle_frame() https://bugzilla.gnome.org/show_bug.cgi?id=760666 2016-01-19 22:49:20 +0100 Thibault Saunier * ext/vpx/gstvpxenc.c: vpxenc: Unref frame on ERROR All code paths for handle_frame() must somehow take ownership of the frame, be it by actually unreffing, forwarding the frame elsewhere or storing it for later. http://bugzilla.gnome.org/show_bug.cgi?id=760666 2016-01-20 18:20:43 +1100 Jan Schmidt * sys/v4l2/gstv4l2deviceprovider.c: v4l2: Don't free props structure twice. gst_v4l2_device_provider_probe_device() frees the passed props structure, don't free it again in the caller. 2016-01-19 15:15:35 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Cleanup uneeded return statement 2016-01-19 15:14:59 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't set colorimetry for non YUV formats Setting colormetry in caps for RGB have no meaning, but worst it confuses the converters downstream. https://bugzilla.gnome.org/show_bug.cgi?id=759624 2016-01-19 13:01:17 +0000 Tim-Philipp Müller * gst/rtp/gstrtpchannels.c: * gst/rtp/gstrtpchannels.h: rtp: fix compiler warnings with gcc-6 In file included from gstrtpL16depay.h:27:0, from gstrtp.c:73: gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable] static const GstRTPChannelOrder channel_orders[] = 2016-01-19 14:57:03 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Don't play anything after the end of the data chunk even when seeking Especially in push mode we would completely ignore the size of the data chunk when not stop position is given for the seek. Instead make sure that the end offset is at most the end of the data chunk if known. Without this we would output anything after the data chunk, possibly causing loud noises if the media file is followed by an INFO chunk or an ID3 tag. 2016-01-19 14:55:57 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Don't do calculations with -1 offsets when handling SEGMENT events We use that to signal "infinity", taking the difference between that and some other value is not going to give us any useful result for the end offsets of segments. 2016-01-18 11:30:45 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling" This reverts commit 271501f6576de4d141e7c2f618e28b9e3b1e5b38. It wasn't meant to be pushed yet as the commit message indicates. 2016-01-12 14:01:21 -0800 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: handle rtcp/srtcp caps properly when using interleaved data We check the stream profile and use the proper RTCP caps: application/x-srtcp if we are using a secure profile and application/x-rtcp otherwise. https://bugzilla.gnome.org/show_bug.cgi?id=760556 2016-01-05 16:15:16 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: WIP: rtpjitterbuffer: Add RFC7273 media clock handling 2016-01-15 11:36:35 +0000 Thibault Saunier * ext/vpx/gstvpxenc.c: vp8enc: Return FLOW_ERROR when an error accures FALSE would mean FLOW_OK https://bugzilla.gnome.org/show_bug.cgi?id=760666 2016-01-08 22:19:06 +0300 Sergey Borovkov * ext/qt/qtitem.cc: qml: Mark material dirty when texture buffer is updated Qt might not redraw the scene otherwise. https://bugzilla.gnome.org/show_bug.cgi?id=758286 2016-01-15 03:57:45 +0530 Nirbheek Chauhan * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: break as soon as the device is found No need to loop further if there's no side-effects for it 2016-01-15 03:56:49 +0530 Nirbheek Chauhan * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: Fix error handling when selecting/opening devices Post an element error when the CoreAudio device cannot be selected or opened. Also ensure that we post a GST_ERROR with more detail. 2016-01-13 23:40:20 +0100 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: When flushing on EOS, don't process more data than the "data" size Even if we have more data queued up when flushing than the size of the data chunk, don't process and output it. If the data size is known, this likely contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just outputting them as if they were data is going to cause unexpected behaviour and unpleasant audio noises. 2014-08-29 15:40:23 +0200 Antonio Ospite * tests/check/pipelines/wavenc.c: tests: fix a thinko in the wavenc example The code is supposed to follow somehow what the comment above says, that is to have one channel with a wave of freq 440 and the other channel with a wave of freq 880, but an off by one error results in frequencies of 0 and 440. https://bugzilla.gnome.org/show_bug.cgi?id=735673 2014-08-29 15:07:58 +0200 Antonio Ospite * gst/interleave/interleave.c: interleave: Fix the example by setting channel-masks in the sink pads The current example does not work, it fails with: ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error. gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: streaming task paused, reason not-negotiated (-4) This is because negotiation with wavenc gets messed up by the missing channel positions configuration. The proper way to define the channel layout when using the interleave element in code would be to set the channel-positions property, but gst-launch-1.0 does not know how to deal with arrays; so the example pipeline works around the issue by setting the channel-masks in the sink pads. Also fix a repetition in the deinterleave example description https://bugzilla.gnome.org/show_bug.cgi?id=735673 2016-01-11 16:29:55 +0000 Tim Sheridan * gst/audioparsers/gstsbcparse.c: sbcparse: Fix frame length calculation SBC frame length calculation wasn't being rounded up to the nearest byte (as specified in the A2DP 1.0 specification, section 12.9). This could cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly calculated frame lengths. Incorrect frame length calculation causes frame coalescing to fail, as subsequent frames in the stream aren't found in the expected locations. https://bugzilla.gnome.org/show_bug.cgi?id=742446 2016-01-10 22:54:12 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: demote warning on wrong reserved value to fixme We are likely just parsing a backward-compatible stream we don't fully support. 2016-01-08 16:27:05 -0300 Thiago Santos * gst/imagefreeze/gstimagefreeze.c: imagefreeze: simplify caps selection The downstream caps query with a filter alraedy gives us the possible intersection so there is no need to check it again with downstream if it is supported. Just try to set it directly. 2016-01-07 20:42:41 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: fix unnecessary sub-buffer creation We create a sub-buffer just to copy over its metas and then throw it away immediately, just use the original input buffer directly. 2016-01-07 20:38:27 +0000 Tim-Philipp Müller * gst/rtp/gstrtpdvdepay.c: rtpdvdepay: fix unnecessary sub-buffer creation We create a sub-buffer just to copy over its metas and then throw it away immediately, just use the original input buffer directly. 2016-01-07 20:34:05 +0000 Tim-Philipp Müller * gst/rtp/gstrtpamrdepay.c: rtpamrdepay: fix unnecessary sub-buffer creation We create a sub-buffer just to copy over its metas and then throw it away immediately, just use the original input buffer directly. 2016-01-07 20:27:29 +0000 Tim-Philipp Müller * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: fix major memory leak and performance issue We call gst_rtp_buffer_get_payload() which creates a sub-buffer of each input buffer, just to copy over metas, and then leak it. https://bugzilla.gnome.org/show_bug.cgi?id=760289 2016-01-08 15:32:47 +0200 Sebastian Dröge * tests/check/elements/rganalysis.c: rganalysis: Fix compiler warnings in the unit test elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined [-Werror,-Wshift-negative-value] push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0)); ~~ ^ elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined [-Werror,-Wshift-negative-value] push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14)); ~~ ^ elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined [-Werror,-Wshift-negative-value] push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14)); ~~ ^ 2016-01-05 18:13:06 +0000 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: don't map buffer multiple times when parsing 2016-01-07 18:20:30 +0200 Steven Hoving * gst/matroska/matroska-read-common.c: matroska: Store subtitle stream count in the correct variable And don't override the video stream count instead. 2016-01-05 18:59:06 +0200 Sebastian Dröge * gst/equalizer/gstiirequalizernbands.c: equalizer: The child-proxy API is GObject based in 1.x Not GstObject anymore. 2015-05-21 17:41:12 +0200 Pablo Anton * sys/v4l2/gstv4l2transform.c: v4l2-*: Configuring output pool correctly for using drivers min_buffer if present. Signed-off-by: Pablo Anton https://bugzilla.gnome.org/show_bug.cgi?id=755736 2015-12-31 15:46:31 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: add debug msg on CRC mismatch while validating frame header 2015-12-31 16:00:49 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: drop unneeded braces at _parse_frame() exit Additionally, drop redundant comment & line break 2015-12-31 15:55:18 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: minor grammar correction 2015-12-31 15:34:57 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: update URLs on pointers to online spec 2015-12-31 14:40:15 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: make buffer DTS setting explicitly unconditional We are setting it to PTS regardless of block_strategy 2015-12-31 14:21:40 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: add actual invalid block type to warning For someone that read the spec is clear the only *invalid* data block type is 127. For the rest, its useful information. Additionally. values 7-126 are currently reserved by the spec so the situation might change in the future. 2015-12-31 14:12:36 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: use shift instead of mask & comp We are only interested on the first bit of the first byte of the metadata block header to figure out whether is marked as the last one. The shift makes it quite clearer. 2015-12-31 12:52:13 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: warn on wishful parsing of weird headers If we get anything from 7 to 126 as type when parsing a metadata block header, we are likely dealing with a FLAC stream version we don't fully understand. Issue a warning if so. Document function assumptions regarding the passed-on type while at this. 2015-12-31 11:33:45 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: show meaningful info on frame CRC check As CRCs are calculated for the comparition already, we might as well (cheaply) inform the user how the numbers differ if a missmatched pair is found. While at it: Rephrase candidate-frame message to make more sense 2015-12-31 02:40:43 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: drop remaining trailing whitespace 2015-12-31 02:15:06 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: drop superflous else clauses 2015-12-31 01:09:51 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: factor out buffer time and offset resetting Avoids multiple occurrences of the same resetting pattern 2015-12-31 00:54:48 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: move block handling by type out of _parse_frame() 2015-10-07 18:51:25 +0900 Hyunjun Ko * gst/rtsp/gstrtspsrc.c: rtspsrc: replace duplicated codes to call new base sdp apis https://bugzilla.gnome.org/show_bug.cgi?id=745880 2015-12-30 12:16:56 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: drop redundant return statement on _header_is_valid() Fix the rather vague error message while at it. 2015-12-30 01:56:26 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: rework gst_flac_parse_frame_is_valid() drop unnecessary nesting looking for end of frame 2015-12-30 00:37:04 -0800 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstflacparse.c: flacparse: factor out context clearing routine 2015-12-29 18:05:56 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Guard against no codec data in prores caps creation CID 1346532 2015-12-29 17:58:38 +0200 Sebastian Dröge * ext/vpx/gstvpxdec.c: vpxdec: Initialize buffer variable to NULL False positive but trivial to fix and possibly causing compiler warnings at some point in the future too. CID 1346535 2015-07-27 15:53:26 +0200 Wim Taymans * sys/v4l2/gstv4l2deviceprovider.c: v4l2deviceprovider: add properties to the device Add properties to the device with exactly the same keys and sematics as what pulseaudio uses as property keys. Also handle the case when a device is probed manually and not through gudev. https://bugzilla.gnome.org//show_bug.cgi?id=759780 2015-12-25 11:41:19 +0100 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: Free the various buffers in GstBaseTransform::stop() Previously we leaked them completely, but as they're specific to the caps freeing them in stop() instead of finalize() makes most sense. 2015-12-24 15:28:06 +0100 Sebastian Dröge * configure.ac: Back to development === release 1.7.1 === 2015-12-24 14:16:21 +0100 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.7.1 2015-12-24 13:19:24 +0100 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2015-12-24 12:22:32 +0100 Sebastian Dröge * po/cs.po: * po/de.po: * po/el.po: * po/hu.po: * po/nb.po: * po/nl.po: * po/pl.po: * po/ru.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: po: Update translations 2015-12-21 09:57:33 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: drop flushes from our own offset seek Prevents downstream from receiving flushes for a seek only in upstream. Those seeks are only to start reading from the right offset when skipping or returning to qt atoms. https://bugzilla.gnome.org/show_bug.cgi?id=758928 2015-11-11 16:53:19 +0100 Thibault Saunier * gst/matroska/matroska-demux.c: matroskademux: Always set the channel mask for PCM streams Just use the gst_audio_channel_get_fallback_mask function for now as the specification is too complicated and nobody implements it. 2015-12-21 11:37:26 +0100 Thomas Roos * sys/directsound/gstdirectsoundsink.c: directsoundsink: Fix sleep for buffer-time lower than 200000 https://bugzilla.gnome.org/show_bug.cgi?id=748680 2015-12-21 12:31:19 +0100 Sebastian Dröge * configure.ac: configure: Use -Bsymbolic-functions if available While this is more useful for libraries, some of our plugins with multiple files and some internal API can also benefit from this. 2015-12-18 15:34:52 +0000 William Manley * gst/debugutils/progressreport.c: * gst/debugutils/progressreport.h: progressreport: add support for using format=buffers with do-query=false This is useful for investigating and debugging pipelines which are producing buffers at a slower/faster rate than you would expect. https://bugzilla.gnome.org/show_bug.cgi?id=759635 2015-12-18 15:49:43 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Update formats table This change add all the new RGB based format. Those format removes the ambiguity with the ALPHA channel. Some other missing multiplanar format has been added with some additional cleanup. 2015-12-18 05:17:15 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Don't write invalid edit list start time. Avoid writing a negative number as a large positive integer in an edit list when the first_ts is smaller than the first_dts - which can happen when the first packet received has a PTS but no DTS. https://bugzilla.gnome.org/show_bug.cgi?id=759615 2015-12-04 23:16:45 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Only update running time when it increases. Don't increment running time from every buffer. The correct logic to only increment when running time advances is a little further down, so delete this left-over line. 2015-11-18 11:01:20 +0100 Thibault Saunier * gst/matroska/matroska-mux.c: matroska-mux: Implement prores support https://bugzilla.gnome.org/show_bug.cgi?id=758258 2015-11-18 16:20:38 +1100 Jan Schmidt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroska-demux: Play ProRes video streams Generate video/x-prores caps for ProRes video streams. Every frame needs an 8 byte header prepended, as described in http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout so do that in a post-processing callback. https://bugzilla.gnome.org/show_bug.cgi?id=758258 2015-12-18 10:18:09 +0530 Ravi Kiran K N * ext/dv/gstdvdec.h: dvdec: Remove unused fields Remove unused fields frame_len and space https://bugzilla.gnome.org/show_bug.cgi?id=759614 2015-12-17 16:03:04 +0100 Vincent Dehors * gst/rtp/gstrtpj2kdepay.c: rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers https://bugzilla.gnome.org/show_bug.cgi?id=758943 2015-12-16 11:43:58 +0000 Luis de Bethencourt * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: dv1394: log error if failed to set socket status flag Log an error message if failed to set write or read socket as non-blocking. CID 1139608 CID 1139609 2015-12-15 17:10:00 +0000 Dave Craig * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: Check for NULL return value of gst_pad_get_current_caps() https://bugzilla.gnome.org/show_bug.cgi?id=759503 2015-12-16 09:35:53 +0100 Sebastian Dröge * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update to git 2015-12-15 19:28:05 -0500 Nicolas Dufresne * ext/qt/Makefile.am: qtsink: Add configured GL cflags to the build We don't directly link to GL in the element, though we use GL headers. For this reason we need to include the proper GL headers path. This prevent this element from using a different GL header then libgstgl. 2015-12-15 14:27:22 -0500 Nicolas Dufresne * ext/vpx/Makefile.am: vpx: Add missing headers in Makefile.am This fixes distcheck. https://bugzilla.gnome.org/show_bug.cgi?id=755510 2015-09-24 12:57:00 +0530 Prashant Gotarne * ext/vpx/Makefile.am: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvp9enc.h: * ext/vpx/gstvpxenc.c: * ext/vpx/gstvpxenc.h: vpx: created common baseclass GstVPXEnc GstVP8Enc and GstVP9Enc has almost 80% code in common. created common baseclass GstVPXEnc for GstVP8Enc and GstVP9Enc https://bugzilla.gnome.org/show_bug.cgi?id=755510 2015-12-15 12:57:53 -0500 Nicolas Dufresne * ext/vpx/gstvp9dec.c: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxdec.h: vpxdec: Remove unneeded add video_meta This also remove copies for VP8, which was not correctly in place in previous related patch. 2015-12-15 09:49:24 +0530 Prashant Gotarne * ext/vpx/Makefile.am: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9dec.h: * ext/vpx/gstvpxdec.c: * ext/vpx/gstvpxdec.h: vpx: created common base class GstVPXdec for vpx decoders Base class for the vp8dec and vp9dec. https://bugzilla.gnome.org/show_bug.cgi?id=755510 2015-12-14 11:09:46 +0900 Vineeth TM * gst/audiofx/gststereo.c: plugins-bad: Fix example pipelines rename gst-launch --> gst-launch-1.0 replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**) fix caps in examples https://bugzilla.gnome.org/show_bug.cgi?id=759432 2015-06-10 09:17:08 -0400 Xavier Claessens * configure.ac: * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Add GTlsInteraction property https://bugzilla.gnome.org/show_bug.cgi?id=750709 2015-12-14 09:05:06 -0500 Evan Callaway * gst/rtsp/gstrtspsrc.c: rtspsrc: Retry connection if tunneling needs authentication Leverage response from gst_rtsp_connection_connect_with_response to determine if the connection should be retried using authentication. If so, add the appropriate authentication headers based upon the response and retry the connection. https://bugzilla.gnome.org/show_bug.cgi?id=749596 2015-12-14 14:19:05 +0000 Luis de Bethencourt * gst/rtsp/gstrtspsrc.c: rtspsrc: check port-range format The string could exist but with a wrong format, in that case we still want to reset the values of client_port_range.min and max like we do if there is no string. CID 1139593 2015-12-14 14:55:12 +0100 Thomas Roos * sys/directsound/gstdirectsoundsink.c: directsoundsink: Check device property and fail if device can't be found Don't use default if a specific device is set but it can't be found. https://bugzilla.gnome.org/show_bug.cgi?id=759452 2015-12-14 14:15:00 +0100 Thomas Roos * sys/directsound/gstdirectsoundsink.c: directsoundsink: Fix handling of the mute property - set mute value at startup - correct set and get mute functions https://bugzilla.gnome.org/show_bug.cgi?id=755106 2015-12-14 13:43:59 +1100 Matthew Waters * ext/qt/gstqsgtexture.cc: glmemory: base classify and add the pbo memory on top The base class is useful for having multiple backing memory types other than the default. e.g. IOSurface, EGLImage, dmabuf? The PBO transfer logic is now inside GstGLMemoryPBO which uses GstGLBuffer to manage the PBO memory. This also moves the format utility functions into their own file. 2015-12-11 11:23:13 +0100 Thomas Roos * sys/directsound/gstdirectsoundsink.c: directsoundsink: Check the return value of GetStatus() too to decide if there was an error If GetStatus() fails, the status itself won't be very meaningful but we also have to look at its return value. This fixes blocking pipelines when removing sound devices or during other errors, where we wouldn't notice the error and then wait forever. https://bugzilla.gnome.org/show_bug.cgi?id=734098 2015-12-10 17:41:46 +0000 Luis de Bethencourt * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: isomp4: remove unused parameters in build_*_extension AtomTRAK parameter is not used by build_mov_alac_extension(), build_jp2h_extension(), or build_mov_alac_extension() and can be removed. 2015-12-10 15:11:07 +0000 Luis de Bethencourt * gst/isomp4/gstqtmux.c: isomp4: replace variable only used once Replace has_shift variable with value since it is only use once. 2015-12-09 12:24:09 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix packet dropping after a big discont We would queue 5 consective packets before considering a reset and a proper discont here. Instead of expecting the next output packet to have the current seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're going to drop all queued up packets. 2015-12-09 11:49:02 +0530 Ravi Kiran K N * gst/interleave/interleave.h: interleave: Remove unsed field Remove unused field collect_event in interleave. https://bugzilla.gnome.org/show_bug.cgi?id=759226 2015-12-07 16:33:14 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Stop pushing data as soon as possible in push-mode When working in push-mode, we attempt to push out everything currently buffered in the adapter. This has two pitfalls: * We could stop earlier (the moment we get a non-ok or non-not-linked) * We return the last combined flow return, which might be completely different from the previous combined flow return 2015-12-07 09:08:09 -0500 Nicolas Dufresne * autogen.sh: * common: Automatic update of common submodule From b319909 to 86e4663 2015-12-07 14:41:51 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Add a warning if an empty RTCP packet is tried to be sent https://bugzilla.gnome.org/show_bug.cgi?id=759119 2015-11-30 19:20:13 -0500 Nicolas Dufresne * configure.ac: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9dec.h: vpxdec: Use GstMemory to avoid copies With the VPX decoders it's not simple to use downstream buffer pool, because we don't know the image size and alignment when buffers get allocated. We can though use GstAllocator (for downstream, or the system allocator) to avoid a copy before pushing if downstream supports GstVideoMeta. This would still cause a copy for sink that requires specialized memory and does not have a GstAllocator for that, though it will greatly improve performance for sink like glimagesink and cluttersink. To avoid allocating for every buffer, we also use a internal buffer pool. https://bugzilla.gnome.org/show_bug.cgi?id=745372 2015-11-30 08:42:35 +0100 Edward Hervey * gst/audioparsers/gstaacparse.c: aacparse: Avoid over-skipping when checking LOAS config There might be multiple LOAS config in a row in a full frame. The first one might be a multi-layer config (which we can't properly parse yet)... but then followed by a valid (single-layer) one. The code was previously skipping whole frames (instead of just the LOAS config we failed to read) resulting in multiple frames (seen up to 6s in some situation) being dropped before finally getting the configuration. https://bugzilla.gnome.org/show_bug.cgi?id=758826 2015-11-25 17:08:56 +0100 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Properly set SPARSE stream flags for subpicture/subtitle And while we're at it, also detect 'DXSA' as being a variant fourcc of 'DXSB' for XSUB 2015-11-30 21:23:52 -0800 Reynaldo H. Verdejo Pinochet * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: grammar fix 2015-11-30 21:01:17 -0800 Reynaldo H. Verdejo Pinochet * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: switch shoutcast stream provider Fixes failing ICY test. Previous provider has streaming disabled outside UK. https://bugzilla.gnome.org/show_bug.cgi?id=758114 2015-11-18 16:10:11 +0100 Michael Olbrich * gst/avi/gstavimux.c: avimux: don't crash if we never got audio caps before stopping auds.blockalign is set once the first caps arrive. If gst_avi_mux_stop_file() is called before this happens then auds.blockalign is zero and gst_avi_mux_audsink_set_fields() cause a crash: [...] avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign; [...] https://bugzilla.gnome.org/show_bug.cgi?id=758912 2015-12-01 18:20:23 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: don't block when resurecting a buffer When we are resurecting a buffer, don't block. instead let us copy a buffer. 2015-12-01 00:30:08 -0300 Thiago Santos * gst/wavparse/gstwavparse.c: wavparse: remove extra variable to improve readability Makes it easier to see that the event is being replaced/unrefed 2015-12-01 00:22:36 -0300 Thiago Santos * gst/wavparse/gstwavparse.c: wavparse: respect seqnum in seek events Propagate the original seek seqnum to events originated from seeking to make sure they have the same value 2015-12-01 00:03:21 -0300 Thiago Santos * gst/wavparse/gstwavparse.c: wavparse: flush upstream when seeking in pull mode Makes sure upstream will unblock and return the thread so that seeking can continue https://bugzilla.gnome.org/show_bug.cgi?id=758861 2015-11-27 09:27:29 +0100 Anton Bondarenko * gst/rtp/gstrtph264pay.c: rtph264pay: add "send SPS/PPS with every key frame" mode It's not enough to have timeout or event based SPS/PPS information sent in RTP packets. There are some scenarios when key frames may appear more frequently than once a second, in which case the minimum timeout for "config-interval" of 1 second for sending SPS/PPS is not sufficient. It might also be desirable in general to make sure the SPS/PPS is available with every keyframe (packet loss aside), so receivers can actually pick up decoding immediately from the first keyframe if SPS/PPS is not signaled out of band. This patch adds the possibility to send SPS/PPS with every key frame. This mode can be enabled by setting "config-interval" property to -1. In this case the payloader will add SPS and PPS before every key (IDR) frame. https://bugzilla.gnome.org/show_bug.cgi?id=757892 2015-11-27 09:03:51 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * tests/check/elements/rtp-payloading.c: rtph264pay: change config-interval property type from uint to int This way we can use -1 as special value, which is nicer than MAXUINT. This is backwards compatible even with the GValue API, as shown by a unit test. https://bugzilla.gnome.org/show_bug.cgi?id=757892 2015-11-26 21:46:11 +0000 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: add support for Opus Add support for demuxing Opus encapsulated in MP4 files, based on the following spec: https://www.opus-codec.org/docs/opus_in_isobmff.html https://bugzilla.gnome.org/show_bug.cgi?id=742643 2015-11-25 22:48:32 +0000 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: use macro for codec_name Use _codec() macro instead of duplicating code. 2015-03-25 16:32:55 +0100 Philipp Zabel * sys/v4l2/gstv4l2videodec.c: v4l2: videodec: choose format from caps https://bugzilla.gnome.org/show_bug.cgi?id=733827 2015-03-27 15:02:33 +0100 Philipp Zabel * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: add gst_v4l2_object_probe_caps Add a variant of gst_v4l2_object_get_caps that bypasses the probed_caps cache. https://bugzilla.gnome.org/show_bug.cgi?id=733827 2015-11-19 17:20:55 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2.c: v4l2-probe: Skip devices without supported formats 2015-11-13 12:35:59 -0500 Nicolas Dufresne * configure.ac: * sys/v4l2/gstv4l2.c: v4l2: Track /dev/video* to triggered required probe If something in /dev/video* get added, removed or replaced, we need to probe the devices again in order to ensure the dynamic devices are up to date. https://bugzilla.gnome.org/show_bug.cgi?id=758085 2015-11-25 14:51:40 +1100 Alessandro Decina * gst/rtpmanager/rtpsession.c: rtpmanager: rtpsession: don't send empty RTCP packets generate_rtcp can produce empty packets when reduced size RTCP is turned on. Skip them since it doesn't make sense to push them and they cause errors with elements that expect RTCP packets to contain data (like srtpenc). 2015-11-24 10:57:28 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: restore the segment on case of soft reset When seeking back to restore the mdat position a flush is pushed through and it resets downstream segment information. Make sure that after the flush (that does a soft reset) a segment will be pushed again Fixes regressions spotted at https://ci.gstreamer.net/job/GStreamer-master-validate/2100/ 2015-11-20 12:44:22 +0000 Graham Leggett * gst/multifile/gstmultifilesink.c: multifilesink: fix spelling of variable https://bugzilla.gnome.org/show_bug.cgi?id=758390 2015-11-20 11:05:51 +0000 Luis de Bethencourt * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: unite duplicate FourCC Unite in fourcc.h the FourCCs that are used twice or more in qtdemux 2015-11-20 11:18:43 +1100 Roman Nowicki * ext/qt/qtitem.cc: qml: reuse existing GstQSGTexture Fixes a memory leak leaking the texture objects. https://bugzilla.gnome.org/show_bug.cgi?id=758286 2015-11-20 11:08:37 +1100 Matthew Waters * ext/qt/gstqsgtexture.cc: qml: activate the wrapped context when binding Mitigates the following critical gst_gl_context_thread_add: assertion 'context->priv->active_thread == g_thread_self ()' failed 2015-11-19 11:55:19 +0100 Roman Nowicki * ext/qt/qtitem.cc: qml: proper initialization if scene is already initialized The scene graph can be initialized when the we receive window handle change notification and so we will not receive a scenegraph initialization notification. Initialize ourself in this case. https://bugzilla.gnome.org/show_bug.cgi?id=758337 2015-11-19 15:33:45 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Fix capture/output-io-mode properties There was some miss-match in the implementation. This makes it concistent, though functionally it worked, except the video decoder output-io-mode getter. 2015-11-19 19:48:06 +0000 Luis de Bethencourt * gst/isomp4/atoms.c: atoms: remove unused argument of build_mov_wave_extension() AtomTrak * trak argument of build_move_wave_extension() isn't used. Removing it. 2015-11-19 19:28:20 +0000 Luis de Bethencourt * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: remove duplicate FourCC Use the available FourCCs in fourcc.h instead of duplicating them. 2015-11-19 18:36:39 +0000 Luis de Bethencourt * gst/isomp4/atoms.c: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: isomp4: centralize all FourCC 10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c already exist in fourcc.h. Don't duplicate these and use them directly. Plus moving 6 to fourcc.h, to centralize them all. 2015-11-19 17:32:12 +0000 Luis de Bethencourt * gst/matroska/webm-mux.c: matroska/webmmux: fix outdated example launch lines Update gst-launch-0.10 lines to gst-launch-1.0 2015-11-16 13:26:50 +0000 Luis de Bethencourt * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: isomp4: add support for Opus in mp4mpux Add support for muxing MP4 files containing Opus. Based on the spec detailed here: https://www.opus-codec.org/docs/opus_in_isobmff.html https://bugzilla.gnome.org/show_bug.cgi?id=742643 2015-11-17 15:23:17 -0800 Reynaldo H. Verdejo Pinochet * tests/examples/gtk/glliveshader.c: Remove unnecessary NULL checks before g_free() g_free() is NULL-safe 2015-11-18 19:10:56 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Replace tabs with spaces 2015-11-18 19:07:53 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers This fixes seeking if the first entries in the samples table are negative. The binary search would always fail on this as the array would not be sorted if interpreting the negative numbers as huge positive numbers. This caused us to always output buffers from the beginning after a seek instead of close to the seek position. Also add a case to the comparison function for equality. 2015-11-18 16:01:48 +0000 Luis de Bethencourt * gst/matroska/matroska-mux.c: matroskamux: remove duplicate check We want 1 or 2 streamheaders, the check if (bufarr->len != 1 && bufarr->len != 2) is enough. Not need to check if bufarr->len is <= 0 or > 255. 2015-11-18 14:48:36 +0900 Vineeth TM * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Fix error leak and handle error g_thread_try_new allows for possiblity of failures. In case it fails, error is not handled and leaked. https://bugzilla.gnome.org/show_bug.cgi?id=758260 2015-11-15 17:16:29 -0800 Josep Torra * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: Properly handle backward compat for event deserialization Actual code is checking for a NULL terminator and a ';' terminator, for backward compat, in a chained way that cause all events being rejected. The proper condition is to reject the events when terminator isn't in ['\0', ';'] set. https://bugzilla.gnome.org/show_bug.cgi?id=758151 2015-11-15 17:11:02 -0800 Josep Torra * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: Test for handling of custom events in rtpgst Add a simple test that checks proper serialization/deserialization of custom events with rtpgstpay and rtpgstdepay. 2015-11-16 16:23:43 -0500 Nicolas Dufresne * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: vpxdec: Use threads on multi-core systems This adds an automatic mode to the threads property of vpxdec in order to use as many threads as there is CPU on the platform. This brings back GStreamer VPX decoding performance closer to what is achieved by other players, including Chromium. https://bugzilla.gnome.org/show_bug.cgi?id=758195 2015-11-16 10:58:32 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: only send initial gaps for non-fragmented streams It would be unusual to have the header segment with an 'edts' atom indicating gaps at the beginning when handling fragmented streams. The header usually doesn't contain any timestamping information, this should come from the playlist/manifest and the segments with media in those scenarios. https://bugzilla.gnome.org/show_bug.cgi?id=758171 2015-11-17 09:41:34 -0300 Thiago Santos * gst/isomp4/qtdemux.c: Revert "Revert "qtdemux: respect qt segments in push-mode for empty starts"" This reverts commit d842ff288a9d01214a046becbfd9cbff3a4acea0. This was reverted by accident 2015-11-17 12:39:05 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: Add "loop" property for enabling/disabling multicast loopback On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it is a setting for the receiver socket. As such we will need it on udpsrc too to allow filtering out our own multicast packets. 2015-11-16 13:52:05 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: Revert "qtdemux: respect qt segments in push-mode for empty starts" This reverts commit 142d8e2d23e5602e7382977af1043d621625f8c8. 2015-11-16 16:56:04 +0900 Vineeth TM * gst/isomp4/qtdemux.c: qtdemux: Fix string memory leak The string got using g_strdup_printf will be allocated memory and should be freed after use. https://bugzilla.gnome.org/show_bug.cgi?id=758161 2015-11-14 21:51:11 -0800 Reynaldo H. Verdejo Pinochet * sys/v4l2/gstv4l2object.c: v4l2/object: remove unnecessary NULL check before g_free() 2015-11-14 21:45:29 -0800 Reynaldo H. Verdejo Pinochet * sys/oss/gstosssrc.c: osssrc: remove unnecessary NULL check before g_free() 2015-11-14 21:43:24 -0800 Reynaldo H. Verdejo Pinochet * sys/sunaudio/gstsunaudiosrc.c: sunaudiosrc: remove unnecessary NULL checks before g_free() 2015-11-14 21:36:30 -0800 Reynaldo H. Verdejo Pinochet * gst/wavparse/gstwavparse.c: wavparse: remove unnecessary NULL checks before g_free() 2015-11-14 21:31:08 -0800 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-mux.c: matroskamux: remove unnecessary NULL checks before g_free() 2015-11-14 21:26:21 -0800 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-read-common.c: matroska/read-common: remove unnecessary NULL checks before g_free() 2015-11-14 20:43:10 -0800 Reynaldo H. Verdejo Pinochet * gst/isomp4/atoms.c: isomp4/atoms: remove unnecessary NULL checks before g_free() 2015-11-14 20:35:54 -0800 Reynaldo H. Verdejo Pinochet * gst/rtp/gstrtptheorapay.c: rtp/theorapay: remove unnecessary NULL checks before g_free() 2015-11-14 20:33:54 -0800 Reynaldo H. Verdejo Pinochet * gst/rtp/gstrtpvorbispay.c: rtp/vorbispay: remove unnecessary NULL checks before g_free() 2015-11-14 20:31:34 -0800 Reynaldo H. Verdejo Pinochet * gst/rtp/gstrtpjpegpay.c: rtp/jpegpay: remove unnecessary NULL checks before g_free() 2015-11-14 20:27:04 -0800 Reynaldo H. Verdejo Pinochet * gst/rtp/gstrtpgstpay.c: rtpgstpay: remove unnecessary NULL checks before g_free() 2015-11-14 20:22:09 -0800 Reynaldo H. Verdejo Pinochet * gst/rtsp/gstrtspsrc.c: rtspsrc: remove unnecessary NULL checks before g_free() 2015-11-14 20:14:25 -0800 Reynaldo H. Verdejo Pinochet * gst/flx/gstflxdec.c: flxdec: remove unnecessary NULL check before g_free() 2015-11-14 20:09:54 -0800 Reynaldo H. Verdejo Pinochet * gst/effectv/gstop.c: effectv/optv: remove unnecessary NULL checks before g_free() 2015-11-14 20:05:03 -0800 Reynaldo H. Verdejo Pinochet * gst/effectv/gstshagadelic.c: effectv/shagadelictv: remove unnecessary NULL checks before g_free() 2015-11-14 20:01:43 -0800 Reynaldo H. Verdejo Pinochet * gst/effectv/gstripple.c: effectv/ripple: remove unnecessary NULL checks before g_free() 2015-11-14 19:56:57 -0800 Reynaldo H. Verdejo Pinochet * gst/effectv/gstradioac.c: effectv/radioac: remove unnecessary NULL checks before g_free() 2015-11-14 19:52:12 -0800 Reynaldo H. Verdejo Pinochet * gst/effectv/gststreak.c: effectv/streak: remove unnecessary NULL check before g_free() 2015-11-14 17:04:55 -0800 Reynaldo H. Verdejo Pinochet * ext/shout2/gstshout2.c: shout2: remove unnecessary NULL checks before g_free() 2015-11-14 16:57:13 -0800 Reynaldo H. Verdejo Pinochet * ext/vpx/gstvp9enc.c: vp9enc: remove unnecessary NULL check before g_free() 2015-11-14 16:54:42 -0800 Reynaldo H. Verdejo Pinochet * ext/vpx/gstvp8enc.c: vp8enc: remove unnecessary NULL check before g_free() 2015-11-14 16:20:33 -0800 Reynaldo H. Verdejo Pinochet * ext/soup/gstsouphttpsrc.c: souphttpsrc: remove unnecessary NULL checks before g_free() 2015-11-13 13:34:02 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: v4l2object: add support of NV16, NV61 and NV24 formats Mapped respectively to V4L2_PIX_FMT_NV16/V4L2_PIX_FMT_NV16M, V4L2_PIX_FMT_NV61,V4L2_PIX_FMT_NV61M and V4L2_PIX_FMT_NV24 v4l2 formats. https://bugzilla.gnome.org/show_bug.cgi?id=758058 2015-11-11 14:10:53 +0900 Vineeth TM * gst/multifile/gstsplitmuxpartreader.c: splitmuxpartreader: Fix GCond leak inactive_cond is not being cleared resulting in memory leak. https://bugzilla.gnome.org/show_bug.cgi?id=757924 2015-08-06 12:44:20 +0900 Vineeth TM * ext/jpeg/gstjpegdec.c: jpegdec: fix output state memory leak When jpeg_finish_decompress is called, output state reference is being created. But if there is any failures in finishing decompress, it jumps to setjmp, and at that point state was not referenced. Resulting in leak of output state. Hence adding another setjmp after output state is referenced. Similarly adding another setjmp to unmap the frame in case error happens before finish_decompress https://bugzilla.gnome.org/show_bug.cgi?id=753087 2015-11-10 12:32:39 +1100 Matthew Waters * ext/gtk/gstgtkglsink.c: gtk: add the overlaycomposition feature to the template caps There is a possibility that the _get_caps impl will be called with the feature in the filter caps which when interecting with the template, will return EMPTY and therefore fail negotiation. https://bugzilla.gnome.org/show_bug.cgi?id=757854 2015-08-10 11:23:45 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: respect qt segments in push-mode for empty starts In push-mode it is hard to support qt segments overall but it is possible to support when the file isn't heavily edited but just contain a segment to indicate a gap at the beginning. This also allows properly timestamping data that has negative DTS in push-mode. It is relevant to support those for 2 scenarios: 1) fragmented streaming 2) HTTP playback of 'regular' mp4 https://bugzilla.gnome.org/show_bug.cgi?id=753484 2015-11-05 18:39:33 +0530 Nirbheek Chauhan * ext/pulse/pulsedeviceprovider.c: pulse: Don't leak caps and structures in the device provider 2015-11-04 19:01:20 +0530 Arun Raghavan * gst/rtpmanager/rtpsession.c: rtpmanager: Document properties that are expressed in bits per second This changed in 928cd110bcea5d143cab3ea747991851d52ecbad and 73c0c2920f9aca96982a4de0c20b3417aa148b81 but was not documented. https://bugzilla.gnome.org/show_bug.cgi?id=747863 2015-11-04 18:51:32 +0530 Arun Raghavan * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: rtpmanager: Trivial gst-indent fixes 2015-08-12 13:35:40 +0200 Philippe Normand * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: support for cenc auxiliary info parsing outside of moof box When the cenc aux info index is out of moof boundaries, keep track of it and parse the beginning of the mdat box, before the first sample. https://bugzilla.gnome.org/show_bug.cgi?id=755614 2015-11-03 20:33:10 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Use codecutils helpers for creating Opus caps Also fix up codec data with values from the container. https://bugzilla.gnome.org/show_bug.cgi?id=757152 2015-11-03 14:51:48 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: There is no multistream field for Opus anymore https://bugzilla.gnome.org/show_bug.cgi?id=757152 2015-11-03 12:42:52 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: * gst/matroska/webm-mux.c: matroska/webmmux: Support Opus in webmmux and VP9 in matroskamux https://bugzilla.gnome.org/show_bug.cgi?id=729950 2015-11-03 12:40:15 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Parse and handle CodecDelay, SeekPreroll and DiscardPadding https://bugzilla.gnome.org/show_bug.cgi?id=727305 2015-11-03 12:18:19 +0200 Sebastian Dröge * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroskamux: Write CodecDelay, DiscardPadding and SeekPreroll for Opus And also adjust timestamps and durations according to the codec delay, both should include it for whatever reason. https://bugzilla.gnome.org/show_bug.cgi?id=727305 2015-11-03 11:49:54 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Opus headers are not in-band https://bugzilla.gnome.org/show_bug.cgi?id=727305 2015-11-03 22:01:07 +0530 Arun Raghavan * sys/v4l2/gstv4l2.c: v4l2: Set O_CLOEXEC on the device fd This is needed to make sure that child processes don't inherit the video device fd which can cause problems with some drivers. 2015-11-03 14:46:30 +0000 Luis de Bethencourt * gst/rtpmanager/gstrtpjitterbuffer.c: rtpmanager: switch G_GINT64_FORMAT for GST_STIME_ARGS No need to use G_GINT64_FORMAT for potentially negative values of GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS. Plus it creates more readable values in the logs. https://bugzilla.gnome.org/show_bug.cgi?id=757480 2015-11-03 14:26:29 +0000 Luis de Bethencourt * gst/rtpmanager/rtpsource.c: rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff No need to manually handle negative values of diff, GST_STIME_ARGS does exactly this. 2015-11-02 16:53:15 +0000 Luis de Bethencourt * gst/videomixer/videomixer2.c: videomixer: use GST_STIME_ARGS for GstClockTimeDiff No need to manually handle negative values of diff, GST_STIME_ARGS does exactly this. 2015-11-02 16:43:46 +0000 Luis de Bethencourt * gst/deinterlace/gstdeinterlace.c: deinterlace: use GST_STIME_ARGS for GstClockTimeDiff No need to manually handle negative values of diff, GST_STIME_ARGS is available for this. 2015-10-30 10:05:37 +0530 Ravi Kiran K N * gst/audiofx/audiochebband.c: audiochebband: Fix typo in example pipeline Fix typo in example pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=757340 2015-10-28 23:47:30 +0530 Nirbheek Chauhan * sys/v4l2/gstv4l2deviceprovider.c: v4l2: fix double-unref in the v4l2 device provider 2015-10-27 10:48:00 +0100 Nicola Murino * gst/matroska/matroska-ids.c: matroskamux: don't drop JPEG frames that only have PTS but no DTS set For the MS/VfW codec ids, we want to write DTS timestamps instead of PTS because that's what everyone else seems to do (and it's also how it is in AVI). So for those input formats we use the buffer DTS instead of the PTS. However, if there's no DTS set but only the PTS then just take the PTS instead of dropping the input buffer. This is useful especially for I-frame only codecs like JPEG and huffyuv, but should also be fine as fallback in general. Fixes regression with input JPEG frames that only have PTS set on them. https://bugzilla.gnome.org/show_bug.cgi?id=756967 2015-10-24 23:57:38 +0200 George Kiagiadakis * tests/check/elements/splitmux.c: tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads https://bugzilla.gnome.org/show_bug.cgi?id=753622 2015-10-24 23:57:29 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxsink.c: splitmuxsink: do not destroy the multiqueue & muxer when going to NULL Instead, delay it until all request pads have been released. This is because the release_pad() vfunc requires the multiqueue and muxer to be there in order to release their request pads as well. If those elements are destroyed earlier, release_pad() does not work, no pads are released and some resources are leaked. https://bugzilla.gnome.org/show_bug.cgi?id=753622 2015-10-20 15:28:10 +0300 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Read buffer timestamp *after* actually setting it https://bugzilla.gnome.org/show_bug.cgi?id=756809 2015-10-24 17:14:07 +0300 Sebastian Dröge * gst/audiofx/gstscaletempo.c: * gst/audiofx/gstscaletempo.h: scaletempo: Fix handling of rate < 0 We have to reverse all samples in a buffer before processing them to properly have continuous data from one buffer to another. As a result we will have a negative applied rate and a rate of 1.0. Also make sure that input buffers are correctly clipped to the segment, otherwise our calculations are going to go wrong. Also copy over the segment event's sequence number to the output segment while we're at it. https://bugzilla.gnome.org/show_bug.cgi?id=757033 2015-10-19 18:04:56 -0300 Thiago Santos * gst/deinterlace/gstdeinterlace.c: deinterlace: break as soon as non-interlaced if found It looks for a non-interlaced entry on the filter caps, break as soon as one is found to avoid wasting cpu 2015-10-19 17:50:28 -0300 Thiago Santos * gst/deinterlace/gstdeinterlace.c: deinterlace: implement accept-caps Implement accept-caps handler to avoid doing a full caps query downstream to handle it. This commit implements accept-caps as a simplification of the _getcaps function, so it exposes the same limitations that getcaps would. For example, not accepting renegotiation to caps with capsfeatures when it was last configured to a caps that it has to deinterlace. 2015-10-19 17:06:28 -0300 Thiago Santos * tests/check/elements/deinterlace.c: tests: deinterlace: fix small typo in comment 2015-10-26 00:41:28 +1100 Jan Schmidt * tests/files/Makefile.am: check: Dist splitvideo0[012].ogg test files. 2015-10-23 20:16:17 +0300 Sebastian Dröge * gst/audiofx/gstscaletempo.c: * gst/audiofx/gstscaletempo.h: scaletempo: Add support for F64 2015-10-22 17:40:38 -0700 Mischa Spiegelmock * docs/plugins/inspect/plugin-rtp.xml: * gst/multipart/multipartdemux.c: * gst/rtp/README: * gst/rtp/gstrtpvp8pay.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/udp/gstudpsrc.c: docs: Minor fixes in various places https://bugzilla.gnome.org/show_bug.cgi?id=756996 2015-10-21 17:43:31 +0100 Luis de Bethencourt * gst/goom/plugin_info.c: goom: remove compiler trick After commit 2cb6cfed22166b262ae50cb58f3ff11dd8ba91f9 there is no need to trick the compiler anymore about the usage of variable cpuFlavour. 2015-10-21 14:35:02 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From b99800a to b319909 2015-10-21 17:41:38 +0530 Ravi Kiran K N * gst/audiofx/audiofxbaseiirfilter.h: audiofx: remove unused variable Remove unsued variable have_coeffs in audiofxbaseiirfilter https://bugzilla.gnome.org/show_bug.cgi?id=756905 2015-10-20 17:29:42 +0300 Sebastian Dröge * configure.ac: Use new GST_ENABLE_EXTRA_CHECKS #define https://bugzilla.gnome.org/show_bug.cgi?id=756870 2015-10-21 14:25:55 +0300 Sebastian Dröge * README: * common: Automatic update of common submodule From 9aed1d7 to b99800a 2015-10-21 11:53:09 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: relax creation time parsing Parse wrong timestamps like we used to write as well, e.g. 10:9:42, and the hour might be without a leading zero in any case. 2015-10-21 11:45:35 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: fix indentation 2015-10-21 11:44:50 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: extract both creation date and time Before we only extracted the date part. 2015-10-21 11:16:01 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: fix writing of creation time Don't write time as e.g. 11:9:42 2015-10-13 12:42:56 -0300 Thiago Santos * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: update fragment offset It was always being set to 0, making the resulting stream broken for the receiver https://bugzilla.gnome.org/show_bug.cgi?id=756422 2015-10-19 15:36:37 +0300 Ryan Hendrickson * gst/isomp4/gstqtmux.c: qtmux: Don't unconditionally use strnlen() It's not available on older OSX and we can as well use memchr() here. https://bugzilla.gnome.org/show_bug.cgi?id=756154 2015-10-19 17:38:32 +0900 Vineeth TM * gst/auparse/gstauparse.c: auparse: Fix event memory leak Free the event after being handled to prevent memory leak. https://bugzilla.gnome.org/show_bug.cgi?id=756799 2015-10-19 09:14:19 +0100 Tim-Philipp Müller * gst/isomp4/gstqtmuxmap.c: qtmux: unify raw audio caps into a single caps structure 2015-10-19 15:15:30 +1100 Matthew Waters * ext/qt/qtitem.cc: gl: be consistent in gobject boilerpate GST_GL_IS_* vs GST_IS_GL_* git grep -l 'GST_GL_IS_' | xargs sed -i 's/GST_GL_IS_/GST_IS_GL_/g' 2015-10-19 15:15:30 +1100 Matthew Waters * ext/gtk/gtkgstglwidget.c: gl: be consistent in gobject boilerpate GST_GL_IS_* vs GST_IS_GL_* git grep -l 'GST_GL_IS_' | xargs sed -i 's/GST_GL_IS_/GST_IS_GL_/g' 2015-10-17 15:26:46 +1100 Matthew Waters * tests/examples/gtk/glliveshader.c: glshaderelement: implement on-demand create-shader signalling One may not have an GstGLContext available or current in the thread where one would need to update the shader. Support this by signalling create-shader whenever the one-shot 'update-shader' is set to TRUE. 2015-10-17 02:40:50 +1100 Matthew Waters * ext/gtk/gstgtkbasesink.c: gtk: separate out the widget/window destroy callbacks Fixes assertion due to the sink_finalize() being run before the widget destroy callback. https://bugzilla.gnome.org/show_bug.cgi?id=755969 2015-10-17 01:08:29 +1100 Matthew Waters * tests/examples/gtk/Makefile.am: * tests/examples/gtk/glliveshader.c: gl/examples: add a live shader demo using the new GstGLSLStage Implemented with videotestsrc ! glshader ! glupload ! gtkglsink Errors on an invalid shader compilation are ignored however any error provided by the glsl compiler is printed to stdout. 2015-10-14 15:42:50 -0700 Reynaldo H. Verdejo Pinochet * gst/isomp4/qtdemux.c: qtdemux: add support for FFV1 coded streams in mov https://bugzilla.gnome.org/show_bug.cgi?id=752495 2015-09-04 16:02:32 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: glshader: port to using GstGLSLStage objects for string management A GstGLShader is now simply a collection of stages that are compiled and linked together into a program. The uniform/attribute interface has remained the same. 2015-10-14 15:53:26 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: EOS immediately if we have an empty seek segment https://bugzilla.gnome.org/show_bug.cgi?id=748316 2015-10-14 10:43:19 +0300 Stavros Vagionitis * ext/soup/gstsouphttpsrc.c: souphttpsrc: Make non-inclusive segment boundaries inclusive The problem is that the filesrc and souphttpsrc are behaving differently regarding the calculation of the segment boundaries. The filesrc is using a non-inclusive boundaries, while the souphttpsrc uses inclusive. Currently the hlsdemux calculates the boundaries as inclusive, so for this reason there is no problem with the souphttpsrc, but there is an issue in the filesrc. The GstSegment is non-inclusive, so the proposed solution is to use non-inclusive boundaries in the hlsdemux in order to be consistent. Make the change in the hlsdemux, will break the souphttpsrc, which will expect inclusive boundaries, but the hlsdemux will offer non-inclusive. This change makes sure that the non-inclusive boundaries are converted to inclusive. https://bugzilla.gnome.org/show_bug.cgi?id=748316 2015-10-11 22:07:54 +0000 Graham Leggett * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpclientsink.h: souphttpclientsink: Add the retry and retry-delay properties These allow a failed request to be retried after the given number of seconds instead of failing the pipeline. Take account of the Retry-After header if present. Add retries parameter that controls the number of times an HTTP request will be retried before failing. https://bugzilla.gnome.org/show_bug.cgi?id=756318 2015-10-14 12:03:15 +0200 Guillaume Desmottes * gst/isomp4/qtdemux.c: qtdemux: fix caps leak If the QtDemuxStream are re-used they may already have caps which used to be leaked. Reproduced using the validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate scenario. https://bugzilla.gnome.org/show_bug.cgi?id=756561 2015-10-14 09:29:50 +0900 Vineeth TM * gst/isomp4/qtdemux.c: qtdemux: Fix taglist memory leak Free the stream and its sub items instead of just the stream https://bugzilla.gnome.org/show_bug.cgi?id=756544 2015-10-11 12:06:26 +0100 Thibault Saunier * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Allow negotiating to S8 as a raw format but stop making it best choice Negotiation to audio/x-raw,format=S8 was not possible because S8 does not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;` https://bugzilla.gnome.org/show_bug.cgi?id=756387 2015-10-11 09:18:40 +0100 Thibault Saunier * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Add prores support https://bugzilla.gnome.org/show_bug.cgi?id=756388 2015-10-12 18:56:32 +0100 Tim-Philipp Müller * tests/check/Makefile.am: tests: add GST_PLUGINS_BASE_LIBS for flvdemux check So it pulls in the right libgsttag-1.0. 2015-10-11 22:27:47 +0100 Julien Isorce * gst/goom/Makefile.am: * gst/goom/gstaudiovisualizer.c: * gst/goom/gstaudiovisualizer.h: * gst/goom/gstgoom.h: * gst/goom2k1/Makefile.am: * gst/goom2k1/gstaudiovisualizer.c: * gst/goom2k1/gstaudiovisualizer.h: * gst/goom2k1/gstgoom.h: goom/goom2k1: remove obsolete left over files They now use the new GstAudioVisualizer base class from gst-plugins-base/gst-libs/gst/pbutils Also fixed undefined reference to gst_audio_visualizer_get_type Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD. https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-10-12 10:48:23 +0900 Vineeth TM * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: Fix buffer memory leak during failures mapped buffer is not being unmapped during failures https://bugzilla.gnome.org/show_bug.cgi?id=756231 2015-10-12 11:18:51 +0900 Vineeth TM * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Check if soup message is created If soup message is not created then the same should not be passed on, which is resulting in segfault. Hence throwing a warning message and returning https://bugzilla.gnome.org/show_bug.cgi?id=755326 2015-10-12 11:15:15 +0900 Vineeth TM * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Check if location being set is valid Adding a check in set_property to find if the location uri is valid and printing warning if not valid. https://bugzilla.gnome.org/show_bug.cgi?id=755326 2015-10-12 11:09:30 +0900 Vineeth TM * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Fix memory leaks during failures freeing streamheader_buffers and sent_buffers during failure cases. https://bugzilla.gnome.org/show_bug.cgi?id=755326 2015-10-12 11:03:17 +0900 Vineeth TM * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Replace redundant free_buffer_list function Removing free_buffer_list and replacing it with already available function g_list_free_full https://bugzilla.gnome.org/show_bug.cgi?id=755326 2015-10-11 16:40:01 +0200 Edward Hervey * tests/check/Makefile.am: check: Don't forget base CFLAGS for flvdemux check elements/flvdemux.c:25:25: fatal error: gst/tag/tag.h: No such file or directory 2015-10-11 11:37:51 +0100 Sebastian Dröge * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: Create a TIME segment when creating streamable output Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which does the same for flvmux. 2015-09-23 13:50:52 +0200 Havard Graff * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: * tests/check/Makefile.am: * tests/check/elements/flvdemux.c: flvdemux: output speex vorbiscomment as a GstTagList This is what speexdec expects. https://bugzilla.gnome.org/show_bug.cgi?id=755478 2015-09-22 22:59:16 +0200 Havard Graff * gst/flv/gstflvmux.c: * tests/check/elements/flvmux.c: flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE Or else flvdemux don't understand it https://bugzilla.gnome.org/show_bug.cgi?id=754435 2015-09-02 10:44:59 +0200 Havard Graff * gst/flv/gstflvmux.c: * tests/check/elements/flvmux.c: flvmux: use time segment and copy timestamps when streamable Add a basic test using speex data to verify timestamping. https://bugzilla.gnome.org/show_bug.cgi?id=754435 2015-09-23 13:14:03 +0200 Havard Graff * gst/flv/gstflvdemux.c: flvdemux: speex is also always 16KHz This is just a cosmetic change for the logs, since the right caps for Speex is being set elsewhere. https://bugzilla.gnome.org/show_bug.cgi?id=755479 2015-07-14 15:19:44 +0200 Stian Selnes * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: rtpmanager: Add 'source-stats' to stats and notify Add statitics from each rtp source to the rtp session property. 'source-stats' is a GValueArray where each element is a GstStructure of stats for one rtp source. The availability of new stats is signaled via g_object_notify. https://bugzilla.gnome.org/show_bug.cgi?id=752669 2015-06-05 17:20:33 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Implement sending of reduced size RTCP packets https://bugzilla.gnome.org/show_bug.cgi?id=750456 2015-10-08 15:01:13 +0530 Ravi Kiran K N * gst/audiofx/audiodynamic.h: audiofx: Remove unused variable Remove unused variable 'degree' in audiodynamic https://bugzilla.gnome.org/show_bug.cgi?id=756234 2015-10-08 14:44:07 +0900 Vineeth TM * gst/isomp4/qtdemux.c: qtdemux: Fix memory leak for corrupted file Free brands before overriding them. https://bugzilla.gnome.org/show_bug.cgi?id=756226 2015-10-08 11:44:04 +0900 Vineeth TM * ext/gdk_pixbuf/gstgdkpixbufdec.c: gdkpixbufdec: Fix pixbuf_loader leak during failures https://bugzilla.gnome.org/show_bug.cgi?id=756219 2015-10-07 23:23:45 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Add missing break 2015-10-07 13:03:02 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations https://bugzilla.gnome.org/show_bug.cgi?id=751311 2015-10-07 13:02:12 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpmanager: add "max-dropout-time" and "max-misorder-time" props https://bugzilla.gnome.org/show_bug.cgi?id=751311 2015-10-07 17:14:57 +0900 Vineeth TM * gst/isomp4/gstqtmux.c: qtmux: Fix date memory leak When getting date from taglist, the memory should be freed after using it. https://bugzilla.gnome.org/show_bug.cgi?id=756171 2015-10-05 11:03:38 +0900 Vineeth TM * gst/isomp4/gstqtmux.c: qtmux: Fix sample memory leak When getting sample from taglist, the memory should be freed after using it. https://bugzilla.gnome.org/show_bug.cgi?id=756068 2015-10-05 13:10:56 +0900 Vineeth TM * gst/cutter/gstcutter.c: cutter: Fix buffer leak Buffer is added to the internal cache, and pushed only when accumulated buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated is not freed. Freeing the cache when the state changes from PAUSED to READY. https://bugzilla.gnome.org/show_bug.cgi?id=754212 2015-08-31 21:10:16 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Use default upstream event handling https://bugzilla.gnome.org/show_bug.cgi?id=752694 2015-08-31 21:05:03 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set https://bugzilla.gnome.org/show_bug.cgi?id=752694 2015-07-22 09:47:22 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * tests/check/elements/rtpmux.c: gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers By not doing this, the muxer is not effectively a rtpmuxer, rather a funnel, since it should be a single stream that exists the muxer. If not specified, take the first ssrc seen on a sinkpad, allowing upstream to decide ssrc in "passthrough" with only one sinkpad. Also, let downstream ssrc overrule internal configured one We hence has the following order for determining the ssrc used by rtpmux: 0. Suggestion from GstRTPCollision event 1. Downstream caps 2. ssrc-Property 3. (First) upstream caps containing ssrc 4. Randomly generated https://bugzilla.gnome.org/show_bug.cgi?id=752694 2015-10-02 22:42:20 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Fixup last commit 2015-10-02 22:21:45 +0300 Sebastian Dröge * configure.ac: * gst/udp/gstudpsrc.c: Update GLib dependency to 2.40.0 2015-06-30 16:56:19 +0200 Miguel París Díaz * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpstats: add utility for calculating RTP packet rate 2015-08-10 18:14:39 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: handle empty segments in seeking adjust If seeking targets an empty segment skip it as there is no media offset to get from it. Instead look for the next one. This doesn't make seeking in push-mode work if you seek to an empty segment but at least won't get you to wrong offsets. https://bugzilla.gnome.org/show_bug.cgi?id=753484 2015-04-17 14:25:43 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: post messages when fragments are being opened and closed This can be useful for applications that need to track the created fragments (to log them in a recording database, for example) https://bugzilla.gnome.org/show_bug.cgi?id=750108 2015-04-29 18:23:28 +0100 Ramiro Polla * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: allow non-video streams to serve as reference In the absence of a video stream, the first stream will be used as reference. https://bugzilla.gnome.org/show_bug.cgi?id=753617 2015-07-22 17:45:12 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxsink.c: splitmuxsink: initialize mux_start_time properly mux_start_time refers to the running_time of the buffer that goes first in the output file. Normally this time is 0, so this variable is initialized to 0 during the state change to PAUSED. However, when dealing with dynamic pipelines and starting a recording while the pipeline has already run for a while, the running_time of the first buffer is > 0 and this causes a problem with detecting the end of the first file(s) when splitting by duration, because the code will later compare the threshold_time with (last buffer running_time - mux_start_time) and will get it wrong until mux_start_time advances enough to make this difference < threshold_time, creating empty files in the meantime. https://bugzilla.gnome.org/show_bug.cgi?id=753624 2015-09-16 16:03:02 +0900 Vineeth T M * gst/avi/gstavidemux.c: avidemux: Reverse playback does not consider segment.start During reverse playback, the media should stop playing at segment.start This does not happen, and avidemux continues to process data even when current timestamp is less that segment.start. https://bugzilla.gnome.org/show_bug.cgi?id=755094 2015-09-23 12:39:35 +0900 Manasa Athreya * gst/isomp4/qtdemux.c: qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream If stream has more than one trex box which is not matched to actual track id, it makes qtdemux crashed. Author : Manasa Athreya (manasa.athreya@lge.com) https://bugzilla.gnome.org/show_bug.cgi?id=754864 2015-09-04 14:24:45 +0530 Ravi Kiran K N * gst/smpte/gstsmpte.c: smpte: get size, stride info using VideoInfo Use VideoInfo data to get size stride and offset, instead of hard coded macros. https://bugzilla.gnome.org/show_bug.cgi?id=754558 2015-09-04 14:18:50 +0530 Ravi Kiran K N * gst/smpte/gstsmpte.c: smpte: free mask Free the memory allocated to 'mask' to avoid memory leak. https://bugzilla.gnome.org/show_bug.cgi?id=754555 2015-08-20 11:02:58 +0900 Vineeth TM * tests/examples/equalizer/demo.c: * tests/icles/equalizer-test.c: * tests/icles/gdkpixbufoverlay-test.c: * tests/icles/gdkpixbufsink-test.c: * tests/icles/test-oss4.c: * tests/icles/videocrop-test.c: gstreamer: good: tests: Fix memory leaks when context parse fails. When g_option_context_parse fails, context and error variables are not getting free'd which results in memory leaks. Free'ing the same. And replacing g_error_free with g_clear_error, which checks if the error being passed https://bugzilla.gnome.org/show_bug.cgi?id=753853 2015-10-02 16:18:15 +0900 Hyunjun Ko * gst/rtpmanager/rtpsource.c: rtpsource: doesn't handle probation and rtp gap in case of sender https://bugzilla.gnome.org/show_bug.cgi?id=754548 2015-10-02 16:16:32 +0900 Hyunjun Ko * docs/plugins/gst-plugins-good-plugins.signals: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals Allows for applications to get internal source's RTP statistics. (eg. sender sources for a server/client) https://bugzilla.gnome.org/show_bug.cgi?id=746747 2015-09-15 03:14:37 +1000 Matthew Waters * ext/qt/gstplugin.cc: * ext/qt/gstqsgtexture.h: * ext/qt/gstqtsink.cc: * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qt: add support for building on osx/ios Including: - Necessary configure checks - Necessary compile time platform checks - Necessary runtime qt iOS/OSX platform detection https://bugzilla.gnome.org/show_bug.cgi?id=755100 2015-10-02 14:17:48 +1000 Jan Schmidt * sys/ximage/gstximagesrc.c: ximagesrc: Gather and coalesce all damaged areas before retrieving. These days the xserver seems to give us the same damage regions over and over for entire windows, and we retrieve them multiple times, which gives time for more damage to appear. Instead, just quickly gather all damaged areas into a region list and copy out once. 2015-10-01 16:24:32 +0100 Luis de Bethencourt * gst/goom2k1/Makefile.am: * gst/goom2k1/gstgoom.h: goom2k1: use the new audiovisualizer base class Rebase to have goom using the GstAudioVisualizer base class in gst-plugins-base/gst-libs/gst/pbutils https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-10-01 16:16:08 +0100 Luis de Bethencourt * gst/goom/Makefile.am: * gst/goom/gstgoom.h: goom: use the new audiovisualizer base class Rebase to have goom using the GstAudioVisualizer base class in gst-plugins-base/gst-libs/gst/pbutils https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-09-30 17:35:33 -0300 Thiago Santos * gst/interleave/deinterleave.c: * tests/check/elements/deinterleave.c: deinterleave: implement accept-caps Avoid using default accept-caps handler that will query downstream and is more expensive. Just check if the caps is compatible with the template and check if the channels are the same. 2015-09-30 09:35:39 -0300 Thiago Santos * tests/check/elements/deinterleave.c: tests: deinterleave: also check for caps query results 2015-09-30 12:30:59 -0300 Thiago Santos * gst/interleave/deinterleave.c: deinterleave: use the caps query filter It was being ignored and would lead to wrong results if the element doing the query would rely on the intersection being made. 2015-09-30 10:00:31 -0300 Thiago Santos * gst/interleave/deinterleave.c: deinterleave: implement a caps query handler for the sinkpad It was missing and apparently code relied on having it there for not allowing a change in the number of channels 2015-09-30 09:05:03 -0300 Thiago Santos * gst/interleave/deinterleave.c: deinterleave: fix caps leak Caps from the pad template are being leaked. In any case it is from a static pad template and will 'leak' in the end, just doing the cleanup for the good practice. 2015-09-29 22:57:52 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: gtk: add some GL debug statements to show up in GL traces 2015-08-28 16:24:24 +0100 Luis de Bethencourt * ext/qt/gstqtsink.cc: qtsink: explicitely fallthrough switch statement In case ret is False, fallthrough to default case. CID #1320705 2015-09-29 11:15:01 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/gdkpixbufoverlay.c: tests: gdkpixbufoverlay: add minimal unit test https://bugzilla.gnome.org/show_bug.cgi?id=755773 2015-09-29 11:12:48 +0100 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufsink: don't leak old pixel buffer when setting a new overlay https://bugzilla.gnome.org/show_bug.cgi?id=755773 2015-09-28 20:25:22 +0100 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: avoid potential string overflow We don't necessarily have full control over the input tags, so it's possible that the ISRC tag contains a longer string than expected, in which case we'd write over the end of the static-size 13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc. Make sure to only copy the ISRC if it's not too long, and make sure the buffer we write to is always NUL-terminated by using g_strlcpy(). CID 1324931. 2015-09-28 18:03:51 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Remove leftover assertion from 0.10 We now allocate memory via GstAllocator and as such can handle arbitrary alignments, not only <= G_MEM_ALIGN. https://bugzilla.gnome.org/show_bug.cgi?id=755708 2015-09-29 00:25:00 +1000 Matthew Waters * ext/gtk/gstgtkbasesink.c: gtk: fix assertion when the element has no peer When proxying keyboard/navigation/mouse events, only unref a successfully retreived peer pad. https://bugzilla.gnome.org/show_bug.cgi?id=755738 2015-08-28 16:35:39 +0100 Luis de Bethencourt * ext/qt/qtitem.cc: qml: remove overwritten value Value in tex is overwritten before being used. Removing it. CID 1320715 https://bugzilla.gnome.org/show_bug.cgi?id=754253 2015-09-02 23:45:07 +1000 Matthew Waters * ext/qt/Makefile.am: * ext/qt/gstqsgtexture.h: * ext/qt/gstqtgl.h: * ext/qt/qtitem.cc: * ext/qt/qtitem.h: qt: add support for building/running on android Including: - Necessary configure checks - Necessary compile time platform checks - Necessary runtime qt android platform detection - Escaping GLsync definition with Qt's GLES2 implementation https://bugzilla.gnome.org/show_bug.cgi?id=754466 2015-09-02 23:40:31 +1000 Matthew Waters * ext/qt/Makefile.am: qt: don't use CPPFLAGS for tools that cannot use them For example moc will bail out when given arguments it does not know about. The moc specific MOC_CPPFLAGS can still be used to pass flags to moc. https://bugzilla.gnome.org/show_bug.cgi?id=754466 2015-09-02 23:39:54 +1000 Matthew Waters * ext/qt/Makefile.am: qt: rename library to include gst prefix libqtsink -> libgstqtsink https://bugzilla.gnome.org/show_bug.cgi?id=754466 2015-09-25 10:01:37 +0200 Guillaume Marquebielle * gst/audioparsers/gstaacparse.c: aacparse: fix uninitialized variables in LOAS config reading On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and 'channels' are pointing to uninitialized values can end on setting random values as rate and channels on src caps. https://bugzilla.gnome.org/show_bug.cgi?id=755611 2015-09-18 00:58:23 +1000 Jan Schmidt * ext/gdk_pixbuf/gstgdkpixbufsink.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: Fix some compiler warnings when building with G_DISABLE_ASSERT Touches rtpmanager and gdkpixbufsink 2015-08-18 14:30:57 +0100 Chris Bass * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: qtdemux: support timed-text subtitle tracks. https://bugzilla.gnome.org/show_bug.cgi?id=752818 2015-09-26 00:12:46 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/rtpmanager/gstrtpjitterbuffer.c: gst: Don't use deprecated gst_segment_to_position() 2015-09-21 13:47:21 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time https://bugzilla.gnome.org/show_bug.cgi?id=755125 2015-09-16 19:28:11 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: rtpbin/session: Allow RTCP sync to happen based on capture time or send time Send time is the previous behaviour and the default, but there are use cases where you want to synchronize based on the capture time. https://bugzilla.gnome.org/show_bug.cgi?id=755125 2015-09-25 23:51:09 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.6.0 === 2015-09-25 23:15:55 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.6.0 2015-09-25 22:57:34 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2015-09-25 14:08:09 +0200 Thibault Saunier * gst/smpte/gstsmptealpha.c: smptealpha: Do not set width/height before comparing with old values Otherwise we end up considering the values did not change and we wrongly work with the old video format (which will lead to wrong behaviour/segfaults). https://bugzilla.gnome.org/show_bug.cgi?id=755621 2015-09-24 18:51:39 +0200 Sebastian Dröge * ext/gtk/gstgtkbasesink.c: gtk: Only run from the main thread in stop() if we created the window We're not doing anything at all from the main thread in other cases. 2015-09-24 15:52:40 +0200 Thibault Saunier * ext/gtk/gtkgstbasewidget.c: gtk: When setting format check if pending format changed In case the format changed fast and the pending format is different than the currently set but the currently set is equal to the pending one we could end up having mismatch between the finally set format and the data stream format. https://bugzilla.gnome.org/show_bug.cgi?id=755542 2015-09-24 15:51:28 +0200 Thibault Saunier * ext/gtk/gstgtkbasesink.c: gtk: Do not forget to release OBJECT_LOCK on error path https://bugzilla.gnome.org/show_bug.cgi?id=755542 2015-09-24 11:37:04 +0200 Thibault Saunier * ext/gtk/Makefile.am: * ext/gtk/gstgtkbasesink.c: * ext/gtk/gstgtkutils.c: * ext/gtk/gstgtkutils.h: * ext/gtk/gtkgstglwidget.c: gtk: Factor out a function to run a function on main thread https://bugzilla.gnome.org/show_bug.cgi?id=755251 2015-09-24 10:51:31 +0200 Thibault Saunier * ext/gtk/gstgtkbasesink.c: gtk: Marshall state changes in the main thread Gtk is not MT safe thus we need to make sure that everything is done in the main thread when working with it. https://bugzilla.gnome.org/show_bug.cgi?id=755251 2015-09-23 20:59:00 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Accumulate segments for edit lists before activating the next segment eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always accumulating segments manually when activating a segment. This is only needed when handling edit lists, not when activating a segment because of a seek. Do the accumulation when switching edit list segments instead. This fixes segment seeks again, while keeping edit lists playback working. https://bugzilla.gnome.org/show_bug.cgi?id=755471 2015-09-23 17:43:51 +0530 Vikram Fugro * gst/spectrum/gstspectrum.c: spectrum: send phase values in the GstMessage for Phase info https://bugzilla.gnome.org/show_bug.cgi?id=755463 2015-09-23 11:42:51 +0200 Thibault Saunier * ext/gtk/gstgtkbasesink.c: gtksink: Do not show window until we reach the PAUSED state https://bugzilla.gnome.org/show_bug.cgi?id=755459 2015-09-22 00:46:01 +1000 Jan Schmidt * gst/matroska/matroska-mux.c: matroska-mux: Don't output a warning on MONO multiview mode. 2015-09-21 10:47:15 +0200 Thibault Saunier * ext/gtk/gstgtkbasesink.c: gtksink: Do not re destroy the GtkWindow if destroyed by the user Otherwise we will get an ASSERT. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755249 2015-09-19 17:02:18 +0200 Sebastian Rasmussen * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: Fix memory leaks The same memory leaks were fixed in identical fashion for vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277 2015-09-19 17:04:07 +0200 Sebastian Rasmussen * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup * use g_list_free_full(), don't iterate elements maually when freeing * call gst_rtp_*_pay_clear_packet(), don't duplicate its code * use gst_buffer_unref() to clarify that it is buffers being released, instead of refering directly to gst_mini_object_unref() Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277 2015-09-19 18:44:22 +0200 Sebastian Dröge * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer https://bugzilla.gnome.org/show_bug.cgi?id=755265 2015-09-19 11:46:37 +0200 Thibault Saunier * ext/gtk/gstgtkbasesink.c: * ext/gtk/gstgtkbasesink.h: * ext/gtk/gstgtkglsink.c: gtkglsink: Hide and clean the GtkWindow we might create When stopping the sink we should always hide the window. https://bugzilla.gnome.org/show_bug.cgi?id=755249 === release 1.5.91 === 2015-09-18 19:33:13 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.5.91 2015-09-18 19:23:57 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2015-09-18 11:50:31 +0200 Sebastian Dröge * po/zh_CN.po: po: Update translations 2015-09-17 10:50:01 +0900 Eunhae Choi * gst/avi/gstavidemux.c: avidemux: Fix taglist leak gst_tag_list_insert() does not take ownership of the inserted taglist. https://bugzilla.gnome.org/show_bug.cgi?id=755138 2015-09-17 13:35:02 +0900 Vineeth T M * ext/gtk/gtkgstglwidget.c: gl: Fix GError leaks during failures https://bugzilla.gnome.org/show_bug.cgi?id=755140 2015-09-16 07:05:36 +1000 Jan Schmidt * gst/audioparsers/gstaacparse.c: aacparse: Skip LOAS AAC until a valid config is seen. It's normal when dropping into the middle of a stream to not always have the config available immediately, so skip LOAS until a valid config is seen without either setting invalid caps or erroring out. https://bugzilla.gnome.org/show_bug.cgi?id=751386 2015-09-13 15:41:38 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: reset just a bit more upon flush_stop 2015-09-13 15:40:09 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: remove dead struct member 2015-09-11 17:09:28 +0900 Vineeth TM * gst/udp/gstmultiudpsink.c: multiudpsink: fix GError memory leak when hostname resolution fails https://bugzilla.gnome.org/show_bug.cgi?id=754869 2015-09-10 15:26:54 -0300 Thiago Santos * gst/matroska/ebml-write.c: matroskamux: drop HEADER flag from output buffers Drop HEADER flag from output buffers if they are not indeed headers. Fixes resending of headers in tcp connection handling https://bugzilla.gnome.org/show_bug.cgi?id=754768 2015-09-10 16:00:50 +0100 Tim-Philipp Müller * gst/matroska/ebml-write.c: matroskamux: fix matroskamux ! matroskademux Don't carry over DISCONT flags from the input buffers to the output buffer, or the demuxer might reset its state when it receives the first data buffer just after parsing the simple block header, and then expect sane data to follow. Fixes matroskamux ! demux erroring out. https://bugzilla.gnome.org/show_bug.cgi?id=754768 https://bugzilla.gnome.org/show_bug.cgi?id=657805 2015-09-09 12:51:40 -0700 Martin Kelly * gst/rtsp/README: rtsp: fix small README typo https://bugzilla.gnome.org/show_bug.cgi?id=754807 2015-09-10 00:07:18 +1000 Matthew Waters * ext/qt/qtitem.cc: gtk, qt: more specifically define the compile time requirements Otherwise we could include headers/configurations that will never been installed. https://bugzilla.gnome.org/show_bug.cgi?id=754732 2015-09-10 00:07:18 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: gtk, qt: more specifically define the compile time requirements Otherwise we could include headers/configurations that will never been installed. https://bugzilla.gnome.org/show_bug.cgi?id=754732 2015-09-10 00:00:11 +1000 Matthew Waters * ext/qt/gstqsgtexture.cc: qt: use our function table instead of directly calling gl functions Otherwise when building with --as-needed we would need to link to a GL or GLES library. https://bugzilla.gnome.org/show_bug.cgi?id=754732 2015-09-04 19:45:37 +0100 Tim-Philipp Müller * gst/audioparsers/gstwavpackparse.c: wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks https://bugzilla.gnome.org/show_bug.cgi?id=752106 2015-09-04 19:34:41 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek flac contains the sample offset in the frame header, so after a seek without index flacparse will know the exact position we landed on and timestamp buffers accordingly. It only set the pts though, which means the baseparse-set dts which was set to the seek position prevails, and since the seek was based on an estimate, there's likely a discrepancy between where we wanted to land and where we did land, so from here on that dts/pts difference will be maintained, with dts possibly multiple seconds ahead of pts, which is just wrong. The easiest way to fix this is to just set both pts and dts based on the sample offset, but perhaps parsed audio should just not have dts set at all. https://bugzilla.gnome.org/show_bug.cgi?id=752106 2015-09-06 16:33:02 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: docs: remove properties and signals that no longer exist https://bugzilla.gnome.org/show_bug.cgi?id=726443 2013-10-11 15:13:00 +0000 George Chriss * gst/flv/gstflvmux.c: flvmux: Make the element count in arrays not include end One-line removal of tags_written++ This should fix rtmp output to crtmpserver, and hopefully noone is expecting that the element count includes the end element, as different bits of documentation say different things about whether it should or not. https://bugzilla.gnome.org/show_bug.cgi?id=661624 2015-07-30 00:59:15 +1000 Jan Schmidt * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Store incoming bitrate tags and send in the metadata Apparently the Microsoft Azure RTMP server requires that the videodatarate and audiodatarate metadata be provided, so set those, even if it's to 0. Use the actual input bitrate tags if available. 2015-09-04 00:06:29 +1000 Jan Schmidt * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't parse key data more than needed. When an auxilliary streams are present in the SDP media, there's no need to re-parse the SDP attributes multiple times. 2015-09-03 20:56:55 +1000 Jan Schmidt * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access. In parse_keymgmt(), don't mutate the input string that's been passed as const, especially since we might need the original value again if the same key info applies to multiple streams (RTX, for example). When a resource is 404, and we have auth info - retry with the auth info the same as if we had receive unauthorised, in case the resource isn't even visible until credentials are supplied. Fix a memory leak handling Mikey data. When generating a random keystring, don't overrun the 30 byte buffer by generating 32 bytes into it. 2015-09-04 15:43:40 +0200 Thibault Saunier * ext/gtk/gtkgstbasewidget.c: gtk: Do not consider GtkEvents as handled Applications might still want to use them after the sink transformed them into GstNavigation events 2015-09-04 15:18:05 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Fix build with GLib < 2.44 G_IO_ERROR_CONNECTION_CLOSED was added in 2.44. 2015-09-04 12:01:52 +0300 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data This happens on Windows if we use the same socket for sending packets, and the remote sends ICMP port/host unreachable messages. https://bugzilla.gnome.org/show_bug.cgi?id=754534 2015-09-02 21:12:41 +0300 Sebastian Dröge * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: rtpvorbis/theoradepay: Fix handling of fragmented packets This was broken in b1089fb520 by not considering the full packet length of a fragmented packet but only the length of the first one. https://bugzilla.gnome.org/show_bug.cgi?id=754417 2015-09-01 15:39:22 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmfsrc: Reply to latency query 2015-08-07 17:27:48 +0530 Nirbheek Chauhan * ext/qt/qtitem.cc: qmlsink: Ensure that at least one windowing system is available Otherwise, we'll just crash at runtime because the gl context is NULL https://bugzilla.gnome.org/show_bug.cgi?id=754108 2015-08-31 16:42:30 -0400 Olivier Crête * tests/check/elements/rtpsession.c: tests: Fix rtpsession test failure The time of the first RTCP packet is semi-random, so sometimes it was produced before enough packets from the second SSRC were received. First drop queued RTCP packets, then advance the clock enough to ensure that at least one new RTCP packet is produced. https://bugzilla.gnome.org/show_bug.cgi?id=750731 2015-08-31 18:06:31 +0100 Tim-Philipp Müller * ext/gtk/gtkgstglwidget.c: gtk, qt, gl: fix typo in debug and error messages 2015-08-31 18:06:31 +0100 Tim-Philipp Müller * ext/qt/gstqtsink.cc: * ext/qt/qtitem.cc: gtk, qt, gl: fix typo in debug and error messages 2015-08-31 13:56:04 +0200 Stefan Sauer * tests/check/elements/level.c: level: improve the test for multi-channel mode Change the test to verify the read-index for multiple messages per buffer. See https://bugzilla.gnome.org/show_bug.cgi?id=754144 2015-08-31 12:46:52 +0200 Jan Alexander Steffens (heftig) * gst/matroska/matroska-demux.c: matroskademux: Align raw video frames to 32 bytes Outputting unaligned video frames causes videoscale et al to crash when attempting SIMD-accelerated conversion. https://bugzilla.gnome.org/show_bug.cgi?id=736965 2015-08-26 23:16:46 +0200 Stefan Sauer * gst/level/gstlevel.c: level: fix level calculations for mutliple channels This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab. 2015-08-27 10:28:55 +0530 Ravi Kiran K N * gst/smpte/gstsmpte.c: smpte: Fix memory leak In gst_smpte_collected(), check upfront if input formats are same or not. This avoids allocation of in1 and in2 buffers and subsequent memory leak when input formats do not match. https://bugzilla.gnome.org/show_bug.cgi?id=754153 2015-08-21 11:52:19 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: don't try to connect to dead radio server 2015-08-21 16:29:16 +0900 Vineeth TM * gst/rtsp/gstrtspsrc.c: rtspsrc: Trivial fix to check correct condition When checking for describe method, because of missing parentheses, wrong condition is being checked, which will result in wrong behavior. https://bugzilla.gnome.org/show_bug.cgi?id=753912 2015-08-21 13:19:02 +0900 Vineeth TM * gst/matroska/matroska-read-common.c: matroska: read: fix tag list memory leak gst_toc_entry_merge_tags makes a new ref of the taglist, so it should be unref'ed as soon as the tags are merged to the tocentry https://bugzilla.gnome.org/show_bug.cgi?id=753904 2015-08-21 12:20:59 +0900 Vineeth TM * ext/wavpack/gstwavpackdec.c: wavpackdec: fix taglist memory leak When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased by audiodecoder and the caller should free the taglist being passed. https://bugzilla.gnome.org/show_bug.cgi?id=753903 2015-08-20 14:45:33 +0200 Jean-Michel Hautbois * sys/v4l2/gstv4l2transform.c: v4l2transform: fix pad closing Signed-off-by: Jean-Michel Hautbois https://bugzilla.gnome.org/show_bug.cgi?id=753875 2015-08-19 13:52:21 +0300 Sebastian Dröge * ext/gtk/gtkgstglwidget.c: gtk/gl: Use our GL function table instead of directly calling GL functions Otherwise we would have to link the plugin to the GL libraries directly. === release 1.5.90 === 2015-08-19 13:29:53 +0300 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.5.90 2015-08-19 12:47:42 +0300 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2015-08-19 11:29:55 +0300 Sebastian Dröge * po/el.po: * po/zh_CN.po: po: Update translations 2015-08-13 17:29:58 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesrc.c: multifilesrc: fix regression with starting from index set via index property When we haven't started yet, set the start_index when we set the index property, so that we start at the right index position after the initial seek. The index property was never really meant to be for writing, but it used to work, so let's support it for backwards compatibility. https://bugzilla.gnome.org/show_bug.cgi?id=739472 2015-08-18 10:52:11 +0100 Alex Ashley * gst/isomp4/qtdemux.c: qtdemux: fix offset calculation when parsing CENC aux info Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for DASH common encryption, however commit bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master shortly before the CENC commit caused the calculation of the CENC aux info offset to be incorrect. The base_offset was being added if present, but if the base_offset is relative to the start of the moof, the offset was being added twice. The correct approach is to calculate the offset from the start of the moof and use that offset when parsing the CENC aux info. 2015-08-17 14:28:24 -0300 Thiago Santos * ext/flac/gstflacenc.c: flacenc: actually return true for accept-caps query handling 2015-08-17 14:07:10 +0900 Hyunjun Ko * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpklvpay.c: rtp: copy metadata in the (de)payloaders which is missed before https://bugzilla.gnome.org/show_bug.cgi?id=753706 2015-08-16 15:21:51 -0400 Dustin Spicuzza * configure.ac: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: directsoundsink: allow specifying audio playback device https://bugzilla.gnome.org/show_bug.cgi?id=753670 2015-08-16 13:51:47 -0300 Thiago Santos * ext/flac/gstflacenc.c: flacenc: remove single entry if from loop Iterate from the 2nd channel on and create the 1 channel struct outside to make loop structure simpler and only slightly faster. 2015-08-16 13:21:41 -0300 Thiago Santos * ext/flac/gstflacenc.c: flacenc: implement proper accept-caps Should just compare with what can be immediatelly accepted by the element. flacenc can't renegotiate so if it has a caps already it should only accept if it is that caps otherwise just use the template caps 2015-08-16 13:03:36 -0300 Thiago Santos * ext/flac/gstflacenc.c: flacenc: improve sink pad template caps Removes the need for custom caps query handling and makes it more correct from the beginning on the template. It is a bit uglier to read because there is 1 entry per channel but makes code easier to maintain. 2015-08-16 12:41:56 -0300 Thiago Santos * gst/y4m/gsty4mencode.c: y4mencode: fix gst-launch version in documentation 2015-08-15 22:32:21 -0300 Thiago Santos * ext/speex/gstspeexenc.c: * ext/wavpack/gstwavpackenc.c: * gst/law/alaw-encode.c: * gst/law/mulaw-encode.c: audioencoders: use template subset check for accept-caps It is faster than doing a query that propagates downstream and should be enough Elements: speexenc, wavpackenc, mulawenc, alawenc 2015-08-15 22:29:41 -0300 Thiago Santos * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngenc.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: * gst/y4m/gsty4mencode.c: videoencoders: use template subset check for accept-caps It is faster than doing a query that propagates downstream and should be enough Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc 2015-08-16 17:21:24 +0100 Tim-Philipp Müller * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: use new baseparse API to fix tag handling https://bugzilla.gnome.org/show_bug.cgi?id=679768 2015-03-17 17:50:37 -0400 Olivier Crête * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: use new base parse API to fix tag handling https://bugzilla.gnome.org/show_bug.cgi?id=679768 2015-08-16 14:37:53 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: use new baseparse API and fix tag handling https://bugzilla.gnome.org/show_bug.cgi?id=679768 2015-08-16 13:04:02 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Use signed integer type to be able to check for negative subtraction results CID 1315829 2015-08-16 11:50:34 +0100 Luis de Bethencourt * gst/rtp/gstrtpvorbisdepay.c: rtpvorbisdepay: remove dead code payload_buffer must be NULL in ignore_reserved. Check will always be false. Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf CID #1316476 2015-08-15 22:45:53 -0300 Thiago Santos * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: alawenc: port to AudioEncoder base class 2015-08-15 22:15:26 -0300 Thiago Santos * ext/twolame/gsttwolamemp2enc.c: audioencoders: use template subset check for accept-caps It is faster than doing a query that propagates downstream and should be enough Elements: amrnbenc, lamemp3enc, twolamemp2enc 2015-08-15 22:15:26 -0300 Thiago Santos * ext/lame/gstlamemp3enc.c: audioencoders: use template subset check for accept-caps It is faster than doing a query that propagates downstream and should be enough Elements: amrnbenc, lamemp3enc, twolamemp2enc 2015-08-15 09:16:23 -0300 Thiago Santos * ext/flac/gstflacdec.c: * ext/speex/gstspeexdec.c: * ext/wavpack/gstwavpackdec.c: * gst/law/alaw-decode.c: * gst/law/mulaw-decode.c: audiodecoders: use default pad accept-caps handling Avoids useless check of downstream caps when handling an accept-caps query Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec 2015-08-15 08:49:57 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: videodecoders: use default pad accept-caps handling Avoids useless check of downstream caps when handling an accept-caps query Elements: jpegdec, pngdec, vp8dec, vp9dec 2015-08-15 11:31:04 -0300 Thiago Santos * gst/law/alaw-decode.c: alawdec: make error handling a bit nicer Print the element along with the debug to make it easier to trace the failures 2015-08-15 11:04:16 -0300 Thiago Santos * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: alawdec: port to audiodecoder base class mulawdec was already ported, alawdec was left behind. 2015-08-15 10:34:14 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: only look for more samples in moofs in pull-mode For playback of some fragmented formats with qtdemux it will try to look for the next moof after finishing one but it is only possible for pull-mode. For playback of streaming fragmented formats such as DASH it should just not try to look for another moof but instead wait for more data. https://bugzilla.gnome.org/show_bug.cgi?id=752602 https://bugzilla.gnome.org/show_bug.cgi?id=752603 2015-08-15 14:31:15 +0200 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: glsink: Enable sync meta on pools we offer As the upload is asynchronous, we need to enable the sync meta to gain correct rendering. The buffer pool receiver don't know about that. 2015-08-15 15:12:27 +0200 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gtkgstglwidget.c: gtkglsink: Add overlay composition support Rendering composition overlay in GL with additional high resolution overlay being added. 2015-08-15 15:08:11 +0200 Nicolas Dufresne * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtkglsink: Fix unsafe handling of buffer life time We need to keep the active buffer (the one we have retreive a texture id from) otherwise it's racy and upstream may upload new content before we have rendered or during later redisplay. 2015-08-14 18:07:15 +0200 Nicolas Dufresne * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: * ext/gtk/gtkgstglwidget.c: gtkglsink: Remove reset path The reset path is bogus and there is no reason to get rid of these things during resize. 2015-08-15 12:58:50 +0200 Sebastian Dröge * gst/audioparsers/gstdcaparse.c: dcaparse: Don't look for a second syncword There are streams out there that consistently contain garbage between every frame so we never ever find a second consecutive syncword. See https://bugzilla.gnome.org/show_bug.cgi?id=738237 2015-08-15 11:12:05 +0100 Tim-Philipp Müller * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vp8enc, vp9enc: reset multipass file index when stopping encoder Fixes multipass encoding when re-using the same element/pipeline for subsequent encoding runs. https://bugzilla.gnome.org/show_bug.cgi?id=747728 2015-08-15 11:09:42 +0100 Tim-Philipp Müller * ext/vpx/gstvp9enc.c: * ext/vpx/gstvp9enc.h: vp9enc: provide support for multiple pass cache files Some files may provide different caps insight of one stream. Since vp9enc support caps reinit, we should support cache reinit too. If more then file cache file will be created, the naming will be: cache cache.1 cache.2 ... Based on patch by: Oleksij Rempel https://bugzilla.gnome.org/show_bug.cgi?id=747728 2015-08-14 11:41:42 -0300 Thiago Santos * tests/check/elements/aacparse.c: tests: aacparse: use caps query instead of accept-caps The accept-caps query just does a shallow check at the current element while at this test we want it to also look at downstream. So use caps query there. https://bugzilla.gnome.org/show_bug.cgi?id=753623 2015-08-14 11:40:22 -0300 Thiago Santos * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: enable accept-template flag Do a quick check with the pad template caps as it is enough. Users should have figured the appropriate full caps on a previous caps query https://bugzilla.gnome.org/show_bug.cgi?id=753623 2015-08-14 15:46:53 +0200 George Kiagiadakis * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: send the User-Agent header Sometimes it is useful to know this information on the server side. Other popular implementations (vlc, ffmpeg, ...) also send this header on every message. This includes a new "user-agent" property that the user can set to use a custom User-Agent string. The default is "GStreamer/" https://bugzilla.gnome.org/show_bug.cgi?id=750101 2015-08-14 15:42:42 +0200 George Kiagiadakis * gst/rtsp/gstrtspsrc.c: rtspsrc: wrap gst_rtsp_message_init_request in a local function This will allow adding common request initialization, like the user agent string, in just one place. 2015-08-14 09:36:09 +0530 Prashant Gotarne * gst/audiofx/audioecho.c: audioecho: make sure buffer gets reallocated if max_delay changes https://bugzilla.gnome.org/show_bug.cgi?id=753490 2015-07-09 09:51:26 +0200 Oleksij Rempel * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: vp8enc: provide support for multiple pass cache files Some files may provide different caps insight of one stream. Since vp8enc support caps reinit, we should support cache reinit too. If more then file cache file will be created, the naming will be: cache cache.1 cache.2 ... https://bugzilla.gnome.org/show_bug.cgi?id=747728 2015-04-15 22:51:51 +0200 Ramiro Polla * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs Use constantDuration to calculate the timestamp of non-first AU in the RTP packet. If constantDuration is not present in the MIME parameters, its value must be calculated based on the timing information from two consecutive RTP packets with AU-Index equal to 0. https://bugzilla.gnome.org/show_bug.cgi?id=747881 2015-08-14 06:43:13 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: souphttpsrc: remove unnecessary if, g_free is null safe 2015-08-14 08:33:56 +0100 Alex Ashley * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: add property to set HTTP method To allow souphttpsrc to be use HTTP methods other than GET (e.g. HEAD), add a "method" property that is a string. If this property is not set, GET is used. https://bugzilla.gnome.org/show_bug.cgi?id=752413 2015-08-14 11:13:01 +0200 Edward Hervey * tests/check/generic/states.c: check: Rename states unit test Makes it easier to differentiate from other modules states unit test 2015-08-14 09:21:25 +0200 Sebastian Dröge * gst/goom/gstaudiovisualizer.c: * gst/goom/gstaudiovisualizer.h: * gst/goom2k1/gstaudiovisualizer.c: * gst/goom2k1/gstaudiovisualizer.h: goom: Rename get_type() function of base class to prevent symbol conflicts This is a problem when statically linking. 2015-08-13 16:32:55 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset Otherwise we will just output buffers without timestamps after a reset if no timestamps are provided by upstream, e.g. when using RTSP over TCP. https://bugzilla.gnome.org/show_bug.cgi?id=749536 2015-08-12 17:16:01 +0530 Ravi Kiran K N * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.h: matroska: Remove unused variable https://bugzilla.gnome.org/show_bug.cgi?id=753556 2015-08-12 00:18:20 +0200 Matthew Waters * ext/gtk/gtkgstbasewidget.c: gtk: fix motion event name s/motion/mouse/ Fixes hover interaction with DVD menus 2015-08-12 00:14:14 +0200 Matthew Waters * ext/gtk/gtkgstbasewidget.c: gtk: correct navigation events for window scaling i.e. take into account the possiblity of scaling in the sink or through GDK_SCALE. Fixes DVD Menus with a scaled gtkwidget 2015-08-11 13:34:59 +0200 Matthew Waters * ext/gtk/gstgtkbasesink.c: * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: gtk: implement GstNavigation interface Now we can push key/mouse input into the pipeline for DVD use cases. 2015-08-04 20:59:17 +0300 Sebastian Dröge * gst/rtp/Makefile.am: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtputils.c: * gst/rtp/gstrtputils.h: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: Copy metadata in the (de)payloader, but only the relevant ones The payloader didn't copy anything so far, the depayloader copied every possible meta. Let's make it consistent and just copy all metas without tags or with only the video tag. https://bugzilla.gnome.org/show_bug.cgi?id=751774 2015-08-10 18:20:15 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: fix small typo in comment 2015-08-10 16:19:18 -0400 Nicolas Dufresne * gst/goom2k1/gstgoom.c: goom2k1/doc: Fixup previous commit 2015-08-10 15:55:19 -0400 Nicolas Dufresne * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom2k1/doc: Use GstGoom2k1 namespace The doc generator isn't happy when we have class name clash. Simply use it's own namespace. 2015-08-10 17:10:42 +0530 Prashant Gotarne * gst/audiofx/audioecho.c: audioecho: removed unused variable in set_property unused local variable 'delay' is removed. https://bugzilla.gnome.org/show_bug.cgi?id=753450 2015-08-10 12:45:27 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: fix suboptimal queue iteration code 2015-08-09 17:25:45 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: don't use glib 2.44-only API 2015-07-29 14:14:50 +0100 Alex Ashley * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux_types.c: qtdemux: add support for ISOBMFF Common Encryption This commit adds support for ISOBMFF Common Encryption (cenc), as defined in ISO/IEC 23001-7. It uses a GstProtection event to pass the contents of PSSH boxes to downstream decryptor elements and attached GstProtectionMeta to each sample. https://bugzilla.gnome.org/show_bug.cgi?id=705991 2015-08-10 14:13:50 +0900 Hyunjun Ko * gst/rtp/gstrtph264depay.c: rtph264depay: checking if depay has sps/pps nals before insertion https://bugzilla.gnome.org/show_bug.cgi?id=753430 2015-08-08 16:44:49 +0100 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: fix outdated comment The default behaviour was changed in the 0.10 -> 1.x transition, but the comment was not updated. 2015-08-08 17:42:22 +0200 Sebastian Dröge * gst/rtp/gstrtptheorapay.c: rtptheorapay: If flushing a packet failed, go out of the loop immediately 2015-08-08 17:41:02 +0200 Sebastian Dröge * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: If flushing a packet failed, go out of the loop immediately 2015-08-08 17:34:50 +0200 Sebastian Dröge * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2 and 4:4:4 formats. 2015-08-08 17:28:03 +0200 Matthew Waters * ext/qt/gstqsgtexture.cc: * ext/qt/gstqsgtexture.h: * ext/qt/qtitem.cc: qml: implement the required multiple GL context synchonisation From GStreamer's GL context into the QML context 2015-08-06 17:46:13 +0200 George Kiagiadakis * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpklvpay.c: rtpklv(de)pay: add "RTP" in the klass string GstRTSPMedia uses this classification to detect the real payloader inside a dynpay bin and asserts if it doesn't find it, therefore it is required https://bugzilla.gnome.org/show_bug.cgi?id=753325 2015-08-05 11:13:09 -0300 Thiago Santos * tests/check/elements/rtpaux.c: tests: rtpaux: use a dynamic pt in the test 1) Tests that using dynamic PT instead of the default ones work 2) If we ever decide to change the codec here we don't need to worry about change the PT for the default one of the new codec in the test https://bugzilla.gnome.org/show_bug.cgi?id=746445 2015-08-05 10:53:15 +0900 Hyunjun Ko * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: print valid type where guint32 is expected https://bugzilla.gnome.org/show_bug.cgi?id=746445 2015-08-06 11:33:37 +0900 Hyunjun Ko * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmupay.c: rtppayload: set standard payload type as default Initialize the PT to the default value of the codec and check if it is still the default before declaring the pt to be dynamic or not when setting the caps. Also use the PT constants from the rtp lib when possible https://bugzilla.gnome.org/show_bug.cgi?id=747965 2015-07-26 12:07:56 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: store the moof-offset also for push mode It will be used in some cases for getting the correct offsets from trun atoms. https://bugzilla.gnome.org/show_bug.cgi?id=752603 2015-07-26 02:09:24 -0300 Thiago Santos * gst/isomp4/atoms.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.h: qtdemux: handle default-base-is-moof flag Handle the flag from the tfhd that signals the base offset to start from the moof atom https://bugzilla.gnome.org/show_bug.cgi?id=752603 2015-07-29 18:54:35 -0600 Glen Diener * gst/matroska/matroska-demux.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroskademux: Preserve forward referenced track tags https://bugzilla.gnome.org/show_bug.cgi?id=752850 2015-08-04 18:07:35 -0300 Thiago Santos * tests/check/elements/rtpaux.c: tests: rtpaux: fix test failure The RTP PT for alaw is 8. Less than 50 packets are received in the length of this test so it would never drop a buffer or would drop only the last buffer and it would fail sometimes when the received wouldn't receive the retransmission packet in time. https://bugzilla.gnome.org/show_bug.cgi?id=746445 2015-08-04 20:59:17 +0300 Sebastian Dröge * gst/rtp/gstrtpstreamdepay.c: rtpstreamdepay: Only allow activation in push mode We need a proper caps event from upstream with the full RTP caps as we can't create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g. a filesrc or any other element that supports pull mode. https://bugzilla.gnome.org/show_bug.cgi?id=753066 2015-08-04 16:28:17 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: soup: fix typo in translated string https://bugzilla.gnome.org/show_bug.cgi?id=753240 2015-08-04 12:25:46 +0300 Sebastian Dröge * gst/rtp/gstrtph264depay.c: rtph264depay: Put the profile and level into the caps 2015-08-04 12:09:12 +0300 Sebastian Dröge * gst/rtp/gstrtph264depay.c: rtph264depay: Only update the srcpad caps if something else than the codec_data changed h264parse does the same, let's keep the behaviour consistent. As we now include the codec_data inside the stream too here, this causes less caps renegotiation. 2015-08-04 11:48:27 +0300 Sebastian Dröge * gst/rtp/gstrtph264depay.c: rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id The spec says: When a picture parameter set NAL unit with a particular value of pic_parameter_set_id is received, its content replaces the content of the previous picture parameter set NAL unit, in decoding order, with the same value of pic_parameter_set_id (when a previous picture parameter set NAL unit with the same value of pic_parameter_set_id was present in the bitstream). 2015-08-03 13:45:59 -0300 Thiago Santos * gst/multifile/gstsplitmuxsink.c: splitmuxsink: remove extra \n at debug message 2015-08-03 13:42:20 -0300 Thiago Santos * gst/multifile/gstsplitmuxsink.c: splitmuxsink: prevent deadlock when states change too fast If the GOP is completed, pads have to start gathering for the next one but it is possible that the the state might go to COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the thread has a chance to wake up and proceed, leaving it trapped in the check_completed_gop loop and deadlocking the other threads waiting for it to advance. To solve it, this patch also checks that tha input running time hasn't changed to prevent this scenario. 2015-08-03 17:55:01 +0300 Sebastian Dröge * gst/rtp/gstrtph264depay.c: rtph264depay: Insert SPS/PPS NALs into the stream h264parse does the same and this fixes decoding of some streams with 32 SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere. This looks like a mistake in the part of the spec about the codec_data. 2015-07-30 11:29:27 +0900 Eunhae Choi * ext/soup/gstsouphttpsrc.c: souphttpsrc: handle empty http proxy string 1) If the system http_proxy environment variable is not set or set to an empty string, we must not set proxy to avoid http connection error. 2) In case of proxy property setting, if user want to clear the proxy setting, they should be able to set it to NULL or an empty string again, so this is fixed too. 3) Check if the proxy string was parsed correctly. https://bugzilla.gnome.org/show_bug.cgi?id=752866 2015-07-29 15:46:20 +0530 Ravi Kiran K N * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: remove unused variable Remove unused variable 'framecount' from dvdemux https://bugzilla.gnome.org/show_bug.cgi?id=753008 2015-07-30 15:32:09 +0900 Vineeth TM * gst/rtsp/gstrtspsrc.c: rtspsrc: assertion error due to wrong condition check In media to caps function, reserved_keys array is being used for variable i, leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed changed it to variable j https://bugzilla.gnome.org/show_bug.cgi?id=753009 2015-07-30 15:21:20 +0900 Vineeth TM * gst/rtp/gstrtpmp4vdepay.c: rtpmp4vdepay: rtpbuffer is being unref'ed twice process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay the refernce should not be removed here https://bugzilla.gnome.org/show_bug.cgi?id=753042 2015-07-29 11:26:46 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Strip keys from the fmtp that we use internally in our caps Skip keys from the fmtp, which we already use ourselves for the caps. Some software is adding random things like clock-rate into the fmtp, and we would otherwise here set a string-typed clock-rate in the caps... and thus fail to create valid RTP caps https://bugzilla.gnome.org/show_bug.cgi?id=753009 2015-07-29 19:28:33 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Support mpegtsmux as a muxer. As a fallback, look for a pad template sink_%d on the muxer when requesting pads, to support mpegtsmux https://bugzilla.gnome.org/show_bug.cgi?id=752999 2015-06-25 01:35:27 +1000 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: splitmuxsrc: Use a separate lock to delay typefind. Don't hold the main splitmux part lock over the parent state change function, as it prevents posting error messages that happen. Since the purpose is to prevent typefinding from proceeding, use a separate mutex just for that. 2015-07-29 13:43:50 +0900 Vineeth TM * gst/matroska/matroska-read-common.c: matroska: fix memory leak After adding to tag list, key_val is not being free'd resulting in memory leak https://bugzilla.gnome.org/show_bug.cgi?id=752992 2015-07-27 13:34:14 +0900 Manasa Athreya * gst/isomp4/qtdemux.c: qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc 'NONE' and 'raw ' fourcc don't always contain U8 audio, it can be more bits as well, in which case it's just like 'twos'. https://bugzilla.gnome.org/show_bug.cgi?id=752613 2015-07-24 15:10:05 +0200 Dimitrios Katsaros * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: v4l2: Allow framerate to be large then 100pfs This limit was arbitrary. We still fixate near 100pfs for compatibility. https://bugzilla.gnome.org/show_bug.cgi?id=752825 2015-07-25 03:25:28 -0400 Olivier Crête * gst/avi/gstavidemux.c: avidemux: Stop without posting error on flushing This could just be a normal pipeline shutdown. 2015-07-23 15:00:08 +0900 Hyunjun Ko * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also https://bugzilla.gnome.org/show_bug.cgi?id=752618 2015-07-22 15:13:48 +0200 Edward Hervey * ext/qt/Makefile.am: qt: Don't dist files that might not exist We only require moc building at build time. 2015-07-22 08:05:04 +0200 Edward Hervey * ext/qt/Makefile.am: qt: Tidy up makefile a bit more Separate generated files, from disted files 2015-07-21 11:23:21 +0100 Julien Isorce * ext/gtk/gtkgstglwidget.c: gstglwidget: use gst_gl_display_create_context Also handle the failure case. https://bugzilla.gnome.org/show_bug.cgi?id=750310 2015-07-16 18:09:30 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/matroskademux.c: tests: add minmal matroskademux test for subtitle output Some of the subtitle chunks will have embedded NUL-terminators (last three), some don't (first three), some will have markup, some won't, some will be valid UTF-8 (all but last), some won't (last stanza). https://bugzilla.gnome.org/show_bug.cgi?id=752421 2015-07-16 18:49:26 +0300 Dimitrios Christidis * gst/matroska/matroska-demux.c: matroskademux: fix for subtitle buffers with NUL terminators Commit 45892ec8 created a regression where g_utf8_validate() would fail if the subtitle buffer had a NUL terminator as part of the data. https://bugzilla.gnome.org/show_bug.cgi?id=752421 2015-07-21 13:31:05 +0200 Stian Selnes * gst/rtp/gstrtpvp8depay.c: rtpvp8depay: Check available bytes before copy Need to check that the number of bytes we want to copy from the adapter actually is available and handle the error case gracefully. This error may happen if malformed packets are received and we don't have a complete frame. https://bugzilla.gnome.org/show_bug.cgi?id=752663 2015-07-16 09:32:36 +0900 Paul Hyunil * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: Support subtitle when track subtype is fourcc_subt https://bugzilla.gnome.org/show_bug.cgi?id=752655 2015-07-20 16:59:40 +0800 Song Bing * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Set timestamp when queue buffer. Should set timestamp when queue buffer. https://bugzilla.gnome.org/show_bug.cgi?id=752618 2015-07-20 11:09:20 +0200 Thibault Saunier * ext/gtk/gtkgstglwidget.c: gtk: Log GDK GL error when failling creating GdkGLContext 2015-07-18 17:19:18 +1000 Matthew Waters * ext/qt/qtitem.cc: glcontext: fix get_current_gl_api on x11/nvidia drivers They require to get_proc_address some functions through the platform specific {glX,egl}GetProcAddress rather than the default GL library symbol lookup. 2015-07-18 17:19:18 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: glcontext: fix get_current_gl_api on x11/nvidia drivers They require to get_proc_address some functions through the platform specific {glX,egl}GetProcAddress rather than the default GL library symbol lookup. 2015-07-17 16:00:01 -0400 Nicolas Dufresne * ext/gtk/gtkgstglwidget.c: gtkgstglwidget: Cleanup unused private member new_buffer has been moved to base class. Also cleanup the properties comment, which are also all moved into the base class. 2015-07-17 15:57:37 -0400 Nicolas Dufresne * ext/gtk/gstgtkbasesink.c: gtksink: "widget" must be access from main thread Document that "widget" property must be accessed from the main thread (where GTK is running). This is the same for state transition on these elements. It is very natural to do so un GTK applications. 2015-07-17 15:08:53 -0400 Nicolas Dufresne * ext/gtk/gtkgstglwidget.c: gtkglsink: Don't leak vertex array and buffers This is now possible since reset is always called from the main thread. https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-17 14:36:56 -0400 Nicolas Dufresne * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: gtkgstbasewidget: Fix black frame on resize This is solved by only applying the new format when the next buffer is to be rendered and on the GTK thread. https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-17 13:05:05 -0400 Nicolas Dufresne * ext/gtk/gstgtkbasesink.c: * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: gtkgstbasewidget: Pass already parsed VideoInfo As the base sink already parse the caps into VideoInfo it makes sense to pass in VideoInfo to the widget instead. https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-16 16:49:32 -0400 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: gtkglsink: Port to GstGtkBaseSink base class https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-16 16:00:37 -0400 Nicolas Dufresne * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: gtksink: Port to GstGtkBaseSink https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-16 15:59:59 -0400 Nicolas Dufresne * ext/gtk/Makefile.am: * ext/gtk/gstgtkbasesink.c: * ext/gtk/gstgtkbasesink.h: gtkbasesink: Create a base class This contains all the common code between the gtkglsink and gtksink. https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-16 14:30:42 -0400 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstglwidget.h: gtkglsink: Port to GtkGstBaseWidget https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-16 12:55:11 -0400 Nicolas Dufresne * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: * ext/gtk/gtkgstwidget.c: * ext/gtk/gtkgstwidget.h: gtksink: Port to GtkGstBaseWidget https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-16 12:51:34 -0400 Nicolas Dufresne * ext/gtk/Makefile.am: * ext/gtk/gtkgstbasewidget.c: * ext/gtk/gtkgstbasewidget.h: gtk: Add GtkGstBaseWidget This is a "pseudo" base class. Basically it's a shared instance and class structure and a shared set of function between the two widget. It cannot have it's own type like normal base class since the one instance will implement GtkGLArea while the other implements GtkDrawingAreay. To workaround this, the parent instance and class is a union of both. https://bugzilla.gnome.org/show_bug.cgi?id=752441 2015-07-15 17:35:22 -0400 Nicolas Dufresne * ext/gtk/gtkgstglwidget.c: gtkgstglwidget: Remove unused gl_caps 2015-07-15 16:56:33 -0400 Nicolas Dufresne * ext/gtk/gstgtksink.c: gtksink: Create a window if the widget is unparented The same way as it's now done with the gtkglsink, create a top level window if the widget is not parented. https://bugzilla.gnome.org/show_bug.cgi?id=751104 2015-07-15 14:35:02 -0400 Nicolas Dufresne * ext/gtk/gstgtksink.c: gtksink: Ensure the copy pasted code remains the same Move back the default property at the same place they are in the other sink. This helps when using a diff viewer to synchronized this unfortunate copy paste. https://bugzilla.gnome.org/show_bug.cgi?id=751104 2015-07-15 14:32:42 -0400 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.c: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtk: Fix race between queue_draw and destroy In GTK dispose can be called before the last ref is reached. This happens when you close the container window. The dispose will be explicitly called, and destroyed notify will be fired. This patch fixes this race by properly tracking the widget state. In the sink, we now set the widget pointer to NULL, so the widget will properly get created again if you set your pipeline to NULL state after the widget was destroy, and set it back to PLAYING. https://bugzilla.gnome.org/show_bug.cgi?id=751104 2015-07-16 15:12:17 +0200 Havard Graff * gst/rtpmanager/gstrtpmux.c: * tests/check/elements/rtpmux.c: rtpmux: handle different ssrc's on sinkpads Do this by not putting the ssrc from the src pads in the caps used to probe other sinkpads, and then intersecting with it later. https://bugzilla.gnome.org/show_bug.cgi?id=752491 2015-07-16 17:19:03 +0100 Tim-Philipp Müller * gst/avi/gstavimux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: Update mailing list address from sourceforge to freedesktop 2015-07-15 13:44:52 +0300 Dimitrios Christidis * gst/matroska/matroska-demux.c: matroskademux: fix trailing '*' displayed with some text subtitles The subtitle buffer we push out should not include a NUL terminator as part of the data, we just add such a terminator for safety, but it should not be included in the buffer size. A NUL terminator is not valid UTF-8, so checks will fail if it's included in the size, and the NUL will be replaced by the fallback character specified when converting, i.e. '*'. https://bugzilla.gnome.org/show_bug.cgi?id=752421 2015-07-15 18:23:05 +0200 Wim Taymans * ext/pulse/pulsedeviceprovider.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: add properties to GstDevice Add the extra properties we get from pulse to the GstDevice we expose with the device monitor 2015-07-15 11:47:51 -0400 Nicolas Dufresne * ext/gtk/gtkgstwidget.c: gtkgstwidget: Add missing break in get_property 2015-07-15 11:44:30 -0400 Nicolas Dufresne * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.h: gtksinks: Remove undefined private structure The classes contains a private structure which are not defined, hence unused. 2015-07-15 17:20:20 +0530 Ravi Kiran K N * gst/audiofx/audioinvert.c: * gst/audiofx/audiowsincband.c: audiofx: Fix typo in example pipelines Fix typo in example pipelines of audiowsincband and audioinvert. https://bugzilla.gnome.org/show_bug.cgi?id=752416 2015-04-15 18:27:04 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxsink.c: splitmuxsink: add a "format-location" signal that allows better control over filenames In certain applications, splitting into files named after a base location template and an incremental sequence number is not enough. This signal gives more fine-grained control to the application to decide how to name the files. https://bugzilla.gnome.org/show_bug.cgi?id=750106 2015-04-15 20:13:27 +0300 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudio.c: osxaudiosrc: no resampling on OS X Unlike Remote IO, AUHAL doesn't have built-in resampling for sources -- confirmed by Core Audio engineer Doug Wyatt: http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html https://bugzilla.gnome.org/show_bug.cgi?id=743758 2015-04-15 18:29:14 +0300 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudio.c: osxaudiosrc: avoid get_channel_layout This only produces a warning and serves no purpose. https://bugzilla.gnome.org/show_bug.cgi?id=743758 2015-04-07 15:40:14 +0530 Arun Raghavan * sys/osxaudio/gstosxcoreaudio.c: osxaudio: Avoid making a duplicate structure in caps for mono/stereo case For 1ch or 2ch devices, we just need to set the caps to allow both options since CoreAudio will up/downmix appropriately. Also fixes the condition for the 2ch case to be exact, rather than at least 2 channels since the downmix will not take place in the >stereo case. 2015-04-06 16:22:34 +0530 Arun Raghavan * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiohal.c: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: Don't set the format on an initialized AudioUnit We need to initialize the AudioUnit early to be able to probe the underlying device, but according to the AudioUnitInitialize() and AudioUnitUninitialize() documentation, format changes should be done while the AudioUnit is uninitialized. So we explicitly uninitialize the AudioUnit during a format change and reinitialize it when we're done. 2015-04-06 15:55:59 +0530 Arun Raghavan * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: osxaudio: Minor spelling fix (unitialize -> uninitialize) 2015-03-21 20:34:25 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: osxaudio: Fix lockup in _audio_unit_property_listener _audio_unit_property_listener is called either from a Core Audio thread or as a result of a Core Audio API (e.g. AudioUnitInitialize) from our own thread. In the latter case, osxbuf can be already locked (GStreamer's mutex is not recursive). We introduce the flag cached_caps_valid and use it instead of nullifying cached_caps when we cannot lock on osxbuf. https://bugzilla.gnome.org/show_bug.cgi?id=743758 2015-03-12 12:15:12 +0200 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudio.c: osxaudio: Invalidate cached caps on format change Listen for changes in hardware stream format and channel layout, and invalidate cached caps (since they contain the preferred caps). https://bugzilla.gnome.org/show_bug.cgi?id=743758 2015-03-09 23:34:06 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiocommon.h: * sys/osxaudio/gstosxcoreaudiohal.c: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: Overhaul of probing caps - Probing caps is unified between source and sink - Hardware stream format is now reported as preferred capabilities (dynamically updated when hardware configuration changes) - Get hardware channel layout from Remote IO just like from HAL - More comprehensive mapping between AudioChannelLabel and GstAudioChannelPosition - Support for unpositioned channel layouts - Announce stereo-mono upmixing/downmixing in caps https://bugzilla.gnome.org/show_bug.cgi?id=743758 2015-03-09 23:15:56 +0200 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudio.c: osxaudio: AudioUnitInitialize on open Call AudioUnitInitialize upon open. Otherwise, we cannot get (hardware) stream format nor channel layout from the outer scope. 2015-07-12 14:27:15 +0100 Tim-Philipp Müller * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvp8depay.c: rtp: depayloaders: implement process_rtp_packet() vfunc For more optimised RTP packet handling: means we don't need to map the input buffer again but can just re-use the mapping the base class has already done. https://bugzilla.gnome.org/show_bug.cgi?id=750235 2015-05-27 19:19:27 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: implement process_rtp_packet() vfunc For more optimised RTP packet handling: means we don't need to map the input buffer again but can just re-use the map the base class has already done. https://bugzilla.gnome.org/show_bug.cgi?id=750235 2015-07-10 14:01:43 +0200 Edward Hervey * ext/qt/qtitem.cc: configure/qt: Fix build without Qt5X11Extras 2015-07-06 23:10:51 +1000 Matthew Waters * ext/qt/.gitignore: * ext/qt/Makefile.am: * ext/qt/gstplugin.cc: * ext/qt/gstqsgtexture.cc: * ext/qt/gstqsgtexture.h: * ext/qt/gstqtsink.cc: * ext/qt/gstqtsink.h: * ext/qt/qtitem.cc: * ext/qt/qtitem.h: * tests/examples/qt/qml/.gitignore: * tests/examples/qt/qml/main.cpp: * tests/examples/qt/qml/main.qml: * tests/examples/qt/qml/play.pro: * tests/examples/qt/qml/qml.qrc: new qt5 qml GL video sink Very much in the same spirit as the Gtk GL sink Two things are provided 1. A QQuickItem subclass that renders out RGBA filled GstGLMemory buffers that is instantiated from qml. 2. A sink element that will push buffers into (1) To use 1. Declare the GstGLVideoItem in qml with an appropriate objectName property set. 2. Get the aforementioned GstGLVideoItem from qml using something like QQmlApplicationEngine engine; engine.load(QUrl(QStringLiteral("qrc:/main.qml"))); QObject *rootObject = engine.rootObjects().first(); QQuickItem *videoItem = rootObject->findChild ("videoItem"); 3. Set the videoItem on the sink https://bugzilla.gnome.org/show_bug.cgi?id=752185 2015-07-10 00:13:32 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix indention 2015-07-09 23:59:10 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Always estimate DTS from the current clock time Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS we would produce wrong DTS. As now the estimated DTS is based on the clock, don't store it in the jitterbuffer items as it would otherwise be used in the skew calculations and would influence the results. We only really need the DTS for timer calculations. https://bugzilla.gnome.org/show_bug.cgi?id=749536 2015-07-09 09:26:09 -0300 Thiago Santos * tests/check/elements/.gitignore: gitignore: ignore rtph263 test 2015-07-09 13:03:23 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtk: add to the generic/states test 2015-06-17 09:36:57 -0400 Xavier Claessens * ext/gtk/gstgtkglsink.c: GstGtkGLSink: Ensure widget has a toplevel parent Checking for a parent is not enough, it must have a toplevel one. If widget has no toplevel parent then add it in a GtkWindow, that make it usable from gst-launch-1.0. https://bugzilla.gnome.org/show_bug.cgi?id=751104 2015-06-17 09:36:40 -0400 Xavier Claessens * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: GstGtkGLSink: Post error if widget gets destroyed https://bugzilla.gnome.org/show_bug.cgi?id=751104 2015-06-16 16:21:26 -0400 Xavier Claessens * ext/gtk/gstgtkglsink.c: GstGtkGLSink: fix possible warning in finalize If the element is finalized before going in READY state the widget could still be NULL. https://bugzilla.gnome.org/show_bug.cgi?id=751104 2015-07-08 23:47:44 -0300 Thiago Santos * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2 Replace static constants with macros to make gcc happy CC elements/elements_rtpjitterbuffer-rtpjitterbuffer.o elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND; ^ elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000; ^ elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000; 2015-07-08 23:40:45 -0300 Thiago Santos * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: run indent and fix some comments Fix indent on this file and break some comment lines into two to make it fit 80 chars per line 2015-07-08 15:02:24 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: rework segment event handling for adaptive streaming When a new time segment is received upstream is going to restart with a new atom. Make the neededbytes and todrop variables reflect that to avoid waiting too much or dropping the initial bytes that contain the header. 2015-07-08 12:35:55 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: push data from adapter before starting new segment The adapter might have data remaining from the previous segment, push it all before clearing the adapter and starting a new segment. It can accumulate data if it had pushed and got not-linked, returning immediately without processing all the data. Before starting a new segment this data should be handled. 2015-07-08 19:59:13 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset https://bugzilla.gnome.org/show_bug.cgi?id=749536 2015-07-08 21:08:36 +0200 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: fix gap-time calculation and remove "late" The amount of time that is completely expired and not worth waiting for, is the duration of the packets in the gap (gap * duration) - the latency (size) of the jitterbuffer (priv->latency_ns). This is the duration that we make a "multi-lost" packet for. The "late" concept made some sense in 0.10 as it reflected that a buffer coming in had not been waited for at all, but had a timestamp that was outside the jitterbuffer to wait for. With the rewrite of the waiting (timeout) mechanism in 1.0, this no longer makes any sense, and the variable no longer reflects anything meaningful (num > 0 is useless, the duration is what matters) Fixed up the tests that had been slightly modified in 1.0 to allow faulty behavior to sneak in, and port some of them to use GstHarness. https://bugzilla.gnome.org/show_bug.cgi?id=738363 2015-06-30 11:21:31 +0200 Stian Selnes * gst/rtpmanager/gstrtpjitterbuffer.c: Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected" This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0. The reverted patch is wrong and introduces a regression because there may still be time to receive some of the packets included in the gap if they are reordered. 2015-07-07 23:53:02 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: flush samples before adding more from moof Avoids accumulating all samples from a fragmented stream that could lead to a 'index-too-big' error once it goes over 50MB of data. It could reach that before 2h of playback so it doesn't take that long. As upstream elements are providing data in time format they should be the ones that have more information about the full media index and should be able to seek if possible. 2015-07-07 23:56:12 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: rename upstream_newsegment to upstream_format_is_time upstream_newsegment isn't really clear on what it means, it is set to TRUE when the upstream element sends a segment in TIME format, so rename it to be more clear about it. It is important to know this because it means that upstream has a notion of time and qtdemux is likely being driven by an upstream element that is reading from a higher level abstraction than a file, such as a DASH, MSS or DLNA element. 2015-07-07 21:31:08 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: fix leak by flushing previous sample info from trak In fragmented streaming, multiple moov/moof will be parsed and their previously stored samples array might leak when new values are parsed. The parse_trak and callees won't free the previously stored values before parsing the new ones. In step-by-step, this is what happens: 1) initial moov is parsed, traks as well, streams are created. The trak doesn't contain samples because they are in the moof's trun boxes. n_samples is set to 0 while parsing the trak and the samples array is still NULL. 2) moofs are parsed, and their trun boxes will increase n_samples and create/extend the samples array 3) At some point a new moov might be sent (bitrate switching, for example) and parsing the trak will overwrite n_samples with the values from this trak. If the n_samples is set to 0 qtdemux will assume that the samples array is NULL and will leak it when a new one is created for the subsequent moofs. This patch makes qtdemux properly free previous sample data before creating new ones and adds an assert to catch future occurrences of this issue when the code changes. 2015-07-07 16:46:33 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: fix index size check and debug message It is allocating samples_count + n_samples, not only n_samples 2015-07-08 17:02:05 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Calculate receive time if we don't have any This is required to properly schedule packet loss timers and make sure all our calculations work properly. https://bugzilla.gnome.org/show_bug.cgi?id=749536 2015-07-08 15:13:17 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations That is, handle DTS==GST_CLOCK_TIME_NONE correctly. https://bugzilla.gnome.org/show_bug.cgi?id=749536 2015-07-08 20:31:42 +0900 Vineeth T M * gst/avi/gstavidemux.c: avidemux: fix event leak when seek fails in avidemux, event is not being freed. https://bugzilla.gnome.org/show_bug.cgi?id=752117 2015-07-08 12:02:22 +0200 Stian Selnes * gst/rtp/gstrtph263depay.c: * tests/check/Makefile.am: * tests/check/elements/rtph263.c: rtph263depay: Make sure payload is large enough Plus new unit test. https://bugzilla.gnome.org/show_bug.cgi?id=752112 2015-07-08 08:59:49 +0900 Vineeth TM * gst/rtp/gstrtpklvdepay.c: rtpklvdepay: fix printf format compiler warning v_len is of type guint64, but while print the value(16 + len_size + v_len) G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT https://bugzilla.gnome.org/show_bug.cgi?id=752100 2015-07-07 20:25:47 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-rtp.xml: docs: add new RTP elements to docs 2015-07-07 20:07:31 +0100 Tim-Philipp Müller * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: add basic unit test for KLV payloading Also make it so that the mtu is always set if specified, not only in case of the rather weird bufferlist test code path. This allows us to easily make the payloader fragment a payload across multiple output packets by setting a small MTU on it. 2015-07-07 19:58:42 +0100 Tim-Philipp Müller * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpklvdepay.h: rtpklvdepay: improve start detection and handle fragmented KLV units 2015-07-05 20:25:10 +0100 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpklvdepay.c: * gst/rtp/gstrtpklvdepay.h: rtp: add SMPTE 336M KLV metadata depayloader http://tools.ietf.org/html/rfc6597 2014-08-09 10:08:42 +0100 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpklvpay.c: * gst/rtp/gstrtpklvpay.h: rtp: add SMPTE 336M KLV metadata payloader http://tools.ietf.org/html/rfc6597 2015-07-07 16:59:20 +0200 Stefan Sauer * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/atomsrecovery.c: * gst/isomp4/properties.h: * gst/matroska/matroska-mux.c: * gst/rtpmanager/rtpsource.c: docs: fix "Symbol name not found at the start of the comment block" Add symbols or change comment into a regular comment. 2015-07-07 16:58:53 +0200 Stefan Sauer * gst/audioparsers/gstamrparse.h: docs: remove outdated doc strings 2015-07-03 23:10:40 +0200 Stefan Sauer * docs/plugins/gst-plugins-good-plugins-docs.sgml: docs: add missing plugins and ensure master doc is sorted 2015-07-07 15:54:41 +0100 Luis de Bethencourt * gst/imagefreeze/gstimagefreeze.c: Revert "imagefreeze: Remove impossible error condition" This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94. pad only handle EOS events but not EOS flow, and will push the buffer again resulting in an assertion error. So we should not handle the buffer and return EOS flow. 2015-07-07 15:50:50 +0100 Tim-Philipp Müller * gst/rtp/gstrtpg729depay.c: rtpg729depay: unmap rtp buffer in error path 2015-07-07 15:48:40 +0100 Tim-Philipp Müller * gst/rtp/gstrtpg729pay.c: rtpg729pay: fix buffer leak The handle_buffer vfunc takes ownership of the input buffer. Fixes elements/rtp-payloading under valgrind. 2015-07-02 08:52:43 +0200 Tobias Mueller * gst/goom/goom_core.c: goom: Initialised variables to remove compiler warnings goom_core.c: In function 'goom_update': goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized] goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur); ^ goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized] goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur); ^ https://bugzilla.gnome.org/show_bug.cgi?id=752053 2015-07-07 09:18:39 +0100 Tim-Philipp Müller * gst/rtp/gstrtph261pay.c: rtph261pay: fix indentation 2015-07-06 19:11:00 +0900 Jimmy Ohn * gst/rtp/gstrtph261pay.c: rtph261pay: Fix uninitialized variable compiler error endpos variable does not correctly understand in the 4.6.3 GCC version. So compile error appears when we do compile rtph261pay using jhbuild. This patch is fixed the compile error in 4.6.3 GCC version. https://bugzilla.gnome.org/show_bug.cgi?id=751985 2015-07-06 19:33:35 +0200 Thibault Saunier * ext/gtk/gtkgstglwidget.c: gtkglsink: Release the widget lock when trying to get the GL context Otherwise we might be waiting for the lock on the main loop (for example in the ->render vmethod) and thus we will deadlock. 2014-11-12 12:08:58 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: flvdemux: Handle seek flags properly Allows for non-keyframe seeks. https://bugzilla.gnome.org/show_bug.cgi?id=738570 2015-02-24 10:50:52 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: avoid looping reading the 'moof' atom forever It gets stuck if it only finds a moof and no mfra/mfro or moov atoms. Skip the moof to continue the parsing to have it either play or error out. https://bugzilla.gnome.org/show_bug.cgi?id=745089 2015-06-26 13:24:17 +0900 Vineeth TM * ext/flac/gstflacdec.c: flacdec: improve error handling for files which have corrupted header, libflac is not able to process the metadata properly. We just try to ignore the error and continue with the processing, since metadata parsing is not making much of a difference to libflac https://bugzilla.gnome.org/show_bug.cgi?id=751334 2015-07-06 20:16:38 +0900 Hyunjun Ko * sys/ximage/ximageutil.c: ximagesrc: add meta transform function ximage metadata can't be transformed or copied, but provide an empty transformation function instead of NULL to allow unconditional calling of metas' transform functions. https://bugzilla.gnome.org/show_bug.cgi?id=751778 2014-06-16 16:14:28 +0200 Stian Selnes * gst/rtp/gstrtph263pdepay.c: rtph263pdepay: init debug category https://bugzilla.gnome.org/show_bug.cgi?id=752012 2014-06-20 10:59:14 +0200 Stian Selnes * gst/rtp/gstrtpvp8depay.c: rtpv8depay: ignore reserved bit in payload descriptor Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that: R: Bit reserved for future use. MUST be set to zero and MUST be ignored by the receiver. https://bugzilla.gnome.org/show_bug.cgi?id=751929 2015-07-04 20:56:42 +0200 Stian Selnes * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261pay.c: rtph261pay: rtph261depay: Add documentation https://bugzilla.gnome.org/show_bug.cgi?id=751982 2015-07-03 21:58:14 +0200 Stefan Sauer * common: Automatic update of common submodule From f74b2df to 9aed1d7 2015-07-03 14:29:16 +0200 Sebastian Dröge * gst/rtp/gstrtph261pay.c: rtph261pay: Fix compiler warning gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init': gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable] GObjectClass *gobject_class; 2015-07-03 14:03:05 +0200 Sebastian Dröge * gst/rtp/gstrtph261depay.c: rtph261depay: Let the base class push the buffer so it can deal with the flow return 2015-07-03 14:11:35 +0200 Sebastian Dröge * gst/rtp/gstrtph261pay.c: rtph261pay: Remove unused adapter 2015-07-03 13:17:24 +0200 Sebastian Dröge * gst/rtp/gstrtpspeexpay.c: speexpay: Directly attach payload to the output buffer instead of copying it 2015-07-03 13:07:20 +0200 Sebastian Dröge * gst/rtp/gstrtpsbcpay.c: sbcpay: Attach payload directly to the output instead of copying 2014-12-01 14:18:40 +0100 Stian Selnes * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtph261depay.c: * gst/rtp/gstrtph261depay.h: * gst/rtp/gstrtph261pay.c: * gst/rtp/gstrtph261pay.h: * tests/check/elements/rtp-payloading.c: rtp: add H.261 RTP payloader and depayloader Implementation according to RFC 4587. Payloader create fragments on MB boundaries in order to match MTU size the best it can. Some decoders/depayloaders in the wild are very strict about receiving a continuous bit-stream (e.g. no no-op bits between frames), so the payloader will shift the compressed bit-stream of a frame to align with the last significant bit of the previous frame. Depayloader does not try to be fancy in case of packet loss. It simply drops all packets for a frame if there is a loss, keeping it simple. https://bugzilla.gnome.org/show_bug.cgi?id=751886 2015-07-03 12:18:52 +0200 Sebastian Dröge * gst/rtp/gstrtpmpvdepay.c: rtpmpvdepay: Don't forget to unmap the input buffer 2015-07-03 12:14:47 +0200 Sebastian Dröge * gst/rtp/gstrtpmpvpay.c: rtpmpvpay: Create buffer lists instead of pushing each buffer individually 2015-07-03 12:03:59 +0200 Sebastian Dröge * gst/rtp/gstrtpmpapay.c: rtpmpapay: Use buffer lists instead of pushing each fragment individually 2015-07-03 10:51:57 +0200 Sebastian Dröge * gst/rtp/gstrtpmp4apay.c: rtpmp4apay: Create buffer lists and don't copy payload memory 2015-06-29 16:14:18 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT When there are a lot of small gaps, we can consider that there is a big gap (too losses) to reset the buffer. https://bugzilla.gnome.org/show_bug.cgi?id=751636 2015-06-29 15:53:52 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: If possible, always update the current time before looping over all timers If we have a clock, update "now" now with the very latest running time we have. If timers are unscheduled below we otherwise wouldn't update now (it's only updated when timers expire), and also for the very first loop iteration now would otherwise always be 0. Also the time is used for the timeout functions, e.g. to calculate any times for the next timeouts and we would otherwise pass too old times there. https://bugzilla.gnome.org/show_bug.cgi?id=751636 2015-07-02 14:34:57 +0100 Luis de Bethencourt * sys/v4l2/gstv4l2transform.c: v4l2transform: fix memory leak tmp needs to be freed before going out of scope in 'done'. CID #1308954 2015-07-02 12:23:45 +0200 Sebastian Dröge * gst/rtp/gstrtph263ppay.c: rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it 2015-07-02 09:48:02 +0200 Sebastian Dröge * gst/rtp/gstrtph263pdepay.c: rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us 2015-07-02 09:17:59 +0200 Sebastian Dröge * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: rtph263pay: Stop using an adapter and directly use the buffer We always pushed one buffer into the adapter, then handled exactly that one buffer and flushed it from the adapter. Now also don't memcpy() the actual payload but just attach the input buffer's data to the output buffer. This code still needs some serious refactoring/rewriting. 2015-07-01 21:57:28 +0200 Sebastian Dröge * gst/rtp/gstrtpgsmpay.c: rtpgsmpay: Remove non-existing includes for now git add -p mistake. 2015-07-01 19:29:07 +0200 Sebastian Dröge * gst/rtp/gstrtpgstpay.c: rtpgstpay: Use the return value of gst_buffer_append() 2015-07-01 19:19:13 +0200 Sebastian Dröge * gst/rtp/gstrtpgsmpay.c: rtpgsmpay: Attach payload to the output buffer instead of copying it 2015-07-01 17:58:56 +0200 Sebastian Dröge * gst/rtp/gstrtpg729pay.c: rtpg729pay: Attach payload directly to output buffers instead of copying 2015-07-01 17:43:51 +0200 Sebastian Dröge * gst/rtp/gstrtpg723pay.c: rtpg723pay: Attach payload buffer to the output instead of copying 2015-07-01 17:30:39 +0200 Sebastian Dröge * gst/rtp/gstrtpdvdepay.c: rtpdvdepay: Map the output buffer once instead of once every 80 bytes 2015-07-01 21:46:46 +0900 Jimmy Ohn * gst/avi/gstavidemux.c: avidemux: fix return type of index_entry_offset_search() It's a compare function and may return a negative value, so should for correctness and consistency return a signed integer. https://bugzilla.gnome.org/show_bug.cgi?id=751780 2015-07-01 14:12:57 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: refactor handle_next_buffer The goal of this patch is making handle_next_buffer function more readable avoiding unnecesary gotos and adding other cosmetic changes. 2015-07-01 15:40:25 +0200 Sebastian Dröge * gst/rtp/gstrtpac3pay.c: rtpac3pay: Attach the payload to the output buffer instead of copying it Might also want to produce buffer lists here if needed. 2015-07-01 15:38:47 +0200 Sebastian Dröge * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpsirendepay.c: rtp: Fix indention 2015-07-01 12:37:11 +0200 Sebastian Dröge * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-VP8-OPUS.sh: * tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh: rtp: Add examples with VTS/ATS for VP8/OPUS Let's have an example with modern codecs. 2015-06-30 18:11:33 +0200 Sebastian Dröge * gst/rtp/gstrtph264pay.c: rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING() 2015-06-30 14:06:20 +0200 Sebastian Dröge * gst/rtp/gstrtpvp8depay.c: vp8depay: Don't lock/map every non-keyframe buffer twice Just copy the complete header instead of first looking at the first byte and then at the remaining 10 bytes. 2015-06-29 16:05:44 +0100 Luis de Bethencourt * sys/v4l2/gstv4l2object.c: v4l2: document fallthrough cases Pacify coverity and document fallthrough cases in switch statements. CID #1308948, #1308947, #1308946 2015-06-29 10:36:58 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout" This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182. If we have multiple immediate timers, we want to first handle the one with the lowest sequence number... which would be broken now. Instead of this we should just use a GSequence for the timers, and have them sorted first by timestamp, and for equal timestamps by sequence number. Then we would always only have to take the very first timer from the list and never have to look at any others. 2015-06-29 10:14:05 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout If we have lots of such immediate timeouts, we would otherwise have quadratic runtime in the number of timeouts. 2015-06-19 18:01:03 -0300 Thiago Santos * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: sticky events are sent automatically from the pad No need to send them explicitly from the element https://bugzilla.gnome.org/show_bug.cgi?id=751240 2015-06-19 18:00:40 -0300 Thiago Santos * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: make sure to push sticky events before adding pad It allows the caps to be set on the pad before being added for dynamic autoplugging to work. https://bugzilla.gnome.org/show_bug.cgi?id=751240 2015-06-26 00:05:29 +0900 Hyunjun Ko * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property Enable to use new ntp-time-source property of rtpbin https://bugzilla.gnome.org/show_bug.cgi?id=751496 2015-06-25 23:19:58 +0900 Hyunjun Ko * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: rtpbin/session: fix description https://bugzilla.gnome.org/show_bug.cgi?id=751496 2015-06-25 10:57:25 +0100 Luis de Bethencourt * gst/imagefreeze/gstimagefreeze.c: * gst/matroska/matroska-demux.c: * tests/examples/shapewipe/shapewipe-example.c: docs: decodebin2 -> decodebin 2015-06-25 10:47:06 +0100 Luis de Bethencourt * gst/deinterlace/gstdeinterlace.c: deinterlace: update example pipeline Update reference to decodebin2 to decodebin 2015-06-25 10:45:35 +0100 Luis de Bethencourt * gst/deinterlace/gstdeinterlace.c: deinterlace: remove dead assignments Values in fields_required and same_buffer are overwritten before used. Removing assignment 2015-06-25 10:06:07 +0100 Tim-Philipp Müller * ext/Makefile.am: * ext/mikmod/Makefile.am: * ext/mikmod/README: * ext/mikmod/drv_gst.c: * ext/mikmod/gstmikmod.c: * ext/mikmod/gstmikmod.h: * ext/mikmod/mikmod_reader.c: * ext/mikmod/mikmod_types.c: * ext/mikmod/mikmod_types.h: * m4/Makefile.am: * m4/libmikmod.m4: * win32/MANIFEST: * win32/vs8/libgstmikmod.vcproj: mikmod: remove ancient unported plugin This hasn't been touched in 11 years, and clearly no one's been missing it. 2015-06-23 20:15:13 +0900 Gilbok Lee * gst/isomp4/qtdemux.c: qtdemux: does not detect orientation Most files don't contain the values for transposing the coordinates back to the positive quadrant so qtdemux was ignoring the rotation tag. To be able to properly handle those files qtdemux will also ignore the transposing values to only detect the rotation using the values abde from the transformation matrix: [a b c] [d e f] [g h i] https://bugzilla.gnome.org/show_bug.cgi?id=738681 2015-06-25 00:04:16 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.5.2 === 2015-06-24 23:30:41 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.5.2 2015-06-24 22:56:12 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2015-06-24 11:15:00 +0200 Sebastian Dröge * po/nl.po: po: Update translations 2015-06-23 18:42:59 -0400 Nicolas Dufresne * tests/check/elements/qtmux.c: qtmux: Correctly test each segments In presence of gaps, qtdemux will emit multiple segments. The second segment start should match the CTTS. https://bugzilla.gnome.org/show_bug.cgi?id=751361 2015-06-23 17:54:31 -0400 Nicolas Dufresne * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Correctly calculate the elst media start The media start has nothing to do with the shift we have applied but with the value of the first PTS. This is defined as: Dt(0) = 0 Ct(0) = Dt(0) + CTTS(0) So the media start is always the first CTTS. https://bugzilla.gnome.org/show_bug.cgi?id=751361 2015-06-23 11:49:32 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: accumulate previous edts entries into segment.base Allows playing edts editted files with proper synchronization of streams. This patch fixes the regression introduced by bf95f93c0189aa04f18e264b86b6527e431c5d53 that was added to fix segment seeks handling. Having the accumulated_base separated from the main segment.base allows handling both segment seeks and edts editted files. https://bugzilla.gnome.org/show_bug.cgi?id=751361 2015-06-23 00:56:16 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: improve some debug messages Those messages are about the stream, use the pad as the debug object to make it clear from the logs https://bugzilla.gnome.org/show_bug.cgi?id=751361 2015-06-22 22:22:09 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: store last_dts of the first buffer Buffers need not to start at running-time 0 so the last_dts needs to be the value of the first buffer's dts as it is used to compute the duration of the buffers. If it was left at 0 the first buffer would have a larger duration when it shouldn't https://bugzilla.gnome.org/show_bug.cgi?id=751361 2015-06-23 17:11:57 +0900 Vineeth TM * gst/audioparsers/gstflacparse.c: flacparse: fix possible memory leak when buffer is stored to seektable, and stop gets called due to corrupt flac file, then the seektable is not being released https://bugzilla.gnome.org/show_bug.cgi?id=751364 2015-06-23 16:28:40 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: Revert "splitmuxsink: Mask async-start/done while switching files." This reverts commit d61e5393f110ed482815d77807245d78b52eff46. Causes failures muxing larger GOP sizes for some reason. Reverting while I figure it out 2015-06-18 23:22:06 +1000 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Fix startup and shutdown races. Fix 2 startup races when things happen too quickly, and 1 at shutdown by holding a ref to the pads in use until the loop functions exit. Handle errors activating file parts and publish them on the bus. https://bugzilla.gnome.org/show_bug.cgi?id=750747 2015-06-18 09:26:13 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Mask async-start/done while switching files. Sometimes, extra async-start/done from the internal sink while the element is still starting up can cause splitmuxsink to stall in PAUSED state when it has been set to PLAYING by the app. Drop the child's async-start/done messages while switching, so they don't cause state changes at the splitmuxsink level. https://bugzilla.gnome.org/show_bug.cgi?id=750747 2015-06-15 16:12:10 +1000 Jan Schmidt * gst/matroska/matroska-demux.c: matroska-demux: Use gst_video_multiview_guess_half_aspect() Use the gst_video_multiview_guess_half_aspect() utility function to set the half-aspect flag (or not) on stereoscopic frame-packed videos. 2015-06-15 16:10:37 +1000 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Move multiview caps calculations, add half-aspect heuristics Move the multiview caps calculations to the configure_stream() function, so the rest of the video info is available, and use the gst_video_multiview_guess_half_aspect() function to determine if the half-aspect flag should be set on frame-packed video. 2015-06-18 16:06:02 -0400 Nicolas Dufresne * gst/isomp4/qtdemux.c: qtdemux: Add cslg support The cslg atom provide information about the DTS shift. This is needed in recent version of ctts atom where the offset can be negative. When cslg is missing, we parse the CTTS table as proposed in the spec to calculate these values. In this implementation, we only need to know the shift. As GStreamer cannot transport negative timestamps, we shift the timestamps forward using that value and adapt the segment to compensate. This patch also removes bogus offset of ctts_soffset, this offset shall be included in the edit list. https://bugzilla.gnome.org/show_bug.cgi?id=751103 2015-06-19 18:37:59 -0400 Nicolas Dufresne * tests/check/elements/qtmux.c: qtmux: Test gaps at start of stream https://bugzilla.gnome.org/show_bug.cgi?id=751242 2015-06-19 18:40:43 -0400 Nicolas Dufresne * gst/isomp4/gstqtmux.c: qtmux: Use PTS to figure-out presence of gaps We need to look at the presentation timestamp in order to conclude if there is a gap at the start of a stream. https://bugzilla.gnome.org/show_bug.cgi?id=751242 2015-06-19 16:45:02 -0400 Nicolas Dufresne * gst/isomp4/gstqtmux.c: qtmux: Set edit list to compensate DTS shift We shift DTS forward to avoid negative timestamps which cannot be represented with version 0 of the CTTS table. To stick with that version (backward compatibility), the spec recommend using an edit list entry to move back the presentation time to where it should be. https://bugzilla.gnome.org/show_bug.cgi?id=751242 2015-06-22 14:35:52 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: flvmux: Insert AVC end of sequence This FLV specific mark is needed to prevent Flow Player (most likely all Flash base player) from going into buffering state when near EOS. https://bugzilla.gnome.org/show_bug.cgi?id=751320 2015-06-22 13:05:29 +0900 Vineeth TM * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: matroska: remove useless check No need to check for context availability while freeing. We are inside inside a code block with a condition that dereferences context. if (context->type == 0 ... https://bugzilla.gnome.org/show_bug.cgi?id=751306 2015-06-22 19:35:57 +0900 Vineeth T M * gst/matroska/lzo.c: lzo: fix memory leak the opened file is not being closed during test, which will result in memory leak. https://bugzilla.gnome.org/show_bug.cgi?id=751306 2015-06-22 19:30:58 +0900 Vineeth T M * ext/mikmod/mikmod_reader.c: mikmod_reader: Possible null pointer dereference: gst_reader variable is being used before actually checking if it allocated properly https://bugzilla.gnome.org/show_bug.cgi?id=751306 2015-06-22 19:45:14 +0900 Sangkyu Park * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Minor clean-up 1. Fix the code which is wrong coding style. 2. Fix a typing error of comment. https://bugzilla.gnome.org/show_bug.cgi?id=751316 2015-06-22 11:28:13 +0200 Jose Antonio Santos Cadenas * gst/rtpmanager/rtpsource.c: rtpsource: Do not try to push NULL buffers If update_receiver_stats() fails, we can't really do anything with this buffer anymore and have to drop it. This happens if there's a big seqnum discontinuity for example. https://bugzilla.gnome.org/show_bug.cgi?id=751311 2015-06-22 13:10:02 +0900 Vineeth TM * gst/flv/gstflvdemux.c: flvdemux: trivial cleanup trivial patch to add proper ( while checking for if(G_UNLIKELY()) https://bugzilla.gnome.org/show_bug.cgi?id=751306 2015-06-22 13:16:08 +0900 Vineeth TM * gst/audioparsers/gstdcaparse.c: dcaparse: initialize size variable size can be used in cleanup without being initialized. Hence setting it to 0 when declaring https://bugzilla.gnome.org/show_bug.cgi?id=751306 2015-06-22 13:13:29 +0900 Vineeth TM * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: initialze bpf variable bpf variable might be used in cleanup without being intialized. https://bugzilla.gnome.org/show_bug.cgi?id=751306 2015-06-19 14:50:59 +0200 Miguel París Díaz * gst/rtpmanager/gstrtprtxqueue.c: rtprtxqueue: reverse pending list before pushing buffers With this we send the RTX buffers in the same order that they were requested. https://bugzilla.gnome.org/show_bug.cgi?id=751297 2015-06-21 19:22:10 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: flvmux: Fix DTS validity check This check was up-side-down, causing a bad timestamp at start and then all timestamp being delayed. https://bugzilla.gnome.org/show_bug.cgi?id=751298 2015-06-17 15:19:47 -0400 Nicolas Dufresne * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_types.c: cslg: Add Composition Shift Least Greatest Atom This simply add fourcc and dump function for the cslg Atom. https://bugzilla.gnome.org/show_bug.cgi?id=751103 2015-06-17 15:18:38 -0400 Nicolas Dufresne * gst/isomp4/qtdemux_dump.c: ctts_dump: Fix signess issues It didn't bug, but use correct signess in traces. The number of entries is unsigned while the offset can be signed according to recent spec. https://bugzilla.gnome.org/show_bug.cgi?id=751103 2015-06-16 17:48:08 -0400 Nicolas Dufresne * common: Automatic update of common submodule From 6015d26 to f74b2df 2015-06-16 11:43:39 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: gst_rtp_buffer_ext_timestamp() modifies its first argument, keep a copy around 2015-06-16 10:30:34 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Compare ext RTP times, not plain RTP time and ext RTP time when calculating elapsed time Otherwise all RTP times after a wraparound would be considered as going backwards, they will always be smaller than the ext RTP time. 2015-06-15 21:32:43 +0200 Sebastian Dröge * ext/gtk/gtkgstglwidget.c: gtkglwidget: Const'ify another array 2015-06-15 21:29:46 +0200 Sebastian Dröge * ext/gtk/gtkgstglwidget.c: gtkglwidget: Calculate the viewport size ourselves Getting the current viewport and modifying it relatively will produce an interesting feedback loop during widget resizing. Over a few frames we will gradually move the viewport a bit until it converged again, adding unnecessary additional borders at the top and left. 2015-06-15 21:24:01 +0200 Sebastian Dröge * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtk: Use the display width/height for the widget's preferred width/height 2015-06-15 20:45:11 +0200 Sebastian Dröge * ext/gtk/gstgtksink.c: * ext/gtk/gtkgstwidget.c: gtksink: Add support for xRGB/BGRx 2015-06-15 20:39:59 +0200 Sebastian Dröge * ext/gtk/gstgtksink.c: * ext/gtk/gtkgstwidget.c: gtk: Cairo color formats are in native endianness, GStreamer's in memory order CAIRO_FORMAT_ARGB32 is ARGB on big endian and BGRA on little endian. 2015-06-15 20:35:38 +0200 Sebastian Dröge * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtk: Implement ignore-alpha property and enable it by default 2015-06-15 20:13:57 +0200 Sebastian Dröge * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtksink.c: gtk: Sync properties from the sink to the widget upon widget creation 2015-06-15 19:25:12 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: The default rtp-profile should be AVP, not AVPF 2015-06-15 18:28:37 +1000 Matthew Waters * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtk: implement pixel and display aspect ratio handling 2015-06-15 14:32:21 +0900 Sangkyu Park * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Minor cleanup 1. Add Null check in 'free_item' function. 2. Fix a typing error of comment. https://bugzilla.gnome.org/show_bug.cgi?id=750965 2015-06-15 14:35:35 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: gtk: silence unused variable warnings for unsupported winsys' 2015-06-15 14:33:08 +1000 Matthew Waters * ext/gtk/gtkgstglwidget.c: gtk: implement basic wayland GL support 2015-06-12 17:44:51 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: flmux: Make sure best_time is initialized 2015-06-12 23:29:19 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: rtpbin/session: Add new ntp-time-source property and deprecate use-pipeline-clock property The new property allows to select the time source that should be used for the NTP time in RTCP packets. By default it will continue to calculate the NTP timestamp (1900 epoch) based on the realtime clock. Alternatively it can use the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's clock time. The latter is especially useful for synchronizing multiple receivers if all of them share the same clock. If use-pipeline-clock is set to TRUE, it will override the ntp-time-source setting and continue to use the running time plus 70 years. This is only kept for backwards compatibility. 2015-04-07 16:03:42 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: test for muxing with DTS outside the segment https://bugzilla.gnome.org/show_bug.cgi?id=740575 2015-06-11 17:26:49 -0400 Nicolas Dufresne * gst/isomp4/qtdemux.c: qtdemux: Adjust segment according to ctts offset In presence of a CTTS, the segment start/stop must be offset so the segment start/stop include the PTS. This is needed since the PTS cannot be negative in this format. This fixes issues where the running time of the first buffer isn't at the start. https://bugzilla.gnome.org/show_bug.cgi?id=740575 2015-04-03 20:34:42 -0400 Nicolas Dufresne * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Handle DTS with negative running time As QT works with duration, simply bring back first DTS to 0 and shift forward the PTS of the same amount. https://bugzilla.gnome.org/show_bug.cgi?id=740575 2015-06-10 18:15:52 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Add negative runtime DTS support This is done by using new feature of the CollectPad clip function which sets the DTS as a gint64 in the collected data. It also simplify the code a bit. https://bugzilla.gnome.org/show_bug.cgi?id=740575 2015-06-12 23:06:24 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Rename some variables and debug output to make more sense Local and remote were mixed up in a few places, and the time we store here is not UNIX time (1970 epoch), but NTP time (1900 epoch) in nanoseconds. 2015-06-12 19:21:10 +0300 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: fix latency property query on RemoteIO AudioUnitGetProperty would fail with kParamErr (-50) every time, simply because size wasn't initialized. Now it returns zero latency, but at least it doesn't fail. https://bugzilla.gnome.org/show_bug.cgi?id=750868 2015-06-12 15:39:56 +0200 Thibault Saunier * ext/gtk/gtkgstglwidget.c: gtk: Do not try to activate a NULL GLContext At that point in the code nothing guarantees it exists 2015-04-07 14:06:16 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Fix mapping of latency parameters to buffer attributes 2015-06-12 15:17:30 +1000 Matthew Waters * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstwidget.c: gtk: implement video aspect-ratio handling For both the software and the GL sink's. Doesn't deal with the pixel-aspect-ratio field at all yet. 2015-06-12 12:40:50 +1000 Matthew Waters * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtksink.c: gtk: fix a couple of typos 2015-06-12 12:29:37 +1000 Matthew Waters * ext/gtk/gstgtkglsink.c: gtkglsink: reset the context/display in READY_TO_NULL Fixes context propagation in pipelines with upstream GL elements. 2015-06-11 12:41:10 -0400 Nicolas Dufresne * tests/examples/gtk/gtkglsink.c: gstgtk: No need to realize the widget The widget already does that. 2015-06-11 12:38:53 -0400 Nicolas Dufresne * tests/examples/gtk/gtkglsink.c: * tests/examples/gtk/gtksink.c: gstgtk: Don't leak the widget g_object_get() returns a ref, gtk_container_add() only ref_sink(). That mean we still need to unref afterward. This leak was hiding a reference bug previously present. 2015-06-11 12:10:23 -0400 Nicolas Dufresne * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtksink.c: gstgtk: Allow doing gst-inspect-1.0 on these elements This patch allow going gst-inspect-1.0 on these elements removing ugly crash that was previously occurring. The method consist of making the widget creation as lazy as possible. This way we don't endup doing gtk_init() before the application. We also ref_sink() the widget, so we don't crash if the parent widget is discarded, and cleanly error out with GL if the widget has no parent window, because calling gtk_widget_realized() can only be done if the widget has been parented to a window). 2015-06-12 01:56:37 +1000 Jan Schmidt * gst/matroska/matroska-demux.c: matroska-demux: Actually set detected 3D info into output caps. Use the information read from the StereoMode info to configure multiview-mode and multiview-flags in the video caps. 2015-06-11 13:36:54 +1000 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: splitmuxsink: Take released-but-not-yet-output bytes into account When deciding whether it's time to switch to a new file, take into account data that's been released for pushing, but hasn't yet been pushed - because downstream is slow or the threads haven't been scheduled. Fixes a race in the unit test and probably in practice - sometimes failing to switch when it should for an extra GOP or two. Also fix a problem in splitmuxsrc where playback sometimes stalls at startup if types are found too quickly. https://bugzilla.gnome.org/show_bug.cgi?id=750747 2015-06-11 15:02:44 +0200 Thibault Saunier * ext/gtk/gtkgstglwidget.c: gtk: Do not try to initialize display if we have not have a GLContext yet 2015-06-11 14:58:27 +0200 Sebastian Dröge * tests/examples/gtk/Makefile.am: gtk: Add missing CFLAGS to example 2014-12-18 17:00:30 +1100 Matthew Waters * ext/gtk/Makefile.am: * ext/gtk/gstgtkglsink.c: * ext/gtk/gstgtkglsink.h: * ext/gtk/gstgtksink.c: * ext/gtk/gstgtksink.h: * ext/gtk/gstplugin.c: * ext/gtk/gtkgstglwidget.c: * ext/gtk/gtkgstglwidget.h: * ext/gtk/gtkgstwidget.c: * ext/gtk/gtkgstwidget.h: * tests/examples/gtk/Makefile.am: * tests/examples/gtk/gtkglsink.c: * tests/examples/gtk/gtksink.c: Implement gtk sinks two sinks are provided. gtksink which is a cairo/software based renderer and gtkglsink which utilises the GL support in gtk and gstreamer. 2015-06-11 01:04:51 -0300 Thiago Santos * gst/isomp4/atoms.c: atoms: remove custom gst_buffer_new function in favor of core version Remove a custom specialized version of gst_buffer_new_wrapped by using gst_buffer_new_wrapped_full inside a macro to simplify parameters and give it a more meaningful name. It is only used to create temporary buffers to have its data copied. 2015-06-11 00:14:41 -0300 Thiago Santos * gst/isomp4/atoms.c: atoms: simplify free form data atoms creation Avoid creating an intermediary buffer or memory area just to copy into an atom's data area. 2015-06-10 22:27:27 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: qtmux: add AC-3 muxing support Adds AC-3 muxing support. It is defined for mp4 and 3gp formats. One extra feature that was added was the ability to add extension atoms after set_caps as the AC-3 extension atom needs some data that has to be extracted from the stream itself and is not present on caps. 2015-06-10 22:36:59 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: qtmux: remove unused type MP4S 2015-06-10 22:29:01 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: remove duplicate attribute value set It is also set a few lines below 2015-06-11 00:22:54 +1000 Jan Schmidt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroska: Implement basic stereoscopic video support Implement support for the packed video formats WebM uses, not all the values that Matroska might use. In practice, it's really hard to find any samples in the wild of any. Supported in both the muxer and demuxer. 2015-06-10 01:26:15 +1000 Jan Schmidt * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_types.c: qtdemux: Add basic support for MPEG-A stereoscopic video The MPEG-A format provides an extension to the ISO base media file format to store stereoscopic content encoded with different codecs like H.264 and MPEG-4:2. The stereo video media information(svmi) atom declares the presence and storage method for the video. Stereo video information for MPEG-A can also be supplied through the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which is not implemented in this patch. Also missing is support for stereo video encoded as separate video tracks for now. Based on a patch by Sreerenj Balachandran https://bugzilla.gnome.org/show_bug.cgi?id=611157 2015-06-02 16:15:35 -0400 Xavier Claessens * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Add tls-database property https://bugzilla.gnome.org/show_bug.cgi?id=750298 2015-06-10 14:33:50 +0200 Sebastian Dröge * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP() The mix between all these in the RTP code is confusing, let's try to be consistent. 2015-06-10 14:49:50 +0300 Ilya Konstantinov * gst/rtpmanager/rtpsource.c: rtpmanager: clarify negative lost packets in stats Also: - Move notes on units before field documentation. - Unify documentation style. https://bugzilla.gnome.org/show_bug.cgi?id=750653 2015-06-10 06:38:39 -0400 Xavier Claessens * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix getter of "ssl-use-system-ca-file" https://bugzilla.gnome.org/show_bug.cgi?id=750298 2015-06-10 09:49:47 +0900 Vineeth TM * gst/isomp4/qtdemux.c: qtdemux: fix reverse playback When performing seek, segment->start is being updated with desired_offset, but in case of reverse playback segment->start should be 0 and segment->stop should be updated with desired offset. https://bugzilla.gnome.org/show_bug.cgi?id=750675 2015-01-21 18:09:03 +0100 Philipp Zabel * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: gstv4l2bufferpool: handle -EPIPE from DQBUF to signal EOS The V4L2 decoder signals EOS by returning -EPIPE from DQBUF after the last buffer. https://bugzilla.gnome.org/show_bug.cgi?id=743338 2015-06-06 21:09:19 -0400 Xavier Claessens * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add a GTlsInteraction property It can be used for TLS client authentication. https://bugzilla.gnome.org/show_bug.cgi?id=750471 2015-01-09 11:36:11 +0100 Enrico Jorns * sys/v4l2/gstv4l2transform.c: v4l2: Allow scaling in the v4l2*convert element This is inspired of videoscale and videoconvert elements. https://bugzilla.gnome.org/show_bug.cgi?id=742917 2015-06-09 19:02:55 +0300 Ilya Konstantinov * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.h: rtpmanager: document units of stats and arguments Also, minor spelling and style corrections. https://bugzilla.gnome.org/show_bug.cgi?id=750653 2015-06-09 14:42:27 +0200 Stefan Sauer * Makefile.am: cruft: add the obsolete tmpl dir to cruft-dirs 2015-06-09 11:30:22 +0200 Edward Hervey * common: Automatic update of common submodule From d9a3353 to 6015d26 2015-06-09 07:04:07 +0200 Edward Hervey * common: Fix common version Was accidently downgraded by 87a4884acd8655a6591d735a1d944ecb5ea3de16 2015-06-08 19:11:41 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Also set colorimetry on output devices This completes the code that set the colorimetry on output device. 2015-06-08 19:10:34 -0400 Nicolas Dufresne * common: * sys/v4l2/gstv4l2object.c: v4l2: Add missing SMTP240M matrix This is missing in the doc, but was in the header. 2015-06-08 23:00:16 +0100 Luis de Bethencourt * gst/goom/goom_core.c: goom: possible uninitialized variables warning Build fails with the latest snapshot of gcc-4.9 because param1 and param2 might possibly be used uninitialized. They are set depending on the cases of a switch statement and the compiler sees this as not a complete guarantee. Set them to 0 if the switch statement falls down to the default case. https://bugzilla.gnome.org/show_bug.cgi?id=750566#c6 2015-06-08 17:24:38 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fully implement colorimetry support This fixes wrong mapping for sRGB as in GStreamer sRGB correctly apply to RGB formats, while in V4L2 it's an alias for sYCC. Also add support for the new quantization (range), ycbcr_encoding (matrix) and xfer_func (transfer) enumeration. 2015-06-08 17:01:15 -0400 Nicolas Dufresne * sys/v4l2/ext/types-compat.h: * sys/v4l2/ext/v4l2-common.h: * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: v4l2: Update kernel headers to latest from media tree This is the latest from media tree. This should enable more development of the v4l2 elements. This includes new flags requires to fix draining path in decoder, colorimetry and much more. 2015-06-08 23:07:55 +0200 Stefan Sauer * common: Automatic update of common submodule From d37af32 to d9a3353 2015-06-08 19:42:30 +0100 Chris Clayton * gst/rtp/gstrtpvp8pay.c: rtpvp8depay: potential access beyond end of array Compiling (with gcc-4.9-20150603) produces an error because of an access beyond the end of an array. This patch fixes the error by initializing the loop control/array index variable (i) to 1 and returning i - 1 when a match is found. Also, because the values stored in the array increase in value as the index increases, the >= test unnecessary, so it is removed. 2015-04-30 02:52:58 +1000 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Don't accumulate more than 2 GOPs Don't allow large amounts of data to queue up - we only need the GOP we're writing, and the GOP we're accumulating. 2015-04-16 10:44:49 +1000 Jan Schmidt * gst/isomp4/gstqtmux.c: isomp4: fsync after sending updates in robust mode Use the new GstBuffer SYNC_AFTER flag to trigger an fsync after updating the moov or mdat atom, and after updating the free atom to make it visible. 2015-04-03 00:57:20 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: isomp4: Only set moov header into streamheader at EOS Only update the moov header into the caps if it's the finalised moov at EOS time. Avoids posting a bogus moov at startup and repeated updates in robust-recording mode 2015-04-03 01:44:15 +1100 Jan Schmidt * tests/check/elements/qtmux.c: tests: Update mp4 mux test for mdat placeholder change The mp4 muxer now writes a place-holder mdat as a free atom followed by a 0-byte mdat that covers the rest of the file, making it possible to rewrite it as 64-bit, or leave it as-is if nothing else is written afterward 2015-04-01 11:15:38 +1100 Jan Schmidt * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/atomsrecovery.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: isomp4: Implement robust muxing using ping-pong strategy Implement a robust recording mode, where the output file is always in a playable state, seeking and rewriting the moov header at a configurable interval. Rewriting moov is done using reserved space at the start of the file, and a ping-pong strategy where the moov is replaced atomically so it's never invalid. Track when tags have actually changed, and don't write them into the moov unless they've changed. Clear any existing tags when re-writing them, so we can do progressive moov updating in robust recording mode. Write placeholder mdat as a free atom plus a 32-bit mdat with '0' size, which means "rest of the file" in the spec. Re-write it later to a full 64-bit extended size atom if needed. 2015-04-01 00:58:52 +1100 Jan Schmidt * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: isomp4: Update edit list when re-writing moov Correctly update any edit lists each time the moov is recalculated, updating existing table entries if they already exist instead of just adding new ones. 2015-04-08 01:41:18 +1000 Jan Schmidt * gst/isomp4/gstqtmux.c: isomp4: Remove an extra bracket in a comment. 2015-03-19 20:29:44 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: Protect total_duration state variable with the object lock. Prevent deadlocks from downstream querying duration from the streaming thread. 2015-06-07 23:06:20 +0200 Stefan Sauer * common: Automatic update of common submodule From 21ba2e5 to d37af32 2015-06-07 19:24:20 +0100 Luis de Bethencourt * gst/goom/gstaudiovisualizer.c: goom: clean dereferences of private structure https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-06-07 19:20:04 +0100 Luis de Bethencourt * gst/goom2k1/gstaudiovisualizer.c: goom2k1: clean dereferences of private structure https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-06-07 17:32:01 +0200 Stefan Sauer * common: Automatic update of common submodule From c408583 to 21ba2e5 2015-06-07 17:01:37 +0200 Stefan Sauer * docs/plugins/Makefile.am: docs: remove variables that we define in the snippet from common This is syncing our Makefile.am with upstream gtkdoc. 2015-06-07 17:16:19 +0200 Stefan Sauer * autogen.sh: * common: Automatic update of common submodule From d676993 to c408583 2015-06-07 16:44:37 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.5.1 === 2015-06-07 10:46:34 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * gst/deinterlace/tvtime-dist.c: * gst/videomixer/videomixerorc-dist.c: * win32/common/config.h: Release 1.5.1 2015-06-07 10:38:28 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2015-06-07 10:32:38 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * tests/check/elements/rtpsession.c: rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property 2015-06-07 09:35:38 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: Update translations 2015-06-05 15:32:10 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Don't warn when optional CID are not implement gst_v4l2_get_attributre() shall only be used when the CID is expected to be supported. Otherwise, we get unwanted warning posted to the bus. 2015-06-05 16:43:08 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc https://bugzilla.gnome.org/show_bug.cgi?id=749581 2015-06-04 14:18:01 +0900 Vineeth TM * gst/interleave/interleave.c: interleave: error when channel-positions-from-input=False self->channels is being incremented only when channel-positions-from-input is set as TRUE. So in case of FALSE self->func is not set and hence creating assertion error. Hence removing the condition to increment self->channels. https://bugzilla.gnome.org/show_bug.cgi?id=744211 2015-06-05 10:33:11 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Add support for receiving reduced size RTCP It worked before but gave warnings, now we just ignore RTCP packets that don't start with a SR. As all we're interested in here are SRs. 2015-06-03 12:22:42 +0200 Jose Antonio Santos Cadenas * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Add support for reduce size rtcp According to RFC 5506, reduce size packages can be sent, this packages may not be compound, so we need to add support for getting ssrc from other types of packages. https://bugzilla.gnome.org/show_bug.cgi?id=750327 2015-06-03 13:14:44 +0200 Jose Antonio Santos Cadenas * gst/rtpmanager/rtpsession.c: rtpsession: Add support for receiving reduced size rtcp See RFC 5506 https://bugzilla.gnome.org/show_bug.cgi?id=750332 2015-06-04 16:09:41 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Add support for channel configurations 11, 12 and 14 and 7 actually has 8 channels ISO/IEC 14496-3:2009/PDAM 4 added 11, 12 and 14. 2015-06-03 08:57:57 -0400 Nicolas Dufresne * gst/rtp/gstasteriskh263.c: asteriskh263: Un-rank clashing depayloader This depayloader clash with the standard one for H263p. It produces an H263p stream with a modified header. It uses encoding-name that is the same as H263p (H263-1998) though the resulting ES is not decodable or parsable in GStreamer, making it unsuable in dynamic pipeline. This patch unrank this specialized depayloader since it can only be used in custom pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=739935 2015-06-02 18:09:48 +0100 Luis de Bethencourt * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom2k1: remove variables not needed anymore https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-06-02 17:52:46 +0100 Luis de Bethencourt * gst/goom2k1/Makefile.am: * gst/goom2k1/gstaudiovisualizer.c: * gst/goom2k1/gstaudiovisualizer.h: * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom2k1: rebase to use the audiovisualizer class Rebase to have goom2k1 using the common GstAudioVisualizer class https://bugzilla.gnome.org/show_bug.cgi?id=742875 2015-06-02 17:29:36 +0100 Luis de Bethencourt * gst/goom/Makefile.am: * gst/goom/gstaudiovisualizer.c: * gst/goom/gstaudiovisualizer.h: * gst/goom/gstgoom.c: * gst/goom/gstgoom.h: goom: rebase to use the audiovisualizer class 2015-06-02 16:31:10 +0200 Edward Hervey * tests/check/pipelines/lame.c: check: Use GST_CHECK_MAIN () macro everywhere Makes source code smaller, and ensures we go through common initialization path (like the one that sets up XML unit test output ...) 2015-06-02 16:27:24 +0200 Edward Hervey * tests/check/elements/aacparse.c: * tests/check/elements/ac3parse.c: * tests/check/elements/apev2mux.c: * tests/check/elements/aspectratiocrop.c: * tests/check/elements/audioamplify.c: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/avimux.c: * tests/check/elements/equalizer.c: * tests/check/elements/flacparse.c: * tests/check/elements/id3v2mux.c: * tests/check/elements/jpegdec.c: * tests/check/elements/jpegenc.c: * tests/check/elements/matroskamux.c: * tests/check/elements/mpegaudioparse.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpsession.c: * tests/check/elements/spectrum.c: * tests/check/elements/videobox.c: * tests/check/elements/videocrop.c: * tests/check/elements/videofilter.c: * tests/check/elements/wavpackdec.c: * tests/check/elements/wavpackenc.c: * tests/check/elements/wavpackparse.c: * tests/check/elements/y4menc.c: * tests/check/pipelines/simple-launch-lines.c: * tests/check/pipelines/tagschecking.c: * tests/check/pipelines/wavpack.c: check: Use GST_CHECK_MAIN () macro everywhere Makes source code smaller, and ensures we go through common initialization path (like the one that sets up XML unit test output ...) 2015-05-26 14:47:31 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Only schedule a timer when we actually have to send RTCP Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no feedback is actually pending and no regular RTCP has to be sent). This improves CPU usage and battery life quite a lot. https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-22 13:44:03 +0300 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Remove useless goto https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-21 12:54:47 +0300 Sebastian Dröge * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-H264-rtx.sh: * tests/examples/rtp/client-rtpaux.c: * tests/examples/rtp/server-VTS-H264-rtx.sh: * tests/examples/rtp/server-rtpaux.c: examples: Set RTP profile to AVPF for rtpaux examples https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-04 16:41:50 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Set RTP profile on the rtpsession objects https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-21 14:13:56 +0300 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: Add rtp-profile property for setting the default profile of newly created sessions https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-04 11:51:41 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Only put RRs and full SDES into regular RTCP packets If we may suppress the packet due to the rules of RFC4585 (i.e. when below the t-rr-int), we can send a smaller RTCP packet without RRs and full SDES. In theory we could even send a minimal RTCP packet according to RFC5506, but we don't support that yet. https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-04 13:51:50 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Keep track of tp/tn and t_rr_last separately Otherwise we can't properly schedule RTCP in feedback profiles as we need to distinguish the time when we last checked for sending RTCP (tp) but might have suppressed it, and the time when we last actually sent a non-early RTCP packet. This together with the other changes should now properly implement RTCP scheduling according to RFC4585, and especially allow us to send feedback packets a lot if needed but only send regular RTCP packets every once in a while. https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-04 11:42:08 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc) And modify our RTCP scheduling algorithm accordingly. We now can send more RTCP packets if needed for feedback, but will throttle full RTCP packets by rtcp-min-interval (t-rr-int from RFC4585). In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is statically set to 1s or 0s by RFC4585. Tmin defines how often we should send RTCP packets at most. https://bugzilla.gnome.org/show_bug.cgi?id=746543 2015-05-30 17:41:05 -0400 Olivier Crête * gst/law/mulaw-decode.c: mulawdec: Let baseclass estimate bitrate This makes playback directly from a file work with the right caps. 2015-05-27 16:31:23 +0100 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: dynudpsink: keep GCancellable fd around instead of re-creating it constantly And create it only when starting the element. 2015-05-27 15:55:56 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly Otherwise we constantly create/close event file descriptors, every time we call g_socket_condition_timed_wait() or g_socket_send_message(s)(), i.e. a lot. Which is not particularly good for performance. Can't create GCancellable in ::start() here because it's used in client_new() which may be called via the add-client action signal which may be called before the element is up and running. 2015-05-19 18:13:16 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: keep GCancellable fd around instead of re-creating it constantly Otherwise we constantly create/close event file descriptors, every single time we call g_socket_condition_timed_wait() or g_socket_receive_message(), i.e. twice per packet received! This was not particularly good for performance. Also only create GCancellable on start-up. 2015-05-26 15:33:37 +0100 Luis de Bethencourt * gst/matroska/matroska-read-common.c: matroska: overwritten value assignment curpos is set and immediately after, set again. Remove the redundant assignment. https://bugzilla.gnome.org/show_bug.cgi?id=749909 2015-05-23 13:47:17 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: don't shadow existing outbuf variable And fix unref of the wrong one which will contain NULL in an error code path. 2015-05-23 13:23:22 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: rtpvrawdepay: map/unmap output frame only once, not for every input packet Map output buffer after creating it and keep it mapped until we're done with it instead of mapping/unmapping it for every single input buffer. 2015-05-25 08:47:47 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: remove fixme from 2006 It has been verified by use over time. 2015-05-23 14:36:41 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: fix reverse playback of fragmented media qtdemux creates a samples array and gets the timestamps for buffers by accumulating their durations. When doing reverse playback of fragments, accumulating samples will lead to wrong timestamps as the timestamps should go decreasing from fragment to fragment and the accumulation will produce wrong results. In this case, when receiving a discont for fragmented reverse playback, the previous samples information should be flushed before new data is processed. 2015-05-23 01:03:18 +0900 Jimmy Ohn * gst/multifile/gstsplitfilesrc.c: splitfilesrc: Implement binary search in find_part_for_offset Implement binary search using gst_util_array_binary_search https://bugzilla.gnome.org/show_bug.cgi?id=749690 2015-05-21 13:26:53 +0300 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Don't crash if we receive FIR/PLI from a source we don't know 2015-05-21 09:35:58 +0200 Santiago Carot-Nemesio * gst/rtpmanager/rtpsession.c: rtpsession: Fix collection of statistics Stats should be collected on the media rtp source not in the sender one. https://bugzilla.gnome.org/show_bug.cgi?id=749669 2015-04-20 10:07:30 +0200 Edward Hervey * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: Add a new max-duration file switching mode This new mode ensures that files will never exceed a certain duration based on incoming buffer PTS (and duration if present) Note: * You need timestamped buffers (duh). If some of the incoming buffers don't have PTS, then it will just accept them in the current file 2015-04-17 16:18:32 +0200 Edward Hervey * gst/multifile/gstmultifilesink.c: multifilesink: streamline the file-switch code a bit Use the same functions regardless of the mode we are using 2015-04-02 13:35:18 +0100 Edward Hervey * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: add "aggregate-gops" property to process GOPs as a whole This property can be used in combination with next-file=max-size (and perhaps a future next-file=max-duration) to make sure that each file part starts cleanly with a key frame and the appropriate headers. In order for this property to work correctly, upstream elements should make sure than any headers that need to be written in a standalone file are: 1) in the streamheader caps field 2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER that are just before the keyframe buffer This is useful for MPEG-TS/MPEG-PS file segmenting in combination with mpegtsmux or mpegpsmux. Original patch by: Tim-Philipp Müller 2015-05-20 16:37:22 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.h: rtspsrc: Use single-include header for the RTSP library 2014-10-24 23:47:21 +0100 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: udp: don't use soon-to-be-deprecated g_cancellable_reset() From the API documentation: "Note that it is generally not a good idea to reuse an existing cancellable for more operations after it has been cancelled once, as this function might tempt you to do. The recommended practice is to drop the reference to a cancellable after cancelling it, and let it die with the outstanding async operations. You should create a fresh cancellable for further async operations." https://bugzilla.gnome.org/show_bug.cgi?id=739132 2015-05-18 20:13:01 +0200 Stefan Sauer * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/cutter/gstcutter.c: * gst/equalizer/gstiirequalizernbands.c: * gst/multifile/gstmultifilesink.c: Revert "doc: Workaround gtkdoc issue" This reverts commit 1797c8f8b12d7f4c7a9444c94f34f4d08ec85945. This is fixed by the gtk-doc 1.23 release. cannot contain : http://www.docbook.org/tdg/en/html/para.html http://www.docbook.org/tdg/en/html/refsect2.html 2015-05-18 16:40:21 +0200 Nicola Murino * gst/rtp/gstrtpg726pay.c: rtpg726pay: fix caps leak https://bugzilla.gnome.org/show_bug.cgi?id=749544 2015-05-18 16:34:13 +0200 Nicola Murino * gst/rtp/gstrtpg726depay.c: rtpg726depay: don't leak input buffer https://bugzilla.gnome.org/show_bug.cgi?id=749543 2015-05-18 17:38:31 +0300 Sebastian Dröge * gst/rtpmanager/rtpsource.c: rtpsource: Queue bad packets instead of dropping them So we can send them out once we found the next, consecutive sequence number in case one is following. 2015-05-18 17:38:14 +0300 Sebastian Dröge * gst/rtpmanager/rtpsource.c: rtpsource: Use g_queue_foreach() to unref all buffers in queues 2015-05-18 17:19:31 +0300 Sebastian Dröge * gst/rtpmanager/rtpsource.c: rtpsource: Refactor seqnum comparison code a bit 2015-05-18 17:08:53 +0300 Sebastian Dröge * gst/rtpmanager/rtpsource.c: rtpsource: Allow sequence number wraparound during probation 2015-05-18 17:07:23 +0300 Sebastian Dröge * gst/rtpmanager/rtpsource.c: rtpsource: Make sequence number comparison code more readable ... by using gst_rtp_buffer_compare_seqnum() and signed integers instead of implictly using effects of integer over/underflows. 2015-04-22 18:54:06 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything It might just be a late retransmission or spurious packet from elsewhere, but resetting everything would mean that we will cause a noticeable hickup. Let's get some confidence first that the sequence numbers changed for whatever reason. https://bugzilla.gnome.org/show_bug.cgi?id=747922 2015-05-16 23:37:06 -0400 Nicolas Dufresne * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/cutter/gstcutter.c: * gst/equalizer/gstiirequalizernbands.c: * gst/multifile/gstmultifilesink.c: doc: Workaround gtkdoc issue With gtkdoc 1.22, the XML generator fails when a itemizedlist is followed by a refsect2. Workaround the issue by wrapping the refsect2 into para. 2015-01-23 13:57:40 +0100 Stefan Sauer * gst/isomp4/qtdemux_types.c: qtdemux: avoid wrong warnings on unknown node types Add 'name' and 'mean' fourccs, as we handle them. Right now each use would trigger a warning. 2015-05-08 19:13:00 +0200 Nicola Murino * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726depay.h: rtpg726depay: add block_align to output caps It is needed to correctly negotiate caps with matroskamux and most other muxers. https://bugzilla.gnome.org/show_bug.cgi?id=749129 2015-05-12 13:41:58 +0300 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Fix time-domain convolution with >1 channels input_samples is the number of frames, but we used it as the number of samples. https://bugzilla.gnome.org/show_bug.cgi?id=747204 2015-05-12 12:13:16 +0300 Sebastian Dröge * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vp[89]enc: Properly convert between GStreamer and encoder timebase ... by switching numerator and denominator when scaling. https://bugzilla.gnome.org/show_bug.cgi?id=749122 2015-05-11 13:33:26 +0300 Sebastian Dröge * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vp[89]enc: Don't set timebase from the framerate The framerate very often is just an indication of the ideal framerate, not the actual framerate of the stream. By just using the framerate, we confuse the rate control algorithm algorithm as multiple frames will map to the same PTS or have durations of 0. https://bugzilla.gnome.org/show_bug.cgi?id=749122 2015-05-10 14:21:04 +0200 Mark Nauwelaerts * tests/check/elements/wavpackparse.c: tests: wavpackparse: fix unit test See also https://bugzilla.gnome.org/show_bug.cgi?id=738237 2015-05-10 11:34:33 +0100 Tim-Philipp Müller * ext/twolame/gsttwolamemp2enc.c: docs: update example pipelines in element docs Mostly gst-launch -> gst-launch-1.0, but also use autoaudiosink/autovideosink in more places and update pipelines a little or flesh out descriptions. 2015-05-10 11:34:33 +0100 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: docs: update example pipelines in element docs Mostly gst-launch -> gst-launch-1.0, but also use autoaudiosink/autovideosink in more places and update pipelines a little or flesh out descriptions. 2015-05-10 11:05:00 +0100 Tim-Philipp Müller * ext/shout2/gstshout2.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpL24pay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtpmanager/gstrtpmux.c: * tests/check/pipelines/wavenc.c: * tests/examples/rtp/client-PCMA.c: * tests/examples/rtp/server-alsasrc-PCMA.c: docs: update example pipelines in element docs Mostly gst-launch -> gst-launch-1.0 Use autovideosink/autoaudiosink more often. Sprinkle some converters here and there. 2015-05-09 19:48:55 +0200 Piotr Drąg * po/POTFILES.in: po: update POTFILES.in https://bugzilla.gnome.org/show_bug.cgi?id=749163 2015-05-10 10:52:18 +0100 Tim-Philipp Müller * gst/multifile/gstsplitmuxsrc.c: splitmuxsrc: minor error message clean-up Don't put filename in error message shown to user. 2015-05-07 16:25:36 +0200 Guillaume Desmottes * gst/audioparsers/gstflacparse.c: flacparse: fix buffer leak when stored to seektable Fix a leak with the validate.file.playback.change_state_intensive.samples_multimedia_cx_flac_Yesterday_flac scenario. https://bugzilla.gnome.org/show_bug.cgi?id=749072 2015-05-07 17:10:37 +0900 Paul Hyunil * gst/isomp4/qtdemux.c: qtdemux: fix example pipeline in docs The gst-launch script for example launch line to test qtdemux is missing a queue before the decodebins, otherwise the gst-launch-1.0 command won't work. https://bugzilla.gnome.org/show_bug.cgi?id=749054 2015-05-07 14:51:45 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active" This reverts commit d22ec496328e6ba8edbf2d071d5608b2af2831e8. Application code might expect that it only gets external sources on those signals, and get confused by this. If anything we would need to add new signals. 2015-03-25 15:27:34 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active Without this it seems impossible for an application to easily get notified about the internal ssrcs that are created, e.g. sender sources, and also to know when they are active and produce RTCP packets. https://bugzilla.gnome.org/show_bug.cgi?id=746747 2015-05-04 19:26:14 +0200 Guillaume Desmottes * ext/jpeg/gstjpegdec.c: jpegdec: fix frame leaks in handle_frame() implementation handle_frame() is supposed to consume @frame, so if we don't call gst_video_decoder_drop_frame() or gst_video_decoder_finish_frame() we have to release it manually. https://bugzilla.gnome.org/show_bug.cgi?id=748909 2015-05-04 16:50:38 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix up last commit 2015-05-04 16:46:02 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Only do RTX when using a feedback profile 2015-05-04 13:50:31 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: The stats min_interval is in seconds, not nanoseconds We have to scale it to compare it against our clock times. 2015-05-04 11:38:27 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Only return TRUE if early feedback was requested already and it's early enough 2015-04-30 15:42:34 +0100 Luis de Bethencourt * gst/matroska/matroska-parse.c: matroska: remove unused property enum items 2015-04-30 12:13:59 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: fix buffer leak on eos in push mode Based on patch by Guillaume Desmottes. scenario: validate.http.playback.seek_with_stop.raw_h264_1_mp4 https://bugzilla.gnome.org/show_bug.cgi?id=748617 2015-04-29 19:41:29 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Check for sizes of the rdrf (redirect) atom before accessing the data and use g_strndup() instead of g_strdup() Thanks to Ralph Giles for reporting this. 2015-04-29 15:52:27 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Only enable retransmissions if there is retransmission info in the SDP Otherwise we're going to send early RTCP and NACKs in non-feedback sessions too, which will confuse servers. https://bugzilla.gnome.org/show_bug.cgi?id=748627 2015-02-11 18:09:24 +0530 Ravi Kiran K N * ext/dv/gstdvdemux.c: dvdemux: extract recording time Extracts the recorded time of the dv file from the metadata and puts it into the global tags. https://bugzilla.gnome.org/show_bug.cgi?id=743657 2015-04-28 15:59:25 +0200 Guillaume Desmottes * gst/matroska/matroska-demux.c: matroskademux: fix seek event leak gst_matroska_demux_handle_seek_event() doesn't consume the event so we have to unref it. https://bugzilla.gnome.org/show_bug.cgi?id=748584 2015-04-28 15:42:49 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroska-demux: Send pending tags when adding a new pad We might've parsed those tags before already and tried to push them to non-existing pads before. Now let's do it for real. 2015-04-23 18:57:37 +0200 Sebastian Dröge * gst/rtpmanager/rtpstats.c: rtpstats: Average RTCP packet size is in bytes, bandwidths in bits We need to convert the size to bits for our calculations. https://bugzilla.gnome.org/show_bug.cgi?id=747863 2015-04-23 18:53:39 +0200 Sebastian Dröge * gst/rtpmanager/rtpstats.c: rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere https://bugzilla.gnome.org/show_bug.cgi?id=747863 2015-04-14 18:41:07 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value https://bugzilla.gnome.org/show_bug.cgi?id=747863 2015-04-23 18:49:37 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s https://bugzilla.gnome.org/show_bug.cgi?id=747863 2015-04-27 16:36:27 +0200 Sebastian Dröge * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix RTX unit test The calculations were a bit off everywhere, even before the changes done recently to the delay for RTX of expected future packets. It only worked by accident, but now the calculations are all correct again. Hopefully. 2015-04-27 11:22:11 +0100 Luis de Bethencourt * gst/avi/gstavimux.c: * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/flv/gstindex.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/id3demux/gstid3demux.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/multifile/gstmultifilesrc.c: * gst/multipart/multipartmux.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtpmanager/gstrtpmux.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstvideotemplate.c: * gst/y4m/gsty4mencode.c: Rename property enums from ARG_ to PROP_ Property enum items should be named PROP_ for consistency and readability. 2015-04-27 10:55:13 +0100 Luis de Bethencourt * gst/audiofx/gststereo.c: Rename property enums from ARG_ to PROP_ Property enum items should be named PROP_ for consistency and readability. 2015-04-25 02:49:58 +0300 Ilya Konstantinov * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix "stats" property docs https://bugzilla.gnome.org/show_bug.cgi?id=748436 2015-04-26 17:54:52 +0100 Tim-Philipp Müller * Android.mk: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debugutils/Makefile.am: * gst/deinterlace/Makefile.am: * gst/dtmf/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flv/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/imagefreeze/Makefile.am: * gst/interleave/Makefile.am: * gst/isomp4/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/rtsp/Makefile.am: * gst/shapewipe/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * gst/y4m/Makefile.am: Remove obsolete Android build cruft This is not needed any longer. 2015-04-24 13:55:08 -0300 Thiago Santos * gst/videocrop/gstvideocrop.c: videocrop: print the property values when set Instead of printing the currently used values. The log is meant to show what the properties changed to, not what is being currently used. 2015-04-24 17:01:10 +0100 Luis de Bethencourt * gst/alpha/gstalpha.c: * gst/audiofx/audiokaraoke.c: * gst/deinterlace/gstdeinterlace.c: * gst/multifile/gstmultifilesink.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: remove unused enum items PROP_LAST This were probably added to the enums due to cargo cult programming and are unused. Removing them. 2015-04-24 00:30:35 +0100 Tim-Philipp Müller * gst/level/gstlevel.c: level: fix infinite loop for very low interval values https://bugzilla.gnome.org/show_bug.cgi?id=745515 2015-04-23 16:08:54 +0100 Tim-Philipp Müller * tests/check/Makefile.am: tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON Make sure the test environment is set up. https://bugzilla.gnome.org//show_bug.cgi?id=747624 2015-04-23 16:08:32 +0100 Tim-Philipp Müller * configure.ac: configure: bump automake requirement to 1.14 and autoconf to 2.69 This is only required for builds from git, people can still build tarballs if they only have older autotools. https://bugzilla.gnome.org//show_bug.cgi?id=747624 2015-04-23 16:06:57 +0100 Tim-Philipp Müller * .gitignore: Update .gitignore 2015-04-23 09:55:59 +0200 Jesper Larsen * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix RTCP caps leak https://bugzilla.gnome.org//show_bug.cgi?id=748353 2015-04-22 20:24:20 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra delay. If jitter is very low, this should prevent unnecessary retransmission requests to some degree. https://bugzilla.gnome.org/show_bug.cgi?id=748041 2015-04-22 19:41:07 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest https://bugzilla.gnome.org/show_bug.cgi?id=748041 2015-04-13 11:20:40 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Add "rtx-next-seqnum" property If this is set to FALSE, rtpjitterbuffer will not request retransmissions for future packets based on when they are estimated to arrive. See also https://bugzilla.gnome.org/show_bug.cgi?id=748041 https://bugzilla.gnome.org/show_bug.cgi?id=739868 2015-04-22 19:29:34 +0200 Sebastian Dröge * gst/rtpmanager/gstrtprtxreceive.c: rtxreceive: Put debug output for retransmission requests at the right place Before it was only ever printed once for every time a ssrc was associated with a specific stream. 2015-04-22 18:05:24 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: don't add the same interlace mode twice Some drivers modify the interlace mode to progressive, no matter what input you give them, make sure that we don't add the same interlace mode twice. 2015-04-21 16:34:21 +0100 Luis de Bethencourt * gst/equalizer/gstiirequalizer.c: equalizer: fix dynamic changes on bands When we are in passthrough, the transform function doesn't run and if the passthrough check is in this function it will never be deactivated. Fix this by checking directly whenever a gain is changed. Also set the passthrough to TRUE at init because the gains default to 0, so we can passthrough until any gain property is changed. https://bugzilla.gnome.org/show_bug.cgi?id=748068 2015-04-22 10:30:52 +0200 Sebastian Dröge * INSTALL: Remove INSTALL file autotools automatically generate this, and when using different versions for autogen.sh there will always be changes to a file tracked by git. 2015-04-22 10:30:14 +0200 Sebastian Dröge * LICENSE_readme: Remove LICENSE_readme It's completely outdated and just confusing, better if people are forced to look at the actual code in question than trusting this file. 2015-04-21 15:21:33 +0100 Luis de Bethencourt * sys/v4l2/v4l2_calls.c: v4l2: cast unused return to void Quell unchecked return value defect by casting the return value to void and making it explicit it is going to be ignored. CID #206031 2015-04-17 13:08:02 -0300 Thiago Santos * ext/vpx/gstvp8dec.c: vp8dec: optimize vpx image to gstbuffer copy when strides match Solving this FIXME. Copy the full plane when strides are the same 2015-04-16 15:11:05 -0300 Thiago Santos * ext/vpx/gstvp9dec.c: vp9dec: optimize vpx image to gstbuffer copy when strides match Solving this FIXME. Copy the full plane when strides are the same 2015-04-17 13:32:54 +0100 Vincent Penquerc'h * gst/audioparsers/gstac3parse.c: ac3parse: fix memory leak 2015-04-17 06:51:46 +0000 Alex O'Konski * gst/icydemux/gsticydemux.c: icydemux: Fix segfault if metadata-interval is 0 Prevents an extra unref of GstBuffer when passing a non-icy stream through icydemux with metadata-interval set to 0. Reproducible with: gst-launch-1.0 filesrc location=~/testsong.mp3 ! \ 'application/x-icy,metadata-interval=(int)0' ! icydemux ! decodebin ! wavenc ! \ filesink location=~/testsong.wav https://bugzilla.gnome.org/show_bug.cgi?id=748024 2015-04-17 11:54:23 +0530 Ravi Kiran K N * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: audiofx: fix typo in example pipelines Fix typo in example pipelines https://bugzilla.gnome.org/show_bug.cgi?id=748022 2015-04-15 18:22:37 +0300 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: fix spelling in debug message https://bugzilla.gnome.org//show_bug.cgi?id=747936 2015-04-16 16:33:44 +0100 Luis de Bethencourt * tests/examples/equalizer/demo.c: tests: selectable amount of bands in equalizer demo Adding an option in the equalizer demo to make the number of bands selectable. 2015-04-16 15:31:25 +0200 Sebastian Dröge * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/rtpsource.c: rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource https://bugzilla.gnome.org/show_bug.cgi?id=747394 2015-04-06 12:56:50 +0530 Arun Raghavan * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/rtpsession.c: rtpsession: Track RTX ssrc caps This is needed so that we can generate SR for RTX stream correctly (the clock rate is required). https://bugzilla.gnome.org/show_bug.cgi?id=747394 2015-04-14 13:56:38 +0200 Sebastian Dröge * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers https://bugzilla.gnome.org/show_bug.cgi?id=747394 2015-04-16 16:01:50 +0100 Luis de Bethencourt * tests/examples/equalizer/demo.c: tests: switch equalizer demo to play from uri Switch the equalizer-nbands demo to use uridecodebin, so users can listen to something more pleasant than white noise. If anybody misses the white noise a uri handler to audiotestsrc can be used. 2015-04-16 11:17:38 +0100 Luis de Bethencourt * tests/examples/equalizer/demo.c: tests: improve readability of equalizer demo Rename variable name to make it more readable, add comments for the three scales created per block, and set the window title. 2015-04-15 17:32:37 +0100 Luis de Bethencourt * tests/examples/equalizer/demo.c: tests: add missing license header for equalizer demo 2015-04-16 13:09:19 +0100 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: fix tag list leaks on error paths 2015-04-16 12:23:38 +0100 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: fix tag list leak on unknown stream type 2015-04-09 13:19:49 +0100 Vincent Penquerc'h * tests/check/gst-plugins-good.supp: suppressions: ignore an apparent bug in strtod A buffer overread. https://bugzilla.gnome.org/show_bug.cgi?id=747554 2015-04-15 11:07:27 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxsink.c: splitmuxsink: do not access property variable without the object lock, use the local stack copy instead 2015-04-14 18:45:44 +0200 George Kiagiadakis * gst/multifile/gstsplitmuxsink.c: splitmuxsink: add probe on the multiqueue's sink pad instead of the ghost pad because _release_pad tries to release it from ctx->sinkpad, which is multiqueue's sink pad, and currently fails because the probe is not installed there 2015-04-14 19:08:24 +0200 Sebastian Dröge * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxsend.c: rtprtx*: Fix typos 2015-04-14 17:24:46 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet 2015-04-14 16:27:18 +0200 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Improve debug output a bit if we can't allow early feedback 2015-04-07 18:00:53 -0400 Olivier Crête * gst/rtp/gstrtpvp8depay.c: rtpvp8depay: When dropping intra packet, request keyframe https://bugzilla.gnome.org/show_bug.cgi?id=747208 2015-04-13 20:25:00 +0200 Sebastian Dröge * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO This also happens in the very beginning when we receive the first packet, a warning would be very confusing here. In all places where we should warn about this, we would've printed a warning already before. 2015-04-02 13:26:41 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: minor docs improvement 2014-11-06 12:08:03 +0100 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Add "rtx-max-retries" property This property allows to limit the maximum number of retransmission for a specific packet. https://bugzilla.gnome.org/show_bug.cgi?id=739868 2014-11-04 15:00:52 +0100 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix expected_dts calc in calculate_expected Right above we consider lost_packet packets, each of them having duration, as lost and triggered their timers immediately. Below we use expected_dts to schedule retransmission or schedule lost timers for the packets that come after expected_dts. As we just triggered lost_packets packets as lost, there's no point in scheduling new timers for them and we can just skip over all lost packets. https://bugzilla.gnome.org/show_bug.cgi?id=739868 2015-03-20 18:21:57 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer Resetting the jitterbuffer drops all packets and other things, and will cause a discontinuity in the packets received by the depayloaders. They should now also flush anything they had pending as the new data will start at a different position. https://bugzilla.gnome.org/show_bug.cgi?id=739868 2015-04-10 09:17:26 +0900 Hyunjun Ko * gst/isomp4/qtdemux.c: qtdemux: Update segment.start after key-unit seek When doing key uint seek, qtdemux calls gst_qtdemux_adjust_seek to get proper offset. And then this offset is set to segment.position and segment.time in gst_qtdemux_perform_seek but segment.start is not updated. After that, application sends segment query, qtdemux sets start and stop to query using gst_segment_to_stream_time. Due to the wrong value in segment.start, the stop position is smaller than it should. https://bugzilla.gnome.org/show_bug.cgi?id=746822 2015-04-07 16:12:40 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: remove useless variable do_pts We always write the CTTS in qtmux. Ideally we only want to do that for streams that need DTS, it should be present on the track information rather than be decided based on each buffer 2015-04-07 00:53:35 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: remove subtraction that makes PTS/DTS start from 0 As qt uses durations, it doesn't matter, only the difference between consecutive buffers is important. Also, collectpads already replaces PTS/DTS with the running times for them. 2015-04-06 22:36:43 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: add tests to verify it handles non-0 segments Both input streams in this test have a segment.start = 10s, so output should start from 0 anyway. Another test has both starting at non-0 segments, but the running time of both streams should still start from 0 2015-04-06 20:03:19 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: simple muxing test Adds a new simple test that verifies that data is properly muxed and preserved. PTS, DTS, duration and caps are verified. 2015-04-10 10:59:26 +0530 Ravi Kiran K N * gst/smpte/gstsmpte.h: smpte: remove unused fields Remove the fields - format and fps from smpte as they are unused. https://bugzilla.gnome.org/show_bug.cgi?id=747597 2015-04-10 10:29:47 +0530 Ravi Kiran K N * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/alpha.c: tests: add test suite for alpha Added test suite for alpha element with test cases 1. alpha 2. chroma keying https://bugzilla.gnome.org/show_bug.cgi?id=747595 2015-04-09 12:58:46 +0100 Vincent Penquerc'h * tests/check/gst-plugins-good.supp: suppressions: add a well known zlib inflate bug 2015-04-09 12:58:26 +0100 Vincent Penquerc'h * gst/multifile/gstsplitmuxsink.c: splitmuxsink: fix mutex leak 2015-04-09 12:58:04 +1000 Jan Schmidt * tests/check/elements/rtprtx.c: tests: Fix rtprtx test by handling buffer lists Commit #1018aa made rtprtxsend handle buffer lists, breaking the test which probes for buffers, but not buffer lists. Use a utility function to run the probe callback on each buffer in the list in turn and remove any buffers that are dropped. 2015-04-01 11:15:38 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: isomp4: Refactor various state variables into a mux_mode var Instead of checking various state variables around the muxer, track the current muxing mode in a single 'mux_mode' enum. Add some implementation notes about the different mux modes 2015-04-08 16:40:02 +0200 Edward Hervey * common: * tests/check/Makefile.am: tests: Use AM_TESTS_ENVIRONMENT Needed by the new automake test runner 2015-04-08 11:17:31 +0200 Edward Hervey * gst/rtp/gstrtph263depay.c: rtph263depay: Fix framesize parsing The string passed to the parsing function only contains a framesize, and not + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416 2015-03-20 12:18:37 +0000 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: clip chunk size above the valid maximum (0x7fffffff) https://bugzilla.gnome.org/show_bug.cgi?id=722567 2015-03-20 09:07:35 +0000 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: clip chunk length to available data (when known) This prevents silly chunk lengths from possibly overflowing (at least when we know the actual data length). https://bugzilla.gnome.org/show_bug.cgi?id=722567 2015-04-06 20:17:52 -0700 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Don't accumulate segment bases manually gst_segment_do_seek() does that for us already, and doing it twice will break non-flushing seeks in interesting ways. Leftover from 1.0 porting. Also copy over segment offset and applied_rate, just in case. 2015-04-06 19:08:10 -0700 Sebastian Dröge * tests/icles/test-segment-seeks.c: icles: Fix waiting for segment-done if it happens too fast Sometimes we can get segment-done before we got async-done. If we waited for async-done only, the segment-done would be dropped and we would wait forever for it a few lines below. 2015-04-06 18:55:08 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: stbl_index is valid from 0 onwards It indicates the last sample parsed, not the next one to parse. As it starts in -1, any value from 0 onwards means that it has some valid data. 2015-04-05 20:06:09 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: docs: make GstRTCPSync enum show up in rtpbin docs https://bugzilla.gnome.org/show_bug.cgi?id=747358 2015-04-05 11:45:45 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: add RTPJitterBufferMode enum to rtpbin docs https://bugzilla.gnome.org/show_bug.cgi?id=747358 2015-04-04 11:55:00 -0300 Thiago Santos * gst/multifile/gstmultifilesink.c: multifilesink: close files before posting message Makes sure the files were properly flushed and closed before the message reaches the application 2015-03-30 13:54:23 -0300 Thiago Santos * tests/check/elements/multifile.c: tests: multifile: increment tests to check for multifile messages Also verify that the multifilesink file messages are being correctly posted to the bus 2015-03-30 12:51:35 -0300 Thiago Santos * tests/check/elements/multifile.c: tests: multifile: handle FIXME for proper checking when test finished Use a GstBus and wait for EOS to finish the tests instead of relying on sleeping 2015-03-30 11:14:09 -0300 Thiago Santos * gst/multifile/gstmultifilesink.c: multifilesink: post file message on EOS When multifilesink is operating in any mode other than one file per buffer, the last file created won't have a file message posted as multifilesink doesn't handle the EOS event. This patch fixes it by using the last position to post a file message when EOS is received. This should ensure at least the time related data and the filename are posted to the application or other elements https://bugzilla.gnome.org/show_bug.cgi?id=747000 2015-04-03 18:57:50 +0100 Tim-Philipp Müller * autogen.sh: * common: Automatic update of common submodule From bc76a8b to c8fb372 2015-04-03 02:08:50 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Guard against 64-bit overflow For large-file atoms, guard against overflow in the size field, which could make us jump backward in the file and cause infinite loops. 2015-04-01 23:46:13 +1100 Jan Schmidt * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * tests/check/elements/qtmux.c: isomp4: Make non-seekable downstream an error in normal mode When not in fast-start or fragmented mode, we need to be able to rewrite the size of the mdat atom, or else the output just won't be playable - the mdat placeholder with size == 0 will cover the rest of the file, including any moov atom we write out. https://bugzilla.gnome.org/show_bug.cgi?id=708808 2014-03-15 15:23:01 +0100 Sebastian Rasmussen * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * tests/check/elements/rtp-payloading.c: rtph263pay/-depay: add framesize SDP attribute Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416 2014-03-15 13:33:56 +0100 Sebastian Rasmussen * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: rtpjpegpay/-depay: Remove incorrectly introduced framesize SDP attribute Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726415 2015-03-27 21:09:44 +0100 Peter Seiderer * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2src: device sequence/offset correction in case of renegotiation The v4l2 device restarts the sequence counter in case of streamoff/streamon, the GST offset values are supposed to increment strictly monotonic, so adjust the sequence counter/offset values in case of caps renegotiation. https://bugzilla.gnome.org/show_bug.cgi?id=745441 2014-11-14 14:18:51 +0100 Peter Seiderer * sys/v4l2/gstv4l2src.c: v4l2src: add frame loss detection In case of v4l2 driver filled offset/sequence values add frame loss detection (and write a warning message). Move offset meta data setting and frame loss checking after the timestamp adjustment code to get proper timestamps for the warning message. https://bugzilla.gnome.org/show_bug.cgi?id=745441 2014-11-14 13:48:51 +0100 Peter Seiderer * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2src.c: v4l2: use v4l2 capture device sequence counter Use the v4l2 capture device sequence counter for setting the GstBuffer offset/offset_end values. https://bugzilla.gnome.org/show_bug.cgi?id=745441 2015-03-30 13:12:35 +0200 Tobias Modschiedler * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: Ask the driver about its requirements for min_buffers before initiating buffer pool. If propose_allocation() had not been called yet, it was possible that the driver was not asked at all. In buffer pool: Consider minimum number of buffers requested by driver when setting config. https://bugzilla.gnome.org/show_bug.cgi?id=746834 2015-04-01 19:30:27 -0400 Olivier Crête * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: rtpvp8depay: Parse width/height/profile from keyframes This makes it possible to mux the result into a container such as matroska. https://bugzilla.gnome.org/show_bug.cgi?id=747208 2015-04-01 19:01:49 -0400 Olivier Crête * ext/vpx/gstvp8enc.c: vp8enc: Expose VP8 width/height limitations in the caps template The VP8 format specification (RFC 6386 section 18.1) specifies that the maximum size is 16383x16383. 2015-03-31 00:20:13 +1100 Jan Schmidt * gst/flv/gstflvdemux.c: flv: When passing seek event upstream, hold a ref. In case upstream can't handle the seek, make sure we keep a ref on the event to attempt to handle it ourselves. 2015-03-26 13:34:53 +0100 Guillaume Desmottes * gst/matroska/matroska-read-common.c: matroska: fix GValue leaks when parsing tags gst_tag_list_add_value() doesn't consume the GValue we pass to it so there is no point copying it. https://bugzilla.gnome.org/show_bug.cgi?id=746810 2015-03-23 20:58:25 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: resurrect some flow return handling https://bugzilla.gnome.org/show_bug.cgi?id=744572 2015-03-23 20:57:56 +0100 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: resurrect some flow return handling https://bugzilla.gnome.org/show_bug.cgi?id=744572 2015-03-23 20:56:41 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: resurrect some flow return handling https://bugzilla.gnome.org/show_bug.cgi?id=744572 2015-03-27 18:58:31 -0300 Thiago Santos * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-read-common.c: matroska: store stream tags and push as updated New tags can be found on different parts of the file, so this patch keeps the stream taglists around for the life cycle of the pad and adds those new tags as found. Then a new tag is found, the pad's is marked with a tags changed flag, making the element push a new tag event on the next check. Before this, we were sending only the newly found tags, as the element was losing its taglist when pushing the event. 2015-03-15 14:40:36 +0100 Ramiro Polla * gst/matroska/matroska-demux.c: matroskademux: send global tags incrementally Instead of sending only new tags once they are found, merge the taglist and send them incrementally. 2015-03-14 17:07:05 +0100 Ramiro Polla * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroskaparse: send global tags Global tags are already being read in matroskaparse, but they are not currently being sent. This patch makes global tags get sent incrementally whenever new ones are found. https://bugzilla.gnome.org/show_bug.cgi?id=746242 2015-02-03 10:18:58 +0530 Vineeth T M * gst/effectv/gstquark.c: quarktv: fix "planes" property range, a value of 0 is not allowed When planes property is set to 0, the pipeline executes in an infinite loop and never exits. Since planes must never be 0, set the minimum value in the property description to 1. https://bugzilla.gnome.org/show_bug.cgi?id=743906 2015-03-26 13:42:02 -0700 David Schleef * gst/wavparse/gstwavparse.c: wavparse: Fix up comments regarding DTS 2015-03-25 15:11:34 -0400 Nicolas Dufresne * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Fix segment in TCP mode It is expected that buffers are time-stamped with running time. Set a segment accordingly. In this case we pick 0,-1 as this is what udpsrc would do. Depayloaders will update the segment to reflect the playback position. https://bugzilla.gnome.org/show_bug.cgi?id=635701 2015-03-26 12:21:25 -0700 David Schleef * gst/wavparse/gstwavparse.c: wavparse: be more strict about typefinding DTS Code now matches comments. 2015-03-25 15:10:53 -0400 Nicolas Dufresne * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove useless function This function didn't do anything special, let's not use a function for that. 2015-03-20 13:03:09 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitter: Account for rtx_retry in overflow check As rtx_retry is part of the substraction, we need to take it into account, otherwise we may endup with a big value. 2015-03-24 23:15:15 +0000 Julien Isorce * sys/osxvideo/cocoawindow.m: osxvideosink: check for deprecated constants prior to OSX 10.10 cocoawindow.m:339:5: error: 'NSOpenGLPFAWindow' is deprecated: first deprecated in OS X 10.9 cocoawindow.m:576:7: error: 'NSOpenGLPFAFullScreen' is deprecated: first deprecated in OS X 10.6 cocoawindow.m:605:24: error: 'setFullScreen' is deprecated: first deprecated in OS X 10.7 2015-03-24 16:51:12 -0400 Nicolas Dufresne * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix seeking query The segment start/stop in the query is meant to represent the seekable portion of the stream. It does not match the segment start/stop. Instead export 0 to duration. 2015-03-24 16:18:53 +0100 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Only set caps once if they don't change Previously we were setting new caps with the same content for every H264 or AAC codec_data we found in the stream, spamming everything and causing renegotiations. 2015-03-24 12:46:19 +0100 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Don't create AAC/H264 caps without codec_data Instead delay creating the caps until we read the codec_data from the stream, or fail if we get normal data before the codec_data. AAC raw caps and H264 avc caps always need codec_data, setting caps on the pad without them is going to make negotiation fail most of the time. Even if we later set new caps with the codec_data, that's usually going to be too late. https://bugzilla.gnome.org/show_bug.cgi?id=746682 2015-03-24 15:39:22 +0100 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Fix indention 2015-03-22 13:23:44 +0200 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudio.h: osxaudio: Fix string format warning on 32-bit UInt32 (Darwin, not C99's uint32_t) is 'unsigned long' on 32-bit platforms. 2015-03-21 17:50:40 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: rtpsession: Fix another instance of sticky event misordering warnings Make sure that the sync_src pad has caps before the segment event. Otherwise we might get a segment event before caps from the receive RTCP pad, and then later when receiving RTCP packets will set caps. This will results in a sticky event misordering warning This fixes warnings in the rtpaux unit test but also in the rtpaux and rtx examples in tests/examples/rtp https://bugzilla.gnome.org/show_bug.cgi?id=746445 2015-03-21 17:18:47 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: rtpsession: Also start the RTCP send thread when receiving RTP or RTCP Before we only started it when either: - there is no send RTP stream or - we received an RTP packet for sending This could mean that if the send RTP pads are connected but never receive any RTP data, and the same session is also used for receiving RTP/RTCP, we would never start the RTCP thread and would never send RTCP for the receiving part of the session. This can be reproduced with a pipeline like: gst-launch-1.0 rtpbin name=rtpbin \ udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \ rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \ fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v Before this change the rtcp_fakesink would never send RTCP for the receiving part of the session (i.e. no receiver reports!), after the change it does. And before and after this change it would send RTCP for the receiving part of the session if the sender part was omitted (the last two lines). 2015-03-19 11:54:12 +0100 Sebastian Dröge * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Add support for buffer lists 2015-03-19 11:39:38 +0100 Sebastian Dröge * gst/rtpmanager/gstrtprtxqueue.c: rtprtxqueue: Implement support for buffer lists 2015-03-18 17:32:36 -0400 Nicolas Dufresne * gst/rtsp/gstrtspsrc.c: rtspsrc: Improve trace readability Change the command number into strings. 2015-01-20 10:18:56 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Don't repeatedly warn after no_more_pads (v2) This can get rather spammy for such a high log level. Only warn once per stream. https://bugzilla.gnome.org/show_bug.cgi?id=746274 2015-03-16 11:23:52 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: flvdemux: Introduce constant for no-more-pads threshold https://bugzilla.gnome.org/show_bug.cgi?id=746274 2015-01-20 10:18:29 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: flvdemux: Fix warning to contain 'video' https://bugzilla.gnome.org/show_bug.cgi?id=746274 2015-03-11 21:25:40 +0100 Nicola Murino * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: for dts only stream set pts=dts for intra only formats https://bugzilla.gnome.org/show_bug.cgi?id=745192 2015-03-14 16:39:09 +0100 Ramiro Polla * gst/matroska/matroska-demux.c: * gst/matroska/matroska-read-common.c: matroskademux: fix sending of tags * Fix critical when new tags are found after segment event has already been sent. * Send global tags before stream tags. * Split sending of tags out of gst_matroska_demux_send_event() into its own function. https://bugzilla.gnome.org/show_bug.cgi?id=745973 2015-03-13 18:26:06 +0000 Ramiro Polla * gst/rtsp/gstrtspsrc.c: rtspsrc: properly escape percent sign in documentation 2015-03-13 18:26:44 +0000 Ramiro Polla * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: properly escape percent sign in documentation 2015-03-13 18:48:03 +0000 Thiago Santos * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2src: delay renegotiation until it is likely buffers were reclaimed Allow renegotiation to happen when buffers have returned after an allocation query. As the allocation query is serialized, all buffers from the pool should have returned and we can stop it to create a new one for the new format https://bugzilla.gnome.org/show_bug.cgi?id=682770 2015-03-13 18:47:55 +0000 Thiago Santos * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: add gst_v4l2_object_try_format Similar to set_format but it uses TRY_FMT instead of S_FMT https://bugzilla.gnome.org/show_bug.cgi?id=682770 2015-03-13 18:38:42 +0000 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: fix crash with GST_DEBUG enabled g_inet_socket_address_get_address() does not give us a ref to the address, so don't unref it. 2015-03-12 13:49:56 +0000 Sebastian Dröge * gst/level/gstlevel.c: level: Don't read over the end of the input memory Previously we advanced the in_data pointer by bps for every channel, and then later again for block_size*bps. This caused us to be one sample further than expected if an input buffer covered two analysis frames. And in the end lead to completely bogus values reported by level. https://bugzilla.gnome.org/show_bug.cgi?id=746065 2015-03-12 01:37:08 +1100 Jan Schmidt * sys/oss/gstossdmabuffer.c: Remove a couple of superfluous trailing semi-colons 2015-03-10 09:31:20 +0000 Tim-Philipp Müller * gst/alpha/gstalpha.c: * gst/avi/gstavidemux.c: * gst/debugutils/gstpushfilesrc.c: * gst/isomp4/gstisoff.c: * gst/rtpmanager/rtpsession.c: * gst/udp/gstmultiudpsink.c: * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxcoreaudiocommon.c: Fix double semicolons 2015-03-10 15:46:40 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsrc.c: splitmux: Shut down element before downward state change Make sure the state change won't hang trying to shut down pads by making sure the streaming has stopped before chaining up. 2015-03-09 22:58:05 +0200 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudio.h: osxaudio: stream format is an SPDIF-only field 2015-03-09 22:53:41 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudiosrc.h: osxaudio: fix spaces 2015-03-09 22:52:46 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudiosrc.h: osxaudio: add type check macro 2015-03-09 22:51:51 +0200 Ilya Konstantinov * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiocommon.h: * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: rename gst_core_audio_set_channels_layout() to gst_core_audio_get_channel_layout(). 2015-03-09 22:30:28 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudioringbuffer.c: osxaudio: remove unused finalize 2015-03-09 16:25:43 +0000 Luis de Bethencourt * ext/vpx/gstvp9enc.c: vp9enc: remove duplicate declaration of function 2015-03-09 16:22:29 +0000 Luis de Bethencourt * gst/rtp/gstrtph264depay.c: rtph264depay: remove unused value CID #1226474 2015-03-09 16:14:34 +0000 Luis de Bethencourt * gst/rtp/gstrtph263pay.c: rtph263pay: fix leak CID 1212156 2015-03-09 15:58:33 +0000 Luis de Bethencourt * gst/rtp/gstrtph263pay.c: rtph263pay: remove uneeded variable We just need to save the ebit information in case there is an error decoding. 2015-03-09 16:46:02 +0100 Sebastian Dröge * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vp[89]enc: Reset the encoder when flushing https://bugzilla.gnome.org/show_bug.cgi?id=745704 2015-03-09 12:51:17 +0000 Luis de Bethencourt * gst/matroska/matroska-parse.c: matroska: error mode if can't push buffer If gst_pad_push() fails, inform and return flow error. 2015-03-09 12:13:34 +0000 Luis de Bethencourt * gst/matroska/matroska-parse.c: matroska: unused value Value set in ret will be overwritten just before exiting the function. CID #1226469 2015-03-09 11:10:35 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base These are outside the expected range of sequence numbers and should be clipped, especially for RTSP they might belong to packets from before a seek or a previous stream in general. 2014-02-27 10:52:16 +0100 Linus Svensson * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't include payload type in the caps for framesize When the sdp media attribute framesize are converted to caps the should not be included. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335 2015-03-09 10:05:14 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams 2015-03-09 11:24:58 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Make sure to filter caps in all cases during CAPS query We were skipping the filter step while returning template caps, for example. 2015-03-08 21:15:53 +0000 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't update buffer for OUTPUT For output device, we should not update the buffer with flags and timestamp when we dequeue. The information in the v4l2_buffer is not meaningful and it breaks the case where the buffer is rendered at multiple places. https://bugzilla.gnome.org/show_bug.cgi?id=745438 2015-03-08 18:04:34 +0100 Sebastian Dröge * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Implement cookies property 2015-03-08 18:02:51 +0100 Sebastian Dröge * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Implement automatic-redirect property 2015-03-08 17:54:07 +0100 Sebastian Dröge * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Implement proxy support The properties were there before, but not used anywhere. 2015-02-21 20:05:24 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: resurrect some flow return handling 2015-03-04 10:27:17 +0100 Nicolas Huet * gst/audioparsers/gstaacparse.c: aacparse: fix LOAS parsing issue Fix missing index in syncword searching https://bugzilla.gnome.org/show_bug.cgi?id=745585 2015-03-05 17:54:43 -0300 Thiago Santos * sys/directsound/gstdirectsoundsink.c: directsoundsink: fix modulo math with ringbuffer parameters To get a multiple of bpf use a subtraction and not an addition https://bugzilla.gnome.org/show_bug.cgi?id=745684 2015-03-07 00:55:47 +1100 Jan Schmidt * gst/multifile/gstsplitmuxsink.c: splitmuxsink: Protect property variables with the object lock. Use the object lock instead of the splitmux lock to protect internal property variables, so they're not locked when switching to a new file. https://bugzilla.gnome.org/show_bug.cgi?id=744420 2015-03-06 11:39:39 +0100 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: check: add jitterbuffer unit test See https://bugzilla.gnome.org/show_bug.cgi?id=745539 2015-03-05 09:18:52 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix handling of interleaved (TCP) streams We need to set up the transport in any case, not just if we have a container stream or a non-interleaved stream. Only if we have an interleaved stream and are retrying, we should not set up the stream again. https://bugzilla.gnome.org/show_bug.cgi?id=745599 2015-03-05 10:00:33 +0100 Sebastian Dröge * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: vp[89]dec: Drop frames that have no output buffer because of errors finish_frame() assumes that there is an output buffer. 2015-03-05 09:56:23 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't unref caps we don't own 2015-03-05 09:46:17 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Push RTCP caps on the RTCP pads Otherwise we will get not-negotiated later from rtpbin, and will never be able to send RTCP packets back to the server. Note that error flow returns from the RTCP pads are ignored, that's why it didn't fail more visible before. 2015-03-05 09:35:32 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Make sure to send SEGMENT events on all pads 2015-03-03 16:23:15 +0100 Santiago Carot-Nemesio * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.h: rtp: Add Full Intra Request (FIR) packets to statistics https://bugzilla.gnome.org/show_bug.cgi?id=745587 2015-03-03 16:01:53 +0100 Santiago Carot-Nemesio * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.h: rtp: Add Packet Loss Indication (PLI) to statistics This is helpful to provide statistics in the format defined in http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members. https://bugzilla.gnome.org/show_bug.cgi?id=745587 2015-03-03 19:19:50 +0100 Nicola Murino * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: Remove duration accumulation logic Duration accumulation can cause rounding errors and generate wrong duration with different buffers that share the same timestamp. https://bugzilla.gnome.org/show_bug.cgi?id=745192 2015-03-03 18:40:16 +0100 Nicola Murino * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroska: Add an helper method to get buffer timestamps ... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method that return PTS or DTS based on stream type. https://bugzilla.gnome.org/show_bug.cgi?id=745192 2015-03-04 11:28:12 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Add explanation why we have space for 32 hash tables And also create only one, there's no need yet to create all 32 until we implement RFC2762. 2015-03-04 11:26:57 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: Revert "rtpsession: Do not use an array of maps if they are not being used" This reverts commit 1591adf4cd843d13d8622a30c619425691a84128. https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1: It's the beginning of an implementation of RFC 2762, which is needed for large multicast groups. The implementation is not yet complete but why not leave what is there and implement RFC 2762 instead? 2015-03-04 10:35:12 +0100 Santiago Carot-Nemesio * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Do not use an array of maps if they are not being used rtpsession declares an array of maps to store srrcs but only the the key 0 is being used. This patch replaces the array of maps for just one map and remove useless parameters in rtpsession https://bugzilla.gnome.org/show_bug.cgi?id=745586 2015-02-27 18:12:09 +0900 Jimmy Ohn * gst/avi/gstavidemux.c: avidemux: remove not needed code In gst_avi_demux_handle_src_query, there is not needed code. We already check about stream is vbr or not at the upper line. o, we don't need to check this condition becase stream is not vbr 100% in this case. https://bugzilla.gnome.org/show_bug.cgi?id=745276 2015-03-03 23:25:35 +0000 Tim-Philipp Müller * tests/icles/gdkpixbufoverlay-test.c: tests: gdkpixbufoverlay-test: replace deprecated function Just avoid using the deprecated function entirely, it's easy enough. Defining the macro is not enough. 2015-03-03 19:04:48 +0000 Tim-Philipp Müller * tests/icles/gdkpixbufoverlay-test.c: tests: gdkpixbufoverlay-test: fix compilation against newer gdk-pixbuf gdk_pixbuf_new_from_inline() has been deprecated in favour of GResource. 2015-03-03 18:39:15 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosrc.c: osxaudiosrc: Allow caps renegotiation The ringbuffer does allow renegotiation, so we do not have to report fixed caps once it is acquired (based on a similar patch for the sink side by Ilya Konstantinov ). 2015-02-21 14:41:08 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Allow renegotiating caps Once osxaudiosink's device is open, it fixates on the initial caps and refuses to accept new caps. This is erroneous since the Audio Unit is can accept a new ASBD, and GstAudioRingBuffer supports reconfiguration as well. https://bugzilla.gnome.org/show_bug.cgi?id=743925 2015-03-02 12:04:00 +0100 Gwenole Beauchesne * sys/v4l2/gstv4l2bufferpool.c: v4l2allocator: fix fd leak in DMABUF import mode. Ensure gst_v4l2_buffer_pool_release_buffer() releases the associated GstV4l2MemoryGroup. In particular, this allows for closing the DMABUF handles prior to instantiating new ones. https://bugzilla.gnome.org/show_bug.cgi?id=745443 2015-03-02 15:06:09 +0100 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Use 0 as duration for the EOS "frame" 2015-03-02 15:02:20 +0100 Sebastian Dröge * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvp9enc.h: vp{8,9}enc: Tell the encoder about actual timestamps and durations of frames ... instead of just counting frames. The values are supposed to be in timebase units, not frame units. This fixes various quality problems with VP8/VP9 encoding and in general makes the encoder behave better. Thanks to Nirbheek Chauhan for noticing this bug. 2015-03-01 13:56:17 -0500 Nicolas Dufresne * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: vpxdec: Fix calculation of width in bytes Right now we only support I420, but vpx seems to support more formats. This will prevent hard to find bug in the future. 2015-03-01 13:52:50 -0500 Nicolas Dufresne * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: vpxdec: Don't memcpy in frame map failed This avoid a crash if mapping the frame failed. 2015-03-01 13:48:45 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Add missing break This is cosmetic change. 2015-03-01 13:46:18 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: Workaround driver not setting field correctly As it's very common, handle driver not setting field in buffers by using the field value from the format. This workaround a long time bug in UVC driver. For even buggier driver, we simply assume progressive as before. We also only warn once, to avoid spamming. 2015-02-28 18:10:06 +0100 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: fix key unit seek Unlike many other seek flags, the KEY_UNIT seek flag is not copied over into the GstSegment, since it's only relevant for the seek itself, so we need to pass it explicitly to the seek handler here. https://bugzilla.gnome.org/show_bug.cgi?id=745339 2015-02-27 09:38:01 +0100 Edward Hervey * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-wavenc.xml: docs/plugins: Updates 2015-02-26 23:41:47 +0100 Nicola Murino * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroskamux/demux: initialize dts_only https://bugzilla.gnome.org/show_bug.cgi?id=745192 2015-02-26 23:28:11 +0100 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: store DTS for V_MS/VFW/FOURCC streams https://bugzilla.gnome.org/show_bug.cgi?id=745192 2015-02-26 19:48:33 +0000 Tim-Philipp Müller * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsrc.c: multifile: attempt to fix docs build issue on build bot 2015-02-27 00:41:46 +0530 Arun Raghavan * gst/interleave/interleave.c: interleave: Drop custom latency query handling This is implemented by the default query handler now. 2015-02-27 00:40:05 +0530 Arun Raghavan * gst/videomixer/videomixer2.c: videomixer: Drop custom latency querying logic This is now implemented in the default latency query handler. 2015-02-26 16:10:41 +0100 Sebastian Rasmussen * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: fix payloader description and author e-mail https://bugzilla.gnome.org/show_bug.cgi?id=745226 2014-09-05 16:34:26 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: v4l2: query crop configuration after each call of S_CROP S_CROP ioctl is write-only and the device can adjust crop rectangle so we query back the crop configuration after each S_CROP to know what has been done. https://bugzilla.gnome.org/show_bug.cgi?id=736133 2015-02-26 02:12:18 +0100 Matej Knopp * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS When such stream is present demuxer should set DTS on buffers instead of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW streams. Sample file https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv https://bugzilla.gnome.org/show_bug.cgi?id=745192 2015-02-25 16:45:11 -0800 Aleix Conchillo Flaqué * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Check corruption flag on the right buffer We where checking the buffer we are copying to instead of the buffer we are copying from. https://bugzilla.gnome.org/show_bug.cgi?id=740040 2015-01-19 15:29:24 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: v4l2object: set colorspace in caps for capture devices This information is set by the driver for a capture device, and so could be forwarded to pipeline by setting the colorimetry in caps. https://bugzilla.gnome.org/show_bug.cgi?id=743186 2014-10-06 17:30:06 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2bufferpool: fix import_userptr() in single-planar API when n_planes > 1 In the V4L2 single-planar API, when format is semi-planar/planar, drivers expect the planes to be contiguous in memory. So this commit change the way we handle semi-planar/planar format (n_planes > 1) when we use the single-planar API (group->n_mem == 1). To check that planes are contiguous and have expected size, ie: no padding. We test the fact that plane 'i' start address + plane 'i' expected size equals to plane 'i + 1' start address. If not, we return in error. Math are done in bufferpool rather than in allocator because the former is aware of video info. https://bugzilla.gnome.org/show_bug.cgi?id=738013 2015-01-23 10:15:46 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: v4l2allocator: let bufferpool calculate image size when importing userptr Offset are relative to the buffer and there is no guarantee substracting them will give us the plane size. So we let bufferpool make the math as it is more aware of video info than allocator and pass a size array to allocator import function. Pointed out by Nicolas Dufresne https://bugzilla.gnome.org/show_bug.cgi?id=738013 2014-12-11 16:13:15 +0100 Philippe De Muyter * sys/v4l2/gstv4l2object.c: v4l2object: recognize and distinguish all bayer arrangements Up to now, v4l2src recognized only "bggr" amongst the bayer arrangements. Recognize now also the "rggb", "gbrg" and "grbg" arrangements. https://bugzilla.gnome.org/show_bug.cgi?id=742363 2015-01-15 16:11:53 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: set v4l2_buffer.field when queuing buffer in an output device According to the current specification, application must set this field for an output device. https://bugzilla.gnome.org/show_bug.cgi?id=743013 2015-02-24 05:57:24 +0200 Ilya Konstantinov * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiocommon.h: osxaudiosrc: iOS resampling causes stuttering Fixes stuttering audio when iOS AU is resampling. To make AU resample, one has to request a rate that differs from AVAudioSession's sampleRate. The resampling itself is not the culprit, but rather our API misuse. AudioUnitRender modifies the mDataByteSize members with the actual read bytes count. Therefore, they must be reinitialized before each AudioUnitRender. (The buffers themselves can be preallocated.) The "stutter" was caused by one AudioUnitRender making the buffer too small for other AudioUnitRender invocations, making them fail with -50 (paramErr). By way of luck, when AU didn't resample, all AudioUnitRender invocations read the same number of bytes. (This patch addresses some non-interleaved audio concerns, but at this moment the elements do not support non-interleaved audio and non-interleaved is untested.) https://bugzilla.gnome.org/show_bug.cgi?id=744922 2015-02-22 01:49:52 +0100 Krzysztof Kotlenga * gst/rtsp/gstrtspsrc.c: rtspsrc: improve error message when unauthorized Make use of NOT_AUTHORIZED error code instead of falling back to generic READ error. https://bugzilla.gnome.org/show_bug.cgi?id=601733 2015-02-23 20:06:25 +0000 Tim-Philipp Müller * sys/ximage/ximageutil.c: ximagesrc: remove pointless g_return_val_if_fail() ximage won't ever be NULL here because the dispose function is called via ximage->dispose(). 2015-02-23 19:40:25 +0100 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: All segment resulting from a seek should have the same seqnum https://bugzilla.gnome.org/show_bug.cgi?id=744983 2015-02-19 23:12:31 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: v4l2: Enable copy when no known allocation params When there is no allocation parameters in the query, enable copy threshold. When this threshold is reached, the buffer pool will start copying when the pool reaches a critical level. If the driver supports CREATE_BUFS, this will be used instead. 2015-02-19 23:08:34 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Update allocator flags When we hit emulated formats, we disable CREATE_BUFS since libv4l2 cope very badly with it. Also clear the allocator flags so we will never try to allocate more buffers. This fixes failure when the copy threshold is reached as we where calling CREATE_BUFS, which lead to libv4l2 instability. 2015-02-19 23:07:23 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Use specific debug category The pool has grown enough that it is now handy to seperate v4l2object trace from v4l2bufferpool trace. 2015-02-19 14:29:02 +0000 Vincent Penquerc'h * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: default encoding name to VP8 https://bugzilla.gnome.org/show_bug.cgi?id=737810 2015-02-19 14:06:51 +0000 Vincent Penquerc'h * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: make caps writable before truncating them https://bugzilla.gnome.org/show_bug.cgi?id=737810 2015-02-05 10:29:26 +0000 Vincent Penquerc'h * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: negotiate encoding name Chrome uses a different one than gstreamer. https://bugzilla.gnome.org/show_bug.cgi?id=737810 2015-02-19 12:35:07 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing Otherwise we will just send buffers on the pad without any events beforehand and will get g_warnings() about that. 2015-02-19 11:20:51 +0000 Luis de Bethencourt * ext/jack/gstjackaudiosrc.c: jack: case missing break statement commit b1098c2ea5eabea7af08ce51d22b867eaed2bbe2 added a new case in gst_jack_audio_src_get_property() but forgot to add the break statement to it. 2015-02-18 19:18:00 +0000 Luis de Bethencourt * sys/v4l2/v4l2_calls.c: Revert "v4l2: fraction is reversed" This reverts commit b91fe36644b15ae070d72b9e8a9c7087e82aef12. 2015-02-18 17:49:29 +0000 Luis de Bethencourt * sys/v4l2/v4l2_calls.c: v4l2: fraction is reversed In the fraction 1 / 2. 1 is the numerator and 2 is the denominator. The arguments of fraction gst_value_set_fractions() are value, numerator and denominator. Also, gst_value_set_fraction() fails if denominator is 0 for obvious reasons. 2015-02-17 20:26:55 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2pool: Deactivate other pool When importing buffers from a downstream pool, we need to deactivate that pool to ensure it will be usable again later. Relying on the refcount to reach zero does not work, since elements like xvimagesink keeps a reference on their proposed pool. 2015-02-18 10:10:53 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: qtmux: remove not needed condition gst_buffer_replace can handle NULL inputs by itself 2015-02-18 09:40:14 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate The tfdt should be more accurate as the buffer timestamp is provided by the fragmented format manifest and it might just be an approximation. 2015-02-17 16:57:55 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will confuse downstream with buffers that come before such events. 2015-02-17 12:20:57 +0100 hark * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: Add property port-pattern to specify which JACK ports to connect to https://bugzilla.gnome.org/show_bug.cgi?id=690719 2015-02-17 12:31:06 +0100 Edward Hervey * gst/isomp4/gstisoff.c: * gst/isomp4/gstisoff.h: * gst/isomp4/qtdemux.c: isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_ We need different symbol names, because these symbols are also present in the fragmented plugin ... which will cause conflicts when doing static linking 2015-02-16 14:31:05 +0000 Luis de Bethencourt * gst/goom2k1/lines.c: goom2k1: use fractional part of float division 2015-02-16 13:59:14 +0000 Luis de Bethencourt * gst/multifile/gstsplitmuxsink.c: splitmuxsin: remove dead code Every instance of goto beach has buf_info equal NULL. Don't check for a condition that never happens. CID #1268399 2015-02-15 21:45:24 -0500 Nicolas Dufresne * tests/check/elements/splitmux.c: splitmux-test: Parse error message The test had a function to print the error, but was not parsing it. This was causing warning about dbg_info being used uninitialized. If the test was testing any errors, this would have crashed. 2015-02-15 21:34:28 -0500 Nicolas Dufresne * gst/spectrum/gstspectrum.c: spectrum: Fix min and max for bands property The number of FFTs is calculated with the following formula: guint nfft = 2 * bands - 2; nfft is passed to gst_fft_f32_new() as the len argument and is of type unsigned integer. This method required that len is at leas 1, then maximum G_MAXINT, as other values would be negative. If we extrapolate from the formula above it means we need "bands" to be between 2 and ((guint)G_MAXINT + 2) / 2). https://bugzilla.gnome.org/show_bug.cgi?id=744213 2015-02-15 15:51:55 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix freeing of shared memory When memory (that has been shared using gst_memory_share()) are freed, the memory (or the DMABUF FD) should not bee freed. These memories have a parent. This also removes the extra _v4l2mem_free function and avoid calling close twice on the DMABUF FD. https://bugzilla.gnome.org/show_bug.cgi?id=744573 2015-02-14 11:11:30 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: do not use sparse streams in push-based seeking Using the sparse streams can make the push-based seeking return too far in the stream. It also can lead to issues as the sparse streams will be ignored when restarting playback and, if the sparse stream is the one that has the earliest sample, it will confuse qtdemux's offsets as one stream will have an earlier offset than the demuxer's one which might lead to early EOS. https://bugzilla.gnome.org/show_bug.cgi?id=742661 2015-02-13 19:43:16 +0900 Jimmy Ohn * ext/pulse/pulsesink.c: pulsesink: Enhance code readability in pulsesink_query In pulsesink_query function, we use a switch for the query type. In the CAPS case, there is no 'break', instead we return right away. Use a break and return at the end of the function instead for better code readability. https://bugzilla.gnome.org/show_bug.cgi?id=744461 2015-02-13 20:40:48 +0000 Tim-Philipp Müller * gst/multifile/gstsplitmuxsink.c: splitmuxsink: flag as sink from the start 2015-02-11 15:30:44 +0100 Philippe Normand * gst/isomp4/Makefile.am: * gst/isomp4/fourcc.h: * gst/isomp4/gstisoff.c: * gst/isomp4/gstisoff.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Initial 'sidx' atom parsing support Parse the 'sidx' atom and update the total duration according to the parser result. The isoff parser code is imported from gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data() function was factored out of the gst_isoff_sidx_parser_add_buffer() function. https://bugzilla.gnome.org/show_bug.cgi?id=743578 2015-02-11 05:06:45 +1100 Jan Schmidt * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: flvdemux: Use gst_video_guess_framerate() Use gst_video_guess_framerate() from libgstvideo to guess sensible common framerates where possible from the floating point fps in the stream. 2015-02-11 13:53:02 +0100 Sebastian Dröge * ext/raw1394/gstdv1394src.c: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: * gst/interleave/interleave.c: * gst/rtsp/gstrtpdec.c: * gst/videomixer/videomixer2.c: Improve and fix LATENCY query handling This now follows the design docs everywhere, especially the maximum latency handling. https://bugzilla.gnome.org/show_bug.cgi?id=744106 2015-02-11 10:29:55 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Handle first RTCP packet and early feedback correctly According to RFC 4585 section 3.5.3 step 1 we are not allowed to send an early RTCP packet for the very first one. It must be a regular one. Also make sure to not use last_rtcp_send_time in any calculations until we actually sent an RTCP packet already. In specific this means that we must not use it for forward reconsideration of the current RTCP send time. Instead we don't do any forward reconsideration for the first RTCP packet. 2015-02-10 18:53:53 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: fix compilation with gcc 5.0 2015-02-10 16:00:07 +0000 Tim-Philipp Müller * gst/multifile/gstsplitmuxsink.c: splitmuxsink: fix example pipeline properly x264enc might not have a max-key-int property, but it has a key-int-max property... 2015-02-10 14:57:55 +0000 Luis de Bethencourt * gst/multifile/gstsplitmuxsrc.c: splitmux: fix typo 2015-02-10 14:56:23 +0000 Luis de Bethencourt * gst/multifile/gstsplitmuxsink.c: splitmux: update example pipeline Element x264enc doesn't have a max-key-int property 2015-02-10 13:29:32 +0000 Luis de Bethencourt * gst/multifile/gstsplitmuxsink.c: splitmux: fix memory leak If execution goes to the beach in line 981, buf_info goes out of scope without the memory being free'd. Handle this case. CID #1268403 2015-02-08 12:03:10 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: fix awkward if clause 2015-02-07 01:41:49 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxsink.c: * tests/check/elements/splitmux.c: splitmux: Add unit test for file splitting Add a unit test for file splitting, and fix the leaks in the splitmuxsink it found 2015-02-06 14:43:22 +0000 Luis de Bethencourt * gst/wavparse/gstwavparse.c: wavparse: fix which stop variable is used in assignment Assignment is done to variable segment.stop when the intention was to assign to local variable stop. Instead of overwriting it, the value is now clamped and segment.stop is set to it soon after. CID #1265773 2015-02-07 00:19:36 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxsrc.c: * tests/check/elements/splitmux.c: splitmux: Fix memory leaks until the test valgrinds clean 2015-02-06 06:42:17 +1100 Jan Schmidt * gst/multifile/gstsplitmuxpartreader.c: splitmux: Handle early EOS during part preparation Handle the case where a short file reaches EOS while we're still waiting for no-more-pads, and make sure we continue to the internal READY state for real playback to work properly later. 2015-02-06 05:03:19 +1100 Jan Schmidt * tests/files/splitvideo00.ogg: * tests/files/splitvideo01.ogg: * tests/files/splitvideo02.ogg: tests: Change splitmux test video files Avoid test failure by changing the stored video resolution from 80x60 to 80x64, which needs bug 741030 to be fixed. 2014-08-01 00:07:53 +1000 Jan Schmidt * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * gst/multifile/Makefile.am: * gst/multifile/gstmultifile.c: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitmuxpartreader.c: * gst/multifile/gstsplitmuxpartreader.h: * gst/multifile/gstsplitmuxsink.c: * gst/multifile/gstsplitmuxsink.h: * gst/multifile/gstsplitmuxsrc.c: * gst/multifile/gstsplitmuxsrc.h: * gst/multifile/gstsplitutils.c: * gst/multifile/gstsplitutils.h: * gst/multifile/test-splitmuxpartreader.c: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/splitmux.c: * tests/files/splitvideo00.ogg: * tests/files/splitvideo01.ogg: * tests/files/splitvideo02.ogg: splitmux: Implement new elements for splitting files at mux level. Implement 2 new elements - splitmuxsink and splitmuxsrc. splitmuxsink is a bin which wraps a muxer and takes 1 video stream, plus audio/subtitle streams, and starts a new file whenever necessary to avoid overrunning a threshold of either bytes or time. New files are started at a keyframe, and corresponding audio and subtitle streams are split at packet boundaries to match video GOP timestamps. splitmuxsrc is a corresponding source element which handles the splitmux:// URL and plays back all component files, reconstructing the original elementary streams as it goes. 2015-02-04 16:32:14 -0300 Thiago Santos * tests/check/elements/souphttpsrc.c: * tests/files/test-cert.pem: * tests/files/test-key.pem: tests: souphttpsrc: update ssl key/cert pair Our ones were expired. The new ones were copied from libsoup's tests files. Also sets the property to use our own cert to validate the server, otherwise the default system certs would be used and it would fail. 2015-02-04 02:25:44 -0300 Thiago Santos * gst/rtp/gstrtph264depay.c: rtph264depay: prevent trying to get 0 bytes from adapter This causes an assertion and would lead to getting a NULL instead of a buffer. Without proper checking this would easily lead to a segfault https://bugzilla.gnome.org/show_bug.cgi?id=737199 2015-02-04 21:50:51 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Simple implementation of GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS When the trickmode key-units flag is set on the segment, simply skip any sample on a video stream that isn't a keyframe 2015-02-03 17:35:52 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix container handling We detect a container correctly now so we need to revert the weird check there was before. Use gst_rtspsrc_stream_push_event() to push the caps event on the right pad. See https://bugzilla.gnome.org/show_bug.cgi?id=739391 2015-02-02 19:46:27 -0300 Thiago Santos * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: store and write stream tags Separate global from stream tags storage and write them to the appropriate tags entry in the output 2015-02-02 13:35:59 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: parse stream tags Keep global and stream tags separately and parse the udta node that can be found under the trak atom. The udta will contain stream specific tags and will be pushed as such https://bugzilla.gnome.org/show_bug.cgi?id=692473 2015-01-31 14:32:34 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: store stream and container tags separately Tags received via events, when marked as stream tags, will be stored on that stream's trak atom instead of being stored in the main tags atom. This allows the resulting file to have global and stream tags stored. https://bugzilla.gnome.org/show_bug.cgi?id=692473 2015-01-31 13:14:44 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: qtmux: refactor tags functions to accomodata UDTA at trak level Refactor the functions that were bound to the 'moov' atom to directly pass the desired 'udta' that should receive the tags. This allows the tags to be written to 'udta' at the 'moov' or the 'trak' level, creating tags that are for the container or for a stream only. https://bugzilla.gnome.org/show_bug.cgi?id=692473 2015-01-31 10:47:40 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: map application name to _swr tag It refers to the application name and version used to create the file https://bugzilla.gnome.org/show_bug.cgi?id=692473 2015-01-31 02:30:40 +1100 Jan Schmidt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: Fix seeking past the end of the file in reverse mode. Snap to the end of the file when seeking past the end in reverse mode, and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling for the stop position by always seeking on a segment in stream time 2015-01-30 18:22:31 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Fix signal name This wasn't meant to be pushed at all yet, but now that it's there already it won't hurt to make it correct at least. 2015-01-30 16:56:35 +0100 Sebastian Dröge * gst/rtpmanager/rtpstats.h: rtpstats: Fix typo in documentation 2015-01-30 16:50:36 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Add new on-receiving-rtcp signal This will be emitted whenever an RTCP packet is received. Different to on-feedback-rtcp, this signal gets every complete RTCP packet and not just the individual feedback packets. 2015-01-28 14:02:15 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: simplify segment.base math Remove a fix for heavily edited files added for fixing https://bugzilla.gnome.org/show_bug.cgi?id=345830 to work with seeks and proper gaps playback. The fix was replaced for a more general solution that bases on using previous segment's duration, just like it works for media segments playback. https://bugzilla.gnome.org/show_bug.cgi?id=743518 2015-01-27 14:00:35 +0000 Luis de Bethencourt * gst/videomixer/videomixerorc-dist.c: videomixer: update orc files 2015-01-26 17:08:12 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: Fix data dropping for fragmented streams For fragmented streams with extra data at the end of the mdat qtdemux was not dropping those bytes and would try to use that extra data as the beginning of a new atom, causing the stream to fail. https://bugzilla.gnome.org/show_bug.cgi?id=743407 2015-01-25 17:30:33 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Deprecate rtcp-immediate-feedback-threshold property It had no effect since quite some time and also is not needed in general, especially not to switch between immediate feedback mode and early feedback mode. The latest understanding of the RFC is that from the endpoint point of view, both modes are exactly the same. RTCP is only allowed to use the bandwidth as given by the RFC constraints, as such it is only ever possible to schedule a RTCP packet early but it's against the RFC to schedule more RTCP packets. The difference between immediate feedback mode and early feedback mode is that the former guarantees that an RTCP packet can be sent for every event "immediately", which means that the bandwidth calculations from the RFC have resulted in an RTCP scheduling interval that is small enough. Early feedback mode on the other hand means that we can schedule some packets early to make that happen, but it's not guaranteed at all that it's possible to schedule an RTCP packet per event (i.e. they need to be accumulated or dropped). 2015-01-22 10:29:39 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Delay the next regular RTCP packet after early RTCP This is required to not exceed the short term average RTCP bitrate when using early feedback as compared to without early feedback. 2015-01-22 10:28:52 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Add new send-rtcp-full signal This indicates with a boolean return value if scheduling a new RTCP packet within the requested delay was possible. Otherwise it behaves exactly like send-rtcp. The only reason for adding a new signal is ABI compatibility. 2015-01-20 00:32:00 +0000 Jimmy Ohn * ext/pulse/pulsesink.c: pulsesink: Free format_info in query_getcaps If we can not create probe stream in query_getcaps function, it will appear memory leakage from format info. The following patch prevent memory leakage in pulsesink. https://bugzilla.gnome.org/show_bug.cgi?id=743178 2015-01-23 17:35:51 +0000 Luis de Bethencourt * gst/matroska/matroska-read-common.c: matroskademux: remove unnecessary check No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the flow is OK or not, the check there will be a break from the switch. Removing the check since the outcome is the same. CID #1265762 2015-01-23 15:16:25 +0100 Edward Hervey * gst/matroska/matroska-mux.c: matroskamux: Avoid using freed variable the name variable might have been attributed to pad_name, make sure we free it only *after* pad_name has been used. Coverity CID : 1265774 2015-01-23 15:13:55 +0100 Edward Hervey * gst/avi/gstavimux.c: avimux: Avoid using freed variable the name variable might have been attributed to pad_name, make sure we free it only *after* pad_name has been used. Coverity CID : 1265775 2014-11-14 12:59:31 +0100 Peter Seiderer * sys/v4l2/gstv4l2object.c: v4l2object: reuse caps framerate if not overwritten by v4l2 device Enables duration setting in v4l2src. https://bugzilla.gnome.org/show_bug.cgi?id=740403 2015-01-22 10:29:24 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: rtpsession: Fix indention 2015-01-21 17:36:26 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: qtdemux_dump: Bypass even more code if debugging is disabled And avoid using variables that won't exist when debugging is disabled 2015-01-21 15:30:33 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: qtdemux: Only traverse/dump nodes if guaranteed to be used __gst_debug_min is the "global" lowest debug level set. There's no guarantee the qtdemux debug category is actually set at that level. 2014-12-20 17:09:14 +0100 Edward Hervey * gst/matroska/ebml-read.c: matroska: Avoid debugging below category threshold This part alone was what made the matroska thread take a full core on an android phone ... 2015-01-21 09:56:41 +0100 Sebastian Dröge * ext/twolame/gsttwolamemp2enc.c: Constify some static arrays everywhere 2015-01-21 09:56:41 +0100 Sebastian Dröge * ext/lame/gstlamemp3enc.c: Constify some static arrays everywhere 2015-01-21 09:55:30 +0100 Sebastian Dröge * ext/dv/gstsmptetimecode.c: * ext/mikmod/mikmod_types.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiopanorama.c: * gst/effectv/gstradioac.c: * gst/isomp4/atoms.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: * gst/rtsp/gstrtspsrc.c: * gst/videofilter/gstvideotemplate.c: * gst/wavparse/gstwavparse.c: Constify some static arrays everywhere 2015-01-19 17:49:54 +0000 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: fix deadlock seeking in files without seek entries A mutex unlock was missing. https://bugzilla.gnome.org/show_bug.cgi?id=739975 2015-01-19 12:34:25 +0000 Vincent Penquerc'h * gst/videomixer/blend.c: videomixer: fix illegal memory access in blend function with negative ypos https://bugzilla.gnome.org/show_bug.cgi?id=741115 2015-01-13 16:49:34 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Proxy getcaps Replace the sink_query with new getcaps() virtual and use the proxy helper with the probed caps. This allow upstream element taking decision base on what is supported downstream. 2015-01-13 19:05:20 +0100 Sebastian Dröge * gst/isomp4/fourcc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Add support for v210 2015-01-13 18:58:01 +0100 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: v210 is v210, not UYVY and yuv2 is YUY2, not I420 Also add a few other raw video formats we support: v308, v216 and add comments for a few others we don't support yet. https://developer.apple.com/library/mac/technotes/tn2162/ 2015-01-12 15:56:29 +0100 Stefan Sauer * common: Automatic update of common submodule From f2c6b95 to bc76a8b 2015-01-10 15:51:16 +0100 Sebastian Dröge * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: Disable hack for NSApp iteration with a special #define The hack causes deadlocks and other interesting problems and it really can only be fixed properly inside GLib. We will include a patch for GLib in our builds for now that handles this, and hopefully at some point GLib will also merge a proper solution. A proper solution would first require to refactor the polling in GMainContext to only provide a single fd, e.g. via epoll/kqueue or a thread like the one added by our patch. Then this single fd could be retrieved from the GMainContext and directly integrated into a NSRunLoop. https://bugzilla.gnome.org/show_bug.cgi?id=741450 https://bugzilla.gnome.org/show_bug.cgi?id=704374 2015-01-08 21:07:05 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: uncork if needed upon commit ... to provide for a running clock. 2015-01-09 16:59:53 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Prevent renegotiation Renegotiation isn't supported, simply prevent it the way we do in v4l2src. 2015-01-06 13:54:25 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Don't unlock the stream lock twice 2015-01-09 11:40:40 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: fix stream time conversion Use the right macro to convert to the correct scale or the segment information will be wrong https://bugzilla.gnome.org/show_bug.cgi?id=742572 2015-01-07 18:48:58 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Add protection against driver bug v4l2loopback driver has a this nasty bug that if the queue is larger then 2 buffers, it returns random index on dqbuf. So far we assumed that the index was always right, which would lead to memory being unref twice, and eventually crash. 2015-01-07 17:58:05 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: v4l2: Don't use allocator size to iterate As the buffer array is fixed size and small, it's safer to simply use this static size to cleanup the buffers. This is also more consistent with the rest. The associated method is no longer required and can be dropped. 2015-01-07 17:55:14 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't clean buffer array in dispose This should already have been done, plus this code is incorrect and may lead to crash. https://bugzilla.gnome.org/show_bug.cgi?id=742074 2015-01-07 17:48:31 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't ref queued output buffer This partly revert to the old 1.2 behavior. Instead of keeping a reference to the output buffer queued, we simply release them but don't forward it to GstBufferPool. This way, the buffer pool don't need to be flushed to be stopped. https://bugzilla.gnome.org/show_bug.cgi?id=742074 2015-01-08 11:37:23 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Never fail on streamoff Failing streamoff prevents allocator from being disposed hence lead to device FD leak. There is no known cases where streamoff may fails for which we'd still be streaming. streamoff is known to fail when a device is being unplugged (in which case errno 19/ENODEV is set). https://bugzilla.gnome.org/show_bug.cgi?id=732734 2015-01-07 21:52:17 -0500 Brad Smith * configure.ac: v4l2: Add support for detecting the presence of V4L2 support on OpenBSD https://bugzilla.gnome.org/review?bug=742503 2015-01-04 15:57:10 +0100 Matej Knopp * gst/audioparsers/gstac3parse.c: ac3parse: request at least 8 bytes to properly parse header https://bugzilla.gnome.org/show_bug.cgi?id=742325 2015-01-07 16:20:03 -0800 Michael Smith * gst/wavparse/gstwavparse.c: wavparse: skip an additional uninteresting chunk type before the fmt chunk. 2015-01-07 18:16:12 +0000 Luis de Bethencourt * gst/audiofx/audiodynamic.c: audiodynamic: assert func_index is inside bounds Bringing back the check removed in the previous commit but have that check be a g_assert. Changing the function to static void since return can never be False, because audio format will never be unkown. 2015-01-07 17:31:39 +0000 Luis de Bethencourt * gst/audiofx/audiodynamic.c: audiodynamic: remove always-true conditional func_index is set by the sum of three ternary operators which add, 0:4, 0:2, and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7. The conditional checking if func_index is >= 0 and < 8 will always be true. Removing it. CID 1226442 2015-01-07 18:05:18 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: If we get a gap with a buffer without DTS, error out We (currently?) can't really handle gaps between RTP packets if they're not properly timestamped. The current code would go into calculations with GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably better to error out cleanly instead. 2014-11-21 11:39:19 -0800 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: set PLAYING state after configuring caps We set to PLAYING after we have configured the caps, otherwise we might end up calling request_key (with SRTP) while caps are still being configured, ending in a crash. https://bugzilla.gnome.org/show_bug.cgi?id=740505 2014-12-30 18:03:22 +0000 Tim-Philipp Müller * tests/icles/gdkpixbufoverlay-test.c: tests: gdkpixbufoverlay-test: remove outdated FIXME 2014-12-30 17:19:42 +0000 Tim-Philipp Müller * tests/check/elements/rtpcollision.c: tests: rtpcollision: use alawenc/dec in these tests instead of Speex They should always be built, while the speex elements are not. Need to check for a smaller number of buffers then (7->4) because speexenc will add 3 header buffers while alawenc will just output as many buffers as it receives as input. https://bugzilla.gnome.org/show_bug.cgi?id=742098 2014-12-30 16:36:02 +0000 Tim-Philipp Müller * tests/check/pipelines/simple-launch-lines.c: tests: simple-launch-lines: only run jpeg/png tests if elements are available 2014-12-30 16:26:58 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't return a buffer when returning not GST_FLOW_OK basesrc assumes that we don't return a buffer if something else than OK is returned. It will just leak any buffer we might accidentially provide here. This can potentially happen during flushing. Maybe fixes https://bugzilla.gnome.org/show_bug.cgi?id=741993 2014-12-30 14:52:42 +0000 Tim-Philipp Müller * tests/check/elements/rtpaux.c: tests: rtpaux: use alawenc/dec in these tests instead of Speex They should always be built, while the speex elements are not. https://bugzilla.gnome.org/show_bug.cgi?id=742098 2014-12-29 15:35:19 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Improve detection of being stuck at the same offset Only error out if we read from the same position again and got the same length. Just the same position is not necessarily enough. 2014-12-29 15:00:02 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't get stuck at the same offset when searching for clusters This could happen if there is an invalid cluster with size 0, and in that case just error out instead of looping forever. 2014-12-25 21:32:40 +0000 Tim-Philipp Müller * gst/isomp4/gstqtmux.c: qtmux: fix ALAC muxing Actually copy the codec data instead of copying nothing and then bombing out because there's no data. Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink https://bugzilla.gnome.org/show_bug.cgi?id=741783 2014-12-25 15:48:04 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: just drop invalid rtp packets instead of erroring out Apparently linphone sends an invalid RTP packet as very first packet. We want to ignore that instead of erroring out (same for any other invalid packets really). https://bugzilla.gnome.org/show_bug.cgi?id=741398 2014-12-25 15:44:15 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: fix 0.10-ism in docs 2014-12-25 14:58:12 +0000 Tim-Philipp Müller * tests/icles/gdkpixbufoverlay-test.c: tests: gdkpixbufoverlay-test: use absolute positioning to fix demo https://bugzilla.gnome.org/show_bug.cgi?id=739566 2014-12-25 14:53:09 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: gdkpixbufoverlay: add "positioning-mode" property to allow absolute positions Set positioning-mode=pixels-absolute to allow positioning with absolute coordinates, meaning negative x/y offsets will be interpreted as being to the left/above the video frame instead of being interpreted as relative to the right/bottom edge of the video frame (which is a silly default, but that's how it is). This means we can nicely slide images into and out of the frame, see gdkpixbufoverlay-test. https://bugzilla.gnome.org/show_bug.cgi?id=739566 2014-12-22 15:33:51 +0100 Sebastian Dröge * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: osxaudio: Directly return the ringbuffer's caps if it is acquired 2014-12-22 12:56:19 +0100 Sebastian Dröge * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: osxaudio: Put all audio formats into the template caps We report the proper caps later from the get_caps() vfunc implementation after probing the selected device. 2014-12-22 12:56:05 +0100 Sebastian Dröge * sys/osxaudio/gstosxaudioringbuffer.c: osxaudio: Also set the big endian flag for floating point samples 2014-12-22 11:45:59 +0100 Sebastian Dröge * MAINTAINERS: MAINTAINERS: Update my mail address 2014-12-22 10:23:01 +0100 Sebastian Dröge * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: osxaudio: Fix deadlock and property change notification in device selection code After creating the ringbuffer we have to set the device on the ringbuffer as it defaults to kAudioDeviceUnknown. At this point it can't have changed to anything else yet and we don't have to notify about changes to the sink/src "device" property. It's also not a good idea because GstAudioBaseSrc has the object lock taken while the ringbuffer is created, which might cause a deadlock if something calls back into the element from "notify::device". Once the base class is done with the NULL_TO_READY state change, it has opened the device via the ringbuffer and this might have chosen a different device. Especially if we initially used kAudioDeviceUnknown. Also notify about this property change as initially intended by this code. 2014-12-19 12:30:03 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2pool: Update configuration size We already update our copy of VideoInfo.size to proper size, now also the configuration so the size matches on release. https://bugzilla.gnome.org/show_bug.cgi?id=741420 2014-12-19 10:57:29 +0100 Edward Hervey * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroska-demux: Cache upstream length Instead of constantly querying upstream, just cache the last duration, and in the unlikelyness we might have gone over query again before deciding we are EOS. Cut 15% cpu off matroskademux streaming thread (srsly...) 2014-12-17 17:36:18 +0000 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroska: mux/demux the OpusHead header This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while it is marked as a draft, this part was confirmed to be correct on IRC), and allows one to determine whether a demuxed stream is multistream or not, and thus set the multistream caps field accordingly. In turn, this means downstream does not have to guess. https://bugzilla.gnome.org/show_bug.cgi?id=740744 2014-12-18 11:50:33 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found CID 1258717 2014-12-18 10:53:39 +0100 Sebastian Dröge * common: Automatic update of common submodule From ef1ffdc to f2c6b95 2014-12-12 23:06:07 +0000 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: udpsink: allocate scratch space for render functions on the heap and not the stack. Our allocations could get a bit too large to be sure it's not going to cause trouble using the stack. 2014-06-24 01:16:37 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: re-use send_buffers() code path for render() function It's like rendering a buffer list, just with one buffer. Has the added advantage that if there are multiple clients we can send the buffer to all the clients in one go. 2014-06-24 01:15:25 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: keep client list consistent during removals We unlock and re-lock the client lock while emitting the removed signal, which causes inconsistencies in the client list vs. the client counts. Instead, remove the client from the list already before emitting the signal and put it into a temporary list of clients to be removed. That way things look consistent to the streaming thread, but signal callbacks can still do things like get stats from removed clients. 2014-06-24 00:56:27 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: fix client count after removal 2014-06-23 18:43:21 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: keep client list sorted by socket family We make use of in the send_buffers() function if we need to use different sockets to send to IPv4 and IPv6 destinations. 2014-06-20 11:36:19 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: add sendmmsg-ready render_list function prototype Add prototype for a render_list() function that can use a sendmmsg-style g_socket_send_messages() function once it lands in GLib. We can use this infrastructure to send multiple buffers made up by multiple memories to multiple clients in one go, which drastically reduces the number of syscalls made when sending high-bitrate video streams. https://bugzilla.gnome.org/show_bug.cgi?id=732152 2014-06-19 19:16:01 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: make udp client structure refcounted Use the refcount for memory management and keep track of the number of duplicate clients in a separate variable. This will be useful later, and means we don't have to hold the OBJECT_LOCK all the time. https://bugzilla.gnome.org/show_bug.cgi?id=732866 2014-06-19 18:31:05 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: keep count of number of unique and non-unique IPv4 and IPv6 clients This will come in handy later. 2014-12-16 15:00:22 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Disable create_buf with libv4l2 Libv4l2 does not work with CREATE_BUFS. Instead of failing on random error caused by libv4l2, disable CREATE_BUFS when an emulated format is detected. 2014-12-09 17:39:12 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Add protection against broken libv4l2 It looks like libv4l2 support for CREATE_BUF is incomplete. That combine with existing bugs may lead to crash in GStreamer. These check will make it robust by: - Checking create buf index isn't an already in used index - Checking that the index out of QUERYBUF matches the requested index 2014-12-16 16:37:24 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Add something to the debug logs if an RTX AUX element can't be added ... because the application already has a signal handler set up here. 2014-11-21 14:13:34 +1100 Matthew Waters * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add retransmission support according to RFC4588 Based on the client-rtpaux example 2014-12-16 13:25:01 +0100 Wim Taymans * sys/osxvideo/osxvideosink.m: osxvideosink: clear rectangle structures before use 2014-12-09 15:09:56 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Always set format Right now we try to be clever by detecting if device format have changed or not, and skip setting format in this case. This is valid behaviour with V4L2, but it's also very error prone. The rational for not setting these all the time is for speed, though I can't measure any noticeable gain on any HW I own. Also, until recently, we where doing get/set on the format for each format we where probing, making it near to impossible that the format would match. This also fixes bug where we where skipping frame-rate setting if format didn't change. https://bugzilla.gnome.org/show_bug.cgi?id=740636 2014-12-15 18:30:01 -0500 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: videocrop: Remove todo about caps filter The filter is already interected. 2014-12-15 18:19:05 -0500 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: Make sure new crop is applied Since "basetransform: Fix caps equality check" commit a7f357, set_info() will not be called anymore if crop didn't change the caps. This is fixed by setting "need_update" boolean when cropping properties has been changed, and then applying these if they where not applied before rendering the next frame. This patch also fixed the locking, dropping un-needed custom lock, and no holding needless lock while doing the operation as we already hold the streaming lock. https://bugzilla.gnome.org/show_bug.cgi?id=740787 2014-12-12 18:10:35 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Prefer filter caps order while getting caps https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-12-09 13:38:26 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Add some error handling around channel layout parsing For now we just spit a warning and ignore the channel layout if we can't support it. https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-12-08 22:38:22 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: osxaudio: Take lock around sink/source before accessing the ringbuffer https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-12-01 21:06:27 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudiosrc: Probe channel layout too https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-12-01 20:32:04 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Only fix up channels/layout for PCM caps while probing It's unlikely that setting a channel layout will do much for AC3/DTS streams. If we find at some point that it does make sense, we can perform the structure copying unconditionally (i.e., the current code is wrong, since AC3/DTS will get two structures now - one with the channel layout, one without). https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-12-01 19:41:35 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: osxaudiosrc: Implement caps probing https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-12-01 19:29:57 +0530 Arun Raghavan * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: Bind audio device to audio unit early We want to bind the device during open so that subsequent format queries on the audio unit are as specific as possible from that point onwards. https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-29 23:16:30 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Fix up caps querying a bit This should make caps queries correct in PAUSED and higher as well. https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-28 22:32:36 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: osxaudio: Move osxaudiosrc-specific code out of the generic path Avoids one layering violation (GstCoreAudio referring to GstOsxAudioSrc). https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-28 22:23:17 +0530 Arun Raghavan * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudioringbuffer.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxcoreaudiohal.c: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: Clean up a GstCoreAudio -> GstOsxAudioSrc/Sink reference Now that device selection has no sink/source-specific bits, we can have generic device selection for this path. We do need to now track state changes so we can look up the final device_id once the device is open, though. https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-28 19:40:52 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Move device caps probing to get_caps() This should be preferred to running the probe at device open time. https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-28 18:37:02 +0530 Arun Raghavan * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: Make some debug code compile conditionally https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-28 15:06:35 +0530 Arun Raghavan * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudioringbuffer.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: osxaudio: Move device selection to ringbuffer->open_device() This is conceptually the right thing to do, and allows us to correctly catch errors in device selection as well, which we could not do while creating the ringbuffer. https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-28 14:34:34 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxcoreaudiohal.c: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: Consolidate input and output code paths a bit https://bugzilla.gnome.org/show_bug.cgi?id=740987 2014-11-21 11:54:18 +0100 Thibault Saunier * gst/deinterlace/gstdeinterlace.c: Deinterlace: in query_caps return only supported formats if filter is interlaced In some cases the currently set GstVideoInfo is not interlaced, but upstream caps are interlaced and the info is passed in the filter, we should take that info into account and make sure that we do not consider that case as a "pass through" case. https://bugzilla.gnome.org/show_bug.cgi?id=741407 2014-12-12 11:06:17 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Fix debug statement It was using the non-increasing offset variable, which made that statement not so useful :) 2014-12-12 11:03:15 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Add macros for the various timescale conversions This helps make the code more readable and avoid future bad usage of scaling function argument order. 2014-12-11 10:16:06 +0100 Patrick Radizi * gst/rtp/gstrtph264pay.c: rtph264pay: fix potential crash when shutting down A race condition in the state change function may cause buffers to be unreffed while they are still used by the streaming thread in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain up to the parent class first in the state change function to make sure streaming has stopped and only then free those buffers. https://bugzilla.gnome.org/show_bug.cgi?id=741381 2014-12-12 00:42:06 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Copy flags of the overall segment to output segments Preserve the segment flags of the overall demux segment on the output segments for each pad. 2014-12-09 02:43:00 +0100 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: use 64bit chunk_offset https://bugzilla.gnome.org/show_bug.cgi?id=741279 2014-12-10 17:39:17 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Fix rounding errors in duration update Make sure we store updated segment stop/duration with the same granularity as the duration timescale. And add more debug 2014-12-10 16:55:44 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Update duration when we get more information When dealing with fragmented files, we will get more accurate duration information via the mfra and moof atoms. In order for playback to not stop at the initial duration (from the moov atom), we need to check and update the various duration variables when we find more information. Fixes playback of fragmented files in pull mode 2014-12-10 15:08:40 +0100 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Remove variable assignments never read As detected by clang/scan-build 2014-12-10 14:56:06 +0100 Edward Hervey * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Use GstClockTime for nanosecond-based time variables/fields Avoids confusion with timescaled-based variables and bytes (offset) variables. And use GST_CLOCK_TIME_NONE where applicable 2014-12-03 14:47:05 +0100 Edward Hervey * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gstpushfilesrc.h: pushfilesrc: Add TIME SEGMENT capability Adds a new set of properties to make pushfilesrc output a TIME SEGMENT (instead of the filesrc BYTE SEGMENT). When time-segment is set to True the following will happen: * Seeks are refused (data starts from the beginning of the file) * The BYTE segment will be replaced by a TIME segment with the values specified in the various properties * The first outgoing buffer will have a timestamp set on it (by default it has a value of GST_CLOCK_TIME_NONE) 2014-12-10 11:35:29 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Also only unref caps if they're not NULL 2014-12-10 11:34:42 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: gst_pad_get_allowed_caps() will return NULL if there is no peer 2014-12-09 16:38:38 +0100 Thibault Saunier * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vpXenc: CLOCK_TIME_NONE is not a valid min_latency value We should just use 0 if we do not have the information 2014-12-03 17:26:56 +0100 Thibault Saunier * gst/rtpmanager/gstrtpsession.c: rtpsession: Use an empty iterator in iterate_internal_link when no links And not a NULL Iterator, so it is consistent with the way it usually works and avoid user to need a different code paths to handle that. 2014-12-09 14:01:50 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: set buffer interlace flags when field is V4L2_FIELD_INTERLACED If v4l2_buffer.field is V4L2_FIELD_INTERLACED, we set corresponding GstVideoBuffer flags depending on the video standard. According to V4L2 specification, M/NTSC transmits the bottom field first, all other standards the top field first. https://bugzilla.gnome.org/show_bug.cgi?id=737603 2014-12-08 21:26:18 +0100 Patrick Radizi * gst/rtp/gstrtph264pay.c: rtph264pay: Fixes buffer leak when using SPS/PPS Fixes a buffer leak that would occurr if the pipeline was shutdown while a SPS/PPS header was being created. https://bugzilla.gnome.org/show_bug.cgi?id=741271 2014-12-09 04:43:29 +0100 Mathieu Duponchelle * gst/effectv/gstaging.c: agingtv: fix memcpy when no color aging requested. video_size is the size in pixels, actual size of the memcpy has to be stride * height. 2014-12-07 17:33:51 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2: Workaround libv4l2 RW emulation bug When libv4l2 emulates RW mode on top of MMAP devices, the queues are only initialized on first read. The problem is that poll() will fail if called before the queues are initialized and streaming. Workaround this by doing a zero size read when pool is started in that IO mode. https://bugzilla.gnome.org/show_bug.cgi?id=740633 2014-12-07 17:27:37 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2: Fix RW io mode In RW, allocator can be null, max_buffers can be zero, and we need not to wait while the queue is empty since there is no queue. https://bugzilla.gnome.org/show_bug.cgi?id=740633 2014-12-03 16:40:49 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Cleanup uneeded check and cases There is nothing in between the break and the "done:" anymore, plus USERPTR and DMABUF_IMPORT case is exactly the same. 2014-12-03 17:07:49 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2pool: Fix CREATE_BUFS support for capture This patch fixes CREATE_BUFS support for capture devices. Initially we would only try and allocate more buffers when the copy threshold is reached. When the threshold was not set (needed) it would never happen. Another problem is that on capture side, acquire returns filled buffer, hence need to pool. We need to set a special flag to force allocation to happen. https://bugzilla.gnome.org/show_bug.cgi?id=741134 2014-12-03 16:27:59 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix CREATE_BUF probing Current for every memory type we where probing MMAP CREATE_BUFS ioct. https://bugzilla.gnome.org/show_bug.cgi?id=741134 2014-11-18 16:52:40 +0100 Nicola Murino * gst/matroska/matroska-demux.c: matroskademux: set framerate 0/1 when duration is not known https://bugzilla.gnome.org/show_bug.cgi?id=740130 2014-12-04 17:25:55 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: More fixes for reverse playback When seeking or finding the previous keyframe, do comparisons against targets and segments using composition time to correctly decide which sample times match. 2014-12-03 11:12:55 +0100 Thibault Saunier * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Use an empty iterator in iterate_internal_link when no links We used to setup an iterator with 1 GValue set with a NULL object pointer which is not the normal way to do that. Instead we should make sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE. 2014-12-03 13:20:57 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Handle seeks past EOS as a seek to the end Fix reverse playback of every frame by making seeks past/to EOS find the last segment and start there. 2014-12-02 15:33:25 -0500 Olivier Crête * gst/rtp/gstrtpmpadepay.c: rtpmpadepay: Relax caps to allow any clock-rate Some Wowza setups seem to send an invalid non-90000 clock-rate. 2014-12-01 21:04:02 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: don't use GST_CLOCK_TIME_NONE in non GstClockTime variables Use -1 instead as those are gint64/guint64 variables and not GstClockTime 2014-11-07 17:06:49 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2allocator.h: v4l2allocator: fix gst_v4l2_allocator_stop prototype gst_v4l2_allocator_stop returns a GstV4l2Return, not a gboolean. https://bugzilla.gnome.org/show_bug.cgi?id=739792 2014-11-07 16:41:52 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: unref pool when v4l2_allocator_new() fails https://bugzilla.gnome.org/show_bug.cgi?id=739791 2014-11-30 17:52:47 -0500 Nicolas Dufresne * sys/v4l2/v4l2_calls.h: v4l2: Remove last include to linux/videodev2.h We now use and update our internal copy so we no longer have to ifdef the entire code for features and defines that where added over the years. https://bugzilla.gnome.org/show_bug.cgi?id=740905 2014-08-24 13:38:08 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: implement seeking in fragmented mp4 files in pull mode based on the mfra table 2014-11-29 15:25:51 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: use track fragment decoding time (tfdt) in parse_trun() for interpolation As fallback if we don't have any existing samples as reference point yet. Based on patch by David Corvoysier 2014-11-29 14:37:25 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: parse mfra random access box for fragmented mp4 files If it's present, and we operate in pull mode. 2014-08-15 14:58:26 +0200 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: stop parsing headers for fragmented mp4s at the first moof Currently during header parsing, we scan through the entire file and skip every moof+mdat chunk for fragmented mp4s, which makes start-up incredibly slow. Instead, just stop at the first moof chunk when have a moov, and start exposing the streams, so we can go and start handling the moofs for real. 2014-11-29 13:59:35 +0000 Tim-Philipp Müller * tests/icles/.gitignore: * tests/icles/Makefile.am: * tests/icles/gdkpixbufoverlay-test.c: tests: add interactive gdkpixbufoverlay test Just need to fix the coordinate system now so that negative offsets are actually negative and not flipped to position things from the opposite border. 2014-11-29 13:53:03 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: gdkpixbufoverlay: add "pixbuf" property So we can set a GdkPixbuf directly instead of reading it from an image file on the file system. 2014-11-29 13:23:50 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/pixbufscale.c: * ext/gdk_pixbuf/pixbufscale.h: gdkpixbuf: remove pixbufscale code that was never ported Don't think we'll need this again. 2014-11-29 18:35:42 -0500 Olivier Crête * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: Use offset when copying header The header is not always at the start of the packet, so we need to compute the offset first. 2014-11-28 13:12:46 +0000 Tim-Philipp Müller * ext/taglib/gstapev2mux.cc: apev2mux: write APE tags at end for wavpack files http://www.wavpack.com/file_format.txt: "Both the APEv2 tags and/or ID3v1 tags must come at the end of the WavPack file, with the ID3v1 coming last if both are present." WavPack files that contain APEv2 tags at the beginning of the files are unplayable on players that use FFmpeg (like VLC) and most other software (except Banshee). Players that use libwavpack directly can play the files because it skips the tags, but does not recognize the tag data at that location. https://bugzilla.gnome.org/show_bug.cgi?id=711437 2014-11-28 10:41:55 +0000 Tim-Philipp Müller * tests/icles/.gitignore: * tests/icles/Makefile.am: * tests/icles/test-segment-seeks.c: tests: add interactive test for gapless playback using SEGMENT seeks Not working too well yet, there are glitches even with WAV or FLAC. https://bugzilla.gnome.org/show_bug.cgi?id=692368 2014-11-26 10:33:09 +0300 Andrei Sarakeev * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstaspectratiocrop.h: aspectratiocrop: Handle resolution changes properly When an caps-event is received, we must immediately change the crop to videocrop correctly changed caps-event dimension, otherwise the videocrop will first use the previous value of the crop that when resizing video to a smaller resolution may cause an error. https://bugzilla.gnome.org/show_bug.cgi?id=740671 2014-11-27 17:10:53 +0100 Edward Hervey * common: Automatic update of common submodule From 7bb2bce to ef1ffdc 2014-11-27 11:20:36 +0000 Tim-Philipp Müller * tests/icles/test-accurate-seek.c: test: use gst_util_uint64_scale_round() for timestamp to sample calculation 2014-11-27 11:16:35 +0000 Tim-Philipp Müller * tests/icles/.gitignore: * tests/icles/Makefile.am: * tests/icles/test-accurate-seek.c: tests: add interactive test for accurate seeking For some audio formats. https://bugzilla.gnome.org/show_bug.cgi?id=655276 2014-11-26 16:04:26 +0100 Edward Hervey * gst/isomp4/qtdemux.c: isomp4: Check presence of mfhd in moof The 'mfhd' atom is mandatory in 'moof'. We can later on check whether the fragment number properly increases 2014-11-26 15:59:36 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: isomp4: Fix mfro and tfra atom dumping mfro was skipping the version/flags tfra had wrong byte_reader return value checks 2014-11-26 15:58:26 +0100 Edward Hervey * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_types.c: isomp4: Add mfhd atom dumping 2014-11-27 00:15:02 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Handle empty segments when seeking in reverse play. Empty segments in an edit list have a media_start time of -1, as they don't actually play any media. Allow for that when aligning to the reference stream in reverse play. 2014-11-24 10:36:54 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: Revert "v4l2allocator: Remove unused variable" This reverts commit ad4480d53408a4d97ab531174ef37f258f3253c0. 2014-11-24 10:36:30 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: Revert "v4l2: move vb_queue probing from allocator to v4l2object" This reverts commit ec6b8b84af719d828ddd91c724e715c0b4a556bc. 2014-11-24 10:33:29 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: Revert "v4l2object: allow to automatic selection of dmabuf" This reverts commit e6c2ad5571e5dedb212287efe238eb450032cd4f. 2014-11-23 16:34:15 +0000 Tim-Philipp Müller * REQUIREMENTS: REQUIREMENTS: update a little People actually look at that it seems. 2014-11-23 16:22:12 +0000 Tim-Philipp Müller * gst/icydemux/Makefile.am: icydemux: does not need to link against zlib 2014-11-22 21:28:35 +0000 Tim-Philipp Müller * configure.ac: * ext/speex/gstspeexdec.h: * ext/speex/gstspeexenc.h: speex: remove support for ancient speex versions 2014-11-21 11:21:18 +0100 Branislav Katreniak * ext/soup/gstsouphttpsrc.c: souphttpsrc: log connection events at info level https://bugzilla.gnome.org/show_bug.cgi?id=739305 2014-10-20 13:00:37 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: ensure rtx_retry_period >= 0 https://bugzilla.gnome.org/show_bug.cgi?id=739344 2014-11-21 11:44:24 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Remove unused variable this was introduced by commit ec6b8b https://bugzilla.gnome.org/show_bug.cgi?id=699382 2014-11-16 12:34:17 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Handle corrupted buffer with empty payload This allow skipping buffer flagged with ERROR that has no payload. This is typical behaviour when a recovererable error occured during capture in the driver, but that no valid data was ever written into that buffer. This patch also translate V4L2_BUF_FLAG_ERROR into GST_BUFFER_FLAG_CORRUPTED. Hence decoding error produce by decoder due to missing frames will now be correctly marked. Finally, this fixes a buffer leak when EOS is reached. https://bugzilla.gnome.org/show_bug.cgi?id=740040 2014-11-21 16:36:15 +0100 Benjamin Gaignard * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2object: allow to automatic selection of dmabuf If the v4l2 queue support dmabuf select this buffer pool mode and update the query with allocator. This patch only concern exporting dmabuf and not importing dmabuf fd from downstream element. https://bugzilla.gnome.org/show_bug.cgi?id=699382 2014-11-21 16:13:05 +0100 Benjamin Gaignard * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: move vb_queue probing from allocator to v4l2object The goal is to make those information available in v4l2_object to be able later to select the best allocation method for the pool https://bugzilla.gnome.org/show_bug.cgi?id=699382 2014-11-20 22:42:59 +0530 Arun Raghavan * gst/rtpmanager/gstrtpbin.h: rtpbin: Fix up new_jitterbuffer signal prototype 2014-11-20 20:19:25 +0530 Arun Raghavan * gst/rtpmanager/gstrtpbin.c: rtpbin: Document how to control per-SSRC retransmission 2014-11-20 20:18:45 +0530 Arun Raghavan * docs/design/design-rtpretransmission.txt: doc: Trivial spelling and consistency update 2014-11-20 13:14:14 +0100 Wim Taymans * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: rtpgstpay: put 0-byte at the end of events Put a 0-byte at the end of the event string. Does not break ABI because old depayloaders will skip the 0 byte (which is included in the length). Expect a 0-byte at the end of the event string or a ; for old payloaders. See https://bugzilla.gnome.org/show_bug.cgi?id=737591 2014-11-20 12:40:28 +0100 Wim Taymans * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: avoid buffer overread. Check that a caps event string is 0 terminated and the event string is terminated with a ; to avoid buffer overreads. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591 2014-11-20 10:45:07 +0000 Tim-Philipp Müller * gst/isomp4/gstqtmuxmap.c: qtmux: don't limit max video resolution to 4096x4096 MAX isn't entirely correct as upper limit either, it should really be MAXUINT32, but it's unlikely to be a problem in the near future. https://bugzilla.gnome.org/show_bug.cgi?id=740407 2014-11-19 15:06:00 -0800 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: fix leak for mikey base64 decoded key-mgmt https://bugzilla.gnome.org/show_bug.cgi?id=740392 2014-11-20 09:01:38 +0100 Wim Taymans * gst/videofilter/gstvideobalance.c: videobalance: fix unhandled format in passthrough In passthrough we can handle all formats. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740387 2014-11-19 16:12:38 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: flvdemux: Restrict resyncing to TS regressions The behavior of resyncing video and audio indepen- dently can cause A/V desyncs. Lets restrict resyncs to jumps backward for now. https://bugzilla.gnome.org/show_bug.cgi?id=736397 2014-11-17 23:16:03 +1100 Matthew Waters * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: fix up QoS handling for live sources Only attempt adaptive drop when we are not live https://bugzilla.gnome.org/show_bug.cgi?id=739996 2014-11-10 22:34:39 +0100 Henning Heinold * tests/examples/rtp/client-PCMA.py: * tests/examples/rtp/server-alsasrc-PCMA.py: examples: port python rtp PCMA client/server tests to 1.0 https://bugzilla.gnome.org/show_bug.cgi?id=739930 2014-06-04 12:11:10 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: set the channel positions using the appropriate API This avoids _set_format setting the unpositioned flag when passed NULL as channel positions, as it would not be cleared when setting actual channel positions later. 2014-11-01 22:39:41 +0100 Aurélien Zanelli * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vpx: mark arnr-type properties as deprecated and set them to no-op ARNR type control in libvpx has been deprecated so this commit mark the vp8enc and vp9enc associated properties as deprecated and change their behavior to just display a warning message. https://bugzilla.gnome.org/show_bug.cgi?id=739476 2014-11-10 13:16:01 +0530 Arun Raghavan * gst/rtpmanager/gstrtpbin.c: rtpmanager: Trivial typo fix 2014-11-09 11:04:33 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroska-mux: Use G_DEFINE_TYPE() to register the pad instead of manually registering it 2014-11-06 15:37:28 +0100 Göran Jönsson * gst/matroska/matroska-mux.c: matroskamux: make GstMatroskamuxPad get_type() function thread-safe https://bugzilla.gnome.org/show_bug.cgi?id=739722 2014-11-07 16:11:24 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2allocator.c: v4l2allocator: fix error message if allocator is already active https://bugzilla.gnome.org/show_bug.cgi?id=739789 2014-11-06 21:21:40 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Improve buffer validation Improve buffer validation by making sure each memory are the right one and that each memory is writable. This fixes tearing issues in case downstream uses gst_buffer_make_writable() or other type of GstBuffer copy where memory are only reffed. https://bugzilla.gnome.org/show_bug.cgi?id=739754 2014-11-06 21:38:43 +0100 Josep Torra * gst/rtsp/Makefile.am: rtsp: fix build in gst-uninstalled setup 2014-10-29 18:44:43 +0100 Thibault Saunier * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Handle seqnums https://bugzilla.gnome.org/show_bug.cgi?id=739366 2014-11-04 08:18:41 +0530 Vineeth T M * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: pngdec: change parse logic Right now in parse logic the signature is checked every time the parse function is called, and the whole data is the scanned each and every time, even though the data is scanned in the previous instance. Changing the logic such that, we skip the bytes which are already scanned in the previous instances of parse. This helps in avoiding multiple scan of already scanned data/signature. https://bugzilla.gnome.org/show_bug.cgi?id=737708 2014-11-03 15:26:06 +0100 Wim Taymans * gst/videomixer/videomixer2.c: videomixer2: reverse order of params for converter 2014-11-03 11:44:28 +0100 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: fix typo in flags https://bugzilla.gnome.org/show_bug.cgi?id=739549 2014-11-02 23:33:23 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2src: fix a couple of minor leaks 2014-11-02 19:42:03 +0000 Tim-Philipp Müller * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom2k1: post QoS messages when dropping frames due to QoS 2014-11-02 19:29:52 +0000 Tim-Philipp Müller * gst/goom/gstgoom.c: * gst/goom/gstgoom.h: goom: post QoS messages when dropping frames due to QoS 2014-11-02 19:02:35 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: tweak writing app tag string a little 2014-11-02 16:51:23 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * gst/isomp4/gstqtmux.c: * gst/level/gstlevel.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: Sprinkle some G_PARAM_DEPRECATED and #ifndef GST_REMOVE_DEPRECATED 2014-11-02 16:58:07 +0000 Tim-Philipp Müller * tests/check/elements/level.c: tests: don't use deprecated property in level unit test 2014-11-02 13:06:33 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property Properties are so much more useful if you can actually set and get their values. 2014-10-30 17:41:19 +0000 Simon Farnsworth * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: v4l2: Clean up interlace support Rather than try and guess interlace support as part of checking supported sizes, look for interlace support specifically in its own function. As a cleanup, use V4L2_FIELD_ANY when probing sizes, which should result in the driver doing the right thing. With my capture setup, this gets me the following sample caps: For 1080i resolution: video/x-raw, format=(string)YUY2, width=(int)1920, height=(int)1080, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)interleaved, framerate=(fraction){ 25/1, 30/1 } For 720p resolution: video/x-raw, format=(string)YUY2, width=(int)1280, height=(int)720, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, framerate=(fraction){ 50/1, 60/1 } For 576i/p resolution (both possible at the point of query): video/x-raw, format=(string)YUY2, width=(int)720, height=(int)576, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string){ progressive, interleaved }, framerate=(fraction){ 25/1, 50/1 } This, in turn, makes 576i work correctly; with the old code, the caps would be interlace-mode=progressive for interlaced video. https://bugzilla.gnome.org/show_bug.cgi?id=726194 2014-11-01 12:18:02 +0100 Aurélien Zanelli * ext/vpx/gstvp8utils.h: vpx: remove compatibility defines We are guaranteed to have VPX_IMG_FMT_I420, VPX_PLANE_Y, VPX_PLANE_U and VPX_PLANE_V as we require libvpx > 1.1.0. https://bugzilla.gnome.org/show_bug.cgi?id=739476 2014-11-01 15:33:23 +0000 Tim-Philipp Müller * configure.ac: * ext/wavpack/gstwavpackcommon.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: wavpack: remove support for ancient API version 2014-11-01 10:14:31 -0400 Nicolas Dufresne * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: rtpvp8: Use VP8 encoding name Both Firefox and Chrome uses VP8 as the encoding in their SDP. Adding this now defacto standard name removes the need for special case in SDP parsing code. https://bugzilla.gnome.org/show_bug.cgi?id=737810 2014-11-01 11:59:26 +0000 Tim-Philipp Müller * gst/rtp/gstrtpmp2tpay.c: rtpmp2tpay: fix up template caps so we can output the default pt 33 Add fixed payload type for mp2t to template caps as well, so our output caps match the advertised default pt. Fixes a regression from 1.2. There's still something wrong with caps negotiation though, rtpmp2tpay payload=96 ! fakesink will not output caps with payload=96. 2014-10-30 15:37:36 -0700 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: mikey related memory leaks https://bugzilla.gnome.org/show_bug.cgi?id=739430 2014-06-10 10:04:07 +0100 Vincent Penquerc'h * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: speexenc: update output segment stop time to match clipped samples This will let oggmux generate a granpos on the last page that properly represents the clipped samples at the end of the stream. 2014-06-10 10:59:13 +0100 Vincent Penquerc'h * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: update output segment stop time to match clipped samples This will let oggmux generate a granpos on the last page that properly represents the clipped samples at the end of the stream. 2014-10-07 15:29:33 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: cleanly handle streamon failure for output device On streamon failure, the queued buffer is not released from the bufferpool class point of view because it is queued to the driver and the flush logic is not performed since we are not in streaming state. It causes the v4l2 bufferpool to always return that stop method failed and to leak v4l2 objects and buffers. This commit solve this by performing the flush logic in error case, ie flushing the allocator and restoring queued buffer state to non-queued. https://bugzilla.gnome.org/show_bug.cgi?id=738102 2014-10-08 10:31:21 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: implement dispose method Unref objects in dispose method rather than in finalize in order to prevent circular reference. https://bugzilla.gnome.org/show_bug.cgi?id=738102 2014-10-08 10:35:14 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: check that allocator is non null when stopping pool Otherwise, we could dereference NULL allocator when the stop method is called by the GstBufferPool's finalize method. https://bugzilla.gnome.org/show_bug.cgi?id=738102 2014-10-09 12:15:05 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2sink.c: v4l2sink: Implement unlock/unlock_stop This will prevent deadlocks, but will also properly flush the pool and allocator when going to READY state. It should also fix issues reported on mailing list when seeking is performed. https://bugzilla.gnome.org/show_bug.cgi?id=738152 2014-10-28 21:32:06 +0000 Tim-Philipp Müller * ext/pulse/pulsedeviceprovider.h: * sys/v4l2/gstv4l2deviceprovider.h: * sys/v4l2/gstv4l2tuner.h: pulse, v4l2: add missing G_END_DECLS in some places 2014-10-27 17:57:20 +0100 Sebastian Dröge * common: Automatic update of common submodule From 84d06cd to 7bb2bce 2014-10-27 11:08:20 +0100 Sebastian Dröge * tests/check/elements/aacparse.c: aacparse: Fix unit test now that we always have profile/level in the caps 2014-10-26 14:55:49 +0000 Tim-Philipp Müller * Makefile.am: Parallelise 'make check-valgrind' Some of the RTP unit tests are very flaky and will fail more often with the CPU maxed out fully. Those tests need to be fixed in any case though, they also fail on slower machines and also occasionally with normal 'make check'. 2014-10-26 11:47:25 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Always set profile/level on the caps We have the information already, so why not use it? 2014-10-25 12:36:02 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix crash on some 32-bit systems Make sure to pass right number of bits to gst_structure_new() which is a vararg function. Fixes elements/rtpaux unit test on ppc32. 2014-10-25 00:56:02 +0100 Tim-Philipp Müller * tests/check/elements/rgvolume.c: tests: fix rgvolume test on big-endian systems 2014-10-25 00:53:39 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/mulawdec.c: * tests/check/elements/mulawenc.c: tests: fix mulawdec/mulawenc test for big endian systems 2014-10-24 23:48:30 +0100 Tim-Philipp Müller * gst/interleave/interleave.c: interleave: intersect result with filter caps in caps query Fixes crash in audiotestsrc because of an unsupported format getting negotiated on big-endian systems with audiotestsrc ! interleave ! audioconvert ! wavenc 2014-10-23 15:46:13 +0100 Tim-Philipp Müller * ext/pulse/pulsedeviceprovider.c: * ext/pulse/pulsedeviceprovider.h: pulse: remove some unused typedefs 2014-10-22 15:28:44 +0200 Ananda * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: Fix segfault when resetting the codecs multiple times https://bugzilla.gnome.org/show_bug.cgi?id=738793 2014-10-22 22:50:54 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Temporarily disable stream status posting We need a mechanism in PulseAudio to allow running code outside the mainloop lock. Then we'd be able to post to the bus (taking the GST_OBJECT_LOCK), without worrying about locking order with the mainloop lock, which is the current cause of deadlocks while trying to post the stream status messages. https://bugzilla.gnome.org/show_bug.cgi?id=736071 2014-10-22 15:04:24 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: limit the retry frequency When the RTT and jitter are very low (such as on a local network), the calculated retransmission timeout is very small. Set some sensible lower boundary to the timeout by adding a new property. We use the packet spacing as a lower boundary by default. 2014-10-22 13:40:58 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpjitterbuffer.c: gstrtpjitterbuffer: add "rtx-min-delay" property This property is useful to set a min time to wait before sending a retransmission event. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378 2014-10-22 13:29:48 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Refactor code Refactor some code dealing with calculating various timeouts. See https://bugzilla.gnome.org/show_bug.cgi?id=735378 2014-10-10 19:50:06 +0200 Miguel París Díaz * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: fix Early Feedback Transmission In early retransmission we are allowed to schedule 1 regular RTCP packet at an earlier time. When we do that, we need to set allow_early to FALSE and ignore/drop (or merge) all future requests for early transmission. We now first check if we can schedule an early RTCP and if we can, actually prepare the data for the next RTCP interval. After we send the next regular RTCP after the early RTCP, we set allow_early to TRUE again to allow more early requests. Remove the condition for the immediate feedback for now. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319 2014-10-21 13:01:32 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From a8c8939 to 84d06cd 2014-10-21 13:10:24 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: make debug line less confusing 2014-10-21 12:58:13 +0200 Stefan Sauer * README: * common: Automatic update of common submodule From 36388a1 to a8c8939 2014-07-02 17:50:35 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: rework resync handling Add a need-resync state, this is when we need to try to lock on to a time/RTPtime pair. Always check the RTP timestamps and if they go backwards, mark ourselves as need-resync. Only resync when need-resync is TRUE and we have a valid time. Otherwise we keep the old values. This avoids locking on to an invalid time and causing us to timestamp everything with -1. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417 2014-10-03 17:28:06 -0700 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: set full stream caps on internal src TCP pads Set the complete stream caps on the TCP internal src pads. Otherwise, ptdemux will not properly detect the caps change. https://bugzilla.gnome.org/show_bug.cgi?id=737868 2014-10-17 22:23:27 +0200 Sjoerd Simons * gst/rtpmanager/gstrtpmux.c: * tests/check/elements/rtpmux.c: rtpmux: Don't set PROXY_CAPS flag on the src pad rtpmux behaves like a funnel in that it forwards whatever upstream is sending buffers. So setting proxy caps doesn't make sense as the upstream don't have to have compatible caps, thus resulting in an empty caps set as a result of a caps query. Instead set fixed caps just as funnel does. https://bugzilla.gnome.org/show_bug.cgi?id=738722 2014-10-20 11:57:38 +0530 Vineeth T M * gst/videobox/gstvideobox.c: videobox: critical error when element properties set as max/min left, right, top, bottom can be set from range of -2147483648 to 2147483647 when i launch the videobox element with that values, it gives a critical error (gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed This happens because min cannot be equal to max. https://bugzilla.gnome.org/show_bug.cgi?id=738838 2014-10-15 17:45:24 +0100 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: Revert "rtp: add h265 RTP payloader + depayloader" This reverts commit d06ba9051f904a7eb482c07a97a1827169158663. This breaks the build, as it depends on parser API in -bad. 2014-10-15 17:34:50 +0200 Jurgen Slowack * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtph265depay.c: * gst/rtp/gstrtph265depay.h: * gst/rtp/gstrtph265pay.c: * gst/rtp/gstrtph265pay.h: rtp: add h265 RTP payloader + depayloader 2014-10-05 21:24:27 +0200 Peter G. Baum * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: Support RF64 format https://bugzilla.gnome.org/show_bug.cgi?id=725145 2014-10-11 11:18:42 +1100 David Sansome * gst/equalizer/gstiirequalizer.c: equalizer: Don't call iirequalizer's transform_ip in passthrough mode It tries to map the read-only buffer with GST_MAP_READWRITE and crashes. https://bugzilla.gnome.org/show_bug.cgi?id=737886 2014-10-10 18:30:07 -0400 Olivier Crête * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsource: Rename seqnum-base to seqnum-offset in caps This was modified back in 1.0 in GstRtpBasePayload 2014-10-10 18:11:19 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: * tests/check/elements/dtmf.c: rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset These were renamed in GstRTPBasePayload in 1.0 2014-10-10 17:30:24 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * tests/check/elements/rtpmux.c: rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset These were renamed in GstRTPBasePayload in 1.0 2014-10-06 14:23:22 +0100 Luis de Bethencourt * gst/goom2k1/filters.c: goom2k1: removing block of code that does nothing The loop in zoomFilterSetResolution is meant to change the values in the zf->firedec[] array. Each iteration writes the value of decc onto the arrya, but no conditions that change the value of decc are ever met and the array is filled with zero for each element. Which is the initial state of the array before the loop begins. The loop does nothing. https://bugzilla.gnome.org/show_bug.cgi?id=728353 2014-10-04 17:17:13 +0200 Stefan Sauer * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: don't log all clock_rate changes as warnings. We never initialize clock_rate explicitly, therefore it is 0 by default. The parameter is a uint32 and the only caller ensure that it is >0, therefore it won't become -1 ever. 2014-10-02 14:26:08 +0530 Nirbheek Chauhan * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Fix lifetime of stream headers and queued buffers Stream headers are updated whenever ::set_caps is called, so we can't assume they'll be valid before the message body is written out. We *can* assume that for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those. Also, add some debug logging for stream header interactions. https://bugzilla.gnome.org/show_bug.cgi?id=737771 2014-10-02 03:26:22 +0200 Matej Knopp * gst/audioparsers/gstaacparse.c: aacparse: fix memory leak when prepending ADTS headers https://bugzilla.gnome.org/show_bug.cgi?id=737761 2014-09-23 10:48:09 +0200 Antonio Ospite * gst/interleave/interleave.c: * gst/interleave/interleave.h: interleave: interleave samples following the Default Channel Ordering In order to have a full mapping between channel positions in the audio stream and loudspeaker positions, the channel-mask alone is not enough: the channels must be interleaved following some Default Channel Ordering as mentioned in the WAVEFORMATEXTENSIBLE[1] specification. As a Default Channel Ordering use the one implied by GstAudioChannelPosition which follows the ordering defined in SMPTE 2036-2-2008[2]. NOTE that the relative order in the Top Layer is not exactly the same as the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users using so may channels are already aware of such discrepancies. [1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx [2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127 2014-10-02 10:10:11 +0300 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Send CAPS event after the pad was activated Otherwise the CAPS event will be dropped and we never configure any caps at all, leading to weird behaviour in many situations. Especially header rewriting is not going to work if a capsfilter is after wavenc. https://bugzilla.gnome.org/show_bug.cgi?id=737735 2014-10-01 23:12:30 +0530 Nirbheek Chauhan * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Add some more useful debug logging 2014-10-01 23:05:03 +0530 Nirbheek Chauhan * ext/soup/gstsouphttpclientsink.c: souphttpclientsink: Free queued buffers in ::reset ::render sets a new callback for writing out new buffers only if there aren't already buffers queued for writing with a previously-scheduled callback. However, if the previously-scheduled callback is interrupted by a state change (either manually or due to an error) and there are still buffers in the queue, restarting the pipeline will result in buffers being queued forever, and no callbacks will ever be scheduled, and no buffers will be written out. https://bugzilla.gnome.org/show_bug.cgi?id=737739 2014-10-01 17:29:29 +0300 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Actually use the correct GstVideoInfo for conversion 2014-10-01 17:24:59 +0300 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Revert the last commit and handle resolutions differences properly This is about converting the format, not about converting any widths and heights. Subclasses are expected to handler different resolutions themselves, like the videomixers already do properly. 2014-10-01 17:12:59 +0300 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: GstVideoConverter currently can't rescale and will assert Leads to ugly assertions instead of properly erroring out: CRITICAL **: gst_video_converter_new: assertion 'in_info->width == out_info->width' failed 2014-09-30 11:35:12 +0300 Sebastian Dröge * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp9enc.c: vp8enc/vp9enc: Protect the encoder with a mutex in all situations 2014-09-30 11:31:43 +0300 Sebastian Dröge * ext/vpx/gstvp9enc.c: vp9enc: Allow caps renegotiation https://bugzilla.gnome.org/show_bug.cgi?id=726329 2014-09-30 11:28:39 +0300 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: finish() and drain() should return a GstFlowReturn 2014-03-14 12:59:02 +0100 Jose Antonio Santos Cadenas * ext/vpx/gstvp8enc.c: vp8enc: Allow caps renegotiation https://bugzilla.gnome.org/show_bug.cgi?id=726329 2014-09-29 11:49:45 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: v4l2object: set colorspace for output devices When the v4l2 device is an output device, the application shall set the colorspace. So map GStreamer colorimetry info to V4L2 colorspace and set on set_format. In case we have no colorimetry information, we try to guess it according to pixel format and video size. https://bugzilla.gnome.org/show_bug.cgi?id=737579 2014-09-29 22:48:16 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: Add some documentation about threading and synchronisation This gives a quick introduction to how the pulsesink/pulsesrc code interacts with the pa_threaded_mainloop that we start up to communicate with the server. 2014-09-29 20:18:08 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Make emitting stream status messages synchronous The stream status messages are emitted in the PA mainloop thread, which means the mainloop lock is taken, followed by the Gst object lock (by gst_element_post_message()). In all other locations, the order of locking is reversed (this is unavoidable in a bunch of cases where the object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get control to take the mainloop lock). The only way to guarantee that the defer callback for stream status messages doesn't deadlock is to either stop posting those messages, or make sure that the message emission is completed before we proceed to any point that might take the object lock before the mainloop lock (which is what we do after this patch). https://bugzilla.gnome.org/show_bug.cgi?id=736071 2014-09-16 12:12:49 +0200 Antonio Ospite * gst/wavenc/gstwavenc.c: wavenc: print channel masks in hexadecimal 2014-09-27 16:01:21 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2deviceprovider.h: v4l2: remove redundant struct declaration 2014-09-26 13:46:16 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix compiler warnings gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type 'GstRTSPResult' [-Werror,-Wenum-conversion] res = gst_sdp_message_new (&sdp); ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~ gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type 'GstRTSPResult' [-Werror,-Wenum-conversion] res = gst_sdp_message_parse_uri (uri, sdp); ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 2014-09-25 15:01:14 +0200 Jonas Holmberg * gst/matroska/matroska-demux.c: matroskademux: make demuxer reusable Remove pads from flow combiner and reset last flow return to FLOW_OK by resetting the flow combiner. This prevents FLOW_FLUSHING when trying to re-use the demuxer after setting it back to NULL/READY state. https://bugzilla.gnome.org/show_bug.cgi?id=737359 2014-09-24 16:46:36 +0200 Wim Taymans * gst/videomixer/Makefile.am: * gst/videomixer/gstcms.c: * gst/videomixer/gstcms.h: * gst/videomixer/videoconvert.c: * gst/videomixer/videoconvert.h: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2pad.h: * gst/videomixer/videomixerorc-dist.c: * gst/videomixer/videomixerorc-dist.h: * gst/videomixer/videomixerorc.orc: videomixer: use video library code instead of copy 2014-09-18 16:39:19 +0530 Sanjay NM * gst/audioparsers/gstmpegaudioparse.c: audioparsers: Added index check before using the index https://bugzilla.gnome.org/show_bug.cgi?id=736878 2014-09-23 23:33:37 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: Do not infer DTS on buffers from sparse streams. DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later. This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap) https://bugzilla.gnome.org/show_bug.cgi?id=737095 2014-09-18 17:08:37 +0530 Sanjay NM * gst/goom/ifs.c: goom: Clarified precedence between % and ? https://bugzilla.gnome.org/show_bug.cgi?id=736887 2014-09-18 17:59:31 +0530 Sanjay NM * gst/rtsp/gstrtspsrc.c: rtsp: clarify expression so operator precedence is clear https://bugzilla.gnome.org/show_bug.cgi?id=736903 2014-09-18 16:04:03 +0530 Sanjay NM * ext/libpng/gstpngdec.c: * gst/alpha/gstalpha.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/gstscaletempo.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/deinterlace/gstdeinterlace.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-mux.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/rtpsession.c: Miscellaneous minor cleanups Fix redundant variables and assignments, and unreachable breaks. https://bugzilla.gnome.org/show_bug.cgi?id=736875 https://bugzilla.gnome.org/show_bug.cgi?id=736876 https://bugzilla.gnome.org/show_bug.cgi?id=736879 https://bugzilla.gnome.org/show_bug.cgi?id=736880 https://bugzilla.gnome.org/show_bug.cgi?id=736881 https://bugzilla.gnome.org/show_bug.cgi?id=736888 https://bugzilla.gnome.org/show_bug.cgi?id=736890 https://bugzilla.gnome.org/show_bug.cgi?id=736892 https://bugzilla.gnome.org/show_bug.cgi?id=736893 https://bugzilla.gnome.org/show_bug.cgi?id=736894 2014-09-24 00:12:14 +0100 Tim-Philipp Müller * gst/videobox/gstvideobox.c: videobox: remove duplicate assignments https://bugzilla.gnome.org/show_bug.cgi?id=736897 2014-09-23 22:55:48 +0300 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Only calculate with durations != -1 2014-09-23 19:08:48 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: collect pad for sparse stream should be created with lock set to false Avoids waiting for buffers from sparse streams https://bugzilla.gnome.org/show_bug.cgi?id=737095 2014-09-23 19:07:25 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: fix subtitle buffer duration and strip null termination Strip the \0 off the subtitle as we already know the size and also remember to set the duration as buffer copying doesn't do it. https://bugzilla.gnome.org/show_bug.cgi?id=737095 2014-09-23 19:06:18 +0200 Matej Knopp * gst/isomp4/atoms.c: qtmux: move subtitle layer above video and set alternate group layer -1 is above video, that is 0 And having all subtitles in alternate group 2 means that only one should be selected at a time. https://bugzilla.gnome.org/show_bug.cgi?id=737095 2014-09-23 09:47:31 +0200 Edward Hervey * tests/check/elements/souphttpsrc.c: check/soup: Temporarily disable G_ENABLE_DIAGNOSTIC The SOUP_SERVER_PORT property has been deprecated in recent libsoup versions. 2014-09-23 09:43:05 +0200 Edward Hervey * tests/check/elements/souphttpsrc.c: check/soup: Define minimum version required To avoid deprecation warnings 2014-09-19 19:14:28 +0200 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: Handle mp4a without ESDS atom https://bugzilla.gnome.org/show_bug.cgi?id=736986 2014-09-22 16:15:27 +0200 Linus Svensson * sys/ximage/gstximagesrc.c: ximagesrc: Fix build problem without XFIXES 2014-09-19 14:34:13 +0530 Sanjay NM * gst/dtmf/gstrtpdtmfdepay.c: dtmf: Removed unused structure members https://bugzilla.gnome.org/show_bug.cgi?id=736883 2014-09-11 13:48:44 -0300 Reynaldo H. Verdejo Pinochet * gst/isomp4/atoms.c: isomp4: fix wrong DAR calculation for PAR <= 1 CID #1226452 https://bugzilla.gnome.org/show_bug.cgi?id=736396 2014-09-18 16:59:52 +0530 Sanjay NM * gst/flv/gstflvdemux.c: flv: Removed unreachable break statements https://bugzilla.gnome.org/show_bug.cgi?id=736884 2014-09-17 16:37:11 +0200 Ognyan Tonchev * gst/rtpmanager/gstrtpbin.c: rtpbin: do not leak encsink pad in error case https://bugzilla.gnome.org/show_bug.cgi?id=736807 2014-09-17 16:23:21 +0200 Ognyan Tonchev * gst/multipart/multipartdemux.c: multipartdemux: do not leak new stream event https://bugzilla.gnome.org/show_bug.cgi?id=736805 2014-09-15 09:08:18 +0530 Ravi Kiran K N * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: y4menc: port y4menc to use GstVideoEncoder base class https://bugzilla.gnome.org/show_bug.cgi?id=735085 2014-09-17 13:55:18 +0300 Sebastian Dröge * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiohal.c: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: OSStatus is not a fourcc, so don't print it as one... 2014-09-16 14:26:08 +0200 Ognyan Tonchev * gst/audioparsers/gstflacparse.c: flacparse: do not leak uid after parsing TOC event https://bugzilla.gnome.org/show_bug.cgi?id=736739 2014-09-16 22:47:13 +0300 Sebastian Dröge * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: Declare some more required caps fields in the sink template caps Now only missing are width and height, which are expressed as strings for RTP... so we can't put them into the template caps. 2014-09-16 16:46:07 +0530 Vineeth T M * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gdk_pixbuf/gstgdkpixbufdec.h: gdkpixbufdec: modify wrong packetized mode logic packetized mode is being set when framerate is being set which is not correct. Changing the same by checking the input segement format. If input segment is in TIME it is Packetized, and if it is in BYTES it is not. https://bugzilla.gnome.org/show_bug.cgi?id=736252 2014-09-16 11:26:22 +0300 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Remove unused variable and use correct decoder variable name 2014-09-16 11:25:42 +0300 Sebastian Dröge * ext/libpng/gstpngdec.c: pngdec: Remove unused variable 2014-09-16 13:24:15 +0530 Vineeth T M * ext/jpeg/gstjpegdec.c: jpeggdec: modify wrong packetized mode logic packetized mode is being set when framerate is being set which is not correct. Changing the same by checking the input segement format. If input segment is in TIME it is Packetized, and if it is in BYTES it is not. https://bugzilla.gnome.org/show_bug.cgi?id=736252 2014-09-16 13:23:16 +0530 Vineeth T M * ext/libpng/gstpngdec.c: pngdec: modify wrong packetized mode logic packetized mode is being set when framerate is being set which is not correct. Changing the same by checking the input segement format. If input segment is in TIME it is Packetized, and if it is in BYTES it is not. https://bugzilla.gnome.org/show_bug.cgi?id=736252 2014-09-15 14:39:41 +0200 Antonio Ospite * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: ximagesrc: Remove unused screen-num property The screen number can be still specified as part of the display-name property (e.g. for screen 1 of display 0 use display-name=":0.1"). https://bugzilla.gnome.org/show_bug.cgi?id=736122 2014-09-04 16:10:51 +0200 Antonio Ospite * sys/ximage/gstximagesrc.c: ximagesrc: Draw the cursor only when it is active in the capturing region Use XQueryPointer to check that the pointer is actually active inside the capturing region. This prevents drawing the cursor when the pointer is partially outside of the captured region but not active inside the region; in particular this avoids drawing the "window resize" cursor shapes to the captured image when the mouse pointer crosses a window border. NOTE that this is not only an optimization, this also happen to fix a serious problem in multi-screen setups. Because XFixes gives no information of what screen the pointer is on, ximagesrc was always drawing the cursor on the captured screen even if the mouse pointer was on another screen. For example, when capturing from screen 1 (i.e. display-name=":0.1") the cursor was drawn in the captured image even when the mouse pointer was actually on screen 0, which is wrong and visually confusing. https://bugzilla.gnome.org/show_bug.cgi?id=690646 2014-09-05 11:33:31 +0200 Antonio Ospite * sys/ximage/gstximagesrc.c: ximagesrc: Fix drawing the cursor when it is outside the capturing region When the cursor is partially or totally out of the capturing region on the top side or on the left side, it gets drawn fully inside of the region with its coordinates rounded up to the left or to the top border. This is immediately noticeable when using the xid property to capture a specific window. To fix the issue, allow negative cx and cx coordinates when checking the boundaries before drawing the cursor. NOTE that the boundaries checking calculations still allows the cursor to be drawn when it is only partially outside of the capturing region, but this makes sense and gives a more pleasing visual behaviour. https://bugzilla.gnome.org/show_bug.cgi?id=690646 2014-09-05 00:15:30 +0200 Antonio Ospite * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: ximagesrc: Fix the destination coordinates of the cursor XFixes provides the cursor coordinates relative to the root window, this is not taken into account when using the xid property to capture a specific window, the result is that the cursor gets drawn at the wrong position. In order to fix this consider the window location when calculating the cursor position in the destination image. https://bugzilla.gnome.org/show_bug.cgi?id=690646 2014-09-15 14:51:24 +0200 Peter Korsgaard * sys/v4l2/gstv4l2allocator.c: v4l2allocator: O_CLOEXEC needs _GNU_SOURCE Similar to 94f3d6fc / bz 709423 On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only defined when _GNU_SOURCE is specified, so do so. https://bugzilla.gnome.org/show_bug.cgi?id=736670 2014-09-15 18:11:37 +0200 Wim Taymans * gst/debugutils/gstcapssetter.c: capssetter: update to 1.0 transform_caps sematics In 1.0, we pass the complete caps to transform_caps to allow for better optimizations. Make this function actually work on non-simple caps instead of just ignoring the configured filter caps. 2014-09-08 14:06:00 +0200 Peter G. Baum * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: use WAVE_FORMAT_EXTENSIBLE for more than 2 channels https://bugzilla.gnome.org/show_bug.cgi?id=733444 2014-09-12 15:06:50 +0300 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Fix parsing of adtl chunks We have to skip 12 bytes of data for the chunk, and the data size passed to the sub-chunk parsing functions should have 4 bytes less than the data size. Also when parsing the sub-chunks, check if we actually have enough data to read instead of just crashing. https://bugzilla.gnome.org/show_bug.cgi?id=736266 2014-09-12 10:55:23 +0530 Sanjay NM * gst/udp/gstudpsrc.c: udp: include string.h for memcmp and memset https://bugzilla.gnome.org//show_bug.cgi?id=736528 2014-09-12 13:36:18 +0530 Anuj Jaiswal * gst/matroska/matroska-mux.c: matroskamux: don't bitwise OR the same flag twice https://bugzilla.gnome.org//show_bug.cgi?id=736543 2014-09-12 10:35:36 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: handle real audio 28_8 Fixes duplicate check for 14_4. https://bugzilla.gnome.org//show_bug.cgi?id=736543 2014-09-11 14:46:09 +0530 Anuj Jaiswal * gst/multifile/gstmultifilesink.c: multifilesink: don't OR the same flag twice https://bugzilla.gnome.org/show_bug.cgi?id=736462 2014-09-11 12:52:11 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: If the server reports "Accept-Ranges: none" don't try range requests 2014-09-10 09:50:45 +0200 Ognyan Tonchev * sys/v4l2/gstv4l2sink.c: v4l2sink: Unref pool after usage https://bugzilla.gnome.org/show_bug.cgi?id=736384 2014-09-09 19:03:50 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Don't rank it for now This will prevent the converter to be picked automatically in case someone implement dynamic converter selection support. I'd like this to be ranked only for known device, as it's hard to be sure a device is a converter suited for general purpose. Re-negotiation is also needed before we can rank it. https://bugzilla.gnome.org/show_bug.cgi?id=733607 2014-09-05 08:29:20 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: Detect bad drivers timestamps Even though the UVC driver do a great deal of effort to prevent bad timestamp to be sent to userspace, there still exist UVC hardware that are so buggy that the timestamp endup nearly random. This code detect and ignore timestamp from these drivers, making these camera usable. This has been tested on both invalid and valid cameras, making sure it does not trigger for valid cameras. https://bugzilla.gnome.org/show_bug.cgi?id=732910 2014-08-29 17:09:30 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Workaround driver that don't support REQBUFS(0) There is still around 18 drivers not yet ported to videobuf2. These driver don't support freeing buffetrs through REQBUFS(0) hence for these the memory type probing fails. In order to gain back our previous behaviour in presence of these, we implement a workaround that assuming MMAP is supported. Note that an allocator is only created for device with STREAMING support in the device capabilities. In such case one of MMAP, USERPTR and DMABUF is required. Though DMABUF came afterward, so is not an option and in practice none of these drivers will only do USERPTR. https://bugzilla.gnome.org/show_bug.cgi?id=735660 Also-by: Hans de Goede 2014-09-04 15:11:40 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2videodec.c: v4l2: Merge min_buffers_for* variable into one Reuse the same min_buffers variable for both capture and output, this reduce the length of lines and make the code more readable. https://bugzilla.gnome.org/show_bug.cgi?id=736072 2014-09-04 18:35:46 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: set min_latency for output device according to required minimum number of buffers Since we can get the minimum number of buffers needed by an output device to work, use it to set min_latency which will determine how many buffers are queued. https://bugzilla.gnome.org/show_bug.cgi?id=736072 2014-09-09 16:10:56 +0100 Tim-Philipp Müller * tests/check/elements/udpsrc.c: tests: udpsrc: add check to make sure multiple memory chunks are used 2014-09-09 15:55:18 +0100 Tim-Philipp Müller * tests/check/elements/udpsrc.c: tests: udpsrc: wait for buffers with GCond instead of sleeping Avoids half-second sleep for no reason. 2014-09-09 15:31:32 +0100 Tim-Philipp Müller * tests/check/elements/udpsrc.c: tests: udpsrc: split out socket setup 2014-09-09 13:46:56 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: more efficient memory handling Drop use of g_socket_get_available_bytes() which is not useful on all systems (where it returns the size of the entire buffer not that of the next pending packet), and is yet another syscall and apparently very inefficient on Windows in the UDP case. Instead, when reading UDP packets, use the more featureful g_socket_receive_message() call that allows to read into scattered memory, and allocate one memory chunk which is likely to be large enough for a packet, while also providing a larger allocated memory chunk just in case the packet is larger than expected. If the received data fits into the first chunk, we'll just add that to the buffer we return and re-use the fallback buffer for next time, otherwise we add both chunks to the buffer. This reduces memory waste more reliably on systems where get_available_bytes() doesn't work properly. In a multimedia streaming scenario, incoming UDP packets are almost never fragmented and thus almost always smaller than the MTU size, which is also why we don't try to do something smarter with more fallback memory chunks of different sizes. The fallback scenario is just for when someone built a broken sender pipeline (not using a payloader or somesuch) https://bugzilla.gnome.org/show_bug.cgi?id=610364 2014-09-09 12:15:43 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: rework memory allocation bits and ensure we always have two chunks of memories to read into First chunk is the likely/expected buffer size, second is as fallback in case the packet is larger in the end. Next step: actually use these. 2014-09-09 09:42:15 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: track max packet size and save allocator negotiated by GstBaseSrc 2014-09-08 16:15:05 +0100 Tim-Philipp Müller * gst/audiofx/audioecho.c: audioecho: fix example command line 2014-09-07 12:46:08 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: fix crash with certain videos This is a regression from 1.2 caused by the port to the pad flow combiner. https://bugzilla.gnome.org/show_bug.cgi?id=736192 2014-09-04 16:21:20 +0300 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-read-common.h: matroska-demux: Don't handle parse errors at the end of file as an error But only if they happen after the Matroska segment. https://bugzilla.gnome.org/show_bug.cgi?id=735833 2014-09-04 12:14:11 +0300 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Include redirection target in error messages Just giving the original URI can give the false impression that e.g. that one failed host name resolution, while actually the redirection target did. 2014-09-02 11:13:44 +0400 Andrei Sarakeev * gst/videomixer/videomixer2.c: videomixer: Fix synchronization if dynamically changing the FPS https://bugzilla.gnome.org/show_bug.cgi?id=735859 2014-09-02 13:52:43 +0530 Ravi Kiran K N * gst/smpte/gstsmpte.c: smpte: Check if input caps are the same and create output caps from video info This makes sure that also properties like the pixel-aspect-ratio are the same between both streams and that the output caps contain all fields necessary for complete video caps. https://bugzilla.gnome.org/show_bug.cgi?id=735804 2014-09-02 17:22:07 +0530 Vineeth T M * gst/imagefreeze/gstimagefreeze.c: imagefreeze: replace with gst_buffer_copy gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer. replacing the same with gst_buffer_copy as the functionality is same. https://bugzilla.gnome.org/show_bug.cgi?id=735880 2014-09-03 23:06:53 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: mark jpeg and png as parsed so avdec_mjpeg can be used too https://bugzilla.gnome.org/show_bug.cgi?id=735971 2014-09-03 11:46:13 +0530 Vineeth T M * ext/gdk_pixbuf/gstgdkpixbufdec.c: gdkpixbufdec: free query after use In gst_gdk_pixbuf_dec_setup_pool(), query is being allocated using gst_query_new_allocation(), but the same is not unreferenced hence calling gst_query_unref() after usage of query. https://bugzilla.gnome.org/show_bug.cgi?id=735950 2014-09-03 23:46:34 +1000 Jan Schmidt * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_types.c: qtdemux: Silence some warnings for normal file contents 2014-09-01 09:56:02 +0200 Nicolas Huet * gst/audioparsers/gstaacparse.c: aacparse: Fix parsing issue when the buffer does not have a complete ADTS/LOAS frame https://bugzilla.gnome.org/show_bug.cgi?id=735520 2014-09-02 09:09:49 +0300 Sebastian Dröge * ext/vpx/gstvp9dec.c: vp9dec: Get input width/height from the codec instead of the input caps They are reported properly by libvpx if the correct struct members are used. This also fixes handling of resolution changes without input caps changes. https://bugzilla.gnome.org/show_bug.cgi?id=719359 2013-10-22 18:49:22 +0100 Tom Greenwood * ext/vpx/gstvp8dec.c: vp8dec: Fix for handling resolution changes when decoding VP8 If the resolution changes in the bitstream without the input caps changing we would previously output corrupted video or crash. https://bugzilla.gnome.org/show_bug.cgi?id=719359 2014-09-02 00:55:17 -0300 Thiago Santos * ext/vpx/gstvp9dec.c: vp9dec: Fix segfault when a new caps is received Remember to unref the output caps when a new caps event is received as it should generate a new one based on the new caps. https://bugzilla.gnome.org/show_bug.cgi?id=734266 2014-09-02 00:54:35 -0300 Thiago Santos * tests/check/elements/vp8dec.c: tests: vp8dec: add test for caps renegotiation Check that vp8dec can properly accept a new caps when upstream changes it https://bugzilla.gnome.org/show_bug.cgi?id=734266 2014-08-05 10:34:39 +0200 Jose Antonio Santos Cadenas * ext/vpx/gstvp8dec.c: vp8dec: Reset output and input states when changing format https://bugzilla.gnome.org/show_bug.cgi?id=734266 2014-09-01 16:39:23 +0530 Vineeth T M * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Don't call gst_caps_unref() on template caps when already unreferenced Adding an extra condition while calling gst_caps_unref (templ) and replacing gst_caps_make_writable (gst_caps_ref (caps)) with gst_caps_copy (caps) in line 177, since the functionality is same. https://bugzilla.gnome.org/show_bug.cgi?id=735795 2014-08-29 12:01:27 +0200 Hans de Goede * sys/v4l2/gstv4l2object.c: v4l2: get_nearest_size: Fix "Unsupported field type" errors Most V4L2 ioctls like try_fmt will adjust input fields to match what the hardware can do rather then returning -EINVAL. As is docmented here: http://linuxtv.org/downloads/v4l-dvb-apis/vidioc-g-fmt.html EINVAL is only returned if the buffer type field is invalid or not supported. So upon requesting V4L2_FIELD_NONE devices which can only do interlaced mode will change the field value to e.g. V4L2_FIELD_BOTTOM as only returning half the lines is the closest they can do to progressive modes. In essence this means that we've failed to get a (usable) progessive mode and should fall back to interlaced mode. This commit adds a check for having gotten a usable field value after the first try_fmt, to force fallback to interlaced mode even if the try_fmt succeeded, thereby fixing get_nearest_size failing on these devices. https://bugzilla.gnome.org/show_bug.cgi?id=735660 2014-08-29 10:57:20 +0200 Hans de Goede * sys/v4l2/gstv4l2object.c: v4l2: get_nearest_size: Always reinit all struct fields on retry They may have been modified by the ioctl even if it failed. This also makes the S_FMT fallback path try progressive first, making it consistent with the preferred TRY_FMT path. https://bugzilla.gnome.org/show_bug.cgi?id=735660 2014-08-29 11:55:26 +0300 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Store size of data tag in a 64 bit integer locally too Otherwise we will clip the DS64 value of RF64 files to 32 bits again. 2014-08-29 11:53:23 +0300 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Use 64 bit scaling functions now that fact is a 64 bit integer 2014-08-27 18:55:18 +0200 Peter G. Baum * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: support rf64 format https://bugzilla.gnome.org/show_bug.cgi?id=735627 2014-08-28 13:48:50 -0600 Jason Litzinger * gst/multipart/multipartdemux.c: multipartdemux: Ensure caps before pad added. This stores the stream-start, sets caps, and then adds the pad, which ensures that the caps are set for the "pad-added" callback. https://bugzilla.gnome.org/show_bug.cgi?id=735626 2014-08-28 15:03:50 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: flvmux: Fallback to PTS if DTS is missing Fixing a regression introduce when fixing: https://bugzilla.gnome.org/show_bug.cgi?id=731352 2014-08-28 16:13:29 +0530 Vineeth T M * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Remove impossible error condition We return EOS after the first buffer, and GstPad will make sure now that we won't get any other buffer afterwards until a flush happens. No need to check for it ourselves. https://bugzilla.gnome.org/show_bug.cgi?id=735581 2014-08-28 13:53:23 +0530 Vineeth T M * ext/gdk_pixbuf/gstgdkpixbufdec.c: gdkpixbufdec: EOS and NOT_LINKED are no errors in general Don't post an error message for them but let upstream handle anything accordingly. https://bugzilla.gnome.org/show_bug.cgi?id=735564 2014-08-27 21:07:26 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Correctly offset timestamp The previous method would break AV sync in the case audio or video didn't start at the same point in running time. https://bugzilla.gnome.org/show_bug.cgi?id=731352 2014-08-27 20:56:12 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: flvmux: Save dts from buffer We no longer set dts in muxed buffer. This would lead to encoding tags with timestamp 0 instead of the timestamp of previous buffer. https://bugzilla.gnome.org/show_bug.cgi?id=731352 2014-07-28 20:58:59 -0400 Nicolas Dufresne * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Ensure Timestamp starts at 0 FLV documentation stipulates that timestamp must start at zero. In order to respect this rule, keep the first timestamp around and offset the timestamp from this value. This allow for longer recording time in presence of timestamp that does not start at 0 already. https://bugzilla.gnome.org/show_bug.cgi?id=731352 2014-06-06 23:17:52 -0400 Nicolas Dufresne * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvmux.c: flv: Tag timestamp are DTS not PTS The tags in FLV are DTS. In audio cases, and for many video format this makes no difference, but for AVC with B-Frames, PTS need to be computed from composition timestamp CTS, with PTS = DTS + CTS. https://bugzilla.gnome.org/show_bug.cgi?id=731352 2014-08-07 21:58:14 -0400 Youness Alaoui * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Allow rtp caps without clock-rate The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate. https://bugzilla.gnome.org/show_bug.cgi?id=734322 2014-08-18 14:05:52 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: avoid crashing on dash streams DASH/fragmented moov might have no samples as those are carried in moof fragments. Avoid crashing or failing the stream because of that. 2014-08-18 10:33:48 +0530 Ravi Kiran K N * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: examples: use 'post-messages' property instead of deprecated 'message' property https://bugzilla.gnome.org/show_bug.cgi?id=734979 2014-08-18 11:45:54 +0200 Víctor Manuel Jáquez Leal * gst/udp/gstudpsrc.c: udp: fix udpsrc documentation udpsrc gtk-doc documentation refers to sockfd and closefd properties which has been removed. This patch replaces those references to socket and close-socket respectively. https://bugzilla.gnome.org/show_bug.cgi?id=734987 2014-08-15 10:09:56 +1000 Jan Schmidt * gst/isomp4/gstqtmux.c: qtmux: Make the default timescale 1/1800 second The old default timescale of 1 millisecond produces irrational numbers for a lot of framerate/audio-packet-duration multiples. 1/1800 is a nicer number, as it tends to produce better fractions and therefore slightly higher accuracy overall 2014-08-15 01:17:27 +1000 Jan Schmidt * gst/matroska/matroska-demux.c: matroska: Use gst_video_guess_framerate() function Remove local framerate guessing function in favour of the new gst_video_guess_framerate() function. 2014-08-15 01:12:20 +1000 Jan Schmidt * gst/isomp4/Makefile.am: * gst/isomp4/qtdemux.c: qtdemux: Improve framerate calculation/guessing Change the way the output framerate is calculated to ignore the first sample (which is sometimes truncated in my testing) and use the new gst_video_guess_framerate() function to recognise common standard framerates better. Remove the code that was sorting the first 20 sample durations and then ignoring the result. 2014-08-14 16:36:44 +0300 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Use the best width/height/etc if downstream can handle that Before it was always using whatever downstream preferred, while the code and documentation claimed something different. https://bugzilla.gnome.org/show_bug.cgi?id=727180 2014-08-14 11:29:00 +0530 Ravi Kiran K N * gst/videomixer/videomixer2.c: videomixer: Avoid double free of VideoConvert https://bugzilla.gnome.org/show_bug.cgi?id=734764 2014-08-13 11:58:35 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: fix indentation 2014-08-13 11:54:26 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: un-break duration querying Commit 2b9493b5 broke this in two ways: a) we should only pass duration queries in TIME format upstream (or at least not those in DEFAULT or BYTE format), and b) we mustn't overwrite the default value of 'res' from TRUE to FALSE and not set it again later. This led to bogus durations being reported for FLV playback from file, because TIME queries would fail (as 'res' had been set to FALSE) and parsers then do a BYTE query as fallback and try to guesstimate something in return, which of course goes horribly wrong since the BYTE size returned is for the muxed file. 2014-08-13 13:23:10 +0300 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Allow any raw caps in passthrough mode, not just the ones we handle 2014-08-13 13:04:21 +0300 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Allow ANY capsfeatures, but only in passthrough mode When changing the properties to not be in passthrough mode anymore, we will only accept caps we can process ourselves, potentially causing a not-negotiated error. https://bugzilla.gnome.org/show_bug.cgi?id=720345 2014-08-12 11:34:30 +0100 Tim-Philipp Müller * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update for git 2014-08-12 11:33:56 +0100 Tim-Philipp Müller * configure.ac: configure: build ximagesrc again when checks succeed Third time lucky, hopefully. 2014-08-11 09:26:17 +0100 Tim-Philipp Müller * configure.ac: configure: fix x11 checks to be non-fatal again Must pass an action-if-not-found argument to PKG_CHECK_MODULES or it will error out when it can't find the module requested. Also fix AC_CHECK_LIB usage, extra libs argument was in the wrong place. 2014-08-07 17:12:38 +0300 George Kiagiadakis * gst/isomp4/qtdemux.c: qtdemux: forward DISCONT from upstream to the output streams This makes sense in DASH reverse playback, where the upstream dashdemux will download DASH segments in reverse order, but push their buffers forward to qtdemux and mark each segment start as DISCONT. This needs to be forwarded downstream to the parser/decoder, otherwise it won't work. https://bugzilla.gnome.org/show_bug.cgi?id=734443 2014-08-10 18:55:07 +0100 Tim-Philipp Müller * configure.ac: configure: use pkg-config to detect x11 and simplify checks AC_PATH_XTRA macro unnecessarily pulls in libSM and libICE. https://bugzilla.gnome.org/show_bug.cgi?id=731047 2014-08-10 12:30:07 +0200 Mark Nauwelaerts * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: adjust test data to avoid NAL chopping ... and correspondingly unexpected buffer sizes. 2014-08-09 14:22:42 +0200 Sebastian Rasmussen * ext/speex/gstspeexenc.c: speexenc: Improve annotation of internal function https://bugzilla.gnome.org/show_bug.cgi?id=734542 2014-08-08 12:54:30 +0200 Sebastian Rasmussen * gst/shapewipe/gstshapewipe.c: * tests/examples/shapewipe/shapewipe-example.c: shapewipe: Unref caps and element after usage https://bugzilla.gnome.org/show_bug.cgi?id=734478 2014-08-09 20:47:30 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: improve debug logging of fourccs If we can't show ASCII, at least show them in big endian order. 2014-08-09 20:46:04 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: add support for 'wma ' mapping as found in some ismv files e.g. To_The_Limit_720_2962.ismv 2014-08-09 18:31:20 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: add support for 'vc-1' mapping as found in some ismv files e.g. To_The_Limit_720_2962.ismv 2014-08-07 16:34:36 +0200 Sebastian Rasmussen * gst/rtp/gstrtph263ppay.c: rtph263ppay: Unref pad template caps after use Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435 2014-08-08 12:36:01 +0200 Sebastian Rasmussen * gst/videomixer/videomixer2.c: videomixer: Unref allowed caps after usage Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734474 2014-08-08 12:40:49 +0200 Sebastian Rasmussen * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Unref pad template caps after usage Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734475 2014-08-08 12:44:09 +0200 Sebastian Rasmussen * gst/debugutils/gstnavseek.c: navseek: Unref peer pad after usage Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734476 2014-08-08 12:29:52 +0200 Sebastian Rasmussen * gst/rtpmanager/gstrtpmux.c: rtpmux: Unref pad template caps after usage Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473 2014-08-05 11:47:39 +0200 Srimanta Panda * gst/rtp/gstrtph264pay.c: rtph264pay: append packetization mode parameter to SDP Append packetization-mode parameter to SDP description. Packetization mode signals the properties of an RTP payload type. https://bugzilla.gnome.org/show_bug.cgi?id=733556 2014-08-08 03:58:14 +1000 Jan Schmidt * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: isomp4/qtmux: Write correct file duration when gaps exist. When writing out a trak with an edit list, make sure the overall file duration is also updated to reflect the lengthening of the stream. Add some more debug to qtdemux to warn about streams that are longer than the file and get truncated. 2014-08-04 15:39:17 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Push the correct segment in TCP mode when seeking 2014-08-03 12:33:32 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: unbreak au aligned byte-stream payloading 2014-07-22 13:24:09 +0200 Srimanta Panda * gst/rtp/gstrtph264pay.c: rtph264pay: append profile-level-id to SDP Append profile-level-id to SDP if available. https://bugzilla.gnome.org/show_bug.cgi?id=733539 2014-07-31 18:47:49 +0200 Edward Hervey * Makefile.am: * common: Makefile: Add usage of build-checks step Allows building checks without running them 2014-07-31 09:53:53 -0400 Nicolas Dufresne * sys/ximage/ximageutil.c: ximagesrc: Fix warning about missing return value 2014-07-24 15:28:09 -0400 Nicolas Dufresne * sys/ximage/gstximagesrc.c: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: ximagesrc: Add missing return value to Buffer dispose function Depending ont he build, the method could return FALSE, hence never free the buffers, or already TRUE and lead to a crash: Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=733695 2014-07-28 16:49:16 +0200 Philippe Normand * gst/interleave/interleave.c: * tests/check/elements/interleave.c: interleave: set output caps layout to interleaved Set output caps layout independently from input caps layout which can be either non-interleaved or interleaved. https://bugzilla.gnome.org/show_bug.cgi?id=733866 2014-07-26 12:06:39 -0300 Thiago Santos * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: clear gcond 2014-07-25 14:30:33 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: Revert "v4l2bufferpool: Workaround elements not requesting any buffers" This was a tempory workaround, we should fix the encoders that do not negotatiate the amount of buffers they need. This reverts commit d03bcba3db15d06dbdea6b776a6f28ed2f03272a. 2014-07-08 14:31:59 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't share own pool if min exceed V4L2 capacity If the minimum required buffer exceed V4L2 capacity, don't share down pool. This allow support very high latency, like with x264enc default encoding settings. https://bugzilla.gnome.org/show_bug.cgi?id=732288 2014-07-25 17:42:20 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: v4l2object: query minimum required buffers for output Some v4l2 devices could require a minimum buffers different from default values. Rather than blindly propose a pool with min-buffers set to the default value, it ask the device using control ioctl. https://bugzilla.gnome.org/show_bug.cgi?id=733750 2014-07-23 18:40:10 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2sink.c: v4l2sink: use directly 'obj' instead of 'v4l2sink->v4l2object' https://bugzilla.gnome.org/show_bug.cgi?id=733616 2014-07-23 18:39:50 +0200 Aurélien Zanelli * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: v4l2: set debug messages according to device type and IO mode https://bugzilla.gnome.org/show_bug.cgi?id=733616 2014-05-24 19:02:59 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Remove is_active checks These checks are no longer required with recent change to the bufferpool. This should allow changing the configuartion, hence the way forward renegotiation support. https://bugzilla.gnome.org/show_bug.cgi?id=728268 2014-07-21 18:11:16 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_lang.c: qtdemux: fix language code parsing for 3-letter codes starting with 'a' And handle special value for 'unspecified' explicitly. https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap4/qtff4.html 2014-07-08 02:18:27 +0200 Nicola Murino * ext/jpeg/gstjpegenc.c: jpegenc: Add support for encoding from NV21 and NV12 https://bugzilla.gnome.org/show_bug.cgi?id=732870 2014-07-19 18:04:38 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.4.0 === 2014-07-19 17:20:34 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.4.0 2014-07-19 16:35:41 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2014-07-19 12:32:22 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: Update translations 2014-07-19 11:30:30 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Don't overwrite the first component with the alpha value for BGRx Instead leave the x component unset when filling the borders. https://bugzilla.gnome.org/show_bug.cgi?id=733380 2014-07-16 17:18:59 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Properly report in the CAPS query that we can convert ADTS<->RAW https://bugzilla.gnome.org/show_bug.cgi?id=733190 2014-07-13 16:05:56 +0200 Sebastian Rasmussen * gst/replaygain/gstrgvolume.c: rgvolume: Avoid taking unnecessary refs Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122 2014-07-13 16:04:23 +0200 Sebastian Rasmussen * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Avoid taking an unnecessary ref Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122 2014-07-15 16:59:06 +0200 Piotr Drąg * po/POTFILES.in: po: update POTFILES https://bugzilla.gnome.org/show_bug.cgi?id=733208 2014-07-11 13:35:10 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Fix copy threshold implementation We cannot allocate new buffer in acquire, otherwise the base class is not aware and get confused. Instead, copy in _process(). This leads to crash on finalize. Fixes regression, see https://bugzilla.gnome.org/show_bug.cgi?id=732912 === release 1.3.91 === 2014-07-11 11:38:57 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.3.91 2014-07-11 10:58:08 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2014-07-10 18:11:20 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: v4l2allocator: Use qdata instead of parenting to DmabufMemory Parenting V4l2Memory to DmabufMemory was in conflict with recent optimization in DmabufMemory to avoid dup(), and didn't work with memory sharing. Instead, use a qdata and it's destroy notify. https://bugzilla.gnome.org/show_bug.cgi?id=730441 2014-07-11 08:52:39 +0200 Sebastian Dröge * po/da.po: * po/de.po: * po/hu.po: * po/id.po: * po/pl.po: * po/ru.po: * po/uk.po: * po/vi.po: po: Update translations 2014-07-08 17:50:47 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Workaround elements not requesting any buffers This is a workaround for element that don't request buffers when they should. https://bugzilla.gnome.org/show_bug.cgi?id=732288 2014-07-06 11:27:36 +0200 Sebastian Rasmussen * tests/icles/videocrop-test.c: tests: fix pipeline leak in videocrop test Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976 2014-07-06 11:26:46 +0200 Sebastian Rasmussen * tests/examples/rtp/client-rtpaux.c: examples: client-rtpaux: Release reference to parent when done Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976 2014-07-10 17:19:42 +0100 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: fix query leak https://bugzilla.gnome.org/show_bug.cgi?id=733003 2014-07-10 12:10:45 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Return not-negotiated if we got no caps or caps negotiation failed And do it always, not inside a g_return_val_if_fail(). See https://bugzilla.gnome.org/show_bug.cgi?id=732939 2014-07-08 13:34:28 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2src.c: v4l2src: Ensure internal pool activation Before we would hit an assertion "'gst_buffer_pool_is_active (bpool)' failed" if the internal pool was not used to push buffer downstrea, hence not given to the baseclass. https://bugzilla.gnome.org/show_bug.cgi?id=732912 2014-07-04 20:22:10 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: videomixer: fix double unlock in segment seek segment code path We only want to unlock if we push an event downstream and jump to done_unlock label afterwards. We would also unlock in case of a segment seek and then unlock again later, and nothing good can come of that. (This code looks a bit dodgy anyway though, shouldn't it also bail out with FLOW_EOS here in case of a segment seek scenario, just without the event?) 2014-07-04 19:45:55 +0100 Tim-Philipp Müller * tests/check/elements/qtmux.c: tests: qtmux: suppress glib criticals caused by testing deprecated dts methods 2014-07-04 03:21:30 +0200 Sebastian Rasmussen * gst/avi/gstavidemux.c: * gst/wavparse/gstwavparse.c: avidemux, wavparse: Print invalid fourcc in hex Previously this was printed as characters which caused later processing of the error message to sometimes warn about non-UTF-8 characters. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732714 2014-07-03 15:21:18 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Pool might be NULL in decide allocation If special stride is needed and downstream don't support VideoMeta, pool might be NULL in order to let the baseclass create a generic pool­. This would lead to assertion with on Exynos with: gst-launch-1.0 -v filesrc location=mov ! qtdemux ! h264parse ! \ v4l2video8dec ! fakesink https://bugzilla.gnome.org/show_bug.cgi?id=732707 2014-07-03 15:29:54 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Handle FD error during poll This will ensure we fail earlier if something unrecoverable happens. 2014-07-03 15:28:45 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Wait before polling if queue is empty In kernel before 3.17, polling during queue underrun would unblock right away and trigger POLLERR. As we are not handling POLLERR, we would endup blocking in DQBUF call, which won't be unblocked correctly when going to NULL state. A deadlock at start caused by locking error in libv4l2 was also seen before this patch. Instead, we wait until the queue is no longer empty before polling. https://bugzilla.gnome.org/show_bug.cgi?id=731015 2014-07-02 16:01:47 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix for mikey api change 2014-06-30 10:29:54 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2: fix probing and enumeration of stepwise frame sizes The code enumerating STEPWISE framesizes would start from (min_w, min_h) and then add (step_w, step_h) to get the next framesize. However, it should really allow any width from min_w to max_w with step_w and same for heights. Secondly, we would add and probe each individual stepped frame size to the caps as separate structure, which would lead to hundreds if not thousands of structs ending up in the probed caps. Use integer ranges with steps instead. This was particularly noticable with the Raspberry Pi Cam. https://bugzilla.gnome.org/show_bug.cgi?id=724521 https://bugzilla.gnome.org/show_bug.cgi?id=732458 https://bugzilla.gnome.org/show_bug.cgi?id=726521 2014-06-27 11:33:06 +0100 Daniel Drake * sys/v4l2/gstv4l2object.c: v4l2object: drop workaround for misbehaving TRY_FMT This workaround from 2011 was causing 25 S_FMT ioctls to be sent to my UVC webcam from under gst_v4l2_object_get_caps as it probes all the formats. In total, this adds up to about 5 seconds of execution time, or a 10 second delay while starting up cheese. These ioctls come from a workaround from 2011 where TRY_FMT might make changes to hardware settings, so S_FMT was used to restore the original config: https://bugzilla.gnome.org/show_bug.cgi?id=649067 The driver bug is now assumed fixed. Remove the workaround to fix the long startup delay. https://bugzilla.gnome.org/show_bug.cgi?id=732326 2014-07-01 12:50:31 +0100 Vincent Penquerc'h * gst/videomixer/videomixer2.c: videomixer: reset QoS on segment event https://bugzilla.gnome.org/show_bug.cgi?id=732540 2014-07-01 15:14:34 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: send gap events instead of segment tricks This fixes missing frames from being time skipped. https://bugzilla.gnome.org/show_bug.cgi?id=732372 2014-06-30 00:00:32 +0200 Sebastian Dröge * tests/check/elements/rtpsession.c: rtpsession: Fix memory leaks in unit test 2014-06-29 23:55:19 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpbin.c: rtpbin: Don't leak caps 2014-06-29 20:02:14 +0200 Sebastian Dröge * ext/pulse/pulsesrc.c: pulsesrc: Fix compiler warning when compiling with G_DISABLE_ASSERT 2014-06-29 19:59:53 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Fix compiler warning when compiling with G_DISABLE_ASSERT 2014-06-29 19:57:57 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Fix compiler warnings when compiling with G_DISABLE_ASSERT 2014-06-29 19:54:44 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: Fix compiler warnings when compiling with G_DISABLE_ASSERT 2014-06-29 17:05:13 +0100 Tim-Philipp Müller * ext/pulse/pulsedeviceprovider.c: pulse: fix compiler warnings when compiling with -DG_DISABLE_ASSERT Compiler complains about uninitialised variables in the impossible 'default' code path in device provider source/sink switch-case. 2014-06-29 17:03:17 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2deviceprovider.c: v4l2: fix compiler warnings when compiling with -DG_DISABLE_ASSERT Compiler complains about uninitialised variables in the impossible 'default' code path in device provider source/sink switch-case. 2014-06-28 17:40:45 +0100 Tim-Philipp Müller * tests/check/elements/matroskaparse.c: tests: matroskaparse: fail on errors and disable pull mode test Actually look for error messages on the bus and fail if there is one before the EOS message. Disable pull mode test which is pointless as long as matroskaparse only supports push mode (pull mode support has not been ported over to 1.0). 2014-06-28 17:37:23 +0100 Tim-Philipp Müller * gst/matroska/matroska-parse.c: matroskaparse: don't error out if there's not enough data in the adapter gst_matroska_parse_take() would return FLOW_ERROR instead of FLOW_EOS in case there's less data in the adapter than requested, because buffer is NULL in that case which triggers the error code path. This made the unit test fail (occasionally at least, because of a bug in the unit test there's a race and it would happen only sporadically). 2014-06-28 16:53:58 +0200 Sebastian Dröge * gst/videomixer/videomixerorc-dist.c: * gst/videomixer/videomixerorc-dist.h: videomixer: Update dist generated ORC files 2014-06-28 16:48:13 +0200 Sebastian Dröge * gst/videomixer/gstcms.c: * gst/videomixer/gstcms.h: * gst/videomixer/videoconvert.c: * gst/videomixer/videoconvert.h: * gst/videomixer/videomixerorc.orc: videomixer: Update videoconvert code from -base And also rename the remaining symbols to prevent conflicts during static linking. https://bugzilla.gnome.org/show_bug.cgi?id=728443 2014-06-28 13:01:46 +0100 Tim-Philipp Müller * gst/autodetect/gstautovideosrc.c: autovideosrc: use videotestsrc as fallback element instead of fakesrc fakesrc doesn't announce video caps, so most video pipelines will just error out with not-negotiated if a fallback element is created. 2014-06-28 12:44:31 +0100 Tim-Philipp Müller * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautodetect.h: autoaudiosrc: use audiotestsrc as fallback element instead of fakesrc fakesrc doesn't announce audio caps, so most audio pipelines will just error out with not-negotiated if a fallback element is created. === release 1.3.90 === 2014-06-28 11:21:15 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.3.90 2014-06-28 11:08:33 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2014-06-26 14:52:57 -0400 Olivier Crête * ext/pulse/Makefile.am: * ext/pulse/plugin.c: * ext/pulse/pulsedeviceprovider.c: * ext/pulse/pulsedeviceprovider.h: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2deviceprovider.c: * sys/v4l2/gstv4l2deviceprovider.h: Rename GstDeviceMonitor to GstDeviceProvider 2014-06-24 09:14:40 +0530 Ravi Kiran K N * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/videobox.c: videobox: Add unit test https://bugzilla.gnome.org/show_bug.cgi?id=732144 2014-06-16 11:35:39 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Declare as Compositor in 'klass' 2014-06-26 13:50:19 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: fix speex caps Decoder complains about "notification: Invalid mode encountered. The stream is corrupted" though, even if it works, so there's probably something wrong with the generated codec headers. 2014-06-26 13:43:33 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: fix speex in FLV Speex in FLV is always mono @ 16kHz, see http://download.macromedia.com/f4v/video_file_format_spec_v10_1.pdf section E.4.2.1: "If the SoundFormat indicates Speex, the audio is compressed mono sampled at 16 kHz, the SoundRate shall be 0, the SoundSize shall be 1, and the SoundType shall be 0" Also see https://bugzilla.gnome.org/show_bug.cgi?id=683622 2014-06-26 05:19:57 +1000 Jan Schmidt * gst/isomp4/qtdemux.c: isomp4: Add object type id and fourcc for DTS/DTS-HD Enables playback for files with DTS audio tracks. Also add an extra AC-3 variant fourcc from Nero 2014-03-13 10:35:30 +0100 David Fernandez * gst/videomixer/videomixer2.c: videomixer2: Solve segmentation fault when src caps are configured Change function pointers to NULL while holding the lock to avoid race conditions https://bugzilla.gnome.org/show_bug.cgi?id=701110 2014-06-25 14:34:21 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve SR packet handling Implement 3 different cases for handling the SR: 1) we don't have enough timing information to handle the SR packet and we need to wait a little for more RTP packets. In that case we keep the SR packet around and retry when we get an RTP packet in the chain function. 2) the SR packet has a too old timestamp and should be discarded. It is labeled invalid and the last_sr is cleared. 3) the SR packet is ok and there is enough timing information, proceed with processing the SR packet. Before this patch, case 2) and 1) were handled in the same way, resulting that SR packets with too old timestamps were checked over and over again for each RTP packet. 2014-06-24 10:47:33 +0100 Tim-Philipp Müller * tests/check/elements/udpsink.c: tests: add udpsink test to check client add/remove 2014-06-23 16:13:27 +0100 Tim-Philipp Müller * tests/check/elements/udpsink.c: tests: port udpsink tests to 1.0 They all seem a bit pointless though. 2014-06-23 19:55:29 -0400 Olivier Crête * gst/avi/gstavimux.c: avimux: Add UYVY format 2014-06-06 11:20:21 +0200 Miguel París Díaz * gst/rtpmanager/gstrtpssrcdemux.c: gstrtpssrcdemux: manage ssrc of RTCP RR packets https://bugzilla.gnome.org/show_bug.cgi?id=731324 2014-06-23 20:53:50 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Update offset after parsing adtl chunk Otherwise we will parse it over and over again without ever getting past it. https://bugzilla.gnome.org/show_bug.cgi?id=731533 2013-07-07 20:18:27 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: remove legacy code for passing a window handle "have-ns-view" and the "embed" property was kept in 0.10 for backwards compatibility but it's no longer used in favor of the GstVideoOverlay interface https://bugzilla.gnome.org/show_bug.cgi?id=703753 2014-06-22 19:36:14 +0200 Sebastian Dröge * configure.ac: Back to development 2014-06-22 19:26:03 +0200 Sebastian Dröge * gst/matroska/matroska-read-common.c: matroskademux: Don't call GST_DEBUG_OBJECT() and other macros with non-GObject objects It will crash with latest GLib GIT and was never supposed to work before either. === release 1.3.3 === 2014-06-22 18:08:03 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.3.3 2014-06-22 17:36:28 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2014-06-22 14:24:24 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: Update translations 2014-06-21 01:32:03 +0100 Tim-Philipp Müller * ext/pulse/pulsedevicemonitor.c: * sys/v4l2/gstv4l2devicemonitor.c: pulse, v4l2: update for device "klass" -> "device-class" rename 2014-06-20 12:21:05 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: optimisation: avoid unnecessary memory ref/unrefs We know the buffer will stay valid and we will also not modify the buffer, we just want to send out the data. 2014-06-19 14:59:48 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: avoid some unnecessary run-time type checks 2014-06-19 16:17:23 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: pass the stream id when asking for crypto params This way the app can choose different parameters for each stream. 2014-05-20 14:58:07 -0700 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add support for key length parameters This patch adds supports for the incoming key management parameters for encryption and authentication key lengths. It also adds a new signal request-rtcp-key that allows the user to provide the crypto parameters and key for the RTCP stream. https://bugzilla.gnome.org/show_bug.cgi?id=730473 2014-06-19 15:25:01 +0200 Wim Taymans * gst/rtp/gstrtpvp8depay.c: vp8depay: fix header size checking Use a different variable name to make it clear that we are calculating the header size. Correctly check that we have enough bytes to read the header bits. We were checking if there were 5 bytes available in the header while we only needed 3, causing the packet to be discarded as too small. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595 2014-05-20 12:39:31 +0200 Guillaume Desmottes * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag Similarly to what we did with the DELTA_UNIT flag, this patch propagates the DISCONT flag to the first RTP packet being used to transfer a DISCONT buffer. https://bugzilla.gnome.org/show_bug.cgi?id=730563 2014-05-06 17:42:14 +0200 Guillaume Desmottes * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag Downstream elements may be interested knowing if a RTP packet is the start of a key frame (to implement a RTP extension as defined in the ONVIF Streaming Spec for example). We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from upstream and propagate it to the *first* RTP packet outputted to transfer this buffer. https://bugzilla.gnome.org/show_bug.cgi?id=730563 2014-05-20 13:58:20 +0200 Guillaume Desmottes * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: gstrtpmp4gpay: propagate the GST_BUFFER_FLAG_DISCONT flag Propagate the DISCONT flag to the first RTP packet being used to transfer a DISCONT buffer. https://bugzilla.gnome.org/show_bug.cgi?id=730563 2014-05-20 13:58:20 +0200 Guillaume Desmottes * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: propagate the GST_BUFFER_FLAG_DISCONT flag Propagate the DISCONT flag to the first RTP packet being used to transfer a DISCONT buffer. https://bugzilla.gnome.org/show_bug.cgi?id=730563 2014-06-18 15:03:25 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: don't leak flow combiner 2014-06-18 14:38:55 +0100 Tim-Philipp Müller * gst/rtp/gstrtpj2kpay.c: rtpjp2kpay: pre-allocate buffer-list of the right size 2014-06-18 14:34:09 +0100 Tim-Philipp Müller * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: pre-allocate buffer list of the right size 2014-06-18 14:19:28 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmp4vpay.c: rtpmp4vpay: pre-allocate buffer list of the right size 2014-06-18 13:44:31 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: allocate bitreader on the stack 2014-06-18 13:29:47 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: post error message on bus on error and don't use g_message() 2014-06-18 13:20:44 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: couple of minor optimisations Pre-allocate buffer list of the right size to avoid re-allocs. Avoid plenty of double runtime cast checks and re-doing the same calculation over and over again in rtp_vp8_calc_payload_len(). Only call gst_buffer_get_size() once. 2014-06-18 08:10:03 +0100 Tim-Philipp Müller * gst/rtp/gstrtpgstpay.c: rtpgstpay: pre-allocate buffer list of the right size To avoid re-allocs. 2014-06-18 07:52:05 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: rtph264pay: pre-allocate bufferlist of the right size To avoid unnecessary re-allocs. 2014-06-16 20:15:43 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: * tests/check/elements/rtp-payloading.c: rtph264pay: push single buffer directly, no need to wrap it in a bufferlist No point in a buffer list if we just have one single buffer to push. Fix up unit test to handle that case as well. 2014-06-16 15:35:12 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: rtpvrawpay: make chunks per frame configurable Bit of a misnomer because it's really chunks per field and not per frame, but we're going to ignore that for the time being. 2014-06-16 14:52:16 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: rtpvrawpay: remove unused variables 2014-06-16 14:44:27 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: pre-allocate buffer lists of sufficient size Avoids unnecessary reallocs when appending buffers to the bufferlist. 2014-06-16 13:51:03 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: micro-optimise variable access in inner loop Store some values that don't change during the execution of the inner loops locally, so the compiler knows that too. 2014-06-16 13:38:47 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: use buffer lists Collect buffers to send out in buffer lists instead of pushing out single buffers one at a time. For HD video each frame might easily add up to a couple of thousand packets, multiply that by the frame rate and that's a lot of push() and sendmsg() calls per second. A good reason to push out buffers as early as possible is latency, so we don't accumulate the whole frame in a single buffer list, but instead push it out in a few chunks, which is hopefully a reasonable compromise. 2014-06-16 16:40:07 +0100 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: udp: improve element descriptions for dynudpsink and multiudpsink 2014-06-16 16:17:16 +0100 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: udp: remove suppression of compiler warnings for deprecated GLib API Not needed any more. 2014-06-17 13:16:27 +0530 Ravi Kiran K N * gst/videobox/gstvideobox.c: videobox: Fix caps negotiation issue Make sure that if AYUV is received it will detect that it can produce both RGB and YUV formats Signed-off-by: Ravi Kiran K N https://bugzilla.gnome.org/show_bug.cgi?id=725248 2014-06-16 12:02:41 +0100 Tim-Philipp Müller * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: fix double frees Fix double-frees introduced to fix another coverity report. CID 1223053 2014-06-13 10:12:07 +0100 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: dynudpsink: return FLUSHING when sendto got canceled, not an error 2014-06-13 09:52:03 +0100 Tim-Philipp Müller * sys/oss/gstosshelper.c: oss: simplify probed caps before returning them Exposes all formats in the first structure if the rest is the same for all of them. 2014-06-13 09:45:28 +0100 Tim-Philipp Müller * sys/oss/gstosshelper.c: oss: make sure 16-bit formats are before 8-bit formats in probed caps Probe supported formats in order of desirability rather than in what order they may happen to be in the formats bitmask. Fixes accidentally exposure of 8-bit formats in caps before 16-bit formats (in case where U16 was not supported S8 might be listed before S16). https://bugzilla.gnome.org/show_bug.cgi?id=706884 2014-06-12 16:36:24 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Cleanly handle v4l2_allocator_new failure 2014-06-12 11:24:15 +0100 Vincent Penquerc'h * gst/rtp/gstrtptheoradepay.c: rtptheordepay: fix leaks Coverity 1212163 2014-06-12 11:16:08 +0100 Vincent Penquerc'h * gst/rtp/gstrtpg729pay.c: rtpg729pay: leak fixes Coverity 1212159 2014-06-12 11:11:38 +0100 Vincent Penquerc'h * gst/rtp/gstrtph263pay.c: rtph263pay: fix leak Coverity 1212157 2014-06-12 10:43:53 +0100 Vincent Penquerc'h * gst/rtp/gstrtph263pay.c: rtph263pay: fix leaks Coverity 1212149 2014-06-12 10:31:47 +0100 Vincent Penquerc'h * gst/rtp/gstrtpdvpay.c: rtpdvpay: catch failures to map buffer Coverity 1139741 2014-06-11 17:43:42 +0100 Vincent Penquerc'h * gst/multipart/multipartdemux.c: multipartdemux: guard against having no MIME type The code would previously crash trying to insert a NULL string into a hash table. It does seem a little broken that indexing is done by MIME type and not by index though, unless the spec says there cannot be two parts with the same MIME type. https://bugzilla.gnome.org/show_bug.cgi?id=659573 2014-06-10 15:42:14 -0400 Nicolas Dufresne * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: Send stream-start event This event was not sent. Send it before caps, this requires the pad to be parented. This removes warning like: "Got data flow before stream-start event". Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731475 2014-06-10 15:33:33 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: avoid looping indefinitely in broken svq3 files Abort if an atom with size 0 is read from within the svq3 stsd atoms https://bugzilla.gnome.org/show_bug.cgi?id=726512 2014-06-10 10:52:23 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: add const where appropriate 2014-06-09 10:39:20 +0200 Edward Hervey * ext/speex/gstspeexenc.c: speexenc: add missing va_end in variadic function Coverity 1139944 2014-06-09 10:04:38 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Attempt upstream seek first If we have an upstream element that can handle the seek (such as rtmpsrc), try to do that first before attempting it ourself. 2014-06-04 11:34:27 +0100 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: do not include codec_data on raw audio caps If the wav header contains an extended chunk, we want to keep the codec_data field, but not for raw audio. This fixes some elements (such as adder) from failing to intersect raw audio caps which would otherwise be intersectable. 2014-06-05 09:38:29 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Query duration upstream first Upstream elements (like rtmpsrc) might be able to provide the duration more accurately than flvdemux. Especially with index-less vod files 2014-05-30 19:37:57 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Cleanup poll method and retry on EINTR/EAGAIN https://bugzilla.gnome.org/show_bug.cgi?id=731015 2014-03-06 16:37:51 +0100 Jan Alexander Steffens (heftig) * gst/flv/gstflvdemux.c: flvdemux: set RESYNC buffer flag when bridging large PTS gaps So downstream gets notified when this happens. https://bugzilla.gnome.org/show_bug.cgi?id=725903 2014-06-03 17:59:32 -0400 Olivier Crête * tests/check/elements/rtprtx.c: rtprtx: Reset state on each iteration Otherwise it didn't wait for the test to finish before checking the results. https://bugzilla.gnome.org/show_bug.cgi?id=728501 2014-05-09 14:22:42 +0100 Tim-Philipp Müller * gst/matroska/matroska-read-common.c: matroskademux: don't leak doctype string in error code path CID 1212145. 2014-05-20 08:20:42 +0200 Edward Hervey * ext/vpx/gstvp9enc.c: vp9enc: Don't dereference NULL checks CID #1197703 2014-05-20 08:23:06 +0200 Edward Hervey * ext/vpx/gstvp8enc.c: vp8enc: Don't dereference NULL variable CID #1139838 2014-05-30 14:32:42 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: upstream handles seek if fragmented and on time segment Otherwise we can reject seeks on local files that contain fragmented-like atoms like 'mvex'. Also improve a message log https://bugzilla.gnome.org/show_bug.cgi?id=730722 2014-05-30 16:43:44 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: h264depay: make sure we call handle_nal for each NAL Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure we correctly extract the SPS and PPS. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730999 2014-05-07 14:09:06 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Add custom sticky event to contain the HTTP request and response headers This can be useful to e.g. get cookie information downstream. https://bugzilla.gnome.org/show_bug.cgi?id=729707 2014-05-26 19:47:39 -0300 Thiago Santos * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: remove stream last flow return GstPad already stores that information https://bugzilla.gnome.org/show_bug.cgi?id=709224 2014-05-26 19:37:46 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: remove last flow return from stream struct It is already stored on GstPad on core https://bugzilla.gnome.org/show_bug.cgi?id=709224 2014-05-26 19:19:45 -0300 Thiago Santos * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Use GstFlowCombiner Use the flow combiner to have the standard combination results and avoid repeating the same code https://bugzilla.gnome.org/show_bug.cgi?id=709224 2014-05-26 13:21:25 -0300 Thiago Santos * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: matroskademux: use GstFlowCombiner Use the flow combiner to have the standard combination results and avoid repeating the same code https://bugzilla.gnome.org/show_bug.cgi?id=709224 2014-05-26 13:04:10 -0300 Thiago Santos * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: use GstFlowCombiner Removes flow return combination code to use the newly added GstFlowCombiner 2014-05-23 17:53:00 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: use GstFlowCombiner Removes the common code to combining flow returns to let it be handled by core gstutils' GstFlowCombiner https://bugzilla.gnome.org/show_bug.cgi?id=709224 2014-05-26 10:59:55 -0400 Julien Isorce * sys/v4l2/gstv4l2sink.c: v4l2sink: implement gstvideosink.show_frame instead of gstbasesink.render It allows to show preroll frame. Especially it allows to update the frame when seeking in PAUSED state. https://bugzilla.gnome.org/show_bug.cgi?id=722303 2014-05-26 10:59:06 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2sink.c: v4l2sink: Cleanup old pad alloc declaration 2014-05-26 12:34:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2sink.c: v4l2bufferpool: Copy already queued buffer This is required as during preroll we pass the first buffer twice, hence already queued. It is also useful, to allow filters replaying a previous rendered buffers. This will require 1 more buffer in sink if last-sample is enabled, since the last sample will not be the same as the currently queued buffer. https://bugzilla.gnome.org/show_bug.cgi?id=722303 2014-05-24 20:20:07 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/v4l2_calls.c: v4l2bufferpool: Port to bufferpool flush_start/stop method Port the buffer pool to use the new flush_start/flush_stop virtual methods added to GstBufferPool. https://bugzilla.gnome.org/show_bug.cgi?id=727611 2014-05-25 17:40:58 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update 2014-05-25 16:54:18 +0200 Piotr Drąg * po/POTFILES.in: po: update POTFILES https://bugzilla.gnome.org/show_bug.cgi?id=726556 2014-05-24 23:51:58 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't queue all the buffers before dequeueing first For output device, we where queuing all the buffers, and then we would dequeue one. This means we only have 1 buffer for the pipeline, no matter the size of the queue. Instead, start dequeued when min_latency is reached. Eventually, this the min_latency should also be affected by control MIN_BUFFERS_FOR_OUTPUT (use by encoders). 2014-05-24 23:49:19 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Simply read back the config to update the query It's easy to get the min/max outdate when hacking decide allocation. In order to avoid this, simply read back the choosen value from the config. 2014-05-24 23:31:24 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2src.c: v4l2: Cleanup and fix calculation of latency Calculation of num_buffers (the max latency in buffers) was up-side-down. If we can allcoate, then our maximum latency match pool maximum number of buffers. Also renamed it to max latency. Finally introduced a min_latency for clarity. 2014-05-24 20:00:14 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/v4l2_calls.c: Revert "v4l2bufferpool: Port to bufferpool flush_start/stop method" This reverts commit 2e0fb42e868fc9f6d98b028def80a3e953527307. Conflicts: sys/v4l2/gstv4l2allocator.c sys/v4l2/gstv4l2bufferpool.c sys/v4l2/gstv4l2videodec.c 2014-05-24 18:56:32 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix configuration of other_pool and importation case Fix the choice of min/max, don't override the min/max with own pool selected size, correct other_pool is_active check, start from other_pool config when configuring the other pool and finally validate the configuration. 2014-05-24 18:45:30 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Use proposed allocator as default 2014-05-24 18:43:28 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Fix USERPTR map flags We need to map READ only for output and write only for capture, we where doing the opposite. This fixing USERPTR with glimagesink https://bugzilla.gnome.org/show_bug.cgi?id=730698 2014-05-24 11:16:35 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: parse tkhd transformation matrix and add tags if appropriate Handle the transformation matrix cases where there are only simple rotations (90, 180 or 270 degrees) and use a tag for those cases. This is a common scenario when recording with mobile devices https://bugzilla.gnome.org/show_bug.cgi?id=679522 2014-05-23 19:10:21 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Prevent num_queued from going negative 2014-05-23 18:25:49 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: don't stop if loop returned FLUSHING The decodeing thread returning flushing isn't an error, we should simply try starting the task again. If it's actually flushing, it will stop again by itself. 2014-05-23 17:54:20 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Handle early task stop 2014-05-23 17:28:13 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Handle gst_pad_start_task() failure 2014-05-23 17:19:07 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Add trace for FLUSH_START/STOP handling 2014-05-23 17:18:16 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Fix use of atomic value 2014-05-23 17:01:53 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Improve debugging No need to use obj->element, the pool now have a significant name. Also don't warn if flushing. 2014-05-23 17:01:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Fix handle_frame error handling 2014-05-23 15:56:24 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Add a trace when _start() is called 2014-05-23 15:56:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Add debug assert to detect calls in the wrong state 2014-05-23 15:55:26 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Reset count when stopped 2014-05-23 15:55:08 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2allocator: Return a GstFlowReturn instead of boolean in alloc 2014-05-23 15:17:27 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't leak config structure 2014-05-23 14:12:10 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/v4l2_calls.c: v4l2bufferpool: Port to bufferpool flush_start/stop method 2014-05-23 03:00:50 -0300 Thiago Santos * gst/isomp4/fourcc.h: * gst/isomp4/qtdemux.c: qtdemux: add tag mappings for _swr, _mak and _mod tags swr -> Application name mak -> device manufacturer mod -> device model 2014-05-20 17:37:49 -0400 Nicolas Dufresne * sys/ximage/gstximagesrc.c: ximagesrc: Fix ximage leaks when buffer has more then one ximage From time to time, when the image_pool list has more then 1 element and I suppose at start, all but 1 pooled ximage are leaked. This is due to broken algorithm in gst_ximagesink_src_ximage_get(). There was also a risk of use after free for the case where the ximage size has changed. https://bugzilla.gnome.org/show_bug.cgi?id=728502 2014-05-21 13:23:27 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.3.2 === 2014-05-21 13:06:35 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * common: * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect-build.stamp: * docs/plugins/inspect.stamp: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.3.2 2014-05-21 12:19:39 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2014-05-21 10:51:10 +0200 Sebastian Dröge * common: Automatic update of common submodule From 211fa5f to 1f5d3c3 2014-05-20 08:23:06 +0200 Edward Hervey * ext/vpx/gstvp8enc.c: vp8enc: Don't dereference NULL variable CID #1139838 2014-05-20 08:20:42 +0200 Edward Hervey * ext/vpx/gstvp9enc.c: vp9enc: Don't dereference NULL checks CID #1197703 2014-05-19 11:26:46 +0200 Sebastian Dröge * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Explicitly cast enum "subtype" to its "supertype" gstv4l2bufferpool.c:608:18: error: implicit conversion from enumeration type 'enum _GstV4l2BufferPoolAcquireFlags' to different enumeration type 'GstBufferPoolAcquireFlags' [-Werror,-Wenum-conversion] params.flags = GST_V4L2_POOL_ACQUIRE_FLAG_RESURECT; ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 2014-05-19 11:24:06 +0200 Sebastian Dröge * gst/goom/tentacle3d.c: goom: Use fabs() instead of abs() to calculate the floating point absolute value tentacle3d.c:268:7: error: using integer absolute value function 'abs' when argument is of floating point type [-Werror,-Wabsolute-value] if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * G_PI))) { ^ 2014-05-19 11:21:36 +0200 Sebastian Dröge * gst/debugutils/tests.c: debugutils: Properly calculate the difference with unsigned types tests.c:161:16: error: taking the absolute value of unsigned type 'unsigned long' has no effect [-Werror,-Wabsolute-value] t->diff += labs (GST_BUFFER_TIMESTAMP (buffer) - t->expected); 2014-05-16 17:46:30 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Handle flush while in start_streaming We need to handle the case where a flush occure while the streaming thread is being brought up. In this case, the flushing state of the poll object is cleared. To solve this, we simply set the capture poll to flushing again, this way we know the thread will exit. The decoder streamlock is used to synchronize with handle frame. 2014-05-16 16:44:37 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Don't trace twice the same message 2014-05-15 11:25:50 -0700 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: rtspsrc: always use a random ssrc for the internal session Use a random SSRC different than 0 for the internal session SSRC. https://bugzilla.gnome.org/show_bug.cgi?id=730212 2014-05-16 16:52:25 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: update last_activity when sending RTP Also update last_activity when doing something with the internal source to make sure don't timeout early. See https://bugzilla.gnome.org/show_bug.cgi?id=730217 2014-05-15 18:08:53 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Cleanup M2M properties M2M devices were sharing the same properties as src and sink. Most of these made no sense. This patch reduces the number of propeties and makes io-mode clearer by having capture-io-mode and output-io-mode. This also accidently fixed a bug in gstv4l2transform io-mode code, where the capture io-mode could not be set. https://bugzilla.gnome.org/show_bug.cgi?id=729591 2014-05-15 17:39:39 +0200 Benjamin Gaignard * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Update pool limit with hardware requiremenst If the driver need more buffers than requested by the config, update the pool min/max values. The minimum value for the pool could be provided either by the driver or by the pool. This is best effort for drivers that don't support CID V4L2_CID_MIN_BUFFERS_FOR_CAPTURE. https://bugzilla.gnome.org/show_bug.cgi?id=730200 2014-05-15 10:44:29 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Handle start_streaming error https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-15 10:39:40 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Print the flow return causing the loop to leave https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-15 10:31:40 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Don't lock the decoder when stopping task That src pad task may need to take the lock when being pulled down. takeing that lock can lead to a deadlock. https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 17:18:52 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Don't leak pool if activation failed https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 17:18:35 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Split flush in start/stop_streaming This allow calling start streaming later for capture device. Currently it breaks in dmabuf-import because downstream is holding a buffer that will only be released after stream-start. https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 15:12:26 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Flush buffer pools on flush stop https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 13:28:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix use of atomic active marker https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 13:05:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Don't deactivate otherpool We should not stop the otherpool unless we also stop our own pool, otherwise it will never get restarted. https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 12:33:58 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Also update num_buffers for import cases https://bugzilla.gnome.org/show_bug.cgi?id=730207 2014-05-14 13:42:25 -0700 Aleix Conchillo Flaqué * gst/rtpmanager/gstrtpbin.c: rtpbin: update rtp encoder/decoder docs Use %u in RTP encoder/decoder pads to match other rtpbin pads. https://bugzilla.gnome.org/show_bug.cgi?id=730146 2013-12-27 11:55:18 +0200 George Kiagiadakis * tests/check/elements/rtpsession.c: tests/check: rtpsession: test internal sources timing out 2013-12-26 17:30:42 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: remove unused if branch 1) sources that have sent BYE in the past cannot be senders, since they would have timed out to being receivers in the meantime... 2) sources that have sent BYE are now being removed earlier inside this function 2013-12-26 17:29:42 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: cleanup sources that have sent BYE 2013-12-26 17:24:51 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: unify nested if clauses 2013-12-26 17:21:44 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: timeout internal sources that are inactive for a long time and send BYE 2014-05-13 12:25:04 -0700 Aleix Conchillo Flaqué * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: don't stop looping if event found in the queue If we are inserting a packet into the jitter queue we need to keep looping through the items until the right position is found. Currently, the code stops as soon as an event is found in the queue. Regarding events, we should only move packets before an event if there is another packet before the event that has a larger seqnum. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730078 2014-04-17 13:04:00 +0000 Adrien SCH * gst/matroska/matroska-mux.c: matroskamux: fix the memory leak of language attribute https://bugzilla.gnome.org/show_bug.cgi?id=728418 2014-05-13 13:44:20 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix regression in offset extrapolation When extrapolating the offset, we need to use the extrapolate stride rather then the base stride. This should fix support for format with more then two planes (I420, Y42B, etc). 2014-05-12 18:03:18 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Use default VideoInfo for frame operation When doing frame operation, we need to use the default VideoInfo and let the frame API read the video meta in order to get the stride and offset right. Currently we where using the specialized VideoInfo which reflects what the HW is setup to. 2014-05-12 17:23:19 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2src: minor GValue handling optimisation in probing code 2014-05-12 17:20:14 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2src: avoid lists with one single framerate in probed caps Simplify framerate field if possible, so we don't end up with e.g. framerate = (fraction) { 30/1 }. Maybe the helper function should be moved to core, but we can do this later. 2014-05-12 16:56:35 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Fix leak of palette_data in error cases CID #1212151 2014-05-12 16:53:32 +0200 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Free node_header in error cases CID #1212134 2014-05-12 13:46:01 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Don't use WARNING for not-linked flow return Pollutes debug logs for no reason. It's only an error if all pads return not-linked 2014-05-12 13:45:06 +0200 Edward Hervey * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Skip unknown tags in push-mode We add a new mode (SKIP) in push-mode to skip tags that we don't known about Partially fixes https://bugzilla.gnome.org/show_bug.cgi?id=670712 2014-05-10 09:14:33 +0200 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Add support for variable block size files and remove dead code This dead code wasn't used since the 1.0 port and would need to be modified heavily for variable block size support. https://bugzilla.gnome.org/show_bug.cgi?id=729894 2014-05-09 12:14:23 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Fix NULL check copy paste error CID 1212129 2014-05-09 12:11:54 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Fix potential deadlock due to missing break CID 1212131 2014-05-09 18:01:28 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: increment accepted packets after loss When we detect a lost packet, expect packets with higher seqnum on the input. Also update the unit test. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729524 2014-05-04 11:12:54 -0600 Jason Litzinger * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: Add new test case. 2014-05-09 16:14:21 +0200 Wim Taymans * tests/check/elements/shapewipe.c: shapewipe: no need to activate pads Activation will happen in the state change 2014-05-09 12:10:04 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't leak config structure this fixes a leak of the config structure and take care of making sure caps can't reach ref 0 before we are done doing our check. CID 1212144 2014-05-09 12:08:11 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Remove uneeded cast for code clarity 2014-05-09 11:56:52 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2pool: Fix leak of config structure in error case CIDs 1212167 and 1212167 2014-05-09 11:51:26 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix use of unitilized pool pointer CID #1212173 2014-05-09 16:48:58 +0200 Eric Trousset * gst/isomp4/qtdemux.c: qtdemux: don't respond to a position query in BYTE format with a TIME position https://bugzilla.gnome.org/show_bug.cgi?id=729553 2014-05-09 14:22:42 +0100 Tim-Philipp Müller * gst/matroska/matroska-read-common.c: matroskademux: don't leak doctype string in error code path CID 1212145. 2014-05-06 13:37:47 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Readback pool config if used within the baseclass 2014-05-06 12:58:59 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Replace miss-use of crop meta in favour of proper offset This moves away from copying information and store everything inside the GstVideoInfo structure. The alignement exposed by v4l2 api is now handled using proper offset. 2014-05-06 12:55:30 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: v4l2object: Style fix 2014-05-05 12:38:33 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Reset imported buffer size with expected size This ensure that the buffer pool won't always discard buffer with these memory when they are released. 2014-05-05 12:37:43 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Reset flushed group This ensure that a flushed group memory are the same size as when they where originally allocated / imported. 2014-05-05 12:07:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Get number of allocated buffers from allocator The value of num_allocated buffer would get confused when buffer are being discarded. 2014-05-05 12:06:44 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: v4l2allocator: Add a method to read number of allocated group 2014-05-04 20:23:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Improve debugging 2014-05-04 19:51:48 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Ensure we don't re-enqueue buffer during flush 2014-05-04 19:13:37 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Initilialize debug category 2014-05-04 16:11:09 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Fix libv4l2 support Need to include config.h, otherwise we endup directly using the ioct/mmap/munmap calls and need to vall v4l2_munmap. 2014-05-01 13:04:08 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Set the flags on the object We where not setting the probed flags on the allocator, which mean even if CREATE_BUFS was supported on some driver, it would endup being ignored. 2014-04-29 16:49:52 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Re-enqueue buffer at stream start 2014-04-29 16:06:00 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: There is not group on error 2014-04-29 14:56:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Handle FLUSH_STOP event 2014-04-29 13:05:41 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2videodec.c: v4l2bufferpool: Acquire cannot return a buffer from another pool Return a buffer from an otherpool has unwanted side effects that lead to leaks and prevents deactivating the pool. Instead, we change the _process() API so it can replace the internal buffer with the buffer from the downstream pool. This implied moving from _fill() to _create() method in the src. 2014-04-29 13:00:32 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Remove unreached acquire code The acquire is done in _prepare now. 2014-04-29 12:57:08 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Sanetize buffer refount handling Buffer refcounting is a bit hard, because of the duality between CAPTURE and OUTPUT mode. In the long term, we should consider having two seperate pool instead of this mess. At least state should be better kept this way. 2014-04-29 12:48:04 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Add more traces 2014-04-28 08:48:26 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: v4l2-allocator: Add S to REQBUFS/CREATE_BUFS enum All enum that has REQBUFS and CREATE_BUFS where missing S, which was confusing since they are supposed to match with associcated ioctl name. This also fixes the yet unused CAN_REQUEST flag check. 2014-04-18 17:51:07 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Enabled QoS 2014-04-18 17:02:50 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: v4l2: Fixup USERPTR/DMABUF capture support 2014-04-18 14:45:00 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Improve selecton of min/max in decide allocation 2014-04-18 13:09:00 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Update config if meta is missing Rather then hard failure, we should update the config with the meta option we need and return false. 2014-04-11 17:10:11 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: Add DMABUF and USERPTR importation 2014-04-17 21:45:58 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Valid FD are bigger or equal to zero 2014-04-16 17:04:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't leak downstream pool in propose_allocation parse_nth_allocation_pool() give a ref on the pool, we need to unref it when done. 2014-04-14 12:19:39 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: Introduce DMABUF_IMPORT IO mode 2014-04-10 16:26:34 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/gstv4l2bufferpool.c: v4l2: Add dmabuf export support This can be enabled sing io-mode=dmabuf. This will enabled mmap base drivers to export the buffers as dmabuf. 2014-04-16 15:51:03 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2allocator.c: v4l2allocator: Guaranty queued state integrety Because of the buf in videobuf2, dqbuf may leave the DONE flag being, which would implied that the buffer is queued. As this has been broken for 4 years, simply guaranty the state flags integrity when doing qbuf/dqbuf. See https://patchwork.linuxtv.org/patch/23641/ 2014-04-15 17:31:42 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Implement open/close 2014-04-15 16:43:41 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Ensure output pool is configured 2014-04-15 16:43:15 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2transform.h: v4l2transform: Check if caps have changes before asserting In set_caps, now checks if caps actually changed and succeed if they didn't change. 2014-04-15 16:41:46 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Ensure pool is configured 2014-04-08 18:54:09 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Always set a size when deciding allocation 2014-04-08 18:20:25 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Improved decide allocation Improve decide allocation so it properly configure both local and downstream buffer pools. Also read back the pool config if it was changed to to driver limitations. 2014-04-15 13:30:02 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Do not pre-configure the pool Pre-configuring the pool is error prone, since it may hide a configuration failure and endup with a pool that is not configured the way it should (e.g. no video meta, wrong queue size, etc.) 2014-04-15 13:23:33 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Preserve downstream minimum even in RW 2014-04-15 13:20:12 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Turn cropmeta into a custom option Turn crop meta into a custom option and make sure it's there is needed. 2014-04-09 12:53:19 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2bufferpool: Early catch short allocation Catch short allocation after saving the format. This is not a catch all, but should catch most of the miss-behaving drivers when doing S_FMT/G_FMT and avoid potential crash. 2014-04-04 22:46:40 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: Port to use GstV4l2Allocator 2014-04-04 22:35:48 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2allocator.c: * sys/v4l2/gstv4l2allocator.h: * sys/v4l2/v4l2_calls.h: Implement V4l2 Allocator This goal of this allocator is mainly to allow tracking the memory. Currently, when a buffer memory has been modified, the buffer and it's memory is disposed and lost until the stream is restarted. 2014-04-16 16:35:49 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't advertise crop meta Currently we advertise crop meta, but not element handle support this meta. 2014-04-08 18:18:57 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Setup pool already send element error 2014-04-08 18:17:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Workaround decoder that set num_planes to 0 in the format Some well known decoder wrongly set num_planes to 0 in their format instead of one. In this case we would endup with no size when deciding buffer allocation. 2014-04-08 17:34:19 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Ensure size before configuring the pool 2014-04-04 22:38:05 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: v4l2object: Set minimum buffers to 2 All the element requires at least two buffers. This is not used for RW mode. 2014-04-04 22:37:14 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: v4l2object: Remove unused MAX_BUFFERS define 2014-04-04 22:36:37 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't hardcode min/max use default instead 2014-04-10 17:49:41 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Install PROP_CAPTURE_IO_MODE with right ID 2014-04-08 18:54:50 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: decide_allocation returns a boolean 2014-04-10 17:49:29 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Install PROP_CAPTURE_IO_MODE with right ID 2014-03-27 13:21:25 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Add propose_allocation This should remove 1 copy between the decoder and the transform. 2014-03-27 13:20:53 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: v4l2: Move propose allocation to v4l2object 2014-03-20 17:26:05 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Fixup caps query 2014-03-20 15:31:22 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2transform.c: v4l2transform: Setup cropping if needed 2014-03-19 17:25:16 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2.c: v4l2transform: Expose BGRA and ARGB formats 2014-03-18 17:33:38 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Ensure output pool is activated That pool may be different then the internal pool. 2014-03-16 19:11:16 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Ensure internal buffer pools actication 2014-03-16 11:36:19 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Move subinstance subclass init near other init 2014-03-15 18:56:51 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2transform.c: v4l2transform: Stop stream before closing the devices. 2014-03-15 16:53:54 +0000 Víctor Manuel Jáquez Leal * sys/v4l2/gstv4l2transform.c: v4l2transform: copy metdata 2014-03-04 18:31:27 -0500 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2transform.c: * sys/v4l2/gstv4l2transform.h: Implement GstV4l2Transform Implement a v4l2 element that wraps HW video converters. 2014-03-27 18:41:07 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: Probe for CREATE_BUFS in order to correctly set pool min/max In order to correctly set the pool min/max, we need to probe for CREATE_BUFS ioctl. This can be done as soon as the format has been negotiated using a count of 0. 2014-03-25 15:21:03 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2videodec.c: v4l2: Move capture eos handling in _process() Now that we might be copying out buffer (e.g. downstream don't support video meta bug we need it) we need to move the EOS handling inside the process method. 2014-03-25 10:49:39 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix support for planar format in 1 v4l2 mplane So far we where only setting saving the first plane stride in the meta. This was leading to wrong values in GstVideoMeta. 2014-03-19 17:52:08 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Cleanly fail if set_format is never called 2014-03-19 17:00:56 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: Expose RGB32 formats with and without alpha As soon a the alpha component can be set, we can expose the RGB32 and BGR32 format as ARGB and BGRA as long we can deterministically set the alpha padding value. 2014-03-18 15:49:49 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: Correctly check if video meta is needed Correctly check if video meta is needed. In buffer pool, trust need_video_meta flag in order to decide if configuration should succeed. 2014-03-18 15:45:18 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix tiled stride request Fix stride request for tiled format and improve logging. 2014-03-18 11:53:57 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2object: Ensure video and crop meta are enabled if needed In certain cases we cannot live without video meta and/or crop meta being enabled in our internal buffer pool. Ensure this is always the case, regardless of having support for allocation query. 2014-03-16 18:39:32 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Ensure internal pool are activated 2014-03-16 17:01:10 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Check that pool where allocated before flushing them Upon error, the pools might not have been allocated yet, hence we should not try and flush them (even though we still want to make sure the processing thread is fully stopped). 2014-03-16 16:55:43 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2sink.c: v4l2bufferpool: Enforce activation outside of process Enforce pool being activate from before calling pool process. This should help catching basic errors in the usage of buffer pool. 2014-03-16 12:44:14 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: don't use own pool if downstream don't support video meta 2014-03-14 00:31:32 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Use obj->n_v4l2_planes for correct number of planes Buffer pool was guessing wrongly the number of planes rather then reading the value from obj->n_v4l2_planes. This was causing format YU12 (I420) to fail upon check. 2014-03-07 16:39:29 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fix handling of contiuous vs non-contiguous support The complex mechanic to try and choose the right thing did not work. Instead, simply probe the non-contiguous format first and then the contiguous one. This is in fact very low overhead, as there is a relatively small number of pixel format supported by each devices. 2014-04-15 15:07:23 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2videodec.h: v4l2: Add initial support for alignment and cropping 2014-03-13 19:24:51 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2videodec.c: v4l2object: Rename setup_format() method into acquire_format() The setup_format() was confusing since it does not set anything, in fact it reads the setup from the driver and save it. 2014-03-13 18:21:41 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Move type declaration to the top 2014-03-12 18:07:38 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Protect NULL pool while going to READY When the pipeline fails early, the pool might be unset before the processing thread has run once. Add protection against that. 2014-03-12 18:01:09 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Fail cleanly if pixel format is unkown or not raw video Certain decoder has been found to not choose a format automatically. Running v4l2videodec on these would assert. This patch will make it fail cleanly instead. 2014-03-12 17:56:18 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Clear the input state pointer after unref If caps are set again, we have a risk od returning from set_format with a input_state pointing to dead memory. Clearing the pointer after unref fix this issue. 2014-03-12 17:11:16 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: handle stop being called without flush Uppon certain downstream error, stop() is called without a flush(). This mean that the streaming thread may still be running even though unlock has been called. Now calling flush to reset the decoder state if we are processing. 2014-03-06 18:13:14 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Default to template in caps query 2014-03-11 14:23:32 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Ensure processing thread has stopped when draining 2014-03-11 14:01:27 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Don't drain if processing thread is inactive 2014-05-08 09:49:24 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Clean up all pending operations from libsoup before unreffing our context When we cancel connection attempts and similar things, there are still some operations pending on our main context from the GCancellables. We should let them all run before unreffing our context, otherwise we leak file descriptors. Unfortunately this requires libsoup 2.47.0 or newer as earlier versions steal our main context from us and we can't use it for cleanup later without assertions and funny crashes. Based on a patch by Dmitry Shatrov . https://bugzilla.gnome.org/show_bug.cgi?id=663944 2014-05-07 15:49:39 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: fix compilation of souphttpsrc test for libsoup 2.40 for real https://bugzilla.gnome.org/show_bug.cgi?id=727329 2014-05-07 13:23:50 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: fix compilation of souphttpsrc test for libsoup 2.40 SOUP_CHECK_VERSION was only added in 2.41, but we only depend on 2.40. https://bugzilla.gnome.org/show_bug.cgi?id=727329 2014-05-07 00:58:15 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: skip PICTURE headers without any image data Fixes warning if the image length is 0. 2014-05-06 09:22:18 +0000 Руслан Ижбулатов * configure.ac: configure: use X11 detection macro from common https://bugzilla.gnome.org/show_bug.cgi?id=729621 2014-04-30 11:13:12 +0200 Guillaume Desmottes * gst/rtp/README: rtp/README: update pipelines to work with 1.0 - Use gst-libav encoders/decoders instead of gst-ffmpeg - gstrtpjitterbuffer -> rtpjitterbuffer - gst-launch-0.10 -> gst-launch-1.0 - Add 'videoconvert' element - xvimagesink -> autovideosink https://bugzilla.gnome.org/show_bug.cgi?id=729247 2014-05-05 14:41:05 +0100 Vincent Penquerc'h * gst/matroska/ebml-write.c: matroska: rejig test to avoid undefined shift behavior Coverity 1195121, 1195120 2014-05-05 14:33:38 +0100 Vincent Penquerc'h * ext/vpx/gstvp9enc.c: vp9enc: do not dereference NULL pointer Coverity 1197703 2014-05-05 14:32:06 +0100 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: ensure we don't dereference a NULL pointer while working out the codec ID. Coverity 1195148 2014-05-05 12:07:25 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2.c: v4l2: minor fix for closing the fd The fd returned by open() could theoretically be 0 as well. Coverity CID 1211823. 2014-05-04 20:23:29 -0400 Olivier Crête * tests/check/elements/rtpaux.c: * tests/check/elements/rtprtx.c: rtpaux/rtprtx: Make tests non-racy Fix the raciness by iterating on a condition instead of using the gmainloop. Don't use the EOS as the target, otherwise the retransmission of the last packets are lost. Also count the retranmissions requests that are dropped. Check the condition before blocking on the GCond https://bugzilla.gnome.org/show_bug.cgi?id=728501 2014-05-04 22:32:54 -0400 Olivier Crête * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxreceive.h: rtprtxreceive: Wait until timeout to clear association requests If two streams request a retranmission for the same SSRC, ignore the second one if the first oen is less than one second old, otherwise time out the first one and ignore the second. 2014-05-04 18:59:33 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * tests/check/elements/rtpmux.c: rtpmux: Always let upstream chose the ssrc if it wishes 2014-05-04 13:37:46 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: avoid stall by corrupted seqnum accounting 2014-05-04 01:14:33 -0400 Olivier Crête * ext/pulse/pulsedevicemonitor.c: * ext/pulse/pulsedevicemonitor.h: pulsedevicemonitor: Index are per facility, not global So need to keep the type of device in the device object 2014-05-04 01:13:24 -0400 Olivier Crête * ext/pulse/pulsedevicemonitor.c: pulsedevicemonitor: pa_subscription_event_t are enums, not flags Coverity 1195132 2014-05-02 22:42:54 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2devicemonitor.c: v4l2devicemonitor: Port to use GstV4l2Iterator https://bugzilla.gnome.org/show_bug.cgi?id=727925 2014-05-02 21:38:30 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: v4l2: Use single pass iterator for M2M probe Instead of having each M2M class do their own probing, use the GstV4l2Iterator and probe all devices in a single pass. https://bugzilla.gnome.org/show_bug.cgi?id=727925 2014-05-02 16:55:05 -0400 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/v4l2-utils.c: * sys/v4l2/v4l2-utils.h: v4l2: Add a common device enumerator This will allow removing code duplication (hence bugs duplication). https://bugzilla.gnome.org/show_bug.cgi?id=727925 2014-03-16 11:38:07 +0100 Nicolas Dufresne * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: v4l2videodec: Simplify sub-instanciation mechanism Simplify sub-instanciation by defining an absract type and using subtype class and instance init callback. This also fixes a bug where the template pads get initialized too late. https://bugzilla.gnome.org/show_bug.cgi?id=727925 2014-05-02 18:18:26 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2.c: v4l2: Cleanup plugin registration There is no plan to introduce special sources for jpeg, te v4l2src works fine for this. https://bugzilla.gnome.org/show_bug.cgi?id=727925 2014-05-03 18:30:20 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * tests/check/elements/rtpcollision.c: rtpsession: Keep local conflicting addresses in the session As we now replace the local RTPSource on a conflict, it's no longer possible to keep local conflicts in the RTPSource, so they instead need to be kept in the RTPSession. Also fix the rtpcollision test to generate multiple collisions instead of one by change the address, as otherwise we detected that it was a single one. 2014-05-03 20:48:30 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.3.1 === 2014-05-03 18:02:23 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * gst/audiofx/audiopanoramaorc-dist.c: * gst/deinterlace/tvtime-dist.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videomixer/videomixerorc-dist.c: * win32/common/config.h: Release 1.3.1 2014-05-03 18:02:01 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2014-05-03 17:22:45 +0200 Sebastian Dröge * po/da.po: * po/de.po: * po/el.po: * po/hu.po: * po/id.po: * po/lv.po: * po/nb.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sr.po: * po/zh_CN.po: po: Update translations 2014-05-03 11:43:21 +0200 Sebastian Dröge * tests/check/elements/shapewipe.c: shapewipe: Send initial events after setting the elements to PLAYING Otherwise we send them too early, and setting the elements to PLAYING afterwards will drop all the events again. 2014-05-03 10:15:03 +0200 Sebastian Dröge * common: Automatic update of common submodule From bcb1518 to 211fa5f 2014-05-02 17:12:29 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Set segment position to the stop position of the buffer 2014-05-02 17:10:18 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Properly report errors before stopping the srcpad task 2014-05-02 17:02:02 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Error out if we have no caps yet 2014-05-02 14:49:27 +0100 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: avoid dividing by a 0 blockalign This can be 0. In that case, do not try to cut off the last few bytes from the last buffer. Coverity 1146971 2014-05-02 14:25:01 +0100 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: do not use uinitialized clut on error If we're missing part of the clut, do not try to use it. It seems very likely the break was meant to break out of the switch rather than from the loop. Coverity 1139878 2014-05-02 14:18:08 +0100 Vincent Penquerc'h * gst/flx/gstflxdec.c: flxdec: fix integer overflow Coverity 1139859 2014-05-02 14:09:02 +0100 Vincent Penquerc'h * gst/rtp/gstrtpqdmdepay.c: rtpqdmdepay: remove pointless check Besides, the pointer was dereferenced earlier anyway. Coverity 1139853 2014-05-02 14:06:25 +0100 Vincent Penquerc'h * gst/rtsp/gstrtspsrc.c: rtspsrc: remove duplicate test item was dereference previously. While there, reorder some test for faster early out. Coverity 1139844 2014-05-02 14:02:52 +0100 Vincent Penquerc'h * ext/vpx/gstvp8enc.c: vp8enc: guard against NULL pointer dereference Coverity 1139838 2014-05-02 13:59:07 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: fix theoretical integer overflow This code isn't actually used at the moment, unsure if I should just remove it or not... Coverity 1139811 2014-05-02 13:33:02 +0100 Vincent Penquerc'h * gst/matroska/ebml-write.c: matroska: blindly fix writing variable length negative values Spotted while fixing something else in the area. Nothing calls this with a negative value. 2014-05-02 13:29:33 +0100 Vincent Penquerc'h * gst/matroska/ebml-write.c: matroska: do not lose the top bits when writing a > 32 bit value Coverity 1139806 2014-05-02 12:10:26 +0100 Vincent Penquerc'h * gst/videofilter/gstvideoflip.c: videoflip: add missing break in switch Coverity 1139755 2014-05-02 11:39:39 +0100 Vincent Penquerc'h * gst/matroska/matroska-parse.c: matroska: do not try to call gst_pad_query_default on a NULL pad gst_matroska_parse_query can be called explicitely with a NULL pad. If we reach this point with a NULL pad, fail the query. Coverity 1139715 2014-05-02 11:28:01 +0100 Vincent Penquerc'h * gst/matroska/matroska-parse.c: matroska: do not return GST_FLOW_OK if we did not get a buffer Coverity 1139714 (which will likely come back in another guise, as the _read_init call can have a failing _map) 2014-05-02 11:20:33 +0100 Vincent Penquerc'h * gst/matroska/ebml-write.c: matroska: catch failure to map buffer Avoids dereferencing NULL. Coverity 1139712 2014-05-02 10:52:44 +0100 Vincent Penquerc'h * gst/avi/gstavimux.c: avimux: refuse caps with invalid framerate Coverity 1139701 2014-05-02 10:21:09 +0100 Vincent Penquerc'h * gst/isomp4/gstqtmux.c: qtmux: handle 0 size packets without dividing by 0 Coverity 1139691 2014-05-02 09:49:32 +0100 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: guard against invalid frame size to avoid division by 0 Coverity 1139690 2014-05-02 09:49:17 +0100 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: trivial typo fix 2014-05-02 09:43:54 +0100 Vincent Penquerc'h * ext/speex/gstspeexdec.c: speexdec: remove dead code fpp can never equal 0 here, or the loop would not execute at all. Zero fpp was possible before as the loop condition was allowing it specifically, but no more. Coverity 1139681 2014-05-02 09:41:19 +0100 Vincent Penquerc'h * sys/oss4/oss4-property-probe.c: oss4: remove dead mixer code This was partly removed in the port to 0.11. If still needed, it's still there in the history. Coverity 1139687 2014-05-02 09:33:51 +0100 Vincent Penquerc'h * sys/oss4/oss4-property-probe.c: oss4: fix a missing unlock and a return-only-when-assertions-enabled Spotted on the side while looking at another issue. 2014-03-07 17:31:29 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Correctly map RGB32 format In v4l2 specification, RGB32 has the alpha, or pading, first, not last. See http://linuxtv.org/downloads/v4l-dvb-apis/packed-rgb.html . https://bugzilla.gnome.org/show_bug.cgi?id=540941 2014-04-30 18:06:40 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: remove dead code For 8 bit width, we always have depth==gdepth==width==8. Coverity 1139678 2014-04-30 17:48:53 +0100 Vincent Penquerc'h * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: remove dead code A stricer check is already done earlier, and integer overflows do not seem possible here. Coverity 1139675 2014-04-30 14:50:44 +0100 Vincent Penquerc'h * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: guard against pathological "no space" condition Even if one woul hope one pixel can fit in a MTU, ensure we do not overwrite a buffer if this is not the case. Spotted while looking at Coverity 1208786 2014-04-30 11:52:10 +0100 Vincent Penquerc'h * gst/rtp/gstrtpjpegdepay.c: rtpjpegdepay: sanity check for NULL qtable Can happen (at least in crafted stream) Coverity 1208778 2014-04-30 01:08:41 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: pass on tags from upstream if there are any Don't just ignore upstream tags from e.g. an ID3 tag before the .wav data, pass them on downstream. https://bugzilla.gnome.org/show_bug.cgi?id=729223 2014-04-29 16:26:53 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: optimize timer update When we are not doing retransmission, we just need to find the current seqnum so we can stop when we found it. 2014-04-29 16:21:44 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: rtpjitterbuffer: small optimizations Small optimizations where we can. Add some more debug. 2014-04-29 16:16:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: signal when next_seqnum changed Signal the pushing thread when the next_seqnum changed and we might be able to push a buffer now. 2014-04-29 16:12:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: only signal event when head changed After adding a buffer, only signal the pushing thread when the head buffer changed or else we cause a useless wakeup. 2014-04-29 15:29:31 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: rework packet insert Rework the packet queue so that the most common action (insert a packet at the tail of the queue) goes very fast. Report if a packet was inserted at the head instead of the tail so that we can know when to retry _pop or _peek. 2014-04-29 16:38:55 +1000 Matthew Waters * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: * tests/examples/gtk/gstgtk.c: gl/examples: move to -bad - fix all the compiler errors - give them their own gl directory 2014-04-28 14:41:10 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtpvraw: use plane pointers when needed Pack/unpack planar formats to/from the first plane. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729058 2014-04-28 09:47:10 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Remember if a redirect is permanent or not and store it in the query 2014-04-27 21:57:31 -0400 Nicolas Dufresne * gst/goom/config_param.c: goom: Remove french comment saying to prefix functions All non-static function in this file are already prefixed with goom_. 2014-04-28 00:20:47 +0100 Tim-Philipp Müller * gst/goom/filters.c: goom: fix compilation on ios-arm7-10.9 and osx-x86_64 uint is not a standard type, and the rest of the code uses Uint which is locally typedefed to unsigned int. https://bugzilla.gnome.org/show_bug.cgi?id=729067 2014-04-27 18:29:11 -0400 Luis de Bethencourt * gst/goom/filters.c: goom: fix undefined behaviour of left-shift Don't left-shift into the sign bit, the result is undefined and potentially an overflow could flip the sign. 2014-04-26 20:51:36 -0400 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: check return from qt_demux_video_caps Now qtdemux_video_caps() can return NULL. We need to check this return before using it's value. https://bugzilla.gnome.org/show_bug.cgi?id=728987 2014-04-26 23:35:17 +0100 Tim-Philipp Müller * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/speex/gstspeexdec.c: * gst/avi/gstavidemux.c: * gst/avi/gstavisubtitle.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/multifile/gstmultifilesink.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstudpsrc.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/wavparse/gstwavparse.c: * sys/osxaudio/gstosxaudiosink.c: docs: remove outdated and pointless 'Last reviewed' lines from docs They are very confusing for people, and more often than not also just not very accurate. Seeing 'last reviewed: 2005' in your docs is not very confidence-inspiring. Let's just remove those comments. 2014-04-25 17:58:42 -0400 Luis de Bethencourt * gst/isomp4/qtdemux.c: qtdemux: initialize caps pointer to null Make sure the caps pointer returns initialized when using it in qtdemux_parse_tree (). https://bugzilla.gnome.org/show_bug.cgi?id=728987 2014-04-22 17:07:38 +1000 Jan Schmidt * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Clear last_pt on flush-stop. Otherwise, we don't recheck the buffer caps for clock-rate properly on the next chain. 2014-04-22 17:29:02 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix compiler warning gstdeinterlace.c: In function 'gst_deinterlace_output_frame': gstdeinterlace.c:1537:57: error: 'pattern.length' may be used uninitialized in this function [-Werror=maybe-uninitialized] This actually is always initialized before it is used there, but let's just silence gcc here. 2014-04-21 15:58:45 +0100 Vincent Penquerc'h * gst/rtpmanager/gstrtpmux.c: rtpmux: fix buffer list drop check While porting to 0.11, the check was mistakenly made constant, instead of testing for the return value of process_buffer_locked. Coverity 1139663 2014-04-21 13:44:15 +0100 Vincent Penquerc'h * gst/matroska/matroska-read-common.c: matroska: fix content encoding scope validity check It's 3 bits, and http://matroska.org/technical/specs/index.html says it can't be 0. Coverity 1139660 2014-04-21 13:34:37 +0100 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: fix PAR fraction sanity check It was checking par_num twice, and never par_denum. Coverity 1139634 2014-04-21 13:32:40 +0100 Vincent Penquerc'h * gst/udp/gstmultiudpsink.c: multiidpsink: warn when setsockopt fails This doesn't seem to be fatal, but it's good to let the user know in the logs. Coverity 1139630 2014-04-21 13:27:24 +0100 Vincent Penquerc'h * gst/interleave/deinterleave.c: interlace: catch failure to create audio info from caps Coverity 1139627, 1139628 2014-03-13 09:37:48 +0100 Göran Jönsson * gst/rtp/gstrtph264pay.c: gstrtph264pay: Reset sps pps variable when state change. Reset last_spspps and sps/pps arrays when state transition GST_STATE_CHANGE_PAUSED_TO_READY. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726015 2014-04-18 11:11:14 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: improve EOS handling Make a new method to disable the jitterbuffer buffering. Rework the update_estimated_eos() method. Calculate how much time there is left to play. If we have less than the delay of the jitterbuffer, we disabled buffering because we might never be able to fill the complete jitterbuffer again. If we receive an EOS event, disable buffering. We will drain the buffer and eventually push the EOS event out. When we reach the estimated NPT timeout and we didn't receive an EOS event, make one and queue it so that it can be pushed. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017 2014-04-18 10:21:27 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: send reconfigure when internal-ssrc changes When the internal-ssrc property changes, we want to send a reconfigure upstream to make payloaders use the new suggested ssrc. Using the internal-ssrc property to change the SSRC of a stream is not a good idea and doesn't work when there are multiple senders, we want to set the SSRC directly on the payloaders. Therefore, deprecate this property. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361 2014-04-18 04:23:26 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: assume a full buffer when eos Rework the logic to make buffering messages a little, make sure we don't make the same message multiple times. Consider the buffer full when EOS was received. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017 2014-04-17 18:07:09 +0200 Sebastian Dröge * tests/check/elements/rtprtx.c: rtprtx: Don't forget to unmap rtp buffer in the test 2014-04-17 17:58:58 +0200 Sebastian Dröge * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Require clock-rate in the caps and handle no ssrc in the caps properly 2014-04-17 17:43:12 +0200 Sebastian Dröge * tests/check/elements/rtprtx.c: rtprtx: Provide an ssrc in the test And increase timeout to allow all tests to run in valgrind. 2014-04-17 17:33:46 +0200 Sebastian Dröge * tests/check/elements/rtpsession.c: rtpsession: Fix memory leaks in test 2014-04-17 17:26:36 +0200 Sebastian Dröge * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix hundreds of memory leaks in the test 2014-04-17 17:00:37 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Unref clock id when waiting for the clock is interrupted 2014-04-17 16:39:59 +0200 Sebastian Dröge * tests/check/elements/rtpcollision.c: rtpcollision: Fix memory leaks in unit test 2014-04-16 21:40:45 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: videomixer: name collectpads object based on videomixer name Makes it easier to track things in debug logs when there are multiple mixers and muxers. 2014-04-16 21:37:12 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: videomixer: better logging of incoming events The pad and parent names are already logged as part of logging the object. Instead log the full event details. 2014-04-16 19:03:47 +0200 Sebastian Dröge * tests/check/elements/videomixer.c: videomixer: Fix memory leak in unit test 2014-04-16 18:49:43 +0200 Sebastian Dröge * gst/level/gstlevel.c: level: Use the correct number of samples to iterate over the input array Fixes invalid memory accesses and accesses to uninitialised data. 2014-04-16 18:00:49 +0200 Sebastian Dröge * gst/icydemux/gsticydemux.c: icydemux: Unref dropped events 2014-04-16 17:29:30 +0100 Vincent Penquerc'h * gst/matroska/ebml-read.c: matroska: fix check for amount of data to read History shows length==0 should set data to NULL and return, so we do that too instead of trying to read nothing. Coverity 206205 2014-04-16 17:25:44 +0100 Vincent Penquerc'h * gst/deinterlace/gstdeinterlace.c: deinterlace: fix sign comparison history_count is unsigned, so the whole comparison will be made as unsigned, and fail to reject what it was meant to. Coverity 206204 2014-04-16 17:04:50 +0100 Vincent Penquerc'h * gst/avi/gstavidemux.c: avidemux: remove dead code sub may not be NULL in this switch, there is a bail out just before it if so. Coverity 206098 2014-04-16 16:59:43 +0100 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: flacparse: remove dead code The block_size == 0 was shortcut earlier, and the variable is not modified in the meantime. Coverity 206097 2014-04-16 16:56:54 +0100 Vincent Penquerc'h * gst/videomixer/videoconvert.c: videomixer: remove dead code While it seems to keep a compile time selection, I traced it to some code copied from videoconvert, where it was removed, with the following comment: Also remove the high-quality I420 to BGRA fast-path as it needs the same fix, which causes an additional instruction, which causes orc to emit more than 96 variables, which then just crashes. This can only be fixed in orc by breaking ABI and allowing more variables. Thus, I remove it here as well. Coverity 206064 2014-04-16 16:50:30 +0100 Vincent Penquerc'h * gst/isomp4/qtdemux.c: isomp4: fix incorrect masking for multiple tags Coverity 206058 2014-04-16 16:45:08 +0100 Vincent Penquerc'h * gst/isomp4/atoms.c: isomp4: fix wrong atom flags set when adding samples Coverity 206057 2014-04-16 16:40:02 +0100 Vincent Penquerc'h * gst/audiofx/audiofxbasefirfilter.c: audiofx: fix comparison of delta time to a threshold Coverity 206055 2014-04-16 16:32:26 +0100 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: do not rely on call failure keeping return data unmodified This is clearer this way too. Coverity 206029 2014-04-16 16:28:49 +0100 Vincent Penquerc'h * gst/isomp4/atomsrecovery.c: isomp4: catch fseek error Coverity 206028 2014-04-16 16:25:44 +0100 Vincent Penquerc'h * gst/isomp4/atoms.c: isomp4: report failures to caller Coverity 206027 2014-04-16 18:05:46 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: refuse serialied query when buffering When we are buffering, we can't block and wait for the serialized query to complete because the jitterbuffer will not try to forward the query while buffering. Instead, just refuse the query. 2014-04-16 16:51:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: don't free the serialized query We should never free a serialized query in the queue, it is the upstream caller that will free it. 2014-04-16 17:35:42 +0200 Sebastian Dröge * tests/check/elements/aacparse.c: aacparse: Fix memory leak in the test 2014-04-16 17:33:46 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Create hashtable only when we actually use it In error cases we previously returned without freeing it. 2014-04-16 17:30:59 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Chain up to the parent class' dispose function 2014-04-16 17:23:27 +0200 Sebastian Dröge * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Initialise ioctl struct with zeroes before passing it to ioctl() 2014-04-16 13:47:43 +0200 Marc Leeman * gst/udp/gstudpsrc.c: udpsrc: correct LOG msg for -1 Signed-off-by: Marc Leeman 2014-04-15 21:36:30 +0200 Sebastian Dröge * gst/interleave/interleave.c: interleave: Fix negotiation to work at all again The caps query handling function for the sinkpads was called for the srcpad, and the sinkpads had none. This commit moves it to the right pad, but nonetheless the negotiation still looks wrong. This makes the test pass again after the recent coverity fix and also allows interleave to work again, but someone should really review the negotiation code and fix it. 2014-04-13 09:03:41 +0200 Edward Hervey * sys/oss4/oss4-audio.c: oss4: Maximum number of channels support is 8 Avoids doing potential overwrites in ch_layout (which only has 8 fields). CID #1139826 2014-04-12 22:16:37 +0200 Sebastian Dröge * sys/osxvideo/osxvideosink.m: osxvideosink: Set rank to MARGINAL If available we prefer using glimagesink over osxvideosink. It supports more formats and in general has more features than osxvideosink. 2014-04-11 18:19:49 +0200 Josep Torra * gst/rtp/gstrtph264depay.c: rtph264depay: only guess AU boundaries when aren't indicated by marker The marker bit isn't mandatory and we had in place code to guess AU boundaries by detecting a new picture start. This guessing code didn't work with interlaced content that has proper marker bits to indicate the AU boundaries. It was leaking the first field buffer and producing a corrupted output. fixes: https://bugzilla.gnome.org/show_bug.cgi?id=728041 2014-04-10 10:38:19 -0300 Rafał Mużyło * ext/libpng/gstpngdec.c: pngdec: enable libpng interlaced picture handling Makes libpng deinterlace Adam7 interlaced pictures by default. It is the only interlaced format available and if the picture isn't interlaced the code should behave as before. https://bugzilla.gnome.org/show_bug.cgi?id=726161 2014-04-11 13:27:42 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Only keep-alive the connection in stop() if we have finished all previous messages After cancelling a request we need to create a new connection. 2014-04-11 11:54:12 +0200 Edward Hervey * ext/dv/gstdvdec.c: dvdec: Don't set bogus timestamp/duration This will happen if we have an incoming stream with a non-TIME segment Could be improved later to figure out proper pts/duration. CID #1199702 CID #1199703 2014-04-11 11:53:42 +0200 Edward Hervey * ext/dv/gstdvdec.c: dvdec: Properly refuse incoming stream without framerate The return value wasn't properly propagated back if the caps didn't contain a framerate 2014-04-10 16:35:28 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Also retry on unexpected network failures 2014-04-10 15:45:41 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: New property to specify the maximum number of retries before we give up 2014-03-13 10:56:11 +0100 Alexander Zallesov * ext/soup/gstsouphttpsrc.c: souphttpsrc: Change default timeout to 15 seconds If nothing happens after 15 seconds, chances are good that our connection will never will work. Stop after 15 seconds instead of waiting until the system's default timeout, which can be > 1 minute. 2014-04-09 17:30:54 +0900 Jimmy Ohn * gst/isomp4/qtdemux.c: qtdemux: replace duplicated variable when parsing trex atom https://bugzilla.gnome.org/show_bug.cgi?id=727878 2014-04-09 10:56:29 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Use GST_FLOW_FLUSHING when flushing, not GST_FLOW_EOS ... and reset it properly after flushing is done. Fixes playback in many cases when buffering is used. https://bugzilla.gnome.org/show_bug.cgi?id=727821 2014-04-09 08:58:04 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Properly return stream flags when parsing trex atom https://bugzilla.gnome.org/show_bug.cgi?id=727867 2014-03-19 19:18:11 +0000 Matthieu Bouron * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: use the video frame API instead of the video meta API https://bugzilla.gnome.org/show_bug.cgi?id=726738 2014-03-19 18:47:39 +0000 Matthieu Bouron * sys/osxvideo/osxvideosink.m: osxvideosink: advertize video meta API support https://bugzilla.gnome.org/show_bug.cgi?id=726737 2014-04-08 11:31:06 +0200 Edward Hervey * gst/interleave/interleave.c: interleave: Add missing break in switch statement The caps query is handled entirely already before. CID #1139757 2014-04-06 18:03:11 -0300 Reynaldo H. Verdejo Pinochet * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: use SoupKnownStatusCode if needed From libsoup docs: Prior to 2.44 SoupStatus was called SoupKnownStatusCode, but the individual values have always had the names they have now. Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=727329 2014-04-07 12:58:23 +0100 Vincent Penquerc'h * gst/avi/gstavidemux.c: avidemux: use frames, not bytes, for position query in VBR streams Coverity 1139648 2014-04-07 12:42:14 +0100 Vincent Penquerc'h * gst/smpte/gstsmpte.c: smpte: fix copy/paste error causing unmap on wrong buffer Coverity 1139647 2014-04-07 12:16:17 +0100 Vincent Penquerc'h * gst/deinterlace/gstdeinterlace.c: deinterlace: guard against finding no suitable pattern The code handles a -1 pattern index, and it seems plausible that a pattern might be found later, so it seems best to not send an element error here. Coverity 1139766 2014-04-04 17:38:14 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: update for new MIKEY API 2014-04-03 17:40:01 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: send sender SSRC in the MIKEY message Allocate a new SSRC for our RTCP messages back to the server and set this in the MIKEY message. 2014-04-03 17:39:30 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: make random number for the CSB As recommended in the RFC 2014-03-26 12:10:44 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't put spaces in keymgmt header 2014-03-25 17:47:49 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: create and send the RTCP encryption key Create and make a key for encrypting the RTCP packets back to the server and wrap this in a MIKEY message that we send as a header in the SETUP request. 2014-04-03 12:18:39 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: free the srtpdec element 2014-04-03 12:16:25 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: cleanup stream_free function There is no reason to NULL all fields, we will free the stream anyway. 2014-04-03 12:07:31 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: demote warning to debug For TCP, it is normal that we don't have timestamps so don't WARN on it. 2014-03-29 19:13:06 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: v4l2: Fix support for caps without width, height, framerate or format For format like mpegts, width and height is rarely in the negotiated caps. This patch fixes failure when setting format, and prevent introducing width, height, framerate and format to the caps when fixating. https://bugzilla.gnome.org/show_bug.cgi?id=725860 2014-03-31 18:34:13 +0200 Thibault Saunier * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Always set PTS=DTS on raw video streams 2014-03-31 18:31:22 +0200 Thibault Saunier * gst/avi/gstavidemux.c: avidemux: Always set pixel-aspect-ratio on raw video streams That field is mandatory in caps and if it is not present in the AVI container, it means square pixels thus 1/1. 2014-03-30 00:35:07 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroska-mux: add mapping for Opus audio Might want to consider adding channels/rate requirement to template caps, but requires fixing up of encoder and parser first. 2014-03-30 00:31:11 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroska-demux: add mapping for Opus audio codec https://bugzilla.gnome.org/show_bug.cgi?id=727305 2014-03-29 17:21:17 -0400 William Manley * sys/v4l2/gstv4l2object.c: v4l2src: Fix support for mpegts streams It seems that GStreamer's mpegts elements (tsdemux, tsparse) require caps `video/mpegts,systemstream=true`. As far as I can see the significance of systemstream is to indicate that this is a container format rather than an elementary stream. As this is the case (and I can't understand how it could not be the case with mpegts) I add systemstream=true to v4l2src's caps. This allows v4l2src to be linked with tsdemux for playback from my Hauppauge HD-PVR with the pipeline: v4l2src ! queue ! tsdemux ! video/x-h264 ! decodebin ! xvimagesink In combination with the next commit this fixes using Hauppauge HD-PVR with GStreamer 1.0+. 2014-01-14 14:48:42 +0000 Vincent Penquerc'h * sys/v4l2/v4l2_calls.c: v4l2: attempt to fix infinite (for small version of infinite) loop 2014-03-29 13:20:30 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: rtpmanager: copy sticky events when exposing pads in more places https://bugzilla.gnome.org/show_bug.cgi?id=724712 2014-03-28 20:11:36 +0100 Rico Tzschichholz * sys/v4l2/Makefile.am: v4l2: fix distcheck Make sure ext/*.h are dist'ed 2014-03-27 19:51:50 +0000 Tim-Philipp Müller * sys/ximage/gstximagesrc.c: ximagesrc: only extrapolate alpha mask for 32-bit depth Instead of passing bogus alpha mask values when there's no alpha. https://bugzilla.gnome.org/show_bug.cgi?id=726833 2014-03-21 13:03:17 -0400 Nicolas Dufresne * sys/ximage/gstximagesrc.c: ximagesrc: Add ARGB/BGRA support 2014-03-20 15:28:26 +0100 Ognyan Tonchev * gst/rtp/gstrtpjpegpay.c: jpegpay: consider header len when calculating payload len Fixed https://bugzilla.gnome.org/show_bug.cgi?id=726777 2014-03-26 08:03:22 +0100 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: All frames are sync points 2014-03-26 08:02:43 +0100 Sebastian Dröge * ext/libpng/gstpngdec.c: pngdec: All frames are sync points 2014-03-22 17:07:46 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: segment closing not needed in 1.x ... as sender should keep track of segment base accumulation. Rather, it may have some adverse effects as a spurious segment event, e.g. in collectpads. 2014-03-22 17:05:17 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: early sending pending codec-data for all streams ... at least before syncing across all streams might cause some gap activity on any of those streams, notably sparse streams. See also #712134 2014-03-22 17:01:27 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: handle both sticky and non-sticky custom event 2014-03-25 11:44:27 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: only expose streams on dataflow Only probe on buffers, we don't want to expose the streams on events. 2014-03-25 11:36:40 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtsp/gstrtspsrc.c: rtspsrc: copy sticky events to ghostpad When we expose internal pads as ghostpads, first copy the sticky events so that we have the caps and segment etc. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724712 2014-03-24 14:25:43 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: srtp handling 2014-03-25 10:23:00 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: set SSRC on caps if known 2014-03-24 16:58:25 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: put caps on udpsrc instead of using the signals Try to avoid using the request-pt-map to get caps but set them directly on the udpsrc element. That way, the caps get nicely transformed as they pass through the different elements in the rtpbin, including the AUX and decoder/encoder elements. 2014-03-24 15:35:09 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use profile to set rtcp caps Use the negotiated profile to set x-rtcp or x-srtcp caps 2014-03-24 15:34:26 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: set udpsrc to READY READY is enough to allocate ports now 2014-03-24 14:25:28 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: improve caps handling Protect caps with the lock. Don't push the caps event from the set_property function but mark the pad for reconfiguration so that it will renegotiate and push the new caps event in the streaming thread. 2014-03-24 15:15:34 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: open/close socket in NULL<->READY state We should open the socket when going to NULL<->READY and not in the start/stop vemthod, which is called in READY<->PAUSED. This makes it possible to allocate a socket without going to PAUSED (and starting the negotiation). 2014-03-24 14:35:01 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: free caps in ptmap array Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726696 2014-03-20 11:12:51 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle NULL rtpmap and parse error better 2014-03-18 00:08:50 +0000 Руслан Ижбулатов * tests/examples/gtk/gstgtk.c: gl: fix the use of always-defined macros After 2a0f0399ae226089c2ba07b1b904741b856f37af GST_GL_* macros are always defined to 0 or 1. Don't use #ifdef ... or #if defined() on them. https://bugzilla.gnome.org/show_bug.cgi?id=726591 2014-03-16 23:46:22 -0400 Olivier Crête * configure.ac: configure: Don't check for gudev if video4linux2 is not present 2014-03-16 23:19:55 -0400 Olivier Crête * configure.ac: configure: Don't fail if gudev is not present PKG_CHECK_MODULES has the bad habit of failing the build if it doesn't get what it wants, prevent that. 2012-11-02 13:33:13 +0100 Olivier Crête * configure.ac: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2devicemonitor.c: * sys/v4l2/gstv4l2devicemonitor.h: v4l2: Implement GstDeviceMonitor subclass https://bugzilla.gnome.org/show_bug.cgi?id=678402 2013-08-12 11:49:21 -0400 Olivier Crête * ext/pulse/Makefile.am: * ext/pulse/plugin.c: * ext/pulse/pulsedevicemonitor.c: * ext/pulse/pulsedevicemonitor.h: pulse: Add device monitors https://bugzilla.gnome.org/show_bug.cgi?id=678402 2014-03-16 19:24:26 -0400 Olivier Crête * sys/v4l2/gstv4l2object.c: v4l2: Remove GstPropertyProbe leftovers 2014-02-19 03:04:03 +0100 Mathieu Duponchelle * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: Port to new collectpads API See: https://bugzilla.gnome.org/show_bug.cgi?id=724705 2014-03-16 15:26:04 +0100 Nicolas Dufresne * sys/v4l2/ext/types-compat.h: * sys/v4l2/ext/videodev2.h: v4l2: Add types compatiblity for other OS Adds type compatiblity with other OS like BSD. This uses types mapping macro to avoid conflict with existing defined types. We resuse glib types as these are already available on supported platforms. This is GCC only because of the le32 type that uses bitwise attribute. https://bugzilla.gnome.org/show_bug.cgi?id=726453 2014-03-16 15:55:00 +0000 Tim-Philipp Müller * ext/pulse/pulseutil.c: pulse: fix format info to caps conversion for mulaw 2013-08-13 12:10:42 -0400 Olivier Crête * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: Make gst_pulse_format_info_to_caps() shared https://bugzilla.gnome.org/show_bug.cgi?id=678402 2014-03-15 18:41:16 +0100 Nicolas Dufresne * sys/Makefile.am: v4l2: Fix typo V4L_DIR intead of V4L2_DIR 2013-12-29 17:29:53 +1100 Matthew Waters * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: * tests/examples/gtk/gstgtk.c: [864/906] examples: update to gtk3 2013-07-17 11:22:02 +0200 Sebastian Dröge * tests/examples/gtk/gstgtk.c: [771/906] gl: Some less long/ulong/gulong usage 2013-07-16 18:27:07 +0200 Mathieu Duponchelle * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [769/906] tests/examples: fix and port some of the examples. Realize widgets, remove glupload element. 2013-07-10 11:24:34 +0200 Sebastian Dröge * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: * tests/examples/gtk/gstgtk.c: [729/906] gl: Include config.h everywhere 2013-06-28 11:00:46 +0200 Sebastian Dröge * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [720/906] examples: Stop using deprecated GLib thread API 2012-11-08 22:53:56 +1100 Matthew Waters * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: * tests/examples/gtk/gstgtk.c: [603/906] update FSF address 2012-08-14 14:41:19 +1000 Matthew Waters * tests/examples/gtk/fxtest/pixbufdrop.c: [560/906] examples: update for bus api changes and glimagesink changes 2012-06-07 00:51:47 +1000 Matthew Waters * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: * tests/examples/gtk/gstgtk.c: [511/906] tests: update for 1.0 2010-09-16 15:00:29 +0300 Stefan Kost * tests/examples/gtk/gstgtk.c: [461/906] xoverlay: require base from git and update to new API 2010-07-12 18:38:59 +0200 Julien Isorce * tests/examples/gtk/fxtest/pixbufdrop.c: [457/906] gtk examples: adapt code since the native-window changes from gtk Fixes bug #599885 2010-01-12 18:32:39 +0300 Руслан Ижбулатов * tests/examples/gtk/fxtest/pixbufdrop.c: [413/906] Fix Windows compiler warning in test/examples/gtk/fxtest/pixbufdrop.c 2009-10-23 01:07:29 +0200 Julien Isorce * tests/examples/gtk/fxtest/pixbufdrop.c: [386/906] pixbufdrop: fix example on win32 2009-07-14 20:36:13 +0200 Filippo Argiolas * tests/examples/gtk/gstgtk.c: [361/906] gstgtk: add missing license and copyright information 2009-07-14 20:25:28 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [360/906] examples: add missing copyright/license to my examples 2009-04-12 20:03:30 -0700 David Schleef * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: * tests/examples/gtk/gstgtk.c: [328/906] Convert gtk examples to use helper library Helper lib implements gst-gtk glue on all platforms 2009-02-10 22:39:14 -0800 David Schleef * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [310/906] Global reindent Indent parameters: INDENT_PARAMETERS="--braces-on-if-line \ --case-brace-indentation0 \ --case-indentation2 \ --braces-after-struct-decl-line \ --line-length80 \ --no-tabs \ --cuddle-else \ --dont-line-up-parentheses \ --honour-newlines \ --continuation-indentation4 \ --tab-size8 \ --indent-level2" 2009-02-05 13:13:51 -0800 David Schleef * tests/examples/gtk/fxtest/pixbufdrop.c: [308/906] Rename glpixbufoverlay to gloverlay 2009-01-23 02:04:23 +0100 Julien Isorce * tests/examples/gtk/fxtest/pixbufdrop.c: [301/906] depends on libpng instead of gdk_pixbuf 2009-02-10 21:57:31 -0800 David Schleef * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [298/906] Revert "Fix indention" This reverts commit 96e4ab18c2cf9876f6c031b9aba6282d0bd45a93. You should have asked first. And you would have been told "no", because it causes people on development branches to do a huge amount of extra work. 2009-02-03 18:33:36 +0100 Sebastian Dröge * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [295/906] Fix indention 2008-10-15 16:18:22 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [247/906] Import xray effect Add xray effect. Maps luma to a negative, slightly cyan tinted, curve, applies some light gaussian blur and multiplies it with its sobel edges. Not sure about the name, likely to change. Probably still needs some tuning. 2008-08-19 22:15:17 +0200 Julien Isorce * tests/examples/gtk/fxtest/pixbufdrop.c: [199/906] add pixbufdrop vs8 project 2008-08-19 21:04:29 +0200 Julien Isorce * tests/examples/gtk/fxtest/fxtest.c: * tests/examples/gtk/fxtest/pixbufdrop.c: [198/906] add fxtest vs8 project 2008-08-19 08:50:14 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/pixbufdrop.c: [195/906] fix gstgldifferencematte and add an example app to test it dragging an image over the video (works with pixbufoverlay too, see pixbufdrop --help) 2008-08-16 17:36:10 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [180/906] minor cleanup in fxtest 2008-08-16 10:15:31 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [178/906] improve fxtest command line option handling, default to videotestsrc if no source bin description is given 2008-08-16 09:13:39 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [175/906] add sin effect (desaturate everything but red shades). still needs some tuning. 2008-08-14 21:29:02 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [173/906] add lumaxpro (desaturate + cross process) effect. nothing too impressive but I like it. 2008-08-14 20:54:54 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [172/906] add support for command line parsing to fxtest (try fxtest videotestsrc ! desired caps ! identity). report a new issue on BUGS. 2008-08-14 20:02:04 +0200 Filippo Argiolas * tests/examples/gtk/fxtest/fxtest.c: [171/906] import fxtest (little gtk app to easily test effects) from cvs branch, fixed rgbtocurve. 2014-03-15 18:05:32 +0100 Nicolas Dufresne * configure.ac: v4l2-build: Set HAVE_GST_V4L2 if headers are present The name of HAVE_ need to match the USE_. Now set HAVE_GST_V4L2 if videodev2.h is found. 2014-03-15 16:47:51 +0100 Nicolas Dufresne * configure.ac: * sys/Makefile.am: v4l2: Actually build the plugin The checks were removed inadvertedly in previous patch and not replaced. Re-introduce the configure checks and some of the checks in order to enable this plugin again. We only check if videodev2.h exist on the platform to avoid building on Windows or OSX, though we build against our own copy. This was breaking the build on built-bot. 2014-03-15 13:47:42 +0100 Nicolas Dufresne * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: translation: PO file changes caused by POTFILE.in update 2014-03-15 13:17:21 +0100 Nicolas Dufresne * configure.ac: * po/POTFILES.in: * po/POTFILES.skip: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2videooverlay.c: * sys/v4l2/gstv4l2videooverlay.h: v4l2: Remove XV support XV support for v4l2 never became upstream and ended up being commented out with an undef for a long time now. 2014-03-15 11:13:05 +0100 Nicolas Dufresne * configure.ac: * gst-plugins-good.spec.in: * sys/Makefile.am: * sys/v4l2/ext/v4l2-common.h: * sys/v4l2/ext/v4l2-controls.h: * sys/v4l2/ext/videodev2.h: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/v4l2_calls.c: * tests/icles/Makefile.am: v4l2: Use a copy of videodev2.h header With years the amount of ifdef have grown up and we are not even sure if the old code path compiles. Each time we need to update the v4l2 framework to add the new feature, we break compilation on older kernel. With exception of two controls in the video orientation control, this patch get rid of all ifdef by including the latest version of videodev2.h inside GStreamer. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723446 2014-03-12 15:32:55 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Add properties for selecting SSL/TLS certificate checking And by default properly check certificates against the system's CA certificates. Everything else is not a good default at all. 2014-03-11 14:56:30 +0100 Per x Johansson * gst/matroska/matroska-demux.c: matroskademux: fix assert on fps lower than 1 Fixes assert caused by gst_duration_to_fraction calling gst_util_uint64_scale_int with a denominator of 0 when fps is less than 1. https://bugzilla.gnome.org/show_bug.cgi?id=726106 2014-03-11 00:46:06 -0300 Thiago Santos * gst/videomixer/videomixer2.c: videomixer2: store video info with buffers to keep it in sync Instead the queued buffer might have an old caps while the pad is already storing the information for a new caps. Mixing those while handling buffers will often lead to issues https://bugzilla.gnome.org/show_bug.cgi?id=725948 2014-03-08 19:29:58 -0500 William Manley * sys/v4l2/v4l2_calls.c: v4l2: Fix typo contol -> control https://bugzilla.gnome.org/show_bug.cgi?id=725632 2014-03-04 01:15:49 +0000 William Manley * sys/v4l2/v4l2_calls.c: v4l2: Normalise control names in the same way as v4l2-ctl V4L2 kernel drivers allow configuration of the hardware settings via a mechanism called controls. These can be referred to by name such as "Brightness" and "White Balance Temperature". The user-space command line client for setting these controls (v4l2-ctl) normalises these names such that they only contain lower case alphanumeric characters and the underscore '_'. e.g: Kernel v4l2-ctl ---------------------------------------------------- Brightness brightness White Balance Temperature white_balance_temperature Focus (absolute) focus_absolute GStreamer seems to want to follow this pattern but failed for controls with more than one consecutive non-alphanum character. e.g. GStreamer would produce "focus__absolute_" rather than "focus_absolute". This commit fixes that issue. Backwards compatibility is preserved by normalising all control names before comparison. https://bugzilla.gnome.org/show_bug.cgi?id=725632 2014-03-07 16:17:29 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Make sure to not return EOS immediately if we finished a range request Only return EOS the next time create() is called, if at all. basesrc should already take care of not calling it again. Also always return immediately if the previous flow return was not OK. This indicates an error somewhere. 2014-03-06 12:06:43 -0500 Olivier Crête * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpspeexdepay.c: rtp: Remove caps restrictions from RTP depayloader sink caps Remove caps restrictions that correspond to the default and are not required in SDP. With the new usage of having pads require a subset of the caps, they will make the negotiation fail. 2014-03-06 11:02:09 -0500 Olivier Crête * gst/rtp/gstrtpspeexdepay.c: rtpspeexdepay: Remove caps restrictions for depayloader The "encoding-params" is optional in the SDP, because we now require a subset of the caps, it would fail caps negotiatioin if it wasn't present. So removed it from the template caps. 2014-03-06 13:38:09 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't forget to quit mainloop after we cancelled when we got data after the stop position 2014-03-06 13:35:47 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: If we had a stop position, allow for the server to finish our connection instead of just cancelling Otherwise keep-alive does not make much sense and also the server will have confusing things in the logs. 2014-03-06 12:24:01 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: skip streams with same control url Keep track of what streams we did the SETUP for. We only need to configure caps, wait for pads and push events on setup streams. We can remove the disabled state of the stream and simplify some checks. After we setup a stream, skip the other streams that have the same control url. Use a skipped flag to mark streams that should be skipped. 2014-03-06 12:22:47 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: remove obsolete code 2014-03-05 16:19:19 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: just use the SDP index as the stream id Use the index of the media stream in the SDP as the stream id instead of keeping a separate counter. 2014-03-05 13:35:19 +0100 Thijs Vermeir * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideo: fix build on Mac OSX Mavericks and put new window in front GetCurrentProcess/SetFrontProcess/TransformProcessType was deprecated and now removed in Mac OSX 10.9. orderFrontRegardless is used to make the video window the most front window. 2014-03-05 17:33:56 +0100 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Add docs directory to spec file 2014-03-05 15:44:25 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle NULL control urls better 2014-03-05 14:28:26 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: session: small cleanups It's nicer to explicitly check for NULL on pointer types to make it clear that it's a pointer and not a boolean. 2014-03-05 14:26:02 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: session: handle unknown SSRC in FIR https://bugzilla.gnome.org/show_bug.cgi?id=725712 2014-03-05 11:39:09 +0100 Alessandro Decina * gst/rtsp/gstrtspsrc.c: rtspsrc: fix seeking Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as non-flushing before sending PAUSE and PLAY with the new npt range. Without this patch, those commands would fail with EINTR as the connections were still flushing. 2014-03-03 16:39:26 -0300 Thiago Santos * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: expose xsub as a subtitle instead of as a video It is placed inside a 'vids' struct, so it was being exposed on a pad named video_%d. XSUB are subtitles and this patch adds an special case for it to be exposed in a subpicture_%d pad 2014-03-03 16:38:45 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: do not try to add a tag with tag_name set to NULL This can happen if there are subtitles in the stream, leading to an assertion 2014-03-04 16:40:34 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add support for multiple payload types A media stream can have multiple payload types. Parse all the payload types and collect the caps information. We then have to store the pt<->caps mapping instead of 1 pt and 1 caps. Parse the profile from the SDP and use that to negotiate the transport instead of always using AVP. Rework how we do some tweaks for ASF and Realmedia. 2014-03-04 11:34:39 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: refactor payload handling 2014-03-03 11:34:00 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: fix buffer level with invalid DTS It is possible that the DTS is invalid (when we receive RTP packets from TCP, for example). As a fallback, use the reconstructed PTS value to calculate the buffer level. 2014-03-02 05:10:13 +0100 Sebastian Rasmussen * .gitignore: .gitignore: Ignore gcov intermediate files https://bugzilla.gnome.org/show_bug.cgi?id=725480 2014-02-28 09:34:46 +0100 Sebastian Dröge * common: Automatic update of common submodule From fe1672e to bcb1518 2014-02-27 23:15:04 -0300 Thiago Santos * gst/audioparsers/gstaacparse.c: Revert "aacparse: put codec data on caps for loas format" This reverts commit e459cf3e01a08f1a3ef1fb954a41cfa36b3e510c. This was pushed by accident, the bug should likely be fixed in libav https://bugzilla.libav.org/show_bug.cgi?id=644 2014-02-27 18:55:04 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: mark all parsed frames as sync points all jpeg frames are sync points, so mark them as such so reverse playback can properly work with the video decoder base class https://bugzilla.gnome.org/show_bug.cgi?id=725104 2014-02-25 01:12:05 -0300 Thiago Santos * gst/audioparsers/gstaacparse.c: aacparse: put codec data on caps for loas format gst-libav audio decoder also needs codec data for LOAS format, otherwise it will complain about not having a decoder config and skip all packets https://bugzilla.gnome.org/show_bug.cgi?id=596772 2014-02-27 00:43:48 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: align raw audio memory to powers of two https://bugzilla.gnome.org/show_bug.cgi?id=725008 2014-02-27 00:37:20 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: calculate alignment properly for audio depths not a multiple of 8 2014-02-23 19:09:24 +0100 Matej Knopp * gst/matroska/matroska-demux.c: matroskademux: fix crash with 24-bit raw audio Do not try to align audio buffers to odd numbers, which will get us a NULL buffer which we then crash on. https://bugzilla.gnome.org/show_bug.cgi?id=725008 2014-02-27 00:11:42 +0000 Tim-Philipp Müller * gst/rtpmanager/Makefile.am: rtpmanager: re-enable -Werror 2014-02-27 00:11:11 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix compiler warning gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop': gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function while (result == GST_FLOW_OK); ^ 2014-02-26 22:11:41 +0100 Stefan Sauer * common: Automatic update of common submodule From 1a07da9 to fe1672e 2014-02-26 21:11:23 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Fix uninitialized variable compiler warning 2014-02-26 07:32:32 -0500 Jake Foytik * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Remove raw comparisons of RTP sequence numbers Several conditional statements perform comparison on RTP sequence numbers without taking the sequence number rollover into account. Instead, use the gst_rtp_buffer_compare_seqnum function to perform the comparison. https://bugzilla.gnome.org/show_bug.cgi?id=725159 2014-02-03 01:44:21 +0100 Sebastian Rasmussen * tests/check/Makefile.am: tests: Don't build disabled plugins' check tests https://bugzilla.gnome.org/show_bug.cgi?id=723502 2014-02-26 11:29:45 +0100 Stefan Sauer * docs/Makefile.am: docs: install prebuilt plugin docs if gtk-doc is disabled Sync to the Makefile.am from gst-plugin-base where it is done right. Fixes #725034 2014-02-25 16:10:54 -0500 Hugues Fruchet * sys/v4l2/gstv4l2object.c: v4l2object: do not emit "parsed" caps for vp8 VP8 doesn't require parsing (vp8parse doesn't exist, so negotiation with demux fails if "parsed" is set in caps). https://bugzilla.gnome.org/show_bug.cgi?id=724636 2014-02-11 16:27:08 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Don't require parser for VP8 Until GStreamer has one (see bug722760), we should not require a parser for VP8. https://bugzilla.gnome.org/show_bug.cgi?id=722128 2014-02-10 17:08:25 -0500 Nicolas Dufresne * sys/v4l2/v4l2_calls.c: v4l2: CAPTURE_MPLANE is well tested now https://bugzilla.gnome.org/show_bug.cgi?id=722128 2013-12-18 09:56:35 +0100 Benjamin Gaignard * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: v4l2videodec: Create one element per device For each videoCdevice probe it input/output capabilities if it match with video decoder requirement register a new element. Signed-off-by: Benjamin Gaignard https://bugzilla.gnome.org/show_bug.cgi?id=722128 2013-12-19 15:26:52 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2videodec.c: v4l2videodec: Calculate latency from device information Decoders or other devices that expose a minimum buffers required produce an first output. We use this information to calculate latency. https://bugzilla.gnome.org/show_bug.cgi?id=722128 2013-11-28 17:14:18 -0500 Nicolas Dufresne * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2videodec.c: * sys/v4l2/gstv4l2videodec.h: * sys/v4l2/v4l2_calls.c: v4l2videodec: Implement v4l2videodec Implement an element that can driver V4L2 M2M decoder device. https://bugzilla.gnome.org/show_bug.cgi?id=722128 2014-02-11 12:41:29 +0100 Göran Jönsson * gst/rtp/gstrtph264pay.c: rtph264pay: only update last_spspps time if all sps/pps got sent successfully This fixes an issue with gst-rtsp-server where no sps and pps are sent for the first intra frame, because the payloader starts working already when receiving DESCRIBE but there is no transports so it tries to send sps and pps, but that fails with a FLUSHING flow. But the time for last sent sps and pps would still be set, so when PLAY arrives and the first intra frame is to be sent there is no sps and pps sent due to that time since last sps pps is less than spspps_interval. https://bugzilla.gnome.org/show_bug.cgi?id=724213 2014-02-25 09:00:45 +0100 Santiago Carot-Nemesio * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix deadlock when task creation is no successful https://bugzilla.gnome.org/show_bug.cgi?id=725124 2014-02-22 20:19:49 +0100 Stefan Sauer * gst/autodetect/gstautodetect.c: autodetect: demote candidate error to warning and plug fake{sink,src} In the case where we have no suitable candidate we post a warning and plug a fake-element. Do the same when non of the candidate work. This is more consistent and plugin the fakesink as a fallback is probably helpful for running unit tests without requiring hardware src/sink elements. Fixes #722981 2014-02-23 12:34:48 +0100 Mark Nauwelaerts * sys/v4l2/v4l2_calls.c: v4l2: make some more controls configurable ... at least if one tries hard enough using extra-controls property. 2014-02-23 10:39:20 +0100 Dan Kegel * configure.ac: v4l2: Require mplanar support for now in configure The code fails to compile without currently, see https://bugzilla.gnome.org/show_bug.cgi?id=723446 It's better to disable it instead of failing compilation until this is fixed properly. 2014-02-23 00:14:04 +0100 Stefan Sauer * ext/jack/gstjackaudioclient.c: jack: add some simple log handlers for jack Add log handlers for jack that write to the gst debug log. This avoids spamming the console when e.g. using autoaudiosink, having the jack elements installed, but not running jack. 2014-02-22 21:31:21 +0100 Mark Nauwelaerts * sys/v4l2/v4l2_calls.c: v4l2src: handle old and odd driver behaviour when listing controls 2013-11-28 16:54:58 -0800 Darryl Gamroth * gst/audiofx/audiofxbaseiirfilter.c: audiofxbaseiirfilter: check if coefficients are provided inside filter lock https://bugzilla.gnome.org/show_bug.cgi?id=719524 2014-02-21 19:46:44 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2bufferpool.c: v4l2src: also unset INTERLACED flag on buffers if frame is not interlaced https://bugzilla.gnome.org/show_bug.cgi?id=724899 2014-02-21 14:31:59 +0000 Simon Farnsworth * sys/v4l2/gstv4l2bufferpool.c: v4l2src: Flag interlaced buffers as interlaced. We correctly indicate the field ordering on interlaced buffers, but fail to flag them as containing interlaced video, which we need to do here because we signal interlace-mode=mixed in our caps. This means that downstream elements (like vaapipostproc from gstreamer-vaapi) don't recognise these buffers as in need of deinterlacing. Fix this by setting the interlaced flag on all interlaced buffers. Signed-off-by: Simon Farnsworth https://bugzilla.gnome.org/show_bug.cgi?id=724899 2014-02-19 13:56:37 -0300 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstaacparse.c: aacparse: be more strict at ADTS header parsing Adds two extra checks: - Sampling frequency on header can't be 15. - Frame size should be at least 9 or 7, depending on whether CRC protection is present. https://bugzilla.gnome.org/show_bug.cgi?id=724638 2014-02-19 13:35:59 -0300 Reynaldo H. Verdejo Pinochet * gst/audioparsers/gstaacparse.c: aacparse: make sure we have enough ADTS data We need at least 6 bytes to pass over to _get_frame_len() but we were just checking for a minimum of 2 bytes for the syncword. https://bugzilla.gnome.org/show_bug.cgi?id=724638 2014-02-20 22:52:57 +0100 Stefan Sauer * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautodetect.h: autodetect: check if the kid has a sync property previously autovideosrc did not have a sync property and v4l2src has none either. 2014-02-19 21:55:52 +0100 Stefan Sauer * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautoaudiosrc.h: * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautodetect.h: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosink.h: * gst/autodetect/gstautovideosrc.c: * gst/autodetect/gstautovideosrc.h: autodetect: use a common baseclass This makes the actual elements super simple. We're using the ELEMENT_FLAG to configure source/sink and a string for the Audio/Video type. 2014-02-14 17:14:42 -0800 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add tls-database property Add support for a new property: tls-database. If the property is set, the certificate database will be given to the rtsp connection if TLS protocol is being used. If the server certificate can't be verified with the default database, this additional database will be used. https://bugzilla.gnome.org/show_bug.cgi?id=724396 2014-02-19 22:21:54 +0100 Thijs Vermeir * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudiosink.c: osxaudio: remove unused variables 2014-02-19 21:26:03 +0100 Stefan Sauer * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautodetect.h: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: autodetect: extract common helper code The function to generate the pretty names is basically the same. Use one and add a parameter. 2014-02-19 21:01:39 +0100 Stefan Sauer * tests/check/Makefile.am: * tests/check/elements/autodetect.c: autodetect: improve the tests Add fake audio/video sinks. Previously running the test might be flaky due to the use of real elements (hardware in use), which we don't want to test here. Add two more tests that check that the fakes are chosen. 2014-02-19 15:19:30 +0100 Branislav Katreniak * ext/soup/gstsouphttpsrc.c: souphttpsrc: do not emit error when connection with unknown size ends Commit 46fd12ae5ec53200b16dfd7f17048d6bc60fbfbc introduced connection recovery. But when server does not specify content-size, souphttpsrc tries to reconnect even after regular end of stream. Http server replies with SOUP_STATUS_REQUESTED_RANGE_NOT_SATISFIABLE but souphttpsrc still emits error instead of EOS. https://bugzilla.gnome.org/show_bug.cgi?id=724717 Signed-off-by: Branislav Katreniak 2014-02-19 11:26:22 +0100 Stefan Sauer * tests/check/elements/autodetect.c: autodetect: fix the disabled test Use a shared helper for both tests. It turns out that the valgrind variant is fine (maybe due to picking up pulsesink though). 2014-02-19 11:05:35 +0100 Stefan Sauer * tests/check/elements/autodetect.c: autodetect: remove cruft from the test Remove the obsolete version check and use the ignore macro for the disabled test. 2014-02-18 22:54:45 +0100 Stefan Sauer * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/level/gstlevel.c: * gst/spectrum/gstspectrum.c: docs: use docbook markup for xi:include It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where the only 4, we're fixing them instead. 2014-02-18 22:35:45 +0100 Stefan Sauer * gst/isomp4/gstqtmux-doc.h: isomp4mux: fix copy and paste This fixes doc warnings. 2014-02-18 21:44:24 +0100 Stefan Sauer * gst/debugutils/gstcapssetter.c: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/level/gstlevel.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrgvolume.c: docs: use the gtk-doc syntax to link to properties Don't use docbook unless needed. Also stip other docbook tags in the the files we fix. 2014-02-18 11:28:18 +0100 Stefan Sauer * ext/pulse/pulsesink.c: pulsesink: fix crash when getting the current-device in NULL->READY The "goto unlock" is wrong as in this code path we haven't take the lock yet. Fixes #724619 2014-02-14 22:50:49 +0100 Sebastian Dröge * configure.ac: soup: We need libsoup >= 2.40 for proper usage of the content decoder Previous versions did not consider our chunk allocator and allocated memory by themselves, which caused crashes and broken behaviour. 2014-02-14 15:27:20 -0500 William Jon McCann * gst/audiofx/audiocheblimit.c: * gst/udp/gstudpsrc.c: docs: fix mismatched para tags newer gtkdoc is more sensitive to mismatched docbook tags. This fixes the build in master. 2014-02-14 15:59:46 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: add support for serialized queries See https://bugzilla.gnome.org/show_bug.cgi?id=723850 2014-02-14 15:53:55 +0100 Wim Taymans * tests/check/elements/souphttpsrc.c: tests: fix typecast to fix compilation 2014-02-14 12:01:00 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: proxy caps and allocation on RTP pads recv_rtp_sink: allow proxying of the allocation query. send_rtp_sink: allow proxying of caps and allocation. This allows us to query caps downstream as well as get an allocator from downstream. send_rtp_src: allow proxy of caps, this makes the caps query do upstream. See https://bugzilla.gnome.org/show_bug.cgi?id=723850 2014-02-13 12:29:13 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: handle tags in mac encoding Check the charset from (C)*** tags and set the charset to convert from MAC encoding if suitable. https://bugzilla.gnome.org/show_bug.cgi?id=723166 2014-02-13 12:09:13 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Use new automatic_eos API from basesrc We want to notice ourselves that we're EOS. Otherwise we will always cancel requests in the very end and confuse the server... and also make it impossible to use persistent connections. 2014-02-13 11:11:13 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Consistently use have_size instead of content_size!=0 2014-02-13 10:30:09 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Free extra headers when finalizing the element It's set as property by the application, we should not just reset properties when going back to READY. 2014-02-13 10:28:13 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Properly close the session when going back to NULL Don't wait for that until the element is disposed. 2013-02-28 12:20:52 +0100 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: add support for keep-alive sessions https://bugzilla.gnome.org/show_bug.cgi?id=699926 2014-02-12 13:00:13 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Add "compress" property to enable/disable automatic gzip/deflate content encoding handling 2014-02-12 12:39:10 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Retry connection if we're finished before the content size only if we actually have a content size https://bugzilla.gnome.org/show_bug.cgi?id=722185 2014-02-12 10:08:50 +0100 Sebastian Dröge * ext/soup/gstsouputils.c: souputils: Fix compiler warning gstsouputils.c:35:25: error: comparison of constant 9 with expression of type 'SoupLoggerLogLevel' is always false [-Werror,-Wtautological-constant-out-of-range-compare] 2014-01-07 23:00:56 -0300 Reynaldo H. Verdejo Pinochet * ext/soup/Makefile.am: * ext/soup/gstsoup.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpclientsink.h: * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: * ext/soup/gstsouputils.c: * ext/soup/gstsouputils.h: souphttp*: add ability to do HTTP session logging This changeset adds the loggin infrastructure and mods both souphttpsrc and souphttclientsink to use it. https://bugzilla.gnome.org/show_bug.cgi?id=721764 2014-02-07 14:00:15 +0100 divhaere * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: add support for GRAY8, BGR and RGB video colourspaces in V_UNCOMPRESSED codec https://bugzilla.gnome.org/show_bug.cgi?id=723849 2014-02-11 13:25:46 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Add mapping for NOT_FOUND and NOT_AUTHORIZED errors 2014-02-11 13:25:22 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't duplicate status_code to GStreamer error mapping 2014-02-09 23:38:44 +0100 Sebastian Dröge * gst/goom/filters.c: * gst/goom2k1/filters.c: goom: Remove unused functions 2014-02-09 23:21:20 +0100 Sebastian Dröge * gst/matroska/matroska-parse.c: matroskaparse: Comment out some unused functions used only from the commented out pull-mode code 2014-02-08 21:01:32 +0100 Sebastian Dröge * ext/taglib/gstid3v2mux.cc: id3v2mux: Fix another compiler warning 2014-02-08 17:43:32 +0100 Sebastian Dröge * tests/check/elements/souphttpsrc.c: souphttpsrc: Fix implicit enum conversion compiler warning error: implicit conversion from enumeration type 'SoupStatus' to different enumeration type 'SoupKnownStatusCode' 2014-02-08 17:41:21 +0100 Sebastian Dröge * tests/check/elements/interleave.c: interleave: Fix unitialized variable compiler warning in test error: variable 'mask' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] 2014-02-08 17:27:51 +0100 Sebastian Dröge * ext/taglib/gstid3v2mux.cc: id3v2mux: Fix unitialized variable compiler warning error: variable 'image_type' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] 2014-02-08 17:25:27 +0100 Sebastian Dröge * sys/oss4/oss4-audio.h: oss4: Fix typo in header include guard error: 'GST_OSS4_AUDIO_H' is used as a header guard here, followed by #define of a different macro [-Werror,-Wheader-guard] 2014-02-08 17:24:06 +0100 Sebastian Dröge * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Fix unitialized variable compiler warning variable 'rtx_ssrc' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] 2014-02-08 17:21:19 +0100 Sebastian Dröge * gst/rtp/gstrtpac3depay.c: rtpac3depay: Remove unused variable 2014-02-08 17:19:19 +0100 Sebastian Dröge * gst/flx/flx_fmt.h: flx: Fix typo in header include guard error: '__GST_FLX_FMT__H__' is used as a header guard here, followed by #define of a different macro [-Werror,-Wheader-guard] 2014-02-07 10:07:41 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: remove have_dts flag from pads It was used in the past in 0.10 when there was no explicit DTS field in buffers, now we have it in 1.x series and we can check it directly with GST_BUFFER_DTS_IS_VALID 2014-02-07 01:49:26 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: improve support for sparse streams Do not try to use subsequent buffer timestamps to calculate sparse streams durations because the stream is sparse and the buffers might not be 'time adjacent'. So rely on the duration and give the option to the pad to provide custom 'empty' buffers to represent the gaps in the stream, this can vary on how the data is represented. Right now, the only sparse stream supported is tx3g subtitles. 2014-02-06 12:15:22 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: add support for text/x-raw subtitles Adds it to mp4mux, qtmux and gppmux. Buffers need to be prefixed with 2 bytes for the text length before being muxed. https://bugzilla.gnome.org/show_bug.cgi?id=581295 2014-02-06 12:09:01 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: qtmux: add support for the TX3G atoms Adds functions for creating and setting values related to the tx3g atom for raw text subtitle support. QTFF spec has information on those atoms https://bugzilla.gnome.org/show_bug.cgi?id=581295 2014-02-05 10:27:54 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/gstqtmuxmap.h: qtmux: add subtitle support to qtmuxmap structures adds basic stubs for subtitle support around the qtmux and qtmuxmap structures. Still no real subtitle implemented, but basic functions in place https://bugzilla.gnome.org/show_bug.cgi?id=581295 2014-01-20 17:31:14 -0300 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: factor out read context init/reset While at this, move _track_reset() to track-ids so it can be called from the common read context reset routine. https://bugzilla.gnome.org/show_bug.cgi?id=722705 2014-02-06 12:21:07 +0100 Wim Taymans * gst/effectv/gstrev.c: effectv: fix doc section of revtv element 2014-02-05 12:46:54 +0100 Edward Hervey * sys/osxvideo/Makefile.am: osxvideo: Fix libtool usage --tag=CC is needed for static build 2014-01-16 11:26:41 +0000 Matthieu Bouron * gst/deinterlace/gstdeinterlace.c: deinterlace: do not try set deinterlace method if passthrough is enabled Fixes an issue with progressive content and unsupported video formats for the deinterlace method. https://bugzilla.gnome.org/show_bug.cgi?id=719636 2014-02-04 21:26:56 +0100 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: order format in template caps by preference To minimise risk of bad fixation, though audioconvert at least should be smart enough to avoid it. 2014-02-02 09:57:03 -0800 Dan Kegel * configure.ac: v4l2: Remove obsolete definition GST_V4L2_MISSING_BUFDECL The only use was removed by 9edc0c0365f79ab07ff2e65461c6696e3931a3f0 https://bugzilla.gnome.org/show_bug.cgi?id=723446 2014-02-04 13:43:56 +0100 Rafał Mużyło * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * gst/cutter/gstcutter.c: gst: Don't use endianness-specific S8 audio format It does not exist. https://bugzilla.gnome.org/show_bug.cgi?id=723331 2014-01-31 14:17:54 +0000 Julien Isorce * ext/cairo/gstcairooverlay.c: cairooverlay: add support for RGB16 https://bugzilla.gnome.org/show_bug.cgi?id=723289 2014-01-30 09:43:50 +0100 Per x Johansson * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: Fix constantly growing used uid list Moves the used uid list to the class to avoid having it grow forever. https://bugzilla.gnome.org/show_bug.cgi?id=723269 2014-01-30 10:44:05 +0100 Edward Hervey * common: Automatic update of common submodule From d48bed3 to 1a07da9 2014-01-24 01:52:08 +0000 Mike Sheldon * gst/wavparse/gstwavparse.c: wavparse: Ignore Broadcast Wave Format (BWF) tags when searching for 'fmt' chunk https://bugzilla.gnome.org/show_bug.cgi?id=723125 2014-01-29 10:37:53 +0100 Edward Hervey * tests/check/elements/rtpaux.c: check: Use fakesink sync=True instead of an audio sink Ensures the test can run on systems without alsa (or any audio output for that matter), and will avoid people running build slaves wondering what the hell was beeping during the night :) 2014-01-27 20:05:42 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: custom get_sink_caps handling for private stream caps ... now that those are transformed rather than parsed, some transforming of caps is required as well to make auto-plugging succeed. 2014-01-25 02:06:00 -0500 Ryan Lortie * sys/v4l2/v4l2_calls.c: v4l2: guard use of ENODATA with #ifdef Not all systems with v4l have ENODATA defined, so check that we have it before attempting to use it. https://bugzilla.gnome.org/show_bug.cgi?id=722953 2014-01-24 12:37:39 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: Revert "rtspsrc: Proxy rtpjitterbuffer do-retransmission property" This reverts commit 9f7b1128b1f00a2b87a232ff890867549ab95ba5. This should be handled automatically be rtspsrc if the AVPF profile is used, and manual enabling of it can be done with the new-manager signal. 2014-01-24 10:21:11 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add signal to notify of new manager So that you can configure and connect to signals on the rtpbin. See https://bugzilla.gnome.org/show_bug.cgi?id=722866 2014-01-23 15:17:58 -0800 Aleix Conchillo Flaqué * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Proxy rtpjitterbuffer do-retransmission property https://bugzilla.gnome.org/show_bug.cgi?id=722866 2014-01-21 17:52:44 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: handle expected packet being an RTX packet If the expected packet (do_next_seqnum is TRUE) is the one we requested for retranmission earlier, do the logic to update the retransmission statistics as well before setting up the timers for the next expected packet. Also reset the retransmission counter if the timer is reused for another seqnum. 2014-01-21 15:48:20 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: add a caps accumulator for the request-pt-map signal Add an accumulator that stops the signal emission as soon as a caps has been retrieved. Otherwise the default handler would continue emitting the signal and possibly overwrite the result with NULL again. 2014-01-21 15:25:54 +0100 Wim Taymans * gst/rtpmanager/gstrtprtxreceive.c: rtxreceive: copy flags and timestamps from original buffer 2014-01-21 15:24:52 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: ignore invalid timestamps in rtt calculation When the input buffer does not have a valid timestamp, don't try to calculate the round-trip-time. 2014-01-16 14:23:13 -0300 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroskaparse: better default caps when none set Uses information gathered during EBML parsing to forge a more suitable set of caps instead of blindly assuming everything is video/x-matroska. For consistency, stream type reset was added to matroska-demux too. https://bugzilla.gnome.org/show_bug.cgi?id=722311 2014-01-15 17:29:35 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: tests: rtprtx::test_rtxreceive_data_reconstruction: remove useless code for triggering retransmission There is no need anymore to push yet another buffer in rtxsend in order to trigger the previously requested retransmissions to actually happen. 2014-01-15 17:27:19 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: tests: rtprtx::test_rtxreceive_data_reconstruction: fix race condition Now with rtprtxsend pushing rtx buffers from a different thread, this is necessary to ensure that the result of the test is deterministic. This code makes use of GstCheck's global GMutex and GCond that are being used inside GstCheck's sink pad chain() function in order to synchronize with it. 2014-01-15 17:17:57 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: tests: rtprtx::test_rtxsender_packet_retention: fix race condition Now with rtprtxsend pushing rtx buffers from a different thread, this is necessary to ensure that the result of the test is deterministic. This code makes use of GstCheck's global GMutex and GCond that are being used inside GstCheck's sink pad chain() function in order to synchronize with it. 2014-01-15 11:26:33 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: tests: rtprtx::test_push_forward_seq: fix race condition Now with rtprtxsend pushing rtx buffers from a different thread, this is necessary to ensure that the result of the test is deterministic. This code makes use of GstCheck's global GMutex and GCond that are being used inside GstCheck's sink pad chain() function in order to synchronize with it. 2014-01-15 09:47:03 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: tests: rtprtx::test_push_forward_seq: fix buffer refcounting 2014-01-21 13:42:38 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: ensure that no rtx buffers are sent after EOS To do that, enqueue the EOS event to be sent from the srcpad task thread and flush the queue right afterwards, so that no more rtx buffers can be sent, even if there are more requests coming in. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722370 2014-01-15 09:46:14 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: rtprtxsend: run a new GstTask on the src pad The reason behind this is to minimize the retransmission delay. Previously, when a NACK was received, rtprtxsend would put a retransmission packet in a queue and it would send it from chain(), i.e. only after a new buffer would arrive. This unfortunately was causing big delays, in the order of 60-100 ms, which can be critical for the receiver side. By having a separate GstTask for pushing buffers out of rtxsend, we can push buffers out right after receiving the event, without waiting for chain() to get called. 2014-01-03 17:47:55 +0000 Tim-Philipp Müller * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2send: error out if no caps were received Instead of assuming that input is ogg. 2014-01-03 17:30:12 +0000 Tim-Philipp Müller * ext/shout2/gstshout2.c: shout2send: accept audio/webm, audio/ogg and video/ogg as well Those are advertised in the template caps, but the setcaps handler didn't handle them. But then oggmux and oggparse at least for now still always output application/ogg anyway, so that wasn't a real problem. 2014-01-20 10:12:45 +0100 Sebastian Dröge * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: Don't leak input buffers https://bugzilla.gnome.org/show_bug.cgi?id=722414 2014-01-19 17:40:56 +0100 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: reset some more audio pad data when needed 2014-01-19 17:38:59 +0100 Mark Nauwelaerts * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: write correct blockalign for vbr audio Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720659 2014-01-16 17:36:12 -0800 Aleix Conchillo Flaqué * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: do not drop serialized events when latency is set Serialized events are now queued in the jitter buffer, so we don't want to drop them even latency is set. https://bugzilla.gnome.org/show_bug.cgi?id=722372 2013-12-11 09:36:22 +0100 Michael Olbrich * gst/avi/gstavimux.c: avimux: don't make the buffer writable unless absolutely necessary https://bugzilla.gnome.org/show_bug.cgi?id=722396 2013-09-12 16:56:56 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: set GST_BUFFER_FLAG_DELTA_UNIT when appropriate https://bugzilla.gnome.org/show_bug.cgi?id=722394 2014-01-17 07:46:09 +0100 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: don't ref the newly created allocator Otherwise the allocator will never be deleted. https://bugzilla.gnome.org/show_bug.cgi?id=712612 2014-01-15 22:47:12 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't skip all video frames until the first keyframe Instead do it like all other demuxers and let parsers and decoders handle that. The keyframe information inside the container might be completely wrong like in the sample file of the bug report, and if it is correct and we push no keyframes, then the parsers and decoders will handle that properly anyway. https://bugzilla.gnome.org/show_bug.cgi?id=682276 2014-01-13 10:08:09 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: remove elst_offset variables They are not used anymore 2014-01-06 21:36:17 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: remember reverse playback when verifying the segment end Check if the rate is positive or negative to correctly compare the current position with the segment to make reverse playback work 2014-01-03 10:59:35 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: do not ignore empty segments Make sure empty segments are used and pushed with a gap event to represent its data (or lack of it) Each QtSegment is mapped into a GstSegment with the corresponding media range. For empty QtSegments a gap event is pushed instead of GstBuffers and it advances to the next QtSegment. To make this work with seeks, need to keep track of the starting 'base' to make sure it remains consistently increasing when pushing new segment events. For example: if a seek makes qtdemux start from 5s, the first segment will have a base=0. When the next segment is activated, its base time will be QtSegment.time - qtdemux.segment_base so that it doesn't include the first 5s that weren't played and shouldn't be accounted on the running time This purposedly will remove the fix made for https://bugzilla.gnome.org/show_bug.cgi?id=700264, at this point it was decided to respect the gaps, even if they cause a delay on playback, because that's the way the file was crafted. https://bugzilla.gnome.org/show_bug.cgi?id=345830 2013-12-12 23:05:43 -0500 Olivier Crête * tests/check/elements/rtprtx.c: tests: Remove usage of the system clock from the rtprtx test 2013-12-12 23:22:41 -0500 Olivier Crête * tests/check/elements/rtpcollision.c: tests: Initial segment in rtpcollision test 2014-01-14 15:56:42 +0100 George Kiagiadakis * tests/examples/rtp/client-rtpaux.c: * tests/examples/rtp/server-rtpaux.c: examples/*-rtpaux: specify payload type association for the audio stream, so that rtx works also for audio 2014-01-14 13:08:18 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: remove wrong check for payload type not having been set 1) pt can be lower than 96 2) there is no point in checking that because rtprtxsend will not even store buffers for payload types that it doesn't know about, so this case will never be reached 2014-01-14 13:01:41 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: fix data locking when creating rtx packets This patch moves the creation of rtx packets to be done early, in the src_event() function, when they are requested. The purpose is to run gst_rtp_rtx_buffer_new() with the object locked to protect internal data, because if it is done at the pushing stage, we would have to lock and unlock multiple times in a row while we are pushing the rtx buffers. Previously there was no locking at all, which was terribly wrong. 2014-01-14 12:50:23 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: lock access to internal data in sink_event() function 2014-01-14 12:44:06 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: remove unnecessary call to reset() from finalize() ...and use _free_full() on the pending buffers queue now that reset() is not being called 2014-01-14 12:38:51 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: remove unused parameter from the internal reset() method 2014-01-14 12:32:38 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Use g_slice_* for allocating internal structures 2014-01-14 12:28:01 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: remove stupid mutex unlock in the middle of chain() 2014-01-14 12:25:36 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: use GST_DEBUG_OBJECT / GST_WARNING_OBJECT instead of GST_DEBUG / g_warning 2014-01-14 12:19:58 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: fix integer format specifiers in GST_DEBUG seqnum in this function is 32-bit, so G_GUINT16_FORMAT would produce undefined output on big endian systems 2014-01-14 12:13:49 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: rtprtxsend: change the rtx_pt_map directly in set_property() instead of delaying it for chain() The same lock is held, so there is no point in complicating it... 2014-01-14 12:07:58 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxreceive.h: rtprtxreceive: change the rtx_pt_map directly in set_property() instead of delaying it for chain() The same lock is held, so there is no point in complicating it... 2014-01-14 11:55:00 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: simplify the code of finalize() 2014-01-14 11:52:21 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxreceive.h: rtprtxreceive: use the GstObject lock instead of a new one 2014-01-14 11:45:52 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: rtprtxsend: use the GstObject lock instead of a new one 2013-12-10 14:29:55 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: Add NV12_64Z32 support https://bugzilla.gnome.org/show_bug.cgi?id=722127 2014-01-14 19:08:49 +0900 Justin Joy * sys/oss/gstosshelper.c: osshelper: Don't leak fd when getting card name https://bugzilla.gnome.org/show_bug.cgi?id=722163 2014-01-14 09:43:33 +0000 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: Revert "aacparse: relax the detection of ADTS" This was pushed by mistake along with the V4L2 fix. This reverts commit 8eb4b032bef444397c4d211f2095c173ba114187. 2014-01-14 15:42:01 +0900 Justin Joy * gst/rtp/gstrtpg726pay.c: rtpg726pay: don't leak encoding_name string https://bugzilla.gnome.org/show_bug.cgi?id=722159 2014-01-13 09:14:00 +0000 Vincent Penquerc'h * sys/v4l2/v4l2_calls.c: v4l2: fix build break using V4L2_CAP_VIDEO_M2M_MPLANE This may not be defined. Since the previous version used only the other define (V4L2_CAP_VIDEO_OUTPUT_MPLANE), fall back on this only when not available. 2013-02-27 01:45:52 +0900 Akihiro Tsukada * gst/audioparsers/gstaacparse.c: aacparse: relax the detection of ADTS According to ISO/IEC 13818-7, "channel_config" field in ADTS header may have value of 0, as in the case of frame with PCE. gst_aac_parse_detect_streams() returned FALSE for those frames and discarded them. 2014-01-07 11:58:23 +0000 Julien Isorce * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: check set_config return value in gst_v4l2_buffer_pool_new https://bugzilla.gnome.org/show_bug.cgi?id=720568 2014-01-10 12:40:31 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Add parsed=1 field for encoded output https://bugzilla.gnome.org/show_bug.cgi?id=720568 2014-01-10 12:39:16 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't leak empty caps https://bugzilla.gnome.org/show_bug.cgi?id=720568 2014-01-08 16:51:21 +0000 Julien Isorce * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: do not stop a stream not previously started https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-12 16:27:21 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't enforce dimension field on encoded formats Don't enforce having width, height and framerate in template caps for encoded formats. These don't always need to be exposed and may break negotiation for decoder and decoding sink. If needed, these field will be automatically added when probed caps are known. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-12 17:09:59 +0000 Julien Isorce * sys/v4l2/gstv4l2object.c: v4l2object: unref downstream pool https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-18 13:37:23 -0500 Julien Isorce * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: add gst_v4l2_buffer_pool_flush STREAMOFF set all v4l2buffers to DEQUEUE state. Then for CAPTURE we call QBUF on each buffer. For OUTPUT the buffers are just push back in the GstBufferPool base class 's queue. But the loop actually looks like the same. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-16 17:29:30 -0500 Benjamin Gaignard * sys/v4l2/gstv4l2object.c: v4l2object: Add vp8 support https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-12 16:46:09 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't force framerate field for OUTPUT If there is nothing that seems to force a certain framerate on output device, it is preferable to simply not set that feild. This allow negotiation with tsdemux in a decoder for example. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-12 14:07:03 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: _v4l2fourcc_to_structure() can be static This function is not used anymore outside v4l2object. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-12 14:22:26 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Add MPEG1/2 support https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-12 12:18:45 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Ask for a decent buffer size when dealing with encoded formats https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-07 14:03:53 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: On warn on size change if n_planes > 1 https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-31 16:38:09 +0000 Julien Isorce * sys/v4l2/gstv4l2object.c: v4l2object: check if translated format is valid Also add a FIXME in gst_v4l2_object_setup_format to note that the whole function has to be improved in order to support ENCODED formats. It requires to have an encoder device which we do not have right now. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-07 10:31:15 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Validate returned dimensions https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 19:36:25 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Ensure max is not smaller then min in decide_allocation https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 19:36:06 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't keep the max paramter when using our own pool https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 19:34:44 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Respect the suggested min buffer https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 18:48:44 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Allocate pool if needed in decide_allocation https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 18:49:19 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Add V4L2_CID_MIN_BUFFERS_FOR_CAPTURE support https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 18:48:15 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: v4l2: Move decide allocation into v4l2object https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-05 13:51:13 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Implement _setup_format() This method allow setting up the object from the currently configured format on the device. This is useful for M2M element where input data decides the format that will be set on capture side. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-10 14:34:17 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Split out saving format from set_format() https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-31 15:37:26 +0000 Julien Isorce * sys/v4l2/gstv4l2object.c: v4l2object: set only one plane for encoded format https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-04 16:49:13 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Move code block where it belongs https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-04 16:26:12 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't check format specific information The number of plane, and the stride does not represent a capability change. Same caps can have different stride from the default GstVideoInfo and the number of planes will never change for 1 format. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-04 16:23:18 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2object: Move the extrapolation of stride at the right place Now that we have a stride array, we should extrapolate only when eeded (non multi-planar buffer). https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-04 15:09:44 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Move back assertions where they should be https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-04 15:09:10 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Move mplane logic into gst_v4l2_object_get_caps_info() It makes the gst_v4l2_object_set_format() slightly simplier and will make that logic reusable. Note that gst_v4l2_object_has_mplane() will always return the same value for one device. There is no need to check against the caps as this has already been done by _open. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-03 18:27:47 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Split _v4l2fourcc_to_video_format https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-02 18:05:11 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Request buffers only once VIDIOC_REQBUFS allocates buffer, it has no place inside set_config. Also, some driver do no allow multiple calls to this ioctl. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-02 15:26:50 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Don't validate dimension for encoded format We set the dimensions just in case but don't validate them afterwards. For some codecs the dimensions are *not* in the bitstream, IIRC VC1 in ASF mode for example. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-11-28 17:10:29 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Quirks for dev without initial format Most M2M have undefined behaviour initially when VIDIOC_G_FMT is called. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-11-28 17:09:26 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Add gst_v4l2_object_open_shared() https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-11-28 17:07:05 -0500 Nicolas Dufresne * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: v4l2object: Implement gst_v4l2_dup() This will duplicated the FD from another object and copy over the probed result. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-11-28 16:59:59 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: make IO_MODE enum public This is to allow adding a second io-mode property on M2M device like decoder so input and output can be controlled separatly. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-06-04 23:42:24 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: better handle quirks activation This way we can activate deactivate those quirks all at once at one place. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-06-04 23:34:04 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2: Fix h264 caps V4L2_PIX_FMT_H264 is documentated as byte-stream (with start code). The ensure proper negotiation with element like h264parse. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2013-12-06 14:44:51 -0500 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2object: Split caps in different categories This is need to correctly expose capabilities on specialized devices like decoders and encoders. https://bugzilla.gnome.org/show_bug.cgi?id=720568 2014-01-10 14:16:00 +0000 Tim-Philipp Müller * gst/matroska/matroska-read-common.c: matroskademux: don't leak TOC chapter list 2014-01-10 08:52:16 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: remove obsolete write-dummy-and-overwrite-on-eos code The need for rewriting apparently is obsolete 0.10 leftover. We now have caps for subtitles when we create the headers, so we always write the correct data in the first place. 2014-01-09 23:55:16 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: remove duplicate assignment Coverity CID 1151680 2014-01-09 18:25:04 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: write subtitle codec ID and data at start when known This avoids issues with writing dummy data first, then having to come back and write correct data later. Doing so prevents the muxed stream from being actually streamable. https://bugzilla.gnome.org/show_bug.cgi?id=712134 2014-01-09 17:32:15 +0100 Sebastian Dröge * configure.ac: configure: Include AvailabilityMacros.h for osxvideo check Otherwise MAC_OS_X_VERSION_MIN_REQUIRED might not be defined 2014-01-09 11:56:31 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: qtmux: respect the HDLR box string format for mov and isomedia Mov spec says it uses a pascal style string, while isomedia uses a null terminated one. Store the current atoms flavor into the HDLR to be able to generate the correct output. https://bugzilla.gnome.org/show_bug.cgi?id=705982 2014-01-08 11:28:04 +0100 Wim Taymans * gst/matroska/matroska-mux.c: Revert "matroskamux: Use the running time for container timestamps, not buffer timestamps" This reverts commit b3aa8755fe07639f22e4104f4932d769d6c9075a. We are already using the running-time because they were placed on the buffers with gst_collect_pads_clip_running_time(). Arguably it would be better to not modify the incomming buffers but collectpads seems to want to use absolute timestamps from the buffers for finding the best buffer (this can be changed with a custom compare function..). 2014-01-08 10:41:24 +0100 Sebastian Dröge * configure.ac: configure: Fix AC_COMPILE_IFELSE usage 2014-01-08 10:31:18 +0100 Sebastian Dröge * configure.ac: osxvideosink: Improve configure check for OSX >= 10.6 https://bugzilla.gnome.org/show_bug.cgi?id=721245 2014-01-07 12:13:51 -0800 Aleix Conchillo Flaqué * gst/rtpmanager/gstrtpbin.c: rtpbin: remove unused list of decoders remove list of decoders, which are already handled by the list of elements. https://bugzilla.gnome.org/show_bug.cgi?id=719938 2014-01-08 09:46:55 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Error out if ADPCM caps don't contain the layout field 2014-01-03 15:25:23 +0100 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: Add support for g726 ADPCM https://bugzilla.gnome.org/show_bug.cgi?id=720995 2014-01-07 15:04:02 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use new method to get media-type Use the new method to get the media type of a transport. 2014-01-06 21:12:17 +0100 Stefan Sauer * tests/check/elements/wavparse.c: wavparse: split the test This way one failure won't shadow the other test and also if one fails we get better disgnostics through the test-name. 2014-01-06 14:54:46 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Add HEVC / h265 support 2014-01-06 14:54:38 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Add HEVC / h265 support 2014-01-06 13:36:38 +0100 Stefan Sauer * gst/wavparse/gstwavparse.c: wavparse: remove ifdef'ed code We do have adtl and cue parse as part of toc handling alreday. The fmt code is a left over from <0.10 times. 2014-01-06 13:32:58 +0100 Stefan Sauer * gst/avi/gstavidemux.c: * gst/wavparse/gstwavparse.c: avidemux, waveparse: more logging for unhandled chunks Always print a warning with the tag and if possible do a memdump. 2014-01-05 22:47:42 +0100 Stefan Sauer * gst/avi/gstavidemux.c: avidemux: expose 'strn' - stream name - as title tag 2014-01-05 22:41:24 +0100 Stefan Sauer * gst/avi/gstavidemux.c: avidemux: parse fuji strd We can get maker, model and capture date from this chunk. Fixes #636143 2014-01-05 21:46:33 +0100 Stefan Sauer * gst/avi/gstavidemux.c: avidemux: ... and use the local api both times 2014-01-05 21:38:14 +0100 Stefan Sauer * gst/avi/gstavidemux.c: avidemux: copy the riff api for ncdt into the element This chunk is avi specific, no need to expose this as public api. 2014-01-05 10:28:21 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Add missing semicolon from last commit 2014-01-05 10:22:37 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Use the running time for container timestamps, not buffer timestamps Buffer timestamps have no real meaning here, and for selecting the next buffer we already use the running time anyway. 2014-01-04 21:34:38 +0100 Stefan Sauer * gst/avi/gstavidemux.c: avi: use new riff api to extract nikon metadata Fixes #636143 2013-11-01 16:41:43 +0000 Julien Isorce * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-rtpmanager.xml: rtprtxsend/rtprtxreceive: generate gtk doc 2013-12-02 11:26:09 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: test/check: Verify rtprtxsend::ssrc-map property works as expected 2013-11-29 19:35:44 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxreceive.h: * tests/check/elements/rtpaux.c: * tests/check/elements/rtprtx.c: * tests/examples/rtp/client-rtpaux.c: rtprtxreceive: modify to use a payload-type map like rtprtxsend 2013-11-29 19:58:26 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: do not keep history of packets with an unknown payload type This allows to disable retransmission per payload type by not putting a certain payload type in the map. 2014-01-02 15:18:52 +0100 Wim Taymans * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: * tests/check/elements/rtpaux.c: * tests/check/elements/rtpcollision.c: * tests/check/elements/rtprtx.c: * tests/examples/rtp/server-rtpaux.c: rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream Conflicts: tests/examples/rtp/server-rtpaux.c 2013-11-25 15:00:45 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: Add an rtx-ssrc property to allow external control of the ssrc This is useful when one needs to know the SSRC beforehands, so that it can be used for SRTP for example. 2013-11-13 15:11:35 -0500 Torrie Fischer * tests/examples/rtp/.gitignore: * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-rtpaux.c: * tests/examples/rtp/server-rtpaux.c: examples: rtp: Add end-to-end rtpbin example with RTX elements This example demonstrates how to use rtpbin with retransmission (rtx) elements set in the place of rtpbin's "aux" elements in order to enable RTP retransmission according to the rules of RFC4588. 2013-11-05 17:35:01 +0000 Julien Isorce * docs/design/Makefile.am: * docs/design/design-rtpauxiliary.txt: doc: add design-rtpauxiliary.txt to describe how rtpbin deals with auxiliary elements 2014-01-02 14:48:49 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: session: also push EOS event to RTCP srcpad 2014-01-02 14:46:11 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: place SSRC in Retransmission event 2013-11-01 16:57:15 +0000 Julien Isorce * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtpaux.c: tests/check: add rtpaux::test_simple_rtpbin_aux It shows how to use "set-aux-receive" and "set-aux-send" properties of rtpbin to set rtprtxsend and rtprtxreceive Build 2 pipelines, one for rtpbin as a sender and one for rtobin as a receive. Then transmit an audio stream. It also drops some packets to activate restransmission and check they are actually retransmited. 2013-11-01 17:09:42 +0000 Julien Isorce * tests/check/elements/rtpcollision.c: tests/check: add rtpcollision::test_rtx_ssrc_collision unit test check that rtxrtpsend changes its retransmission ssrc when collision happens 2013-11-06 12:34:13 +0200 George Kiagiadakis * tests/check/elements/rtprtx.c: tests/check: add rtprtx::test_rtxreceive_data_reconstruction This unit test verifies that retransmitted rtp packets coming out of rtprtxreceive are the same as the original ones. 2013-11-05 09:33:51 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: use a realistic limit for the value of max-size-packets G_MAXINT16 is chosen because if the queue contains more than G_MAXINT16 packets, seqnum comparison will not work properly. 2013-11-04 20:05:03 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: rtprtxsend: use a GSequence to implement the buffer queue This has the advantage that searching the queue to find the buffer with the requested seqnum is done with binary search. 2013-11-04 18:38:24 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: * tests/check/elements/rtprtx.c: rtprtxsend: retransmit packets in the same order as the rtx requests 2013-11-02 19:56:44 +0200 George Kiagiadakis * tests/check/elements/rtprtx.c: tests/check: Add unit test for rtxsend's max_size_time property 2013-10-29 18:27:00 +0100 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: rtprtxsend: Handle the max_size_time property This property allows you to specify the amount of buffers to keep in the retransmission queue expressed as time (ms) instead of buffer count (which is the max_size_buffers property). 2013-11-02 15:21:08 +0200 George Kiagiadakis * gst/rtpmanager/gstrtprtxsend.c: rtprtxsend: keep important buffer information in a private structure This is to avoid mapping a buffer every time we need to read a seqnum or a timestamp. 2013-11-01 11:58:47 +0100 George Kiagiadakis * tests/check/elements/rtprtx.c: tests/check: Add rtprtx::test_rtxsender_packet_retention This unit test verifies that the rtxsend element correctly maintains a buffer of already transmitted rtp packets and that it can re-transmit all of them correctly on demand. It also verifies that the limit of this buffer (max-size-packets property) is respected. 2013-11-01 16:22:13 +0000 Julien Isorce * tests/check/elements/rtprtx.c: tests/check: add rtprtx::test_drop_multiple_sender unit test Several senders / one receiver Similar than test_drop_one_sender but with multiple senders mixed through the funnel element. It drops some packets and checks that they are retransmited correctly. 2013-11-01 16:21:00 +0000 Julien Isorce * tests/check/elements/rtprtx.c: tests/check: add rtprtx::test_drop_one_sender unit test Test for one sender / one receiver Build the pipeline videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink and drop some buffers between rtprtxsend and rtprtxreceive Then it checks that every dropped packet has been re-sent. It also checks that not too much requests has been sent. 2013-11-01 16:17:51 +0000 Julien Isorce * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtprtx.c: tests/check: add rtprtx::test_push_forward_seq add simple unit test that manually push buffers in rtprtxsend connected to rtprtxreceive. Drops some buffers and make sure they are retransmisted. 2013-11-01 15:52:03 +0000 Julien Isorce * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtprtxreceive.c: * gst/rtpmanager/gstrtprtxreceive.h: * gst/rtpmanager/gstrtprtxsend.c: * gst/rtpmanager/gstrtprtxsend.h: rtpmanager: add new rtprtxsend / rtprtxreceive elements The purpose of the sender RTX object is to keep a history of RTP packets up to a configurable limit (in time). It will listen for custom retransmission events from downstream. When it receives a request for retransmission, it will look up the requested seqnum in its list of stored packets. If the packet is available, it will create a RTX packet according to RFC 4588 and send this as an auxiliary stream. The receiver will listen to the custom retransmission events from the downstream jitterbuffer and will remember the SSRC1 of the stream and seqnum that was requested. When it sees a packet with one of the stored seqnum, it associates the SSRC2 of the stream with the SSRC1 of the master stream. From then on it knows that SSRC2 is the retransmission stream of SSRC1. This algorithm is stated in RFC 4588. For this algorithm to work, RFC4588 also states that no two pending retransmission requests can exist for the same seqnum and different SSRCs or else it would be impossible to associate the retransmission with the original requester SSRC. When the RTX receiver has associated the retransmission packets, it can depayload and forward them to the source pad of the element. RTX is SSRC-multiplexed Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084 2013-11-05 16:36:46 +0000 Julien Isorce * docs/design/Makefile.am: * docs/design/design-rtpretransmission.txt: doc: add design for rtp retransmission Describe how rtprtxsend and rtprtxreceive generally work but also how the association algorithm is implemented. 2014-01-02 20:23:05 -0300 Reynaldo H. Verdejo Pinochet * ext/soup/gstsouphttpsrc.c: souphttpsrc: use status code macro instead of 407 Rest of the code is using the _PROXY_AUTHENTICATION_REQUIRED macro too. Easier to understand if you don't recall HTTP error codes by heart. 2013-12-31 21:31:43 -0300 Reynaldo H. Verdejo Pinochet * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2send: change audio_format field to format This element and the underlying libshout2 library can handle video media files too. The code already handles video/webm so the name gets confusing. Also add and use DEFAULT_FORMAT macro Instead of hardwiring SHOUT_FORMAT_VORBIS at init https://bugzilla.gnome.org/show_bug.cgi?id=721342 2013-12-31 20:09:29 -0300 Reynaldo H. Verdejo Pinochet * ext/shout2/gstshout2.c: shout2send: clarify meaning of the URL prop https://bugzilla.gnome.org/show_bug.cgi?id=721342 2013-12-27 12:27:32 -0300 Reynaldo H. Verdejo Pinochet * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/shout2/gstshout2.c: shout2send: docs, add a sample pipeline And finish adding shout2send to the docs while at it https://bugzilla.gnome.org/show_bug.cgi?id=721342 2013-12-31 15:00:22 -0300 Reynaldo H. Verdejo Pinochet * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: remove spurious @see_also 2013-12-06 17:08:54 +0000 Matthieu Bouron * gst/deinterlace/gstdeinterlace.c: deinterlace: support any video formats and any caps features if deinterlace mode allows it https://bugzilla.gnome.org/show_bug.cgi?id=719636 2013-12-31 13:31:52 +0100 Sebastian Rasmussen * sys/v4l2/gstv4l2object.c: v4l2: Handle v4l2_ioctl() errors even in error handling Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721268 2014-01-01 12:11:43 -0800 Jeremy Huddleston Sequoia * sys/osxvideo/Makefile.am: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideo: unifdef -DRUN_NS_APP_THREAD 2014-01-01 12:10:01 -0800 Jeremy Huddleston Sequoia * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: osxvideo: Assume SDK and deployment target are at least Snow Leopard 2014-01-01 12:23:50 -0800 Jeremy Huddleston Sequoia * configure.ac: configure: Disable osxvideo on Leopard and earlier This also moves the "other platforms" check in OS X video to before the variable is read https://bugzilla.gnome.org/show_bug.cgi?id=721245 2013-12-31 14:57:27 +0100 Wim Taymans * tests/check/elements/rtpbin.c: tests: add AUX receiver unit test 2013-12-31 13:20:01 +0100 Wim Taymans * tests/check/elements/rtpbin.c: tests: improve rtpbin test 2013-12-31 13:16:46 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: add some docs about AUX elements 2013-12-31 13:01:22 +0100 Wim Taymans * tests/check/elements/rtpbin.c: tests: add AUX sender unit test 2013-12-31 12:31:25 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: add support for AUX sender and receiver AUX elements are elements that can be inserted into the rtpbin pipeline right before or after 1 or more session elements. The AUX elements are essential for implementing functionality such as error correction (FEC) and retransmission (RTX). Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087 2013-12-31 12:22:39 +0100 Wim Taymans * tests/check/elements/rtpbin.c: tests: add decoder test 2013-12-30 17:36:42 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: make request_element method internally We can use the same method to create encoder and decoder elements, they are just internal elements that we create. 2013-12-31 10:25:28 +0100 Stéphane Cerveau * gst/wavparse/gstwavparse.c: wavparse: Skip id3 tag Skip id3 tag during wav parse. https://bugzilla.gnome.org/show_bug.cgi?id=721241 2013-12-31 10:10:05 +0100 Sebastian Dröge * sys/osxaudio/gstosxcoreaudio.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: osx: Make OSX version checks more consistent And especially also consider update versions, e.g. 10.5 with updates will be 1051 or similar and thus bigger than MAC_OS_X_VERSION_10_5 but still won't have the API we want to use. 2013-12-31 10:07:22 +0100 Jeremy Huddleston * sys/osxvideo/osxvideosink.h: osxvideosink: Fix build on updated OS X Leopard https://bugzilla.gnome.org/show_bug.cgi?id=721245 2013-12-30 17:23:22 +0100 Edward Hervey * gst/avi/gstavimux.c: avimux: Add missing break I guess no-one noticed we no longer could mux WMV3 ... COVERITY CID 1139759 2013-12-30 17:20:37 +0100 Edward Hervey * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: Add missing break COVERITY CID 1139762 2013-12-30 17:00:45 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: internal-ssrc is no longer deprecated 2013-12-30 16:59:20 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: add Since tags 2013-12-30 16:52:28 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: add signal for new jitterbuffer Emit a signal when a new jitterbuffer is created so that the app can have a chance to configure it. 2013-12-30 16:28:57 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * tests/check/elements/rtpbin.c: rtpbin: handle multiple encoder instances Keep track of elements that are added to multiple sessions and make sure we only add them to the rtpbin once and that we clean them when no session refers to them anymore. 2013-12-30 15:16:09 +0100 Wim Taymans * tests/check/elements/rtpbin.c: tests: add unit test for encoder element 2013-12-30 15:15:43 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: fix memory leaks 2013-12-30 15:03:34 +0100 Wim Taymans * tests/check/elements/rtpbin.c: tests: fix leak 2013-12-30 15:00:50 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: expect the pads on the encoders Don't use request pads for the encoder elements, the signal handler should request the pads and make sure they are available with the right name. 2013-12-30 14:56:07 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: request-rtp-encoder are no action signals The request-rtp-encoder signals are not action signals so mark them correctly and use an accumulator to collect the result value. 2013-12-30 14:36:45 +0100 Stefan Sauer * gst/wavparse/gstwavparse.c: wavparse: emit midi-base-note tag from data in 'smpl' chunk Add parsing of the 'smpl' chunk. Right now we only grab the midi-base-note and emit it as a tag. 2013-12-26 12:05:19 +0200 George Kiagiadakis * gst/rtpmanager/gstrtpsession.c: gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision When a collision is found on the internal ssrc, we have to change it. Ideally, we want also the payloader upstream to follow this change and use the new internal ssrc. Ideally we want this condition to be always met: if there is one payloader sending on this session, its ssrc should match the internal ssrc. 2013-12-26 11:04:29 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: allow setting internal-ssrc again 2013-12-30 13:31:45 +0100 Edward Hervey * gst/y4m/gsty4mencode.c: y4mencode: Remove dead code set/get property isn't used 2013-12-30 13:30:24 +0100 Edward Hervey * gst/rtp/gstrtpqcelpdepay.c: rtpqcelpdepay: Remove uneeded variable 2013-12-05 15:53:52 -0800 Aleix Conchillo Flaqué rtpbin: allow dynamic RTP/RTCP encoders/decoders * gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder and request-rtcp-decoder). The user will be able to provide encoders or decoders dynamically. The encoders must follow the srtpenc API and the decoders the srtpdec API. Having separate signals for RTP and RTCP allows the user to use different encoders/decoders or provide the same one (e.g. that would be the case for srtpenc). Also, rtpbin now allows application/x-srtp in its pads. https://bugzilla.gnome.org/show_bug.cgi?id=719938 2013-12-27 16:51:32 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: dynamically recalculate RTX parameters Use the round-trip-time and average jitter to dynamically calculate the retransmission interval and expected packet arrival time. Based on patches from Torrie Fischer Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412 2013-12-27 16:50:52 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: calculate average jitter 2013-12-27 16:48:48 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: rtpsession: use RTT from the Retransmission event Place the estimated RTT in the Retransmission event and let the session manager use that instead of the hardcoded value. 2013-12-27 15:57:39 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: take more accurate running-time for NACK Don't use the current time calculated from the tmieout loop for when we last scheduled the NACK because it might be unscheduled because of a max packet misorder and then we don't accurately calculate the current time. Instead, take the current element running time using the clock. 2013-12-30 11:06:38 +0100 Sebastian Dröge * tests/check/elements/wavpackdec.c: wavpackdec: Send a CAPS event in the unit test 2013-12-27 02:14:02 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: improve mss_mode/fragmented special handling Make it clear what should be handled purely by mss mode: 1) Expose the streams on the first moof as there are no moov atoms 2) Properly cleanup streams on flushes Add a note about the meaning of upstream_newsegment and mss_mode for future reference. Make all other special fragment handling shared for both dash and mss streams. 2013-12-12 10:50:27 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: drain the adapter before pushing EOS In a fragmented scenario, qtdemux is operating in push mode and it gets a fragmented buffer. While processing its data downstream gets unlinked (or a input-selector changes its active pad and returns not-linked). Qtdemux stops processing this fragment and returns not-linked upstream, leaving the remaining data in its adapter. When it gets an EOS it should make sure that all the data it had received is pushed before pushing EOS. 2013-12-26 23:21:47 -0300 Reynaldo H. Verdejo Pinochet * ext/shout2/gstshout2.c: shout2send: drop IP only requirement for _set_host() libshout2 (we require > 2.0 at config time) supports both IP and hostname for _set_host(). Dropped an outdated FIXME regarding this limitation, adjusted some comments and changed the param blurb to reflect this too. 2013-12-26 21:43:34 -0300 Reynaldo H. Verdejo Pinochet * ext/shout2/gstshout2.c: shout2send: Retarget FIXME to 2.0 2013-12-26 11:21:36 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN Use the aggregate control instead of the original request url to perform PAUSE/PLAY and TEARDOWN. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003 2013-12-24 14:40:25 +0100 Sebastian Dröge * gst/debugutils/rndbuffersize.c: rndbuffersize: Proxy CAPS, ALLOCATION, SCHEDULING and srcpad events properly 2013-12-24 00:43:39 +0100 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: adpcm max block align is 8192 2013-12-23 12:23:27 -0600 Brendan Long * configure.ac: vp9dec: Require vpx >= 1.3.0 for building vp9dec and vp9enc Previous versions did not have a stable bitstream for VP9. https://bugzilla.gnome.org/show_bug.cgi?id=720986 2013-12-23 15:46:48 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Use correct codec id for ADPCM/DVI 2013-12-23 15:44:30 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Check for the correct size of codec_data in the ACM case 2012-01-14 19:58:17 +0100 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: basic adpcm support https://bugzilla.gnome.org/show_bug.cgi?id=664339 2013-12-20 11:45:38 +0100 Sebastian Dröge * gst/isomp4/descriptors.c: qtdemux: Fix calcuation of descriptor length https://bugzilla.gnome.org/show_bug.cgi?id=720813 2013-12-22 22:33:39 +0000 Tim-Philipp Müller * autogen.sh: * common: Automatic update of common submodule From dbedaa0 to d48bed3 2013-12-22 21:56:03 +0000 Tim-Philipp Müller * po/Makevars: po: set gettext domain in Makevars so we don't have to patch the generated Makefile.in.in https://bugzilla.gnome.org/show_bug.cgi?id=705455 2013-12-19 16:50:10 +0000 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: on receive error only unmap and unref buffer if one was alloced and mapped coverity CID 1139866. 2013-12-19 12:47:22 +0000 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: fix misleading comment Those are not allocated on the stack. 2013-12-17 18:28:25 +0100 Sebastian Dröge * configure.ac: vpx: Mark VP9 support as non-experimental There was a libvpx release with VP9 support now and the bitstream is frozen too. 2013-12-15 21:04:11 -0800 Todd Agulnick * gst/deinterlace/gstdeinterlace.c: Some compiler warning fixes to satisfy XCode compiler https://bugzilla.gnome.org/show_bug.cgi?id=720513 2013-12-16 16:17:07 +0100 Sebastian Dröge * ext/taglib/gstid3v2mux.cc: id3v2mux: Set picture type in the APIC frames 2013-12-16 16:14:52 +0100 Sebastian Dröge * ext/taglib/gstid3v2mux.cc: id3v2mux: Set image-description from the info struct, not the caps 2013-12-16 10:02:37 +0100 Sebastian Dröge * gst/audioparsers/gstwavpackparse.c: * gst/audioparsers/gstwavpackparse.h: wavpackparse: Post AUDIO_CODEC tag 2013-12-16 10:00:37 +0100 Sebastian Dröge * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstsbcparse.h: sbcparse: Post AUDIO_CODEC tag 2013-12-16 09:58:31 +0100 Sebastian Dröge * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: Post AUDIO_CODEC tag https://bugzilla.gnome.org/show_bug.cgi?id=720512 2013-12-16 09:56:29 +0100 Sebastian Dröge * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: dcaparse: Post AUDIO_CODEC tag 2013-12-16 09:54:38 +0100 Sebastian Dröge * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstamrparse.h: amrparse: Post AUDIO_CODEC tag 2013-12-16 09:49:48 +0100 Sebastian Dröge * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: Post AUDIO_CODEC tag 2013-12-16 09:46:16 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: Post AUDIO_CODEC tag 2013-12-16 09:41:14 +0100 Sebastian Dröge * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: Use pbutils functionality to create the AUDIO_CODEC tag 2013-12-13 17:36:36 -0500 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Add error message if the app tries to set the internal-ssrc 2013-12-13 16:08:35 -0500 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Only count nacks when a nack packet is received Not when any RTCP feedback packet is. 2013-12-12 23:22:41 -0500 Olivier Crête * tests/check/elements/rtpcollision.c: tests: Initialize segment in rtpcollision test 2013-12-13 15:57:36 -0500 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Process PSFB FIR requests which lack the media ssrc According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL have a media_ssrc field set to 0. The actual media ssrc is in the FCI. So in that case, we ignore the retained feedback and just let it through to the rtp_session_process_fir() function which will check for the actual SSRC inside the FCI. Fixes a regression introduced by commit 57c27ec3 2013-11-14 16:19:29 +0200 George Kiagiadakis * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders Previously, when the session had multiple internal sender SSRCs, it would issue SR reports with RB blocks only on the first RTCP timeout and afterwards SR reports would be sent empty. This was because the "generation" number in RTPSource would increase more than once during the same cycle and afterwards it would always be greater than the session's generation, which would cause it to be skipped from being included in RBs. This commit fixes this problem by: 1) Increasing the RTPSource generation only at the end of each cycle, which essentially fixes the problem but only when the internal senders are less than GST_RTCP_MAX_RB_COUNT. 2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's SR the given RTPSource has been reported in, which also fixes the problem when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is necessary because of the fact that any RTPSource is marked as reported in itself's SR and makes it impossible to know if it has been reported in other SRs too or not, and which. 2013-11-14 16:23:35 +0200 George Kiagiadakis * tests/check/elements/rtpsession.c: tests/check: add an rtpsession unit test to verify all RBs are included in all SRs, roundrobin This test checks that when we have multiple internal sender sources in rtpsession, SRs contain RBs for every other sender source, and that they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT, which is the max number of RBs that can fit in a SR. 2013-12-12 16:01:10 +0100 Wim Taymans * docs/design/design-rtpcollision.txt: docs: improve docs 2013-11-05 18:03:48 +0000 Julien Isorce * docs/design/Makefile.am: * docs/design/design-rtpcollision.txt: doc: add design-rtpcollision.txt that explains when GstRTPCollision is created It also talks about "BYE only the corresponding source, not the whole session." 2013-11-05 12:31:54 +0000 Julien Isorce * tests/check/elements/rtpcollision.c: tests/check: improve rtpcollision::test_master_ssrc_collision to ensure that a collision does not BYE the whole session Conflicts: tests/check/elements/rtpcollision.c 2013-11-01 17:07:57 +0000 Julien Isorce * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtpcollision.c: tests/check: add rtpcollision::test_master_ssrc_collision unit test It checks the payloader changes its ssrc when collision happens 2013-12-12 10:38:43 +0100 George Kiagiadakis * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: keep extra stats for scheduling BYE Keep an extra stats structure for scheduling the BYE packets. When we decide to schedule BYE, make a copy of the current stats into the bye_stats. Then while we schedule the BYE, update and use only the bye_stats. When we finished scheduling the BYE packet, we use the regular stats again. 2013-12-12 10:34:38 +0100 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: when we schedule BYE, only deal with BYE sources When we are doing the RTCP timeout to schedule BYE packets, don't generate RTCP for all sources but only for the sources marked as BYE. 2013-12-12 10:32:48 +0100 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: reset state after scheduling BYE After we do RTCP, we are not scheduling bye anymore. 2013-12-12 10:31:38 +0100 George Kiagiadakis * gst/rtpmanager/rtpsession.c: rtpsession: also count NACKS when no signal was pending 2013-12-12 10:09:25 +0100 George Kiagiadakis * gst/rtpmanager/rtpsession.c: session: ignore RTCP packets for the BYE sources When we are scheduling BYE packets, ignore all RTCP for the sources that are scheduling a BYE packet. Other sources that are not scheduling BYE should continue receiving RTCP packets as usual. 2013-11-04 11:48:21 +0000 Julien Isorce * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: determine if the session is doing point-to-point In this case T_dither_max is set to 0 according to RFC 4585 2013-12-10 11:57:37 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: serialize events in the buffer Serialize events into the jitterbuffer by inserting them with a -1 seqnum. Update unit test to expect events from the streaming thread. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986 2013-12-10 11:04:06 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: detect -1 seqnum Keep the seqnum as a full guint so that we can check for -1 entries and deal with them correctly. Immediately try to push -1 seqnum. 2013-12-10 11:01:03 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: reorganize jitterbuffer items Keep the oldest item at the head and the newest items on the tail. This makes it easier to deal with -1 seqnums. 2013-12-09 23:34:10 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: correctly check for invalid values Check for -1 on the guint from the buffer item instead of on the guint16 or guint32. Also insert -1 seqnum at the head of the jitterbuffer. 2013-12-08 16:49:55 +0100 Alessandro Decina * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideosink: fix segfault when dealing with padded frames Fixes crashes with vtdec ! osxvideosink where VideoToolbox outputs padded UYVY 2013-12-06 17:58:13 -0500 Olivier Crête * gst/audiofx/gststereo.c: stereo: Port to GStreamer 1.0 API 2013-12-05 12:15:29 +0100 Sebastian Dröge * gst/law/mulaw-decode.c: mulawdec: Require caps to be set before accepting any data 2013-12-05 12:15:19 +0100 Sebastian Dröge * ext/wavpack/gstwavpackdec.c: wavpackdec: Require caps to be set before accepting any data 2013-12-05 12:13:33 +0100 Sebastian Dröge * ext/speex/gstspeexdec.c: speexdec: Require caps to be set before accepting any data 2013-12-05 12:13:10 +0100 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Require caps to be set before accepting any data 2013-12-05 11:42:15 +0100 Sebastian Dröge * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: vpx: Use new gst_video_decoder_set_needs_format() API 2013-12-04 16:23:43 -0500 Olivier Crête * ext/pulse/pulsesink.c: pulsesink: Free device_info in accepts caps https://bugzilla.gnome.org/show_bug.cgi?id=719811 2013-12-04 21:57:48 +0100 Sebastian Dröge * gst/rtp/gstrtptheorapay.c: rtptheorapay: Don't send headers twice if we got them from the caps already 2013-12-04 21:57:04 +0100 Sebastian Dröge * gst/rtp/gstrtptheorapay.c: rtptheorapay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:55:53 +0100 Sebastian Dröge * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: Don't send headers twice if we got them from the caps already 2013-12-04 21:54:16 +0100 Sebastian Dröge * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:17:03 +0100 Sebastian Dröge * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpstreamdepay.c: * gst/rtp/gstrtpstreamdepay.h: rtpstreamdepay: Add RFC4571 RTP stream depayloading element https://bugzilla.gnome.org/show_bug.cgi?id=719829 2013-12-04 10:12:46 +0100 Sebastian Dröge * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpstreampay.c: * gst/rtp/gstrtpstreampay.h: rtpstreampay: Add RFC4571 RTP stream payloading element https://bugzilla.gnome.org/show_bug.cgi?id=719829 2013-12-03 15:08:25 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: improve fragment-start tracking Some buffers can have multiple moov atoms inside and the strategy of using the gst_adapter_prev_pts timestamp to get the base timestamp for the media of the fragment would fail as it would reuse the same base timestamp for all moofs in the buffer instead of accumulating the durations for all of them. Heres a better explanation of the issue: qtdemux receives a buffer where PTS(buf) = X buf -> moofA | moofB | moofC The problem was that PTS(buf) was used as the base timestamp for all 3 moofs, causing all buffers to be X based. In this case we want only moofA to be X based as it is what the PTS on buf means, and the other moofB and moofC just use the accumulated timestamp from the previous moofs durations. To solve this, this patch uses gst_adapter_prev_pts distance result, this allows qtdemux to calculate if it should use the resulting pts or just accumulate the samples as it can identify if the moofs belong to the same upstream buffer or not. https://bugzilla.gnome.org/show_bug.cgi?id=719783 2013-11-21 12:29:28 +0000 Julien Isorce * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: add support for multi-planar V4l2 API in DMABUF mode Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754 2013-11-19 17:16:27 +0000 Julien Isorce * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: refactor by emulating one v4l2_plane in non-MPLANE mode so that the buffer informations can be retrieved the same way in both MPLANE and non-MPLANE mode. Here "emulating" means "manually fill in the plane". Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754 2013-11-13 12:05:40 +0000 Julien Isorce * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: v4l2: add support for multi-planar V4L2 API This api is in linux kernel since version 2.6.39, and present in all version 3. The commit that adds the API in master branch of the linux kernel source is: https://github.com/torvalds/linux/commit/f8f3914cf922f5f9e1d60e9e10f6fb92742907ad v4l2 doc: "Some devices require data for each input or output video frame to be placed in discontiguous memory buffers" There are newer structures 'struct v4l2_pix_format_mplane' and 'struct v4l2_plane'. So the pixel format is not setup with the same API when using multi-planar. Also for gst-v4l2, one of the difference is that in GstV4l2Meta there are now one mem pointer for each maped plane. When not using multi-planar, this commit takes care of keeping the same code path than previously. So that the 2 cases are in two different blocks triggered from V4L2_TYPE_IS_MULTIPLANAR. Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754 2013-12-04 09:12:07 +0100 Wim Taymans * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: don't leak template caps 2013-12-03 21:41:28 +0100 Wim Taymans * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: * tests/check/elements/aacparse.c: audioparsers: use ACCEPT_INTERSECT flag The parser can accept input that is not completely specified. Use the ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to check for intersection only. This allows us to proxy downstream constraints while still allowing non-subset caps as input. We can then also remove the appended template caps workaround. Make a unit-test to check the new feature. This reverts commit 26040ee38cb9e7c42f3d9a0282b3e5cace7ca42d Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024 2013-12-03 21:36:54 +0100 Wim Taymans * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: remove fields from filter We need to remove the fields from the filter when we can convert between them. 2013-12-03 21:29:13 +0100 Wim Taymans * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: refactor code to remove caps fields 2013-12-02 00:10:43 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: microoptimisation: avoid some unnecessary GValue copies 2013-12-01 23:32:20 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix off-by-one crash when downstream caps contain a list of framerates https://bugzilla.gnome.org/show_bug.cgi?id=719544 2013-11-29 11:26:05 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: Use the timestamp of the moof as the base fragment start In SmoothStreaming fragmented scenario, the timestamps are calculated starting from the fragment buffer timestamp. When there is a not-linked return from downstream, qtdemux will return upstream and will keep the non-pushed data into its adapter. On a new fragment buffer pushed to qtdemux, the new buffer timestamp would overwrite the previous one that should be used on the still to be pushed buffers. Because of this, this patch will also update the fragment_start timestamp from the adapter last pts to make sure the moof and timestamps are in sync and will result in correct timestamps for all fragments. 2013-11-15 08:54:07 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: avoid re-reading the same moov and entering into loop In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux could read the moov again after the mdat because it was considering the media as a fragmented one. To avoid this loop this patch makes it store the last processed moov_offset to avoid parsing it again. And it also checks if there are any samples to play before resturning to the mdat, so that it knows there is new data to be played. https://bugzilla.gnome.org/show_bug.cgi?id=691570 2013-11-15 00:52:53 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: do not free streams if they were not created locally When parsing a trak only free streams on failures if those streams were created locally. They could have been created from a previous fragment, in this case we they have valid info from the other fragment. Including pads. https://bugzilla.gnome.org/show_bug.cgi?id=691570 2013-11-29 19:57:46 +0100 Sebastian Dröge * gst/videomixer/blend.c: videomixer: Simplify NV12/21 blending code macros 2013-11-29 19:50:24 +0100 Sebastian Dröge * gst/videomixer/blend.c: videomixer: Fix segfault when filling the background of a UYVY frame https://bugzilla.gnome.org/show_bug.cgi?id=712401 2013-11-29 09:21:52 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: fix compilation with gst debuging disabled qtdemux.c:9452:1: error: label at end of compound statement 2013-11-27 17:02:00 +0100 Jonas Holmberg * gst/rtp/gstrtph264pay.c: rtph264pay: Map inbuffer once only Do not call gst_buffer_extract() twice since each call will map and unmap the biffer. https://bugzilla.gnome.org/show_bug.cgi?id=719434 2013-11-28 11:58:42 -0500 Nicolas Dufresne * tests/check/elements/videofilter.c: videoflip: Add unit test for the 'automatic' method These new tests send a tag event before seding the buffer. Tested case are an empty tag list, a tag list with orientation-180 set and an invalid orientation value. https://bugzilla.gnome.org/show_bug.cgi?id=719497 2013-11-28 16:09:04 +0000 Tim-Philipp Müller * gst/videofilter/gstvideoflip.c: videoflip: don't crash on tag events without orientation tag Would crash in g_free() trying to free an uninitialised pointer. https://bugzilla.gnome.org/show_bug.cgi?id=719497 2013-11-28 16:50:42 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: don't unref buffer twice Cleaning the packet info will already unref the buffer. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078 2013-11-28 22:35:02 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Add HydrogenAudio ReplayGain tags Identical to the itunes (tm) version, but labelled with org.hydrogenaudio.replaygain as the producer. 2013-11-27 16:15:12 +0100 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: explicitly fail when alpha information would have been lost. 2013-05-29 16:06:05 -0400 Nicolas Dufresne * .gitignore: gitignore: Updated to ignore *.swp and .dirstamp 2013-11-26 11:17:42 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroska-demux: Allow a bit more variation when detecting common framerates Instead of +/- 1ns we allow 2ns now. Due to rounding errors there are some Matroska files out there with 33.333331ms per frame for 30fps. 2013-11-26 10:20:31 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroska-demux: Use gst_util_double_to_fraction() instead of GValue magic 2013-11-25 14:03:21 -0500 Nicolas Dufresne * gst/videofilter/gstvideoflip.c: videoflip: Set default method at contruction Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712333 2013-05-29 15:57:09 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.c: v4l2object: Use space instead of tabs https://bugzilla.gnome.org/show_bug.cgi?id=712754 2013-05-29 15:44:31 -0400 Nicolas Dufresne * sys/v4l2/gstv4l2object.h: v4l2object: Fix header indentation so it's readable again It's unfortunate to have to do this, but with the mix of tabs and space, plus all the random indentation this header has become very hard to read. https://bugzilla.gnome.org/show_bug.cgi?id=712754 2013-11-25 17:38:06 +0100 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: check: fix jitterbuffer check Don't advance the clock to 240ms too early. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710013 2013-11-25 11:45:33 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: deprecate max-errors The property wasn't use internally, let the base class handle the number of errors to tolerate. 2013-11-25 15:49:07 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: improve clear-pt-map handling Don't reset the expected output seqnum when clearing the pt map because this could stall the jitterbuffer forever. Add a unit test for this. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800 2013-10-28 21:33:22 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: let the base class decide when to return an error The base videodecoder class has an error counting feature to tolerate a few errors before posting an error message. So don't force the error and let the base class decide when it should happen https://bugzilla.gnome.org/show_bug.cgi?id=710762 2013-10-28 21:28:33 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: Add data skipping on input Add missing bytes skipping when bad input is received. https://bugzilla.gnome.org/show_bug.cgi?id=710762 2013-11-25 12:13:43 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: qtdemux: Discard 2 byte subpicture packets As for text subtitles and as suggested in #712643, throw away the 2 byte terminator packets that some encoders insert. This will make things better when remuxing and causes generation of gap events. 2013-11-25 00:34:21 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix wake-up when new buffers come in after running empty Spotted by 'gratias' on IRC. Probably introduced in recent refactoring. https://bugzilla.gnome.org/show_bug.cgi?id=715039 2013-11-23 12:15:40 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: correctly handle negative relative timestamps ... rather than scaling these as unsigned. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712744 Based on patch by Krzysztof Kotlenga 2013-09-14 03:27:09 +0200 MathieuDuponchelle * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer2: Merge tag events to send them in collected. Otherwise there were race conditions where we would send tags on a flushing srcpad. We have a test for that in GES, but this should be tested systematically with harness in the future as I believe it is useful for exactly that kind of cases. https://bugzilla.gnome.org/show_bug.cgi?id=708165 2013-11-14 17:29:50 -0300 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Use GstVideoInfo helper to create caps for raw video This way we do not miss mandatory fields in caps. At the same time use the gst_pb_utils_get_codec_description helper to get codec description. https://bugzilla.gnome.org/show_bug.cgi?id=712335 2013-11-14 16:11:38 -0300 Thibault Saunier * gst/matroska/Makefile.am: * gst/matroska/matroska-demux.c: matroskademux: Use GstVideoInfo helper to create caps for raw video This way we do not miss mandatory fields in caps. At the same time use the gst_pb_utils_get_codec_description helper to get codec description. https://bugzilla.gnome.org/show_bug.cgi?id=712328 2013-11-13 20:18:17 -0300 Thibault Saunier * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: multifilesrc: Implement seeking in case of multiple images https://bugzilla.gnome.org/show_bug.cgi?id=712254 2013-11-22 12:26:21 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: pass downstream flowreturn to upstream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722 2013-11-18 14:27:48 +0100 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: clear cached caps on close A different device with different caps may be used for the next open. https://bugzilla.gnome.org/show_bug.cgi?id=712611 2013-11-21 15:30:34 +0000 Tim-Philipp Müller * ext/wavpack/gstwavpackcommon.c: * ext/wavpack/gstwavpackstreamreader.c: * gst/apetag/gstapedemux.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/dtmf/gstrtpdtmfsrc.c: * gst/isomp4/atoms.c: * gst/matroska/matroska-demux.c: g_memmove() is deprecated Just use plain memmove(), g_memmove() is deprecated in recent GLib versions. https://bugzilla.gnome.org/show_bug.cgi?id=712811 2013-11-21 11:32:15 +0100 Wim Taymans * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: rtpvorbisdepay: handle packets > 0xffff Handle input packet sizes larger than 16 bits in the depayloader. Remove size restrictions on the payloader. 2013-11-21 11:30:28 +0100 Wim Taymans * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: rtptheoradepay: handle packets > 0xffff Reorganize some things in the depayloader so that it can handle packets larger than 16 bits. Remove the size restriction on the payloader. 2013-11-21 02:28:27 +1100 Jan Schmidt * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_types.c: isomp4: Handle mp4s subpicture streams better. Clean up the handling of mp4s streams. Use the generic esds descriptor function to extract the palette, instead of hard coding a wrong magic offset. Add some more size safety checks when parsing ES descriptors, and replace magic numbers with the descriptive constants that are already defined. Enhance dump output for stsd atoms. Streams from both bug 712643 and historic bug 568278 now both work correctly. Fixes: #712643 2013-11-20 22:08:25 +1100 Jan Schmidt * gst/isomp4/fourcc.h: qtdemux: Sort fourcc declarations and remove duplicates 2013-11-20 21:41:47 +1100 Jan Schmidt * gst/isomp4/Makefile.am: * gst/isomp4/atoms.h: * gst/isomp4/fourcc.h: * gst/isomp4/ftypcc.h: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_types.c: qtdemux: Merge all the fourcc headers into one Remove qtdemux_fourcc.h and ftypcc.h and put it all in fourcc.h 2013-11-19 10:10:51 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: avoid mapping the buffer Reuse the parsed structure to get the timestamps. 2013-11-18 17:13:49 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: fix 'make check' Fix generic/states check. Also, g_return_if_fail() is not for internal state checking. 2013-11-18 14:44:36 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiopanorama.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/deinterlace/gstdeinterlace.c: * gst/flv/gstflvmux.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * gst/multifile/gstsplitfilesrc.c: * gst/multipart/multipartdemux.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstmultiudpsink.c: * gst/videobox/gstvideobox.c: * gst/wavparse/gstwavparse.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/v4l2/gstv4l2object.c: * sys/ximage/gstximagesrc.c: docs: get rid of 'Since: 0.10.x' markers And some gtk-doc markup fixes. 2013-11-16 12:15:14 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: rtpmanager: fix Since markers Should be next stable release series version 2013-11-15 13:48:07 +0200 George Kiagiadakis * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: Fix stats property field names and documentation 2013-11-15 15:20:14 +0100 Torrie Fischer * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: gstrtpsession: Implement a number of feedback packet statistics Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693 2013-11-13 17:11:08 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: remove math operation from loop The elst_offset doesn't change inside the loop, so compute it outside 2013-11-14 20:54:32 +0100 Stefan Sauer * gst/isomp4/qtdemux.c: qtmux: fix playback regression In ae1150e85cf99d7482933aa6f7e4f012fe45a3ec flipping a condition misaligned the else branch, where for there condition that was change there is none. Fixes #712303 2013-11-14 09:20:06 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: rename property to 'stats' This makes the unit test work. We can later also add more stats, not specific to retransmission. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411 2013-11-12 11:19:25 -0500 Torrie Fischer * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: implement rtx statistics 2013-11-13 10:42:21 +0000 Marc Leeman * sys/v4l2/gstv4l2object.c: v4l2object: print FOURCC_FORMAT when enumerating https://bugzilla.gnome.org/show_bug.cgi?id=712206 2013-11-06 12:40:06 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: advance expected seqnum after dropping After dropping a buffer, move our expected seqnum Conflicts: gst/rtpmanager/gstrtpjitterbuffer.c 2013-11-04 15:46:22 +0100 Wim Taymans * gst/rtp/gstrtpgstpay.c: gstpay: only send one caps Only send one caps in a packet. Two caps can happen when setcaps is called and the config-interval expires at the same time. 2013-11-13 10:23:19 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Use the synced buffer mode in auto mode if a clock provider is in the SDP 2013-11-08 11:09:21 +0000 Marc Leeman * sys/v4l2/gstv4l2bufferpool.c: v4l2: init v4l2_buffer to 0x0 before ioctl https://bugzilla.gnome.org/show_bug.cgi?id=712137 2013-11-11 15:27:18 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: remove collision reconfigure event Remove bogus reconfigure event on collision, we don't want to send the event on the receiving RTP pad and the collision event is now handling this case. See https://bugzilla.gnome.org/show_bug.cgi?id=711560 2013-11-01 17:04:28 +0000 Julien Isorce * gst/rtpmanager/gstrtpsession.c: gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad See https://bugzilla.gnome.org/show_bug.cgi?id=711560 2013-11-11 14:25:51 +0100 Wim Taymans * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/rtpsession.c: check: add rtpsession test Add a basic rtpsession test to ensure that RR blocks are generated when multiple SSRC senders are active. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711270 2013-11-11 13:17:25 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: correctly handle timestamps when parsing x-private1-ac3 ... the way it has always worked fine in a52dec. 2013-11-05 10:48:33 +0200 George Kiagiadakis * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost The problem here was that the jitterbuffer lock was unlocked to push the event, but that caused another thread to remove the timer currently being processed, probably because the amount of rtx events (and therefore timers) was getting too high. The solution is to unlock and push the event only after timer processing has finished. fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131 2013-10-24 13:16:42 +0200 Per x Johansson * gst/matroska/matroska-demux.c: matroskademux: Avoid division by zero assert in gst_matroska_demux_search_pos https://bugzilla.gnome.org/show_bug.cgi?id=711829 2013-11-08 17:59:24 +0100 Philippe Normand * gst/wavenc/gstwavenc.c: wavenc: generate a non-empty data header Restore the behavior of the element to the state before commit db29522a430e44450415ca3676abd1b77ee923d9. A non-empty header is generated and when the EOS event is received the header is generated again, this time with the correct size. https://bugzilla.gnome.org/show_bug.cgi?id=711699 2013-11-07 16:17:16 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpsource: update receiver stats for sender An internal sender in a session is also a receiver of its own packets so update the receiver stats. Other senders in the session will use this info to generate correct RB blocks in their SR reports. 2013-11-07 16:13:16 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: refactor receiver stats update 2013-10-25 18:22:00 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: handle fragmented files with mdat before moofs Assume a file with atoms in the following order: moov, mdat, moof, mdat, moof ... The first moov usually doesn't contain any sample entries atoms (or they are all set to 0 length), because the real samples are signaled at the moofs. In push mode, qtdemux parses the moov and then finds the mdat, but then it has 0 entries and assumes it is EOS. This patch makes it continue parsing in case it is a fragmented file so that it might find the moofs and play the media. https://bugzilla.gnome.org/show_bug.cgi?id=710623 2013-10-25 11:42:37 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: When using a buffered mdat, store all received data for later use In push mode, when qtdemux can't use a seek to skip the mdat buffer it has to buffer it for later use. The issue is that after parsing the next moov/moof, there might be some trailing bytes from the next atom in the file. This data was being discarded along with the already parsed moov/moof and playback would fail to continue after the contents of this moov/moof are played. This is particularly bad on fragmented files that have the mdat before the corresponding moof. So you'd get: mdat|moof|mdat|moof ... When a moof was received, it usually came with some extra bytes that would belong to the next mdat (because upstream doesn't care about atoms alignment). So those bytes were being discarded and playback would fail. This patch makes qtdemux store those extra bytes to reuse them later after the mdat is emptied. https://bugzilla.gnome.org/show_bug.cgi?id=710623 2013-11-07 09:49:55 +0100 Sebastian Dröge * gst/udp/gstmultiudpsink.c: multiudpsink: Also use the bind-port property if no bind-address was given 2013-11-07 00:51:12 +0100 Andoni Morales Alastruey * sys/osxaudio/gstosxcoreaudiohal.c: osxaudiosink: fix segfault when we can't get the channels layout 2013-11-05 17:26:49 +0100 Sebastian Dröge * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: Make Picture ID mode configurable and default to no picture ID Some implementations (linphone) only support no picture at all in the stream and will fail if one is provided. https://bugzilla.gnome.org/show_bug.cgi?id=711497 2013-11-05 11:18:34 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 865aa20 to dbedaa0 2013-01-29 10:51:07 +0100 Paul HENRYS * gst/rtp/gstrtph264pay.c: Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event https://bugzilla.gnome.org/show_bug.cgi?id=692787 2013-11-02 22:50:47 +0100 Rico Tzschichholz * gst/rtsp/Makefile.am: * gst/rtsp/gstrtspsrc.h: rtsp: Add missing gio-2.0 deps and includes 2013-11-01 18:31:36 +0100 Sebastian Dröge * gst/audiofx/audioiirfilter.c: audioiirfilter: Fix initialization coefficient handling Broke unit test. 2013-10-31 14:05:43 -0700 Aleix Conchillo Flaque * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: allow setting tls certificate validation flags Added a new property "tls-validation-flags". If the url transport is TLS, the validation flags will be set to the rtsp connection. https://bugzilla.gnome.org/show_bug.cgi?id=711230 2013-10-31 22:43:49 +0100 Sebastian Dröge * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioiirfilter.c: audioiirfilter: Don't crash if no filter coefficients are provided ...and by default use a identity filter. https://bugzilla.gnome.org/show_bug.cgi?id=710215 2013-10-31 19:15:12 +0100 Sebastian Dröge * ext/wavpack/gstwavpackenc.c: wavpackenc: Fix writing of MD5 sums and other metadata blocks These don't have the FINAL_BLOCK flag set. 2013-10-31 13:02:11 -0200 Djalma Lúcio Soares da Silva * ext/raw1394/gsthdv1394src.c: hdv1394src: Make it possible to select a camera by its GUID The source hdv1394src has the guid property that permits select a camera connected from its GUID number. However when this property is setted the selected camera is not changed. The source continues using the default camera. This problem was solved using the function iec61883_cmp_connect. The reference for the function could be found here: http://www.dennedy.org/libiec61883/API-iec61883-cmp-connect.html The solution came from dvgrab source code. https://bugzilla.gnome.org/show_bug.cgi?id=710415 2013-10-31 13:20:41 -0300 Thiago Santos * tests/check/elements/souphttpsrc.c: tests: souphttpsrc: add explicit cast to silence warning Silencing this warning: elements/souphttpsrc.c:533:14: error: comparison between ‘SoupKnownStatusCode’ and ‘enum ’ [-Werror=enum-compare] if (status != SOUP_STATUS_OK && !send_error_doc) With gcc 4.8.2 (debian) 2013-10-31 10:38:35 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.h: * gst/rtsp/gstrtspsrc.c: rtspsrc: proxy new buffer mode 2013-10-30 16:49:36 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: add new timestamp mode Add a new timestamp mode that assumes the local and remote clock are synchronized. It takes the first timestamp as a base time and then uses the RTP timestamps for the output PTS. 2013-10-30 22:12:45 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroska-demux: Fix compiler warning matroska-demux.c: In function 'gst_matroska_demux_add_stream': matroska-demux.c:1379:7: error: format '%u' expects argument of type 'unsigned int', but argument 4 has type 'guint64' [-Werror=format=] "%03u", context->uid); ^ 2013-10-28 13:21:15 +0000 Matthieu Bouron * gst/videomixer/videoconvert.c: videomixer: remove unneeded guint comparaison https://bugzilla.gnome.org/show_bug.cgi?id=711010 2013-10-28 14:13:12 +0000 Matthieu Bouron * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: osxvideosink: fix missing selector name warnings The spaces matter in ObjC https://bugzilla.gnome.org/show_bug.cgi?id=711013 2013-10-28 13:31:34 +0000 Matthieu Bouron * gst/y4m/gsty4mencode.c: y4menc: fix uninitialized variable warning https://bugzilla.gnome.org/show_bug.cgi?id=711011 2013-10-25 11:30:36 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: check if the end_time is defined before using it Avoids sending EOS too soon because of overflow. Can happen on fragmented mp4 playback. 2013-10-23 13:38:20 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: use correct unref function Events aren't GstObjects, but GstMiniObjects 2013-10-15 08:16:20 +0200 Stefan Sauer * gst/isomp4/qtdemux.c: qtdemux: rename chunks_are_chunks to chunks_are_samples and flip the logic As the variable name suggests, sometimes chunks are chunks. Rename the variable to tell what they are when they are not chunks. 2013-10-09 08:04:20 +0200 Stefan Sauer * gst/isomp4/qtdemux.c: qtdemux: fix typos and add more logging for unhandled parts 2013-10-14 16:23:25 +0200 Ognyan Tonchev * gst/udp/gstmultiudpsink.c: multiudpsink: Fix memory leak Unmap all GstMemory of the current buffer when flushing. https://bugzilla.gnome.org/show_bug.cgi?id=710110 2013-10-12 20:44:31 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: fix broken sample pipeline which was muxing raw audio and video into flvmux, which won't work, even if there were converters. 2013-10-12 20:37:41 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field raw implies that it's framed already. Fixes .. ! faac ! flvmux 2013-10-07 14:27:21 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: do not emit EOS when connection drops If the pipeline is stalled for too long, souphttpsrc will block and stop fetching data from the network. This can cause the connection to drop and souphttpsrc would handle it as an EOS. This patch makes it persist and try to fetch more data until the end of the content length or until receiving an error that it is beyong limits in case the content is unknown. https://bugzilla.gnome.org/show_bug.cgi?id=683536 2013-10-10 13:52:35 +0200 Sebastian Dröge * ext/dv/gstdvdec.c: * ext/dv/gstdvdec.h: dvdec: Don't send segment event before caps https://bugzilla.gnome.org/show_bug.cgi?id=709728 2013-10-09 17:46:33 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Send stream-start, caps and segment events in the right order https://bugzilla.gnome.org/show_bug.cgi?id=709728 2013-10-08 11:28:04 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields https://bugzilla.gnome.org/show_bug.cgi?id=709614 2013-10-07 12:54:11 +0200 Sebastian Dröge * gst/deinterlace/tvtime/greedyh.c: deinterlace: Fix handling of planar video formats in greedyh method https://bugzilla.gnome.org/show_bug.cgi?id=709507 2013-10-06 10:01:26 -0700 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-mux.c: matroska: Trivial grammar fix on debug msg 2013-10-06 09:17:00 -0700 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/webm-mux.c: matroskamux: Add context flag for WebM WebM has a couple of specific requirements we need to handle. Idea is to set this flag once and just rely on mux->is_webm at run time instead of repeatedly figuring this out from GST_MATROSKA_DOCTYPE_WEBM (which requires a strcmp()). 2013-10-04 14:42:59 -0700 Reynaldo H. Verdejo Pinochet * gst/matroska/matroska-mux.c: matroska: Do not write SegmentUID for WebM mux WebM spec states SegmentUID is Unsupported. Files produced with gstreamer without this change will spit an error like this when passed to mkvalidator: ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192 2013-10-05 00:00:03 +0200 Matej Knopp * gst/matroska/matroska-demux.c: matroskademux: make dvd palette change event sticky So they don't get lost. https://bugzilla.gnome.org/show_bug.cgi?id=709454 2013-10-03 16:39:26 -0400 Nicolas Dufresne * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: videoflip: Add automatic flip mode driven by image-orientation tag https://bugzilla.gnome.org/show_bug.cgi?id=709312 2013-10-04 13:34:09 +0200 Peter Korsgaard * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: O_CLOEXEC needs _GNU_SOURCE On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only defined when _GNU_SOURCE is specified, so do so. _GNU_SOURCE needs to be defined before any system headers are included, so move the fcntl.h section up. https://bugzilla.gnome.org/show_bug.cgi?id=709423 2013-10-04 12:11:56 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix race in flush-start/flush-stop When flush-stop arrives before we process the result of the _push() in the loop function, we might pause even though we are not flushing anymore. Fix this race by waiting for the srcpad loop function to completely pause after doing the flush-start. 2013-10-03 22:38:43 +0200 Mathieu Duponchelle * gst/videomixer/videoconvert.c: videomixer: Update videoconvert copy https://bugzilla.gnome.org/show_bug.cgi?id=709390 2013-10-03 21:36:34 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: Check if the pad needs reconfiguration in collected https://bugzilla.gnome.org/show_bug.cgi?id=709384 2013-10-03 14:39:35 +0100 Matthieu Bouron * ext/jpeg/gstjpegdec.c: jpegdec: Relax sink caps Since jpegdec already parse the jpeg stream, the sink caps could be relaxed. This will allow jpegdec to be selected in more case and in particular when the jpeg typefinder does not find the width and height. https://bugzilla.gnome.org/show_bug.cgi?id=709352 2013-10-03 18:33:01 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2src: print probed caps as caps again in debug log This got lost during refactoring. 2013-10-03 11:59:25 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Add support for the mp2v fourcc for MPEG-2 video https://bugzilla.gnome.org/show_bug.cgi?id=709270 2013-10-02 15:56:53 +0200 Ognyan Tonchev * gst/matroska/matroska-demux.c: matroskademux: Fix memory leak https://bugzilla.gnome.org/show_bug.cgi?id=709266 2013-09-30 12:31:42 +0300 Sreerenj Balachandran * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_types.c: qtdemux: Add HEVC support https://bugzilla.gnome.org/show_bug.cgi?id=709093 2013-09-30 12:24:32 +0200 Ognyan Tonchev * gst/rtp/gstrtpgstpay.c: rtpgstpay: Fix memory leak We were leaking the GList nodes of the pending buffers. https://bugzilla.gnome.org/show_bug.cgi?id=709079 2013-09-30 12:31:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: fix race when updating the next_seqnum If we were not waiting for the missing seqnum when we insert the lost packet event in the jitterbuffer, we end up not updating the next_seqnum and wait forever for the lost packets to arrive. Instead, keep track of the amount of packets contained by the jitterbuffer item and update the next expected seqnum only after pushing the buffer/event. This makes sure we correctly handle GAPS in the sequence numbers. 2013-09-30 12:30:23 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: small debug improvement 2013-09-30 11:53:08 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: reset skew does not reset clock-rate Don't reset the clock-rate when we reset the skew correction algorithm. Reset the skew correction algorithm when we change the clock-rate. 2013-09-30 11:16:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: pause timer when PAUSED Also pause the timer when we go to the PAUSED state. It is possible that we don't have a clock or base-time in PAUSED to perform the timeouts. 2013-09-30 11:15:25 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: improve debug 2013-09-26 20:41:26 +0200 Hans Månsson * gst/isomp4/gstqtmuxmap.c: mp4mux: Do not require framerate in peer video caps Remove the framerate restriction on the caps. Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864 2013-09-27 15:05:04 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: also go into the loop function after connect When we have opened the stream, go into the loop function so that we can receive messages from the server. 2013-09-27 12:53:06 +0200 Matej Knopp * gst/matroska/matroska-demux.c: matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8 https://bugzilla.gnome.org/show_bug.cgi?id=707933 2013-09-26 16:20:04 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: don't calculate skew without rtptime Skip trying to calculate the skew when we don't have an rtptime. It causes problems when lost packet events are placed in the jitterbuffer. 2013-09-25 23:46:14 +0100 Tim-Philipp Müller * configure.ac: configure: get rid of AS_SCRUB_INCLUDE Should not be needed any more. https://bugzilla.gnome.org/show_bug.cgi?id=707658 2013-09-25 17:42:02 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: disable checks when linking pads We know the pad links will work (and we don't check the return value anyway). 2013-09-25 17:36:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: avoid some pad link checks Link pads without checks, we know it will work. 2013-09-25 12:55:21 +0200 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Don't error out if downstream is not seekable for non-fragmented variants Doing so would be a regression over 1.0 and breaks the unit test. However the result will be most likely unusable, so let's post a warning message on the bus. 2013-09-24 04:02:09 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: calculate some stats 2013-09-23 17:05:44 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: move send_lost_event function Move the send_lost_event function to the do_lost_event handling, there is no need to have a separate function. 2013-09-16 11:20:51 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: add code to parse creation time earlier than 1970 Use g_date_time seconds manipulation to allow to cover the quicktime spec for creation_time. It uses seconds since 1904. Both paths could be done using the generic approach of seconds since 1904 with GDateTime handling, but the first path using seconds from 1970 should be more commonly found and avoids a few objects creation and ref/unref, so keep it there for performance. Additionally, the code for handling seconds since 1970 changed from > to >= because having 0 seconds since 1970 is also a valid case for that path to handle. https://bugzilla.gnome.org/show_bug.cgi?id=707975 2013-09-21 00:55:26 +0200 Matej Knopp * gst/matroska/matroska-demux.c: matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily https://bugzilla.gnome.org/show_bug.cgi?id=708505 2013-09-24 18:30:04 +0100 Tim-Philipp Müller * README: * common: Automatic update of common submodule From 6b03ba7 to 865aa20 2013-09-24 15:05:24 +0200 Sebastian Dröge * configure.ac: configure: Actually use 1.3.0.1 as version to make configure happy 2013-09-24 15:00:24 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.2.0 === 2013-09-24 14:21:08 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.2.0 2013-09-24 14:20:51 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2013-09-20 19:43:21 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.m: osxvideosink: fix segfault releasing the sink show_frame is deferred to the main thread and can be called when the sink has been released, so we need to keep an extra ref on ObjectiveC object helper. https://bugzilla.gnome.org/show_bug.cgi?id=708501 2013-09-19 17:11:34 -0400 Robert Krakora * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Restore original GstMemory in buffer if it has been changed https://bugzilla.gnome.org/show_bug.cgi?id=706083 2013-09-23 16:34:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: rtpmanager: update docs 2013-09-23 15:36:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: docs: update docs with 1.0 element names 2013-09-23 14:13:30 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: add test for retransmission because of reordering 2013-09-23 14:12:03 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: always store lost event in jitterbuffer Always prepare a lost event in the jitterbuffer, it is to wake up and make the pushing thread continue. We drop the event when we are not supposed to push lost events downstream. 2013-09-23 11:18:46 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: schedule lost event differently Schedule the lost event by placing it inside the jitterbuffer with the seqnum that was lost so that the pushing thread can interleave and push it properly. 2013-09-23 11:17:34 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: remove timeouts from check Timeouts make the test unreliable and are not needed. 2013-09-23 11:15:30 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: remove list debug 2013-09-23 11:14:01 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: add type to the item So that the upper layer can know what data is contained in the item. 2013-09-23 09:58:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: fix flush Pass function to flush to properly free the queue items. 2013-09-21 00:08:20 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: append seqnum -1 packets 2013-09-20 23:48:20 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpjitterbuffer: use structure to hold packet information Make the jitterbuffer operate on a structure containing all the packet information. This avoids mapping the buffer multiple times just to get the RTP information. It will also make it possible to store other miniobjects such as events later. 2013-09-20 17:48:52 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: update expected timer when possible When we receive a packet and we have some missing packets, we can update their estimated arrival times based on the timestamp difference. 2013-09-20 17:18:27 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix order of timeout events Improve the order of the timeout events, if there are timers with the same timeout, we want to trigger the lowest seqnum first. For this we need to loop over the complete array of timers to find the best one before triggering the timeout. 2013-09-20 16:58:38 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: send lost event before signaling next buffer First send the lost event, then update the next_seqnum counter and then send the signal to the pushing thread that it can retry to push a buffer. This avoids pushing out buffers before the lost event is pushed. 2013-09-20 15:35:25 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: configure clock-rate on jitterbuffer Add a get and setter to configure the clock-rate in the jitterbuffer instead of passing it as an argument to the insert method. 2013-09-20 12:29:39 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: add test for packet delay and retransmission 2013-09-20 12:27:26 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: add option to reset retransmission timers 2013-09-20 12:25:43 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: stop the timer thread The timeout code could release the lock so we need to check if we are allowed to wait for the clock some more. 2013-09-20 12:25:12 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: unlock only once 2013-09-20 11:30:04 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: check both PTS and DTS 2013-09-20 10:55:03 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: add unit-test for multiple missing packets Check if multiple missing packets generate retransmission events and that the retranmission requests are canceled when the missing packet arrives. 2013-09-20 10:53:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: improve flush and shutdown There is no need to unschedule the timer in flush-start, flush-stop will remove the timers and unschedule. Unschedule the current timer before attempting to join the timer thread. 2013-09-20 10:43:53 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: improve debug 2013-09-20 10:42:27 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: set correct expected time When we already have a timer for a packet, skip it but don't forget to adjust the dts to the expected dts of the next packet. 2013-09-20 10:41:59 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve debug 2013-09-19 16:55:25 +0200 Wim Taymans * gst/alpha/gstalpha.c: alpha: use POFFSET instead of OFFSET Use the more correct POFFSET macro to get the offset of a component in its plane. The offset macro gives the offset of the component relative to the start of the frame. 2013-09-21 18:46:29 +0200 Sebastian Dröge * gst/goom/mmx.h: goom: Fix MMX assembly compilation with clang clang does not want or need a clobber list for emms: error: clobbers must be last on the x87 stack Patch taken from the FreeBSD ports, provided by Dan McGregor 2013-09-20 16:16:57 +0200 Edward Hervey * common: Automatic update of common submodule From b613661 to 6b03ba7 2013-09-20 10:19:22 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroska-demux: Make sure that subtitle buffers are \0-terminated https://bugzilla.gnome.org/show_bug.cgi?id=707933 2013-09-17 12:17:54 +0200 Andoni Morales Alastruey * gst/isomp4/gstqtmux.c: qtmux: handle issues correctly when downstream is not seekable The streamable property only make sense for fragmented formats. For regular MP4, when downstream is not seekable we can't rewrite the headers, so qtmux can only work with fast-start=TRUE, where the headers are written finishing the file. For fragmented MP4, when streamable is not seekable and the streamable property is FALSE, we must enforce streamable=TRUE warning the user about this change https://bugzilla.gnome.org/show_bug.cgi?id=707242 2013-09-17 12:06:06 +0200 Andoni Morales Alastruey * gst/isomp4/gstqtmux.c: qtmux: make "streamable" TRUE as default The most common use case for fragmented MP4 (Dash and Smooth Streaming) is producing streamable content (even for VOD). streamable=FALSE would only be used to generate fragmented MP4 with and index of MOOF's that could be reproduced without a playlist/manifest https://bugzilla.gnome.org/show_bug.cgi?id=707242 2013-09-17 12:01:30 +0200 Andoni Morales Alastruey * gst/isomp4/gstqtmux.c: qtmux: deprecate the streamable property for non-fragmented MP4 The streamable property only makes sense for fragmented MP4. https://bugzilla.gnome.org/show_bug.cgi?id=707242 2013-09-19 17:08:19 -0400 Olivier Crête * sys/v4l2/gstv4l2bufferpool.h: v4l2: Remove commented out line 2013-09-19 18:43:08 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 74a6857 to b613661 2013-09-19 17:35:27 +0100 Tim-Philipp Müller * autogen.sh: * common: Automatic update of common submodule From 098c0d7 to 74a6857 2013-09-19 16:50:44 +0200 Wim Taymans * gst/alpha/gstalpha.c: alpha: don't assume planar formats have just 1 block Don't assume planar formats have just one memory block with the data but use the macros to access the right memory block where a component can be found. 2013-09-19 14:14:52 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: add retransmission jitterbuffer test Store both DTS and PTS on buffers. Make a queue for srcpad events. Activate pads after linking so that we don't get RECONFIGURE events. Add test for retransmission. 2013-09-19 14:12:18 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: keep delay as a separate variable in timer Keep a separate delay in the timer so that we still know the original timestamp of the packet that this timer refers to. We can then place the correct running-time in the Retransmission event. 2013-09-19 14:08:56 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix writability of properties 2013-09-19 11:34:57 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.1.90 === 2013-09-19 10:50:23 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.1.90 2013-09-19 10:21:42 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2013-09-19 09:45:18 +0200 Sebastian Dröge * po/cs.po: * po/nl.po: * po/pl.po: * po/uk.po: * po/vi.po: po: Update translations 2013-09-11 14:27:02 -0400 Olivier Crête * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: dmabuf is not a singleton anymore https://bugzilla.gnome.org/show_bug.cgi?id=707793 2013-09-16 13:53:45 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: souphttpsrc: do not do http requests in READY HEAD requests to discover if the server is seekable shouldn't be done in READY as it might lock the main thread that is doing the state change. https://bugzilla.gnome.org/show_bug.cgi?id=705371 2013-09-18 16:32:28 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: reevaluate the current timer after timeout When we trigger the timeout logic of a timer, reevaluate it because it is possible that it still has the lowest timeout. 2013-09-18 16:31:26 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: don't update time when unscheduled Don't try to estimate the current time when we got unscheduled. 2013-09-18 16:29:37 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: init packet spacing on first buffer Already init the packet spacing variables on the first buffer so that we can calculate the spacing on the second buffer already. 2013-09-18 15:08:45 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: fix comments 2013-09-18 14:57:09 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: push the lost event from the timer thread Instead of pushing the lost event from the chain function, schedule a timeout that will push the lost event from the timer thread. This avoid blocking the upstream thread while we push and sync the event. 2013-09-18 14:23:55 +0200 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: add another test The test is modified slightly because the late lost packets are only generated now when a large gap is received. 2013-09-18 14:12:47 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: round gap duration to multiple of duration Make sure the gap duration in the lost event is a multiple of the packet duration. Enable another test. 2013-09-18 12:29:38 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/Makefile.am: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: keep track of duration Keep track of the estimated duration of missing packets and use it in the lost event. Enable another unit test 2013-09-18 11:59:28 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer: handle large gaps with one lost event When we have a large number of missing packets, generate one lost event for all the packets that have no chance of being pushed out in time. Fix and activate unit test for large gaps. 2013-09-18 11:56:38 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: refactor lost event sending Also make sure we only increment the expected seqnum and last output timestamp. 2013-09-17 23:21:09 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor timeout triggers 2013-09-17 23:03:45 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: simplify the timeout code Keep track of the current time in the timeout loop. Loop over all timers and trigger all the expired ones, we can do this in the same loop that selects the new best timer. 2013-09-17 23:01:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: rearrange timer update code Also update the timers when retransmission is disabled. We need to do this because when we added LOST timers when we detected missing packets and we need to remove those timers when the packet finally arrives. 2013-09-17 22:02:04 +0100 Tim-Philipp Müller * gst/videomixer/Makefile.am: videomixer: link to libm for maths stuff Fixes undefined references to rint and pow on ubuntu build bot. 2013-09-17 15:19:42 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: release lock on shutdown 2013-09-17 15:11:41 +0200 Wim Taymans * tests/check/Makefile.am: check: change for videomixer renamed orc file 2013-09-14 16:03:20 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: remove MAX_TOLERATED_LATENESS https://bugzilla.gnome.org/show_bug.cgi?id=707411 2013-09-16 15:54:37 +0200 Wim Taymans * tests/examples/rtp/client-H264-rtx.sh: examples: we don't need the queue anymore 2013-09-16 15:53:47 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: use separate thread for timeouts Use a separate thread for scheduling the timeouts instead of using the downstream streaming thread that might block at any time. 2013-09-14 15:56:04 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: set first_ts to DTS for streams that have DTS https://bugzilla.gnome.org/show_bug.cgi?id=707340 2013-09-14 15:55:22 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: make sure duration is a valid number for last buffer https://bugzilla.gnome.org/show_bug.cgi?id=707340 2013-09-14 15:54:29 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: use segment.start or last buffer end time in case of missing DTS https://bugzilla.gnome.org/show_bug.cgi?id=707340 2013-09-03 18:14:04 +0200 Matej Knopp * gst/isomp4/gstqtmux.c: Revert qtmux: Use buffer PTS if DTS is not set" This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d. https://bugzilla.gnome.org/show_bug.cgi?id=707340 2013-09-16 11:03:06 +0200 Sebastian Dröge * gst/videomixer/videomixerorc-dist.c: * gst/videomixer/videomixerorc-dist.h: videomixer: Update orc generated files https://bugzilla.gnome.org/show_bug.cgi?id=708131 2013-09-13 16:25:49 +0200 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Demux RTCP buffers from the RTP stream If there are RTCP buffers in the RTP stream, process them as RTCP. This way, we want receive streams following RFC 5761 https://bugzilla.gnome.org/show_bug.cgi?id=687657 2013-09-13 23:26:21 +1000 Jan Schmidt * gst/rtp/gstrtpL24depay.c: rtp: Remove bogus extra caps from L24 template. The extra caps entry in the template was making it sometimes get plugged for any dynamically allocated payload type. 2013-09-13 12:40:41 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: rtpbin: use PacketInfo for the sender Avoid mapping the packet multiple times when sending RTP. 2013-09-13 12:22:36 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: rtpbin: store more in the PacketInfo Store all info in the PacketInfo so that we can avoid mapping the packet multiple times. 2013-09-13 11:32:52 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.h: session: store more in the PacketInfo structure 2013-09-13 11:08:55 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: rtpbin: RTPArrivalStats -> RTPPacketInfo Rename a structure because we are also going to use this for the sender bits. 2013-09-13 10:55:31 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: source: small cleanups 2013-09-12 13:31:01 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: only update stop position if seek requests it Check for GST_SEEK_TYPE_NONE for stop poistion and only update the stop time if it is requested. Otherwise just maintain whatever was stored at the segment https://bugzilla.gnome.org/show_bug.cgi?id=707530 2013-09-13 08:53:25 +0200 Rico Tzschichholz * gst/rtp/Makefile.am: rtp: Add missing headers tp fix make dist In addition to a956a6ceb2deb87cc1361aee1d6626449f46dab2 2013-09-12 15:07:48 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Make sure we have enough data to read image tags Thanks to iputinei for reporting this on IRC. 2013-09-12 15:01:36 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: handle segments with non-0 start We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to transform it back to a buffer timestamp before pushing out the buffer. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931 2013-09-11 13:11:58 -0600 Seán de Búrca * gst/matroska/matroska-demux.c: matroskademux: Fix off-by-one in validation of UTF-8 https://bugzilla.gnome.org/show_bug.cgi?id=707933 2013-09-11 14:32:17 -0300 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Do not check if caps are empty when they are NULL In the case the caps are actually NULL, we should just concider it the same way as empty caps in that case. 2013-09-10 16:44:53 -0600 Seán de Búrca * gst/videomixer/videomixerorc-dist.c: * gst/videomixer/videomixerorc-dist.h: videomixer: fix build if orc is not installed https://bugzilla.gnome.org/show_bug.cgi?id=707886 2013-09-10 17:57:49 -0300 Thiago Santos * gst/matroska/matroska-demux.c: matroskademux: Preserve seqnum when pushing seek upstream After converting a seek from time to bytes, use the same seqnum on the event that goes upstream 2013-09-05 00:17:16 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: track streams that are EOS on push mode to finish earlier When the segment has a defined stop position, qtdemux should check when streams reach this position and mark those as EOS. When all streams are EOS it will return GST_FLOW_EOS to upstream to allow the pipeline to finish instead of continuously consume buffers from upstream that are not useful for the segment. https://bugzilla.gnome.org/show_bug.cgi?id=707530 2013-09-04 15:34:35 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: preserve stop of segment when doing seeks in push mode When handling seeks in push mode, qtdemux converts the seek to bytes and pushes upstream. It needs to keep track of the seek and the subsequent segment to be able to map them back to the requested seek time and properly preserve the segment stop of the seek. This is done by using the start offset in bytes of the seek, that should be the same of the segment from upstream. And this is also backwards compatible with what qtdemux already was using. https://bugzilla.gnome.org/show_bug.cgi?id=707530 2013-07-26 19:40:53 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2pad.h: videomixer: Add colorspace conversion https://bugzilla.gnome.org/show_bug.cgi?id=704950 2013-08-06 15:38:39 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: Don't send reconfigure event when formats or PAR are different It is racy with multiple pads. https://bugzilla.gnome.org/show_bug.cgi?id=704950 2013-07-25 13:49:57 +0200 Mathieu Duponchelle * gst/videomixer/Makefile.am: * gst/videomixer/blend.c: * gst/videomixer/blendorc.orc: * gst/videomixer/gstcms.c: * gst/videomixer/gstcms.h: * gst/videomixer/videoconvert.c: * gst/videomixer/videoconvert.h: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixerorc.orc: videomixer: Bundle private copies of videoconvert code Ideally, this would be part of libgstvideo. Prefixes videoconvert symbols with videomixer_. https://bugzilla.gnome.org/show_bug.cgi?id=704950 2013-08-22 00:03:48 +0200 Mathieu Duponchelle * sys/v4l2/gstv4l2bufferpool.c: v4l2: Use newly #defined metadata names. 2013-09-09 15:11:51 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: only wait if we flushed Only wait for the STREAM_LOCK when we flushed something when sending a command for PAUSED or PLAYING. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611 2013-09-09 15:09:41 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: return when a flush was issued Make gst_rtspsrc_loop_send_cmd() return TRUE when the current action has been flushed 2013-09-09 11:16:40 +0200 David Holroyd * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL24depay.c: * gst/rtp/gstrtpL24depay.h: * gst/rtp/gstrtpL24pay.c: * gst/rtp/gstrtpL24pay.h: * tests/check/elements/rtp-payloading.c: rtp: add L24 pay and depayloader Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734 2013-09-09 14:46:42 +0200 Sebastian Dröge * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Fix missing condition in previous commit 2013-09-09 14:44:58 +0200 Sebastian Dröge * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Also fix strides for other semi-planar video formats 2013-09-09 14:41:42 +0200 Andreea Fulger * sys/v4l2/gstv4l2bufferpool.c: v4l2bufferpool: Fix stride for NV12/NV21 https://bugzilla.gnome.org/show_bug.cgi?id=707758 2013-09-07 16:37:03 +0200 Matej Knopp * gst/matroska/matroska-read-common.c: matroskademux: fix leaking buffer and caps https://bugzilla.gnome.org/show_bug.cgi?id=707688 2013-09-05 19:46:37 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: fix build on win32 gstudpsrc.c:855:15: error: #if with no expression 2013-09-04 15:50:42 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: handle unseekable streams Handle streams that we can't seek in and ignore them in the seek logic. 2013-09-04 15:25:39 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: only check video compression for video streams Or else we might deref a stream with a NULL strf.vids and segfault 2013-06-18 13:27:20 +0100 Alex Ashley * gst/isomp4/atoms.c: * gst/isomp4/fourcc.h: * gst/isomp4/ftypcc.h: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_types.c: qtdemux: Add support for the avc3 sample entry format of the AVC file format Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new structure for fragmented MP4 called "avc3". The principal difference between AVC1 and AVC3 is the location of the codec initialisation data (e.g. SPS, PPS). In AVC1 this data is placed in the initial MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data goes in the first sample of every fragment (i.e. the first sample in each mdat box). The principal reason for avc3 is to make it easier for client implementations, because it removes the requirement to insert the SPS+PPS in to the decoder pipeline every time there is a representation change. This commit adds support for the "avc3" atom, which is almost identical to the "avc1" atom, except it does not contain any SPS or PPS data. https://bugzilla.gnome.org/show_bug.cgi?id=702004 2013-09-04 00:27:50 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: Don't set EOS to FALSE when the collectpad *is* EOS https://bugzilla.gnome.org/show_bug.cgi?id=707238 2013-09-03 17:32:41 +0200 Matej Knopp * gst/audioparsers/gstflacparse.c: flacparse: cleanup on error after state change https://bugzilla.gnome.org/show_bug.cgi?id=707229 2013-09-03 11:23:24 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: Bind to multicast addresses on non-Windows systems On Windows it's not possible to bind to a multicast address but the OS will make sure to filter out all packets that arrive not for the multicast address the socket joined. On Linux and others it is necessary to bind to a multicast address to let the OS filter out all packets that are received on the same port but for different addresses than the multicast address And deprecate the multicast-group property and replace it with the address property. https://bugzilla.gnome.org/show_bug.cgi?id=707042 2013-09-03 10:10:01 +0200 Matej Knopp * gst/audioparsers/gstflacparse.c: flacparse: Free GstBaseParseFrame if pushing a header failed 2013-09-02 16:02:37 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Refactor address resolval into its own function 2013-09-02 23:00:29 +0100 Tim-Philipp Müller * gst/replaygain/gstrganalysis.c: replaygain: fix taglist leak in rganalysis And add some FIXMEs. 2013-09-02 22:50:58 +0100 Tim-Philipp Müller * tests/check/elements/rganalysis.c: tests: rganalysis: rename function for clarity 2013-03-18 14:32:07 +0100 Christoph Reiter * tests/check/elements/rganalysis.c: tests: fix skipped rganalysis tests In 0.10 elements would post tag messages on the bus directly, and rganalysis would only post a tag message when it changed tags. In 1.0, only sinks post tag messages when they receive the serialised tag event. This means that we get an additional tag message on the bus now where we didn't expect one before. https://bugzilla.gnome.org/show_bug.cgi?id=695090 2013-09-02 11:46:52 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Properly propagate downstream flow returns upstream https://bugzilla.gnome.org/show_bug.cgi?id=707229 2013-09-01 21:18:38 +0100 Tim-Philipp Müller * ext/shout2/gstshout2.c: * gst/avi/gstavi.c: * gst/isomp4/isomp4-plugin.c: * gst/rtsp/gstrtsp.c: * sys/sunaudio/gstsunaudio.c: * sys/v4l2/gstv4l2.c: Don't use setlocale in plugins() Only apps should call setlocale(), not libraries. 2013-08-29 13:15:15 +0200 Wim Taymans * gst/rtp/gstrtpmpvpay.c: rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay RTP buffer allocation should not be done with padding for the specific MPEG2 header as the padding is done at the end of the buffer and the last byte is the size of the padding. https://bugzilla.gnome.org/show_bug.cgi?id=706970 2013-08-28 10:51:32 +0200 Bernhard Miller * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosink.h: autovideosink: add sync property https://bugzilla.gnome.org/show_bug.cgi?id=706955 2013-08-28 07:15:00 +0200 Bernhard Miller * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosink.h: autoaudiosink: introduce sync property https://bugzilla.gnome.org/show_bug.cgi?id=706955 2013-08-27 17:33:40 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: push buffers after segment stop until reaching a keyframe This should make decoders able to precisely push buffers until the stop time in case they need the next keyframe to do it. Also, according to gst_segment_clip, it should only push a buffer that the starting ts is strictly smaller than the segment stop, so we change the min < comparison for <= 2013-08-28 13:26:47 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.1.4 === 2013-08-28 12:52:25 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * gst/audiofx/audiopanoramaorc-dist.c: * win32/common/config.h: Release 1.1.4 2013-08-28 12:52:16 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2013-08-28 12:32:10 +0200 Sebastian Dröge * po/pt_BR.po: po: update translations 2013-08-27 15:25:16 +0200 Wim Taymans * gst/matroska/matroska-mux.c: matroska-mux: remove framerate restriction Remove the framerate restriction on the caps. 2013-08-27 09:38:16 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: only update next check time when reconsidering Don't update the next RTCP check time in all cases but only when we reconsidered. This avoids delaying sending a full RTCP packet when we are doing early feedback. 2013-08-27 09:37:33 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add more debug 2013-08-27 09:34:46 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: jitterbuffer: fix types of the retransmission event 2013-08-27 09:33:03 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: only timeout EXPECTED timers on gap Only timeout the EXPECTED timers when we detect a large seqnum gap. 2013-08-26 13:47:53 +0200 Sebastian Dröge * configure.ac: configure.ac: Don't set BZ2_LIBS if bz2 is not found 2013-08-26 11:50:27 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtsession: fix locking We need to take the session lock when getting and manipulating the source. 2013-08-26 11:50:13 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: add some more debug 2013-08-20 22:12:03 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: don't send flush_stop twice. If we get flush start and a seek we need to only send flush_stop once. More info at #706441 2013-08-23 15:56:43 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: propagate discont 2013-08-23 15:49:47 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: multipartdemux: remove dynamic sourcpads when going from PAUSED to READY 2013-08-23 15:29:28 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last https://bugzilla.gnome.org/show_bug.cgi?id=637754 2013-08-23 15:47:25 +0200 Wim Taymans * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtxqueue: add property to configure queue size 2013-08-23 12:07:55 +0200 Wim Taymans * tests/examples/rtp/client-H264-rtx.sh: * tests/examples/rtp/server-VTS-H264-rtx.sh: tests: add retransmission example 2013-08-23 11:55:02 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: proxy jitterbuffer do-retransmission property 2013-08-23 11:17:45 +0200 Michael Olbrich * gst/avi/gstavimux.c: avimux: unmap the correct buffer The audio buffer was mapped so unmap it and not the video buffer https://bugzilla.gnome.org/show_bug.cgi?id=706642 2013-08-18 23:32:22 -0400 Olivier Crête * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Add property to find out the device currently in use https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 23:31:15 -0400 Olivier Crête * ext/pulse/pulsesink.c: pulsesink: De-duplicate code to get the current sink input info https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 22:27:37 -0400 Olivier Crête * ext/pulse/pulsesink.c: pulsesink: Implement changing the device while playing https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 23:32:22 -0400 Olivier Crête * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: Add property to find out the device currently in use https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 23:31:15 -0400 Olivier Crête * ext/pulse/pulsesrc.c: pulsesrc: De-duplicate code to get the current source output info https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 22:27:37 -0400 Olivier Crête * ext/pulse/pulsesrc.c: pulsesrc: Implement changing the device while playing https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-22 14:55:14 +0200 Sebastian Dröge * configure.ac: configure: Fix bz2 configure check for Windows Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2. https://bugzilla.gnome.org/show_bug.cgi?id=465924 2013-02-22 20:57:00 +0900 Akihiro Tsukada * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulsesink: Add support for AAC pass-through https://bugzilla.gnome.org/show_bug.cgi?id=694445 2013-06-24 17:29:37 +0200 Kishore Arepalli * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: crashes if any property changes during playback when location property is not set https://bugzilla.gnome.org/show_bug.cgi?id=702988 2013-08-21 14:54:26 -0400 Olivier Crête * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.h: pulse: Share static caps definition between src and sink The src was also missing 24-bit sample formats 2013-08-21 16:53:59 +0200 Wim Taymans * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtx: various improvements Use locking Don't push from the event handler, collected packets in a queue and push from the chain function. Clear queues on shutdown. 2013-08-21 16:50:59 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: session: generate events correctly Do correct shifting of the bitmask for lost packets. 2013-08-21 16:47:40 +0200 Wim Taymans * gst/rtpmanager/gstrtpmanager.c: rtp: register rtx element better 2013-08-21 16:32:50 +0200 Sebastian Dröge * sys/directsound/gstdirectsoundsink.c: directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others Probably fixes https://bugzilla.gnome.org/show_bug.cgi?id=705477 2013-08-21 13:03:34 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegenc.c: jpegenc: don't ignore return value from _finish_frame() gst_video_encoder_finish_frame() will return FLOW_OK here if there's no output buffer. 2013-08-21 12:56:35 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: jpegdepay: add some more debug 2013-08-21 12:10:00 +0200 Wim Taymans * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: rtpgstdepay: only push events when they changed Keep track of the STREAM_START and TAG events and only push them when they changed. 2013-08-21 10:52:59 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: taglists should not be merged in 1.0 2013-08-21 10:28:50 +0200 Wim Taymans * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: flush on FLUSH_STOP event 2013-08-21 10:03:52 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: reset on state change Do full reset on state change to READY 2013-08-21 09:55:20 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: reset on FLUSH_STOP Clear the adapter and pending buffer list on FLUSH_STOP. 2013-08-21 09:39:30 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: don't use clock for config interval We can't use the clock to time our config-interval because we are not live (or there might not be a clock or the clock might not be running). Instead just simply take the timestamp diff. 2013-08-21 09:33:04 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.h: rtpgstay: don't use // comments 2013-08-08 11:55:22 -0400 Youness Alaoui * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix response argument in handle-request signal 2013-08-08 11:54:41 -0400 Youness Alaoui * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add sdes property and proxy it to rtpbin 2013-08-07 09:47:35 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs 2013-07-25 21:12:05 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags. 2013-07-25 21:10:10 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time 2013-07-25 21:03:34 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps 2013-07-25 20:54:50 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush. 2013-07-25 17:56:38 -0400 Youness Alaoui * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START 2013-07-25 17:52:16 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3 2013-08-20 14:36:59 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: handle EOS When the queue is empty, and we received EOS, pause and push an EOS event downstream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387 2013-08-20 10:26:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update docs 2013-08-20 10:25:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update all timers Keep looping over all registered timers so that we can mark them lost instead of stopping as soon as we find the timer for the current seqnum. 2013-08-20 08:55:50 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: remove unused variables 2013-08-19 21:10:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reorganize timer handling Restructure handling of incomming packet and the gap with the expected seqnum and register all timers from the _chain function. Convert a timer to a LOST packet timer when the max amount of retransmission requests has been reached. 2013-08-19 21:37:44 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor packet spacing calculation 2013-08-19 21:34:38 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: keep track of last seqnum and dts 2013-08-19 21:29:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: small cleanups 2013-08-19 21:21:08 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reset retransmission timers in add/reschedule Reset the retransmission timers when adding and rescheduling a timer. 2013-08-19 21:12:13 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: rename variables for packet spacing 2013-08-19 14:58:01 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: remove lost timer when we get the packet When we receive a packet, also remove the LOST timer for it. 2013-08-19 14:56:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: expected seqnum must increase Only update the expected seqnum when it is bigger than the previous expected seqnum. 2013-08-19 14:55:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add more debug 2013-08-12 16:15:54 +0200 Wim Taymans * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtxqueue: add retransmission queue element 2013-08-12 14:53:33 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add some docs 2013-08-06 16:29:54 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: handle NACK feedback and generate events Handle and parse the feedback NACK packets and generate a Retransmission event for each NACKed packet 2013-08-19 13:19:42 -0400 Olivier Crête * sys/v4l2/gstv4l2object.c: v4l2: Add forward declaration for gst_v4l2_object_get_format_list 2012-10-22 17:58:07 -0400 Olivier Crête * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: De-duplicate caps probing between src and sink 2013-08-13 17:32:17 -0400 Olivier Crête * ext/pulse/Makefile.am: * ext/pulse/pulseprobe.c: * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulse: Remove unused GstPulseProbe 2013-08-19 12:46:45 -0400 Olivier Crête * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/tuner.c: * sys/v4l2/tunerchannel.c: * sys/v4l2/tunernorm.c: v4l2: Use G_DEFINE_ macros for added thread safety 2013-08-17 11:28:13 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: Do not send flush_stop ourself after a flush_start When we receive a flush_start, we should wait for the next flush_stop and foward it, not create a flush_stop ourself. 2013-08-16 17:10:31 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: h264depay: init debug category early Init the debug variable when we register the element because it is also used by the payloader element when it calls the add_sps_pps method. 2013-08-16 13:26:28 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Properly set headers via the base class instead of just pushing them downstream Prevents buffers from being send before the caps and segment events. 2013-08-15 10:59:10 +0100 Chris Bass * gst/isomp4/qtdemux.c: qtdemux: check denominator isn't zero before scaling duration. When gst_qtdemux_configure_stream sets fps_d, check that n_samples is non-zero before using it as a denominator to scale the stream duration. https://bugzilla.gnome.org/show_bug.cgi?id=706076 2013-08-15 15:08:05 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngdec.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: ext: Use new flush vfunc of video codec base classes and remove reset implementations 2013-08-14 16:19:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: forward flush before stopping dataflow First forward the flush event and then stop our loop function. 2013-08-14 13:10:32 +0100 Tim-Philipp Müller * configure.ac: configure: require libsoup >= 2.38 Bump libsoup requirement for newer API used, like headers_get_one(). 2.38 is from early 2012 and is in linen with our GLib requirement. 2013-08-14 11:54:19 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: soup: don't use deprecated soup_message_headers_get() API 2013-08-13 17:44:50 +0200 Edward Hervey * .gitignore: .gitignore: Ignore files from automake test-driver 2013-08-12 15:28:34 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Use the SPS/PPS handling function from the depayloader Remove duplicated copies https://bugzilla.gnome.org/show_bug.cgi?id=705553 2013-08-12 15:26:08 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: Make the SPS/PPS deduplication function generic Make it not touch any internals of the depayloader https://bugzilla.gnome.org/show_bug.cgi?id=705553 2013-08-13 14:09:20 +0100 Chris Bass * gst/audioparsers/gstaacparse.c: aacparse: allow conversion from raw AAC to ADTS This patch will prepend ADTS headers to raw AAC audio frames, allowing upstream elements to link to decoders that only support AAC in ADTS format. Note that no error correction bits are added to ADTS frames in this code. https://bugzilla.gnome.org/show_bug.cgi?id=615740 2013-08-13 12:44:11 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Only free GCheckSum after its last usage https://bugzilla.gnome.org/show_bug.cgi?id=705760 2013-08-13 12:02:29 +0200 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix critical setting a NULL uri redirection 2013-07-13 01:50:56 +0200 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: add redirection to the URI query 2013-07-31 10:42:07 +0200 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: elst should offset samples instead of buffers The current approach where buffers are offset is not ideal, as during seek and loop current time is compared to sample times. https://bugzilla.gnome.org/show_bug.cgi?id=700264 2013-08-07 19:32:07 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: * tests/check/elements/videomixer.c: videomixer: Send EOS if buf_end >= segment.stop That means the whole segment is already played, and we are sure we are EOS at that point. Also handle segment seeks, and do not send EOS in that case. 2013-08-04 14:40:38 +0200 Matej Knopp * gst/avi/gstavidemux.c: avidemux: send proper stream_start event https://bugzilla.gnome.org//show_bug.cgi?id=705449 2013-08-08 11:51:17 +0200 Sebastian Dröge * gst/matroska/ebml-read.c: * gst/matroska/matroska-demux.c: matroskademux: Don't print warnings during flushing and stop as soon as possible https://bugzilla.gnome.org//show_bug.cgi?id=705442 2013-08-07 11:14:38 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvp8depay.c: rtpvp8depay: mark key frames and delta frames properly https://bugzilla.gnome.org/show_bug.cgi?id=705550 2013-08-05 23:23:57 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add NACK feedback in RTCP 2013-08-05 23:22:16 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: source: add methods to register NACK Add a method to register a missing packet for an ssrc along with methods to get the missing packets and clear them. 2013-08-04 23:05:36 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: handle Retransmission event and schedule NACK Handle the retransmission event from downstream and use it to schedule a NACK request. 2013-08-05 23:20:29 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: pass data to remove func Pass the data to the remove function because we are going to deref it when there is pli or fir. 2013-08-06 15:28:50 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Fix compilation 2013-08-06 15:17:44 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE 2013-08-06 11:58:38 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Make sure to send EOS if the buffer end time equals the segment end time Otherwize EOS never gets sent in that particular case. 2013-08-05 08:49:50 +0200 Sjoerd Simons * gst/goom/gstgoom.c: goom: Ensure src caps are writable In some cases the src caps determined by goom weren't writable, causing a bunch of assertion failures and failed caps. Fixed by always explicitely making the caps writable https://bugzilla.gnome.org/show_bug.cgi?id=705475 2013-08-04 23:18:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: use common send_rtcp method Reuse the send_rtcp method that already asks for the current time when requesting a keyframe. 2013-08-04 23:12:50 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: Don't use ClockTimeDiff for unsigned delays 2013-08-04 16:52:15 +0200 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Use buffer PTS if DTS is not set Avoids ending up with completely bogus scaled duration/pts when new buffers have invalid DTS. 2013-08-04 14:32:47 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: skip https test if there's no TLS support in soup/glib 2013-08-04 11:20:41 +0100 Tim-Philipp Müller * gst/rtsp/gstrtpdec.c: rtpdec: use generic marshaller 2013-08-04 10:52:33 +0100 Tim-Philipp Müller * Makefile.am: * sys/v4l2/.gitignore: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2-marshal.list: * sys/v4l2/tuner-marshal.list: * sys/v4l2/tuner.c: * sys/v4l2/tuner.h: * win32/MANIFEST: * win32/common/tuner-enumtypes.c: * win32/common/tuner-enumtypes.h: * win32/common/tuner-marshal.c: * win32/common/tuner-marshal.h: v4l2: remove unused enumtypes and use generic marshaller 2013-08-04 10:47:38 +0100 Tim-Philipp Müller * Makefile.am: * gst/udp/.gitignore: * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-enumtypes.h: * win32/common/gstudp-marshal.c: * win32/common/gstudp-marshal.h: udp: remove unused marshal and enumtypes files 2013-08-04 09:38:19 +0100 Tim-Philipp Müller * Makefile.am: * gst/rtpmanager/.gitignore: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpsession.c: * win32/MANIFEST: * win32/common/gstrtpbin-marshal.c: * win32/common/gstrtpbin-marshal.h: rtpmanager: use generic marshaller 2013-08-04 00:13:07 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: send event in right direction 2013-08-02 17:38:34 -0700 David Schleef * configure.ac: * tests/check/Makefile.am: tests: create/remove orc directory at proper time Before automake creates .deps directories, and during distclean. 2013-08-03 00:25:44 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add FIR and PLI like other RTCP packets Add the FIR and PLI packets like the other RTCP packet instead of from the on-sending-rtcp default signal handler. 2013-08-02 17:22:55 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix property ranges 2013-08-02 16:42:52 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: push retransmission events 2013-08-02 14:12:16 +0200 Lubosz Sarnecki * configure.ac: build: add subdir-objects to AM_INIT_AUTOMAKE Fixes warnings with automake 1.14 https://bugzilla.gnome.org/show_bug.cgi?id=705350 2013-08-02 14:54:56 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add support for retransmission retry When we didn't receive a packet after requesting retransmission, retry asking for retransmission for a certain period. 2013-08-02 14:19:54 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add properties Add properties to control retransmission parameters 2013-08-02 12:44:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: use corrected timeout when rescheduling When we recalculate the timeout, use the corrected timeout value depending on the timer type. 2013-08-02 12:43:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update timers after queueing Else we might update the timer needlessly for duplicates. 2013-08-02 12:42:08 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: move method up 2013-08-02 06:28:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: small cleanup 2013-08-01 23:26:06 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: unschedule old expected packets When we receive a new packet, unschedule old outstanding packets when their seqnum is too far away. 2013-08-01 23:29:23 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor timer update 2013-08-01 23:24:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update timers when removing Update the timers when we remove a timer. Handle canceled timers, make them unschedule the current timer and trigger the timeout code. 2013-08-01 23:22:02 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix typo 2013-08-01 15:40:52 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve timeout management If we change the seqnum of an existing timer and we were waiting for that timer, unschedule it. If we change the timeout of an existing timer and we were waiting on it, only unschedule when the new time is smaller. 2013-08-01 15:05:35 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: install timer for expected arrival Install a timer that is triggered when the expected arrival time of a packet expired. 2013-08-01 14:56:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve unschedule of timers Conflicts: gst/rtpmanager/gstrtpjitterbuffer.c 2013-08-01 12:21:53 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: move code around 2013-08-01 12:07:11 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: estimate inter packet spacing When we see two packets with consecutive seqnums and a different RTP time, use the DTS difference as the inter packet spacing estimate. 2013-08-01 12:01:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: keep track of current timeout 2013-08-01 11:49:10 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: cleanup timer handling 2013-08-01 11:40:41 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reset is only possible with a GAP 2013-08-01 11:29:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: operate on DTS Make the jitterbuffer schedule the timeouts based on the DTS instead of the PTS. This makes it all smoother with reordered frames and gives the decoder time to reorder the frames in time. 2013-08-01 11:14:12 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: rename timout variable 2013-07-31 17:08:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: small cleanup 2013-07-31 16:59:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: block output in paused or buffering 2013-07-31 16:59:09 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: store pts in timer Only store the pts in the timer so that we can both do timeouts with timings on the input and output of the jitterbuffer. 2013-07-30 23:14:24 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: refactor jitterbuffer Refactor the jitterbuffer code. Make separate function for peeking a buffer, pushing the next buffer, waiting for timeouts and handling the timeouts. The main loop now tries to push as many buffers as it can until it runs out of buffers or when it detects a seqnum discont. Then it will wait for some event to happen before attempting to push more buffers. Make methods to register timeouts in an array. These timeouts are registered when we detect a missing packet, sync for the first packet or when we find an estimation for the end-of-stream. This greatly simplifies and clarifies the code and also makes it possible to register more complicated timeout schemes later. 2013-07-30 18:52:58 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: use NULL to ignore percent If we pass NULL to pop and push we ignore the percent result. 2013-07-30 07:00:19 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor Move eos estimation into separate function 2013-07-30 14:28:19 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: don't leak stream_id string https://bugzilla.gnome.org/show_bug.cgi?id=705142 2013-07-29 19:53:52 +0100 Tim-Philipp Müller * po/LINGUAS: * po/da.po: * po/de.po: * po/el.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/ja.po: * po/nb.po: * po/nl.po: * po/pl.po: * po/ru.po: * po/sl.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: po: update translations 2013-07-29 19:48:54 +0100 Tim-Philipp Müller * tests/check/elements/.gitignore: tests: ignore new test binaries 2013-07-29 14:47:49 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.1.3 === 2013-07-29 13:42:18 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.1.3 2013-07-29 13:42:05 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2013-07-29 12:12:41 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: gst: Don't swap start/stop for negative rates in the SEGMENT query 2013-07-29 11:18:40 +0200 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: Check for data size when parsing h264 codec data from strf atom 2013-07-29 10:53:54 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Implement SEGMENT query 2013-07-29 10:53:47 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Implement SEGMENT query 2013-07-29 10:50:59 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Implement SEGMENT query 2013-07-27 18:10:22 +0200 Matej Knopp * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: Support H264 fourcc https://bugzilla.gnome.org/show_bug.cgi?id=704996 2013-07-28 18:09:33 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Fix handling of image tags The caps should be used to get the mimetype and there is only an info structure for the GstSample if the image-type is not NONE. 2013-07-28 18:04:32 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Don't crash if there is no image tag information https://bugzilla.gnome.org/show_bug.cgi?id=705018 2013-07-28 17:38:56 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Fix duration reporting in push mode https://bugzilla.gnome.org/show_bug.cgi?id=700933 2013-07-28 17:32:27 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Don't forget unmapping and unreffing buffer 2013-07-26 21:06:17 +0200 Matej Knopp * gst/avi/gstavidemux.c: avidemux: unmap buffer https://bugzilla.gnome.org/show_bug.cgi?id=704951 2013-07-26 22:31:41 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: don't make buffer writable prematurely There is no reason to make the SR buffer writable at this point. This is better delayed until needed. 2013-07-26 22:25:50 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: ignore RTCP for inactive sources 2013-07-26 22:25:17 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: small cleanup 2013-07-26 17:17:31 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: session: handle partial RTCP report blocks When we have more SSRCs to report than what fit in an RTCP packet, use a generation counter to make sure all of them end up in a packet eventually. 2013-07-26 17:23:10 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: create SSRC before doing session cleanup Make the internal source before we do session cleanup 2013-07-26 17:21:08 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: reorganize the report block code 2013-07-26 16:02:01 +0200 Matej Knopp * gst/matroska/matroska-demux.c: matroskademux: fix memory leak in check_subtitle_buffer https://bugzilla.gnome.org/show_bug.cgi?id=704921 2013-07-26 14:21:40 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: refactor active and sender checks 2013-07-26 12:06:35 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: remove internal sources on timeout When an internal source times out and becomes a receiver, remove it. 2013-07-26 11:47:56 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: create an internal source for RTCP When we need to do RTCP and we don't have an internal source yet, make one. 2013-07-26 10:47:28 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: session: remove old code to change SSRC Remove code used to change the SSRC after a collision. We now send a RECONFIGURE event upstream to make the upstream element change the SSRC. 2013-07-26 10:42:44 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: source: don't update packet SSRC Remove the code to update the SSRC in packets, it can never be called now that we always use a source with matching packet SSRC. 2013-07-26 10:24:22 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: delay allocation of internal source Allocate the internal source when we receive a caps with the SSRC or when we see a buffer with the SSRC. 2013-07-26 10:00:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: session: generate reconfigure on collision When we detect a collision, change the SSRC that we suggest upstream and trigger RECONFIGURE. This should make upstream select a new SSRC. 2013-07-26 09:37:24 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: produce RTCP for all internal sources Loop over all the internal sources and produce RTCP. We also need to queue the RTCP packets and send them when we are finished. 2013-07-26 01:40:20 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: deprecate internal source and ssrc properties Deprecate the internal source and internal ssrc properties. There might be more than one internal source. 2013-07-26 01:29:08 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: internal sources don't use probation 2013-07-26 01:24:07 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: session: give caps to session Let the session parse the caps and update its SSRC when needed. 2013-07-26 01:14:04 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: make method to suggest available SSRC Make a method to suggest the best available SSRC. This is the SSRC of the last created internal source and is used to instruct upstream to produce this SSRC. 2013-07-26 01:01:49 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: keep SDES and set on new internal sources Keep track of the SDES ourselves and set it on all newly created internal sources. 2013-07-26 00:48:25 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: make method to make internal sources Add a method to obtain an internal source and use it to create our internal source 2013-07-26 00:29:41 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.h: session: count internal sources and how many are senders 2013-07-26 00:14:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: separate BYE marking and scheduling First mark sources with BYE and then schedule the BYE RTCP message. 2013-07-25 23:56:46 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: get SSRC from RTCP packet itself Get the SSRC from the RTCP packet instead. 2013-07-25 23:51:34 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: fix bandwidth calculation We iterate over all sources and the internal one is also in the hashtable so avoid adding it twice. 2013-07-25 23:38:08 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add some docs 2013-07-25 23:11:05 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: Rearrange RTCP reporting a little Make a function to generate an RTCP packet for a source, pass the source as a parameter. Move timeout of collisions to session cleanup phase. 2013-07-25 22:39:04 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: move check for is_early around Move the check for the early RTCP to where it is needed and used. 2013-07-25 17:35:02 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: parse packet outside of the session lock 2013-07-25 17:34:06 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: do nicer checks for internal sources 2013-07-25 17:15:37 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: session: let source keep track if it sent BYE 2013-07-25 17:06:22 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: source: reset more 2013-07-25 16:49:41 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: source: also use the source for bye_reason Store the BYE reason in our internal source object. Rename the methods on the source object a little because now the BYE can be received in RTCP or set when the session wants to send BYE. 2013-07-25 16:24:04 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: session: configure sdes with structure only Remove code to configure the SDES with methods and types, only allow configuration with GstStructure 2013-07-25 15:56:39 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: refactor add and find source Make functions to find and add a source to the hashtable. 2013-07-25 15:43:11 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: remove source from sync_rtcp We don't need to know the sender source of the session in the callback, the SR packet is for all participants in the session. 2013-07-24 14:18:14 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add some more debug 2013-07-15 17:11:45 +0100 Vincent Penquerc'h * gst/audioparsers/Makefile.am: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: allow conversion from ADTS to raw AAC Some muxers (eg, qtmux) only support raw AAC, so this allows linking an encoder that outputs ADTS only to those muxers. The conversion is simple (omit the first 7 or 9 bytes of the frame), but has to be done in pre_push instead of handle_frame as 1.0 does not seem to allow skipping bytes there as 0.10 used to. Other conversions are not supported (yet). 2013-07-15 17:15:44 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: fix object_type parsing off-by-one in ADTS frame According to http://wiki.multimedia.cx/index.php?title=ADTS, the value stored in ADTS headers is one less than the object type of the AAC stream. A look at ffmpeg shows it also adds 1 to the value read off the ADTS header. Note that this might break other things that happen to have an inverse off by one to match the existing code. 2013-07-25 11:13:01 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: fix seqnum handling for seeks Use the same seqnum as the seek for flushes/segments that are caused by the seek. Also do the same for segment events Fixes #676242 2013-07-25 01:39:58 -0300 Thiago Santos * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: fix seqnum handling for seeks Use the same seqnum as the seek for flushes/segments that are caused by the seek. Also do the same for segment events Fixes #676242 2013-07-25 01:11:31 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: correctly handle seqnum for seeks and segments Use the same seqnum on messages and events for derived events. Fixed for flushes / stream-start / segment after a seek, and segment after a segment. Fixes #676242 2013-07-12 20:01:42 +0200 Arnaud Vrac * ext/soup/gstsouphttpsrc.c: souphttpsrc: always ignore HEAD errors https://bugzilla.gnome.org/show_bug.cgi?id=704241 2013-07-25 14:26:07 +0200 Sebastian Dröge * ext/jpeg/gstjpegenc.c: jpegenc: Clean up reset/start/stop handling 2013-07-25 14:13:10 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: Use base class error handling function instead of replicating it here 2013-07-25 14:12:56 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Clean up handling of reset/start/stop 2013-07-25 10:41:22 +0100 Tim-Philipp Müller * tests/files/id3-407349-1.tag: * tests/files/id3-407349-2.tag: * tests/files/id3-447000-wcop.tag: tests: fix test ID3 tags up not to rely on dodgy typefinding code Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix the typefinder. https://bugzilla.gnome.org/show_bug.cgi?id=681368 2013-07-25 08:22:45 +0200 Alessandro Decina * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: intersect the probed caps with the filter passed to get_caps() 2013-07-24 14:17:45 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: bin: fix compilation 2013-07-24 12:42:31 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: fix UYVP format 2013-07-24 12:41:58 +0200 Wim Taymans * gst/rtp/gstrtpvrawpay.c: vrawpay: fix UYVP format 2013-07-24 12:41:44 +0200 Wim Taymans * gst/rtp/gstrtpvrawpay.c: vrawpay: fix caps 2013-07-24 10:49:03 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix locking Take the lock earlier so that we do things that follow with the right locking. 2013-07-23 17:40:02 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: don't use invalid times in RTCP timeouts An invalid timeout can be calculated when we disabled RTCP by setting the bandwidth to 0. Make sure all code can handle this case. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626 2013-07-23 17:38:20 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: lock session when changing bandwidth Take the session lock when changing the bandwidth properties so that we don't end up with inconsistent behaviour. 2013-07-23 17:37:05 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: reset some RTCP variables The early_send time was set to 0 and always triggering an early RTCP packet. 2013-07-23 15:03:31 +0200 Edward Hervey * gst/isomp4/qtdemux.c: qtdemux: Add all the mpeg XDCAM variants This should cover all known XDCAM variants (which are all mpeg2 video) Fixes #672227 2013-07-03 18:41:42 +0200 Carlos Rafael Giani * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: added custom downstream sync event rtpbin can now send a custom in-band downstream event which informs downstream that the bin has received an RTCP SR packet. This is useful for applications which want to drop the initial unsynchronized received RTP packets. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560 Signed-off-by: Carlos Rafael Giani 2013-07-22 18:00:16 +0100 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix on-the-fly changing of "mode" and "fields" properties We call setcaps() to reconfigure ourselves, but we need to pass the current *sink* caps, not the source caps then. Also fix a caps leak. https://bugzilla.gnome.org/show_bug.cgi?id=641599 2013-07-22 15:23:39 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Add support for group-id in the stream-start event 2013-07-22 15:23:20 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Add support for group-id in the stream-start event 2013-07-22 15:23:11 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: rtpsession: Add support for group-id in the stream-start event 2013-07-22 15:22:55 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: Add support for group-id in the stream-start event 2013-07-22 15:22:47 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Add support for group-id in the stream-start event 2013-07-22 15:22:36 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Add support for group-id in the stream-start event 2013-07-22 15:22:16 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Add support for group-id in the stream-start event 2013-07-22 15:21:49 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Add support for group-id in the stream-start event 2013-07-19 22:59:15 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: use gst_util_uint64_scale*_round. There could be a case where: 1) you do a new set_caps after buffers have been processed. 2) ts_offset gets set to a different value, eg 0.033333333 3) your pads get EOS, but the check dor that doesn't work because you use ts_offset + a truncated value < segment.stop 4) so in the next collected, you end up comparing for example: 0.9999999999 > 1., which is false and means you don't send EOS. Also adds scale_round in two other places where it potentially could have caused problems. 2013-07-15 17:55:19 -0400 Olivier Crête * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: Add WRLE support 2013-07-19 19:35:26 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: make files from Vivotek camera play Skip tracks of 'vivo' subtype with empty stsd instead of erroring out saying that the file is broken. https://bugzilla.gnome.org/show_bug.cgi?id=699791 2013-07-19 17:14:06 +0100 Tim-Philipp Müller * gst/isomp4/gstqtmux.c: qtmux: when streaming don't try to seek when stopping It might cause errors in sinks that are not seekable and have reported this (like e.g. fdsink) https://bugzilla.gnome.org/show_bug.cgi?id=696228 2013-07-19 17:26:54 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: simplify some helpers Some helper functions are not needed anymore or can be simplified. 2013-07-19 17:12:37 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: for non-raw video, move palette in caps We only need to append the palette to raw video buffers, non-raw video has the palette in the caps still. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292 2013-07-19 01:49:20 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: nitpicking in esds parsing 2013-07-19 01:49:07 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: set proper caps for mpeg-1 audio Remove AAC specific fields from mpeg-1 audio caps, remove assumption that the mpeg1 audio layer is 3, and set `parsed' field. https://bugzilla.gnome.org/show_bug.cgi?id=704548 2013-06-17 21:27:37 +0200 Arnaud Vrac * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp9dec.h: * ext/vpx/gstvp9enc.h: vpx: fix compilation when encoder or decoder headers are not installed https://bugzilla.gnome.org/show_bug.cgi?id=704547 2013-07-16 20:41:15 -0400 Nicolas Dufresne * tests/check/elements/videocrop.c: videocrop: Fix unit for GRAY16 formats 2013-07-16 22:17:17 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: remove chapter stream Remove all streams that are actually table of contents, since we will never need the data after parsing them. 2013-07-16 21:59:37 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: send gap event for sparse streams in push mode This allows to pre-roll at least if the next subtitle buffer is far away. 2013-07-16 21:56:07 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: do not use indexes from sparse stream when seeking in push mode This makes seeking more accurate in push mode, since the previous keyframe on a sparse stream might be far away. 2013-07-16 21:04:07 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: advertise subtitle streams as sparse 2013-07-17 17:11:44 +0200 Arnaud Vrac * gst/matroska/matroska-demux.c: mastrokademux: do not push discont buffers if they aren't discont Unset the discont flag instead of posssibly pushing a buffer with a flag that's still set. https://bugzilla.gnome.org/show_bug.cgi?id=682110 2013-07-17 15:10:00 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: extract the palette from stsd Sometimes a palette is inside the stsd, extract it instead of always using the default one 2013-07-17 14:30:16 +0200 Sebastian Dröge * gst/goom2k1/gstgoom.c: goom2k1: Fix event handling and negotiate as soon as possible 2013-07-17 14:27:57 +0200 Sebastian Dröge * gst/goom/gstgoom.c: goom: Fix event handling and negotiate as soon as possible 2013-07-11 19:45:17 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.m: osxvideosink: warn about the future deprecation of the "embed" property 2013-07-17 09:56:01 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: add support for WRAW Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292 2013-07-17 09:54:58 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: palette is appended to buffers, not in caps Fix the palette handling, in 1.0 we append the palette to the buffer instead of placing it on the caps. See also https://bugzilla.gnome.org/show_bug.cgi?id=704292 2013-07-16 15:37:49 -0400 Olivier Crête * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders 2013-07-15 16:24:07 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: reset segment on flush stop cca2f555d14 introduces a regression, where the demux segment is not reset on flush stop, so the next upstream segment event will calculate an invalid base time on the new segment to be sent downstream. https://bugzilla.gnome.org/show_bug.cgi?id=704255 2013-07-06 17:20:49 +0200 Matej Knopp * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: offset samples according to edit list https://bugzilla.gnome.org/show_bug.cgi?id=700264 2013-07-14 12:50:13 +1200 Douglas Bagnall * tests/examples/spectrum/spectrum-example.c: level: Fix the spectrum example for 1.0 The "message" property has been replaced by "post-messages". Pre-patch output: (test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist: object class `GstSpectrum' has no property named `message' New spectrum message, endtime 0:00:00.100000000 (test_spectrum:23101): GStreamer-CRITICAL **: gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)' failed [...] Post-patch: New spectrum message, endtime 0:00:00.100000000 band 0 (freq 400): magnitude -65.988777 dB phase 1.533397 band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900 band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611 band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615 [...] https://bugzilla.gnome.org/show_bug.cgi?id=704179 2013-07-13 20:56:26 +0200 Matej Knopp * gst/audioparsers/gstaacparse.c: aacparse: be less verbose when parsing LOAS streams https://bugzilla.gnome.org/show_bug.cgi?id=704162 2013-07-12 12:31:39 +0200 Wim Taymans * ext/pulse/pulsesink.h: sink: alaw/mulaw caps don't have a layout property 2013-07-12 12:27:53 +0200 Wim Taymans * ext/pulse/pulseutil.c: pulse: relax mulaw and alaw format checks The audio library considers them as encoded formats and does not fill in the sample width. The audio ringbuffers identifies the format as alaw/mulaw and that is always 8 bits. 2013-07-11 16:13:05 +0200 Matej Knopp * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_types.c: qtdemux: unselect instead of ignoring disabled track, detect chapter track https://bugzilla.gnome.org/show_bug.cgi?id=704007 2013-07-11 20:41:23 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: souphttpsrc: ignore errors from HEAD request HEAD requests are used to check the server headers to see if it seekable. Ignore errors from those requests as they shouldn't be critical. https://bugzilla.gnome.org/show_bug.cgi?id=704053 2013-07-12 03:24:08 +0800 Kyosuke Nekomura * gst/audiofx/audioecho.c: audioecho: Fix handling of delay property in PLAYING/PAUSED state https://bugzilla.gnome.org/show_bug.cgi?id=703901 2013-07-09 17:56:57 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Enable proxy caps on the src pads 2013-07-11 16:57:15 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.1.2 === 2013-07-11 15:58:51 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.1.2 2013-07-11 15:58:29 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2013-07-09 15:34:04 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: defer the window handle setup to the main thread 2013-07-09 15:33:18 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.m: osxvideosink: default to the main in case we are not setup yet 2013-07-07 22:16:05 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.m: osxvideosink: close the internal window correctly 2013-07-07 21:14:22 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: only create the NS app thread for Cocoa once The helper thread for Cocoa, in case no NS run loop is running, should be started only once and shared across all the instances running 2013-07-09 19:10:17 +0200 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: correct argument order in gst_util_uint64_scale_int_round https://bugzilla.gnome.org/show_bug.cgi?id=703350 2013-07-09 17:42:59 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Keep caps order from the peer or the filter 2013-07-09 12:42:17 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Fix handling of buffers without a duration We'll have to pop buffer from collectpads and store it internally only to get the timestamp of the next buffer. If we continue to keep it in collectpads, no new buffer to calculate the end time will ever arrive. https://bugzilla.gnome.org/show_bug.cgi?id=703743 2013-07-09 11:53:07 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Fix negotiation with 0/1 framerates https://bugzilla.gnome.org/show_bug.cgi?id=703743 2013-07-09 11:17:59 +0200 Jonas Holmberg * gst/matroska/matroska-demux.c: matroskademux: Unlock stream lock after use Stream lock of sink pad was not unlocked after non-updating seek. 2013-06-27 13:26:31 +0200 Ognyan Tonchev * gst/multipart/multipartmux.c: multipartmux: Re-set need_segment flag after FLUSH_STOP https://bugzilla.gnome.org/show_bug.cgi?id=703182 2013-07-05 11:51:04 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: bufferpool: don't forget to release buffer on error If the pool is stopped while gst_v4l2_buffer_pool_dqbuf() waits for a buffer then the return value is GST_FLOW_FLUSHING. In this case the buffer to queue must also be released. Otherwise is will never be deleted or returned to its pool. https://bugzilla.gnome.org/show_bug.cgi?id=703764 2013-07-08 14:15:10 +0200 Sebastian Dröge * tests/check/elements/rtp-payloading.c: rtp: Fail payloading unit test if an error message is received 2013-07-08 14:09:37 +0200 Sebastian Dröge * gst/rtp/gstrtph263ppay.c: rtph263ppay: Don't pass upstream filter caps to downstream Downstream usually can't accept video/x-h263 but only application/x-rtp, so we would always get an empty intersection here. https://bugzilla.gnome.org/show_bug.cgi?id=702632 2013-07-05 22:00:37 +0200 Piotr Drąg * po/POTFILES.in: po: update POTFILES.in https://bugzilla.gnome.org/show_bug.cgi?id=703685 2013-07-02 11:13:25 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid some strdup 2013-07-02 10:37:50 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add select-stream signal Add a signal to let the app select what streams will be selected. See https://bugzilla.gnome.org/show_bug.cgi?id=634419 2013-07-02 10:37:35 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid strdup 2013-07-02 10:12:17 +0200 J. Rick Ramstetter * gst/rtp/README: * gst/rtpmanager/gstrtpbin.c: rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin https://bugzilla.gnome.org/show_bug.cgi?id=703426 2013-07-01 16:55:01 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: don't extract data from caps twice gst_video_info_from_caps() always extract width, height, interlace mode and framerate now. It is no longer necessary to do it again for encoded formats. https://bugzilla.gnome.org/show_bug.cgi?id=703399 2013-06-20 09:41:48 -0300 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: also consider stop positions in seeks Use seek stop position as range end for requests https://bugzilla.gnome.org/show_bug.cgi?id=702206 2013-06-19 14:06:40 -0300 Thiago Santos * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: allow seeks in ready On is_seekable, check if the server's headers have already been received. If not, do a HEAD request to get them before responding to basesrc. https://bugzilla.gnome.org/show_bug.cgi?id=702206 2013-07-01 17:28:55 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add signal to notify of the SDP This way, the app can look and modify the SDP. 2013-06-21 18:10:28 +0200 Kishore Arepalli * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: Allow negative offsets to specify offset from bottom/right https://bugzilla.gnome.org/show_bug.cgi?id=702826 2013-06-30 21:01:20 +0200 Matej Knopp * gst/isomp4/Makefile.am: * gst/isomp4/qtdemux.c: qtdemux: compute framerate from average sample duration https://bugzilla.gnome.org/show_bug.cgi?id=703350 2013-06-25 21:16:38 +0200 Alban Browaeys * gst/flv/gstflvdemux.c: flvdemux: Add flvversion 1 to the flash-video caps This allows using avdec_flv which requires this field to be present in the caps. FLV only supports flash-video version 1 right now. https://bugzilla.gnome.org/show_bug.cgi?id=703076 2013-07-01 11:37:00 +0200 Sebastian Dröge * gst/interleave/deinterleave.c: deinterleave: Don't hold object lock while sending events downstream Based on a patch by Kishore Arepalli https://bugzilla.gnome.org/show_bug.cgi?id=703114 2013-07-01 10:59:07 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Add MPEG4 video profile/level to the caps 2013-07-01 10:56:28 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Add AAC profile/level to the caps https://bugzilla.gnome.org/show_bug.cgi?id=703312 2013-06-28 15:21:56 +0200 Wim Taymans * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvorbispay.h: vorbispay: add support for config-interval Align code with the theora payloader and add support for the config-interval to periodically send out the config headers. 2013-06-28 15:21:12 +0200 Wim Taymans * gst/rtp/gstrtptheorapay.c: theorapay: small cleanups 2013-06-28 12:08:19 +0200 Wim Taymans * gst/rtp/gstrtptheorapay.c: theorapay: handle streamheaders as well 2013-06-28 12:06:21 +0200 Wim Taymans * gst/rtp/gstrtpvorbispay.c: vorbispay: always collect headers on data When we see a data packet, always check if we need to collect any previous headers. 2013-06-28 11:43:17 +0200 Wim Taymans * gst/rtp/gstrtpvorbispay.c: vorbispay: handle streamheader as well Take config strings from the streamheader when we can Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312 2013-06-27 07:40:29 +0200 David Svensson Fors * gst/rtp/gstrtph264pay.c: rtph264pay: avoid double buffer unmap on error Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171 2013-06-27 17:02:14 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: reset-sync before play Call reset-sync on the rtpbin before we go to playing. This makes us require SR packets for all streams again before we attempt to sync them. If we don't reset, it might be that we combine SR packets from before and after the PAUSE/PLAYING state change and end up with huge bogus offsets. 2013-06-27 16:23:20 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve sync on first packets Don't throw away the first RTCP packet if it arrives before the first RTP packet but remember and use it to signal sync once we get the RTP packet. See https://bugzilla.gnome.org/show_bug.cgi?id=691400 2013-06-27 16:15:45 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: only signal loop when active Only signal the loop function when it is active. 2013-06-27 16:13:37 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: signal timestamp discont We can now use the RESYNC buffer flag to mark a timestamp discont when we update the ts-offset property. 2013-06-26 20:49:41 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: jpegpay: turn some errors into warnings Turn some errors into warnings, we can continue processing so this should not be fatal. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079 2013-06-26 14:58:53 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: avoid some flushes 2013-06-26 14:41:00 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle data message when waiting for reply When we are waiting for a server reply, handle data messages instead of ignoring them. 2013-06-26 14:27:34 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle data messages in separate method Refactor and make a method to handle a data message. 2013-06-25 20:36:18 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add some more docs to handle-request signal See https://bugzilla.gnome.org/show_bug.cgi?id=702705 2013-06-10 17:20:30 -0400 Youness Alaoui * gst/rtsp/gstrtspsrc.c: Send a clock_provide message on the bus when we get a netclock 2013-06-10 17:20:14 -0400 Youness Alaoui * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Expose use-pipeline-clock property 2013-06-24 17:11:35 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: udpsink: bind to the given interface Actually call BINDTODEVICE to bind to the interface as given by the property. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819 2013-06-22 10:59:17 +0200 Sebastian Dröge * ext/vpx/gstvp8dec.c: vp8dec: Error out gracefully if we get an unsupported color format In theory we can only get I420 though, just to be on the safe side. 2013-06-22 10:57:41 +0200 Sebastian Dröge * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9enc.c: vp9: Add support for YV12, Y42B and Y444 color formats The encoder does not work with Y42B and Y444 yet it seems. 2013-06-22 10:26:18 +0200 Sebastian Dröge * ext/vpx/gstvp9dec.c: vp9dec: Update default postproc settings from vp9_dx_iface.c 2013-06-21 13:11:32 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/webm-mux.c: matroska: Add initial VP9 support 2013-06-21 13:07:30 +0200 Sebastian Dröge * configure.ac: * ext/vpx/Makefile.am: * ext/vpx/gstvp9dec.c: * ext/vpx/gstvp9dec.h: * ext/vpx/gstvp9enc.c: * ext/vpx/gstvp9enc.h: * ext/vpx/plugin.c: vpx: Add initial, experimental VP9 support 2013-06-21 10:32:30 +0200 Youness Alaoui * gst/rtsp/gstrtspsrc.c: rtsp: go back into the loop after doing pause After we do a pause request, go back to loop mode so that we can listen for server messages again. See https://bugzilla.gnome.org/show_bug.cgi?id=702705 2013-06-20 23:16:17 -0400 Olivier Crête * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: Wait after the caps to forward the other events First forward the stream-start, then the caps, then the rest 2013-06-21 00:42:02 +0100 Tim-Philipp Müller * sys/ximage/gstximagesrc.c: ximagesrc: clear dts on buffer acquired from pool When setting timestamps on outgoing buffers, clear the dts explicitly, otherwise it may end up being set to a bogus value from last time it was used. Avoids every second or so buffer's dts being set to 0. Not that it should matter for raw video. 2013-06-20 15:35:11 +0200 Wim Taymans * sys/v4l2/gstv4l2.c: v4l2: don't redefine the PERFORMANCE debug variable It is already defined in core. fixes https://bugzilla.gnome.org/show_bug.cgi?id=702732 2013-06-20 14:43:47 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix race in state change to paused When we go to paused, we first flush the connection and then send the pause command. As a result of the flushing, the scheduled paused command can get lost. Wait until the connection is completely flushed and the rtsp task is waiting before issuing the paused or playing request. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705 2013-06-20 11:31:22 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: handle SEGMENT query 2013-06-19 12:37:31 +0200 Sjoerd Simons * sys/v4l2/gstv4l2src.c: v4l2: Optimize negotiation by removing the query filter As cameras tend to have a quite specific set of capabilities (specific framerates for each resolution), getting the peer caps filtered by our probed caps can cause a big increase in the caps size which slows down things quire a bit. As for negotiation v4l2 iterates through the caps of the peer to find the first intersection with the probed caps, getting the fully expanded intersection of capabilities is not useful. Using the same testcase as for bug #702632, adding this patch on top of the patches suggested there speeds up getting the inital frame from around ~14-15 seconds to around ~3-4 seconds. https://bugzilla.gnome.org/show_bug.cgi?id=702638 2013-06-19 10:30:56 +0200 Kishore Arepalli * gst/avi/gstavidemux.c: avidemux: duration query returns zero for DV video in avi https://bugzilla.gnome.org/show_bug.cgi?id=702625 2013-06-19 11:06:37 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Disable usage of allocation queries This can only reliably work if demuxers have a separate streaming thread per srcpad. This should be done in a demuxer base class, which integrates parts of multiqueue https://bugzilla.gnome.org/show_bug.cgi?id=701856 2013-06-11 15:02:21 +0100 Alex Ashley * gst/isomp4/qtdemux.c: Avoid skipping moov atoms for fragmented MP4 files. bug #700505 Following a representation change that causes a resolution change, the video decoder fails to decode correctly. Dashdemux detects the representation change and pushes a new caps event and an initialization segment (a new moov atom) to the downstream qtdemux, but it doesn't handle this new moov yet, it will only parse the first one it receives. This commit changes qtdemux to accept a new moov in a dash bitstream switching scenario. 2013-06-19 00:42:54 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: send stream-start only once for each stream Do not send stream start again when reconfiguring a pad for new caps. That is common for adaptive streams 2013-06-05 17:02:49 +0200 Andoni Morales Alastruey * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideosink: fix support in VM's without hardware acceleration 2013-06-15 12:29:31 +0200 Jens Georg * gst/rtp/gstrtpmp2tdepay.c: rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES instead of MP2T, so accept that as well for compatibility reasons. https://bugzilla.gnome.org/show_bug.cgi?id=702457 2013-06-16 05:40:13 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: manage element state ourselves Lock the state of the all our elements and manage their states outselves. Because we are working async, we can't rely on the state change function to set the state at the right time or to return the right return value from the state change function. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046 2013-06-14 14:09:50 +0200 Bruno Gonzalez * gst/matroska/matroska-demux.c: matroskademux: Don't unlock stream lock without locking it first https://bugzilla.gnome.org/show_bug.cgi?id=702167 2013-06-13 16:00:33 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Use the right hashtable to calculate bandwidth Don't use an unused hashtable to iterate source to calculate bandwidth. Remove unused code. 2013-06-12 16:27:24 -0600 Brendan Long * configure.ac: pulsesink: Require PulseAudio >= 2.0 This is needed for pa_format_info_get_prop_* functions. https://bugzilla.gnome.org/show_bug.cgi?id=686459 2013-06-13 14:23:08 +0530 Arun Raghavan * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: Revert "pulsesink: Make 2.0 dependency optional" This reverts commit 01457027e0d384aca3e551ae684e0aa074ee5498. We'll just depend on PulseAudio 2.0 or above instead of having the bug partially fixed based on the installed libpulse version. 2013-06-13 12:40:15 +0530 Arun Raghavan * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: pulsesink: Make 2.0 dependency optional The getcaps function we added uses some pa_format_info_get_prop... accessor functions that were only added in 2.0, so we only have our getcaps implementation exist if we're compiling against libpulse 2.0 or above. Eventually, we could bump the minimum requirement to 2.0 or above. https://bugzilla.gnome.org/show_bug.cgi?id=686459 2013-06-12 18:23:46 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: Revert "videomixer: When all sinkpads are eos, update output segment stop and forward it" This reverts commit 2d3910fc7901b5f29e16c0fdd4e9067a6d7f66fe. It's not solving any problem and instead causes code to fall apart. https://bugzilla.gnome.org/show_bug.cgi?id=701519 2013-01-09 09:39:33 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: mark subtitle streams as sparse in stream-start event And also mark the streams that should be selected by default if marked so in the headers. https://bugzilla.gnome.org/show_bug.cgi?id=600648 2013-06-11 22:12:58 +0200 Stefan Sauer * gst/audiofx/audiopanoramaorc-dist.c: * gst/audiofx/audiopanoramaorc-dist.h: audiopanorama: add prebuilt files 2013-06-11 20:27:51 +0200 Stefan Sauer * tests/check/elements/audiopanorama.c: audiopanorama: cleanup and expand the tests Split out two more tests. Extract more common code into helpers. Add coverage for float. 2013-06-10 21:15:20 +0200 Stefan Sauer * gst/audiofx/audiopanorama.c: audiopanorama: cleanup of transform() Only map input if we are reading it. Cleanup the logging and the comments a bit. 2013-06-09 20:35:18 +0200 Stefan Sauer * gst/audiofx/Makefile.am: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiopanorama.h: * gst/audiofx/audiopanoramaorc.orc: audiopanorama: use orc to speedup processing Use special variants for the case when we don't change the panorama (pan=0.0). Simplify the processing functions by passing the panorama value directy instead of the instance. Use orc for clearing buffers too. 2013-06-11 19:24:49 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: check last end_time after conversion to running segment The last end_time was saved after conversion, so the comparison had to be made after conversion for it to make sense. https://bugzilla.gnome.org/show_bug.cgi?id=701385 2013-06-11 19:22:20 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: add mix->segment.start to output_end_time When the segment start is not 0, this created a situation where the output_end_time is inferior to output_start_time, and the duration of the next buffer ended up underflowing. https://bugzilla.gnome.org/show_bug.cgi?id=701385 2013-06-11 13:54:53 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Send stream headers after the segment event https://bugzilla.gnome.org/show_bug.cgi?id=700799 2013-06-11 12:26:24 +0200 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Do allocation query after exposing all pads and no-more-pads Also configure video streams as early as possible. Related https://bugzilla.gnome.org/show_bug.cgi?id=701856 but not fixing that. 2013-06-11 12:25:46 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Don't forward CAPS events from upstream Just use the default pad event handler. https://bugzilla.gnome.org/show_bug.cgi?id=701976 2013-05-26 08:18:04 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Cache the getcaps/acceptcaps probe stream getcaps is called frequently during stream setup, and creating a new stream each time is very inefficient. There's some more room for optimisation by caching the queried sink formats as well, but this needs some more changes to listen for format changes on the sink (for when supported formats change between probe stream creation and sink querying). https://bugzilla.gnome.org/show_bug.cgi?id=686459 2013-05-23 21:39:08 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulsesink: Add a getcaps function This allows us to have more fine-tuned caps in READY or above. However, this is _really_ inefficient since we create a new stream and query sink for every getcaps in READY, which on a simple gst-launch line happens about 35 times. The next step is to cache getcaps results. https://bugzilla.gnome.org/show_bug.cgi?id=686459 2013-05-10 11:32:44 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Take a lock on the ringbuffer in acceptcaps This is needed as a concurrent state change could pull the context or stream out from under our feet. https://bugzilla.gnome.org/show_bug.cgi?id=686459 2013-06-09 20:29:09 +0200 Stefan Sauer * gst/audiofx/audiopanorama.c: * gst/audiofx/audiopanorama.h: audiopanorama: move the enum to the header and use instead of gint Move the enum for the processing method to the header so that we can use the type for the instance struct. 2013-06-09 20:32:22 +0200 Stefan Sauer * tests/check/elements/level.c: level: rework the tests to cover other formats too 2013-06-05 16:32:30 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: make sure the element is not deleted before the pool The pool accesses data from the v4l2object so it must exist at least as long as the pool. Refcount the element which controls the object live-time. https://bugzilla.gnome.org/show_bug.cgi?id=701650 2013-06-07 15:38:25 +0200 Sebastian Dröge * ext/libpng/Makefile.am: png: Link with libgstbase for GstByteReader and GstAdapter 2013-06-07 15:15:15 +0200 Sebastian Dröge * gst/wavenc/Makefile.am: wavenc: Link with libgstbase for GstByteWriter 2013-06-07 13:26:35 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Push stream-start event in pull mode before anything else 2013-05-10 12:09:19 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Get rid of acceptcaps side-effects The sink info callback should not have side-effects on the GstPulseSink object since we are sometimes using with a dummy stream in acceptcaps. https://bugzilla.gnome.org/show_bug.cgi?id=686459 2013-06-05 18:36:40 +0200 Sebastian Dröge * configure.ac: Back to development === release 1.1.1 === 2013-06-05 17:58:51 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * common: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: * win32/common/config.h: Release 1.1.1 2013-06-05 16:35:19 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2013-06-05 15:50:04 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Fix taglist ref handling that made the unit test fail 2013-06-05 15:14:54 +0200 Sebastian Dröge * common: Automatic update of common submodule From 098c0d7 to 01a7a46 2013-06-03 09:17:43 +0200 Michael Olbrich * sys/v4l2/v4l2_calls.c: v4l2: iterate controls with V4L2_CTRL_FLAG_NEXT_CTRL if possible In v2.6.18 control classes where added to the v4l2 API. Iterating over CIDs starting with V4L2_CID_BASE will only find controls for the first control class. By iterating with V4L2_CTRL_FLAG_NEXT_CTRL all controls are found. This is necessary to make controls from other control classes available in the extra-controls property. If V4L2_CTRL_FLAG_NEXT_CTRL is not defined at compile time or not supported at runtime then the old mechanism for iterating is used. https://bugzilla.gnome.org/show_bug.cgi?id=701540 2013-06-05 12:12:53 +0200 Wim Taymans * gst/udp/gstudpsink.c: udpsink: avoid leaking the host Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701586 2013-06-04 08:26:33 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: improve pixel aspect ratio handling Instead of just assuming a aspect ratio of 1/1 use VIDIOC_CROPCAP to ask the device. This also add a pixel-aspect-ratio property to overwrite the value from the driver and a force-aspect-ratio property to ignore it. https://bugzilla.gnome.org/show_bug.cgi?id=700285 2013-06-04 17:04:11 +0200 Stirling Westrup * sys/v4l2/v4l2_calls.c: v4l2: Fix compilation with older kernels https://bugzilla.gnome.org/show_bug.cgi?id=701595 2013-06-03 17:07:10 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: call VIDIOC_REQBUFS with count = 0 in pool_finalize Without this the following sequence fails: - set_caps() - object_stop() (does nothing) - set_format() -> VIDIOC_S_FMT - set_config() -> VIDIOC_REQBUFS with count = N - set_caps() - object_stop() - pool_finalize() - set_format() -> VIDIOC_S_FMT => EBUSY Usually the pool is started after set_config(), in which case object_stop() will result in a pool_stop and therefore VIDIOC_REQBUFS with count = 0 but that is not guaranteed. Also calling VIDIOC_REQBUFS with count = 0 in pool_finalize() if necessary fixes this problem. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701543 2013-05-28 19:14:15 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: rework sink buffer refcounting This is a followup patch for #700781, which is not quite correct. The buffer handling is quite complicated here. The original code intended to the the following: - gst_v4l2_buffer_pool_process() calls QBUF and adds the buffer to the local list. - The sink calls gst_buffer_unref() which returns the buffer to the pool but not the 'free list'. - Some time later DQBUF returns the buffer and gst_v4l2_buffer_pool_release_buffer() puts in on the 'free list'. If the buffer must be copied then (parent_class)->acquire_buffer() is called directly to keep the buffer in the pool. This has two problems: 1. If gst_v4l2_buffer_pool_release_buffer() is called before the buffer is returned to the pool, then the buffer is put on the 'free list' twice. This can happen if a reference to the buffer is kept outside the sink, of if DQBUF returns the buffer, that was just queued with QBUF. 2. If buffers are copied, then all buffers are in the pool at all times. As a result gst_v4l2_buffer_pool_stop() and gst_v4l2_buffer_pool_dqbuf() can access pool->buffers at the same time, which can lead to memory corruption. The patch for #700781 fixes those problems, but with the side effect that there are always buffers outside the pool (because they are queued) and the pool is never stopped. This patch fixes this by releasing the reference to the buffer after handling it (to avoid problem 2.) so it can be returned to the pool. gst_v4l2_buffer_pool_release_buffer() is only called if the buffer is already in the pool (to avoid problem 1.). Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701375 2013-06-02 15:24:38 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: make sure taglist is writable before adding tags Avoids assertions 2013-05-30 19:24:13 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: effectively skip tracks that weren't listed on the 1st moov Without this, stream is NULL and the code will try to access it, leading to segfaults. 2013-05-30 19:23:50 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: skip redundant check !got_moov is already checked the line above 2013-06-02 13:03:40 +0200 Stefan Sauer * tests/check/elements/level.c: tests: cleanup level tests Split out a few more tests to avoid checking the same stuff over and over again. 2013-06-01 21:33:46 +0200 Stefan Sauer * gst/level/gstlevel.h: level: remove unused variables in instance struct 2013-05-31 18:13:02 +0200 Stefan Sauer * tests/check/elements/level.c: level: add a test for continous timestamps A test that checks that msg[n].ts + msg[n].dur == msg[n+1].ts. 2013-04-12 16:02:44 +0300 Anton Belka * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: add tags & toc support Write tags as LIST INFO chunk. Format the toc as cue + LIST adtl chunk. Remove old #ifdef'ed code. 2013-05-31 15:12:08 +0200 Wim Taymans * gst/rtp/gstrtph264pay.c: Revert "rtph264pay: Restructuring to allow for adding optional caps" This reverts commit 61666898cfe89a1b21d3e6850ab44f5b1633ed79. This commit changes what the set_sps_pps() function does, not it doesn't set caps anymore (and should have been renamed). The main problem is that not all call sites are updated and thus leak the string. 2013-05-31 15:11:12 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: * tests/check/elements/rtp-payloading.c: Revert "rtph264pay/depay: Add frame dimensions a payloaded caps" This reverts commit 3dca756a5dba55266256f239e3e12a3d058e185a. The H264 RTP spec has no attributes for width and height. 2013-05-31 15:09:51 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: Revert "rtph264pay/depay: Add optional framerate caps for use in SDP" This reverts commit d8825e2a5c0bfb883ff88e2c9da499c800ebca0a. There is no framerate attribute in the h264 RTP spec. 2013-05-31 15:08:16 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: Revert "rtpjpegpay/depay: Replace framesize caps with width/height" This reverts commit 0075d111b475ca27895ee9476154260b6902940b. Extra application/x-rtp are SDP fields, which are strings. 2013-05-31 15:05:51 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * tests/check/elements/rtp-payloading.c: Revert "rtpjpegpay/depay: Replace framerate caps field with fraction" This reverts commit 9fd25a810b859e0ec205176578735100d83de4af. We deal with sdp attributes in application/sdp, which are always strings. 2013-05-31 12:33:21 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add extra TLS url protocols We also support TLS protocols now. 2013-05-30 14:48:42 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer: Add FIXME comment about the DURATION query from adder Currently the code just takes with maximum upstream duration, which is wrong. It should be the maximum upstream duration in running time. 2013-05-30 21:20:59 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: Set a reference to mix->current_caps as the QUERY_CAPS result. 2013-05-30 17:37:13 +0200 Stefan Sauer * gst/level/gstlevel.c: level: misc cleanups Fix some oudated comments. Sort out some confusion of interval_frames and num_frames. 2013-05-29 20:35:41 +0200 Sebastian Dröge * sys/v4l2/v4l2_calls.c: v4l2: Only conditionally use V4L2_CTRL_TYPE_INTEGER_MENU, it's not available in older versions 2013-05-20 16:45:37 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: v4l2: add a property for arbitrary v4l2 controls This makes it possible to set any controls that can be set with VIDIOC_S_CTRL. The controls are set when the property is set (if the device is open) and when the device is opened. https://bugzilla.gnome.org/show_bug.cgi?id=698837 2013-05-28 18:31:07 +0200 Stefan Sauer * gst/level/gstlevel.c: level: fix discontinuities in timestamps 2013-05-28 15:46:43 +0200 Sebastian Dröge * ext/gdk_pixbuf/gstgdkanimation.c: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gdk_pixbuf/gstgdkpixbufdec.h: gdkpixbufdec: Keep serialized events in order, and don't send SEGMENT before CAPS 2013-05-28 15:45:49 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: create and push stream-start in TCP mode 2013-05-28 15:10:07 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: remove some obsolete code It is not needed to do a state change from the _play() function on ourselves. The state change function already did that and we don't want to interfere with that (or use hacks to avoid interference). 2013-05-28 12:24:37 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: set RTCP caps on the RTCP pads 2013-05-28 12:23:37 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: send stream-start and segment events Also send stream-start and segment event on the RTCP pad. We don't need to send anything on the sync_src pad because we already forwarded all incomming events. 2013-04-25 15:25:06 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add signal to handle server requests Add a signal to be notified of a server request. The signal handler can then construct the response message for the server. See https://bugzilla.gnome.org/show_bug.cgi?id=632207 2013-05-27 22:43:25 -0400 Nicolas Dufresne * gst/videomixer/videomixer2.c: videomixer: Maintain z-order when new pad are added https://bugzilla.gnome.org/show_bug.cgi?id=701109 2013-03-06 13:17:54 +0000 Tom Greenwood * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: vp8enc: Add property to manually specify the timebase of the encoder https://bugzilla.gnome.org/show_bug.cgi?id=695709 2013-05-25 12:17:40 -0400 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Always handle flush_stop_pending atomically It is not protected with the COLLECT_PADS_STREAM_LOCK anymore 2013-05-23 18:14:17 -0400 Thibault Saunier * tests/check/Makefile.am: * tests/check/elements/videomixer.c: tests: videomixer: Add a testsuite for videomixer This is mostly copy pasted from -base/tests/check/elements/adder.c 2013-05-25 10:57:02 -0400 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Do not take COLLECT_PADS_STREAM_LOCK when unnecessary Collectpad takes the lock itself when receiving serialized events and we should not take it for not serialized ones 2013-05-24 19:34:05 +0200 Sebastian Dröge * gst/flx/gstflxdec.c: flxdec: Properly skip non-frame chunks 2013-05-24 19:31:14 +0200 Sebastian Dröge * gst/flx/gstflxdec.c: flxdec: Flush data from adapter after reading it Otherwise we're going in an infinite loop, reading the same data over and over again. 2013-04-24 15:39:54 +0000 Andoni Morales Alastruey * gst/goom2k1/Makefile.am: goom2k1: fix more duplicated symbols 2013-05-22 02:40:52 +0200 Sebastian Rasmussen * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * tests/check/elements/rtp-payloading.c: rtpjpegpay/depay: Replace framerate caps field with fraction The previous implementation had the formatting of SDP attributes happen in each RTP payloader, now instead the constituent values are propagated as caps fields. This allows for applications to do SDP offer/answer based on caps negotiation. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748 2013-05-22 01:58:57 +0200 Sebastian Rasmussen * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: rtpjpegpay/depay: Replace framesize caps with width/height The previous implementation had the formatting of SDP attributes happen in each RTP payloader, now instead the constituent values are propagated as caps fields. This allows for applications to do SDP offer/answer based on caps negotiation. Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay to be backwards compatible with previous payloaders. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748 2013-05-22 03:18:07 +0200 Sebastian Rasmussen * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: rtph264pay/depay: Add optional framerate caps for use in SDP This allows for applications to format SDP attributes and still do SDP offer/answer based on caps negotiation. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749 2013-05-22 03:09:44 +0200 Sebastian Rasmussen * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: * tests/check/elements/rtp-payloading.c: rtph264pay/depay: Add frame dimensions a payloaded caps This allows for applications to format SDP attributes and still do SDP offer/answer based on caps negotiation. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749 2013-05-20 22:14:44 +0200 Sebastian Rasmussen * gst/rtp/gstrtph264pay.c: rtph264pay: Restructuring to allow for adding optional caps Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749 2013-05-23 18:42:09 +0200 Sebastian Dröge * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: (dyn|multi)udpsink: Add properties to specify the bind address and port By default we use the any addresses and a random port for binding the socket. 2013-05-23 18:05:07 +0200 Sebastian Dröge * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: (dyn|multi)udpsink: Bind socket before using it https://bugzilla.gnome.org/show_bug.cgi?id=700878 2013-05-23 17:25:29 +0200 Sebastian Dröge * gst/udp/gstmultiudpsink.c: (multi)udpsink: Add missing getters for socket-v6 and used-socket-v6 properties 2013-05-22 21:01:48 -0400 Nicolas Dufresne * gst/videomixer/videomixer2.c: videomixer: Don't hold stream-lock while pushing non-serialized events https://bugzilla.gnome.org/show_bug.cgi?id=700868 2013-05-22 21:00:45 -0400 Nicolas Dufresne * gst/videomixer/videomixer2.c: videomixer: Don't hold object lock while sending events https://bugzilla.gnome.org/show_bug.cgi?id=700868 2013-05-22 17:32:33 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: The return value of gst_pad_set_caps() is not relevant anymore Caps can fail to be set because the pad is not linked yet for example. 2013-05-15 16:39:36 -0700 David Schleef * gst/isomp4/qtdemux.c: qtdemux: Add error if file has playready drm 2013-05-18 15:06:49 -0400 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Send a reconfigure event upstream if sinkpad caps are not usable https://bugzilla.gnome.org/show_bug.cgi?id=684237 2013-05-21 12:02:51 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: keep a reference to all queued buffers Without this, a queued buffer may be required, filled and queued before it is dequeued. Calling gst_buffer_pool_acquire_buffer() ensures that the buffer is set up correctly and gst_buffer_unref() calls buffer_release(). https://bugzilla.gnome.org/show_bug.cgi?id=700781 2013-05-21 13:33:59 +0200 Alexander Schrab * gst/law/mulaw-decode.c: mulawdec: Handle NULL buffers in handle_frame https://bugzilla.gnome.org/show_bug.cgi?id=698894 2013-05-20 21:44:13 +0200 Sebastian Rasmussen * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: rtpjpegpay/depay: Add framesize caps for use in SDP The format of the value adheres to RFC6064 and it is meant to be parsed and included in the SDP sent by gst-rtsp-server to its clients. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748 2013-05-20 21:34:13 +0200 Sebastian Rasmussen * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Add optional framerate caps for use in SDP The format of the value adheres to RFC4566 and it is meant to be parsed and included in the SDP sent by gst-rtsp-server to its clients. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748 2013-05-20 19:59:13 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: When all sinkpads are eos, update output segment stop and forward it https://bugzilla.gnome.org/show_bug.cgi?id=699793 2013-05-20 19:51:07 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: Don't reset the output segment on flush stop Only init it when getting from READY to PAUSED, and change it on seek events. https://bugzilla.gnome.org/show_bug.cgi?id=699793 2013-05-17 10:16:48 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: v4l2: Don't stop streaming when set_caps is called with unchanged caps This can happen if other parts of the pipeline are reconfigured. Stop streaming even for a short amount of time can be quite visible, so it should be avoided if possible. https://bugzilla.gnome.org/show_bug.cgi?id=700503 2013-05-18 15:39:36 -0400 Thibault Saunier * tests/check/pipelines/simple-launch-lines.c: tests: Re-enable videomixer test https://bugzilla.gnome.org/show_bug.cgi?id=684237 2013-05-18 14:36:39 -0400 Thibault Saunier * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: Send caps event from the streaming thread This way we avoid races in caps negotiation and we make sure that the caps are sent after stream-start. https://bugzilla.gnome.org/show_bug.cgi?id=684237 2013-05-05 20:25:20 +0100 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Do not send flush_stop when receiving a seek There is no reason to send a flush-stop when receiving a seek event. In the case of a flushing seek, we could eventually want to, but in the code path were we check if the seek is "flushing", we have the following comment that makes sense: "we can't send FLUSH_STOP here since upstream could start pushing data after we unlock mix->collect. We set flush_stop_pending to TRUE instead and send FLUSH_STOP after forwarding the seek upstream or from gst_videomixer_collected, whichever happens first." https://bugzilla.gnome.org/show_bug.cgi?id=684237 2013-05-05 20:24:49 +0100 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer2: Protect flush_stop_pending with the collectpad stream lock And make sure to expect a flush-stop after a flush-start https://bugzilla.gnome.org/show_bug.cgi?id=684237 2013-05-17 12:37:59 +0200 Michael Olbrich * gst/rtp/gstrtpmp4apay.c: rtpmp4apay: clear config buffer before using it This is necessary because parts of the memory are only modified with "|=" https://bugzilla.gnome.org/show_bug.cgi?id=700514 2013-05-14 17:30:07 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: Do not expect EOS after a segment event if upstream is mss In case qtdemux is handling a mss stream, do not mark the stream to wait for EOS after a segment. Even if it seems to be the last one according to the current streams information. MSS handling is different here because there is another demuxer driving the pipeline 2013-05-14 16:32:51 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: only set channels and rate if qtdemux knows it Setting both of those to 0 is pointless and means that qtdemux doesn't know the real value. Avoid setting it in this case. 2013-05-14 15:23:08 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: set alac caps using info from codec buffer The samplerate field in the STSD atom is not right for some ALAC files (usually when audio is 96kHz/24bits), so the audio caps must be extracted from the codec data. https://bugzilla.gnome.org/show_bug.cgi?id=700382 2013-05-15 11:13:12 +0200 Arnaud Vrac * gst/avi/gstavidemux.c: avidemux: do not push discont buffers if they aren't discont https://bugzilla.gnome.org/show_bug.cgi?id=682110 2013-05-15 10:51:38 +0200 Sebastian Dröge * common: Automatic update of common submodule From 5edcd85 to 098c0d7 2013-05-14 10:28:10 -0400 Joshua M. Doe * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: videocrop: Add support for GRAY16_LE/GRAY16_BE https://bugzilla.gnome.org/show_bug.cgi?id=700331 2013-05-14 17:29:58 +0200 Sebastian Dröge * gst/replaygain/gstrgvolume.c: rgvolume: Send all events through the proxypads instead of just sending to the target Otherwise the sticky events are missing on the proxypads. 2013-05-14 17:29:18 +0200 Sebastian Dröge * tests/check/elements/rgvolume.c: rgvolume: Fix event handling in the unit test 2013-05-14 16:34:54 +0200 Sebastian Dröge * tests/check/elements/rglimiter.c: rglimiter: Fix event handling in unit tests 2013-05-14 16:31:57 +0200 Sebastian Dröge * tests/check/elements/rganalysis.c: rganalysis: Fix event handling in unit test 2013-05-14 16:08:54 +0200 Sebastian Dröge * tests/check/elements/qtmux.c: qtmux: Fix event handling in unit test 2013-05-14 16:00:58 +0200 Sebastian Dröge * tests/check/elements/multifile.c: multifile: Fix event handling in unit test 2013-05-14 13:58:01 +0200 Sebastian Dröge * tests/check/elements/mulawdec.c: * tests/check/elements/mulawenc.c: mulaw: Fix event handling in unit test 2013-05-14 13:52:18 +0200 Sebastian Dröge * gst/matroska/matroska-parse.c: matroskaparse: Make sure to send a segment event before dataflow 2013-05-14 10:52:19 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: only add interlace-mode to the caps for raw formats https://bugzilla.gnome.org/show_bug.cgi?id=700280 2013-05-14 12:03:03 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: copy and set the actual size of the content https://bugzilla.gnome.org/show_bug.cgi?id=700282 2013-05-14 10:25:56 +0200 Sebastian Dröge * tests/check/elements/interleave.c: interleave: Fix event handling in unit test 2013-05-14 09:45:12 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Improve handling of min/max buffer numbers of the buffer pool 2013-05-14 03:42:59 +0200 Matej Knopp * gst/deinterlace/gstdeinterlace.c: deinterlace: set caps for buffer pool config 2013-05-13 13:30:38 -0400 Olivier Crête * gst/multifile/gstmultifilesink.c: multifilesink: Let the base class do get_times This will make sync=TRUE work, the default is still sync=FALSE 2013-05-11 23:08:23 -0400 Nicolas Dufresne * gst/interleave/interleave.c: interleave: Send stream-start before caps event 2013-05-11 23:24:36 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * tests/check/elements/rtpmux.c: rtpmux: Send stream-start before caps 2013-05-11 23:28:12 -0400 Nicolas Dufresne * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffer-test: Send stream-start before caps followed by segment 2013-05-11 23:34:36 -0400 Nicolas Dufresne * tests/check/elements/rtpbin.c: rtpbin-test: Send missing stream-start and segment events 2013-05-13 15:36:19 +0200 Sebastian Dröge * tests/check/elements/level.c: * tests/check/elements/matroskamux.c: tests: Fix some more event handling in tests 2013-05-13 15:19:36 +0200 Sebastian Dröge * tests/check/elements/icydemux.c: icydemux: Fix event handling in unit test 2013-05-13 15:19:25 +0200 Sebastian Dröge * gst/icydemux/gsticydemux.c: icydemux: Fix sticky event handling 2013-05-13 15:06:03 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Push sticky events in the right order 2013-05-13 14:55:14 +0200 Sebastian Dröge * tests/check/elements/deinterleave.c: deinterleave: Fix event handling in test 2013-05-13 14:07:11 +0200 Sebastian Dröge * gst/interleave/deinterleave.c: deinterleave: Fix sticky event handling 2013-05-13 13:55:44 +0200 Sebastian Dröge * gst/interleave/deinterleave.c: deinterleave: Code style fixes 2013-05-13 10:43:32 +0200 Sebastian Dröge * gst/rtp/gstrtpgstpay.c: rtpgstpay: First let baseclass handle events, then put them into the stream Fixes handling of sticky events. https://bugzilla.gnome.org/show_bug.cgi?id=700213 2013-05-09 22:05:24 -0400 Nicolas Dufresne * tests/check/elements/shapewipe.c: shapewipe-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 18:32:23 -0400 Nicolas Dufresne * tests/check/elements/spectrum.c: spectrum-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 18:25:17 -0400 Nicolas Dufresne * tests/check/elements/videofilter.c: videofilter-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 18:23:30 -0400 Nicolas Dufresne * tests/check/elements/wavpackparse.c: wavpackparse-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 18:21:54 -0400 Nicolas Dufresne * tests/check/elements/y4menc.c: y4menc-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-10 14:00:33 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: multipartdemux: fix example pipeline Need jpegparse. 2013-05-10 13:34:16 +0200 Sebastian Dröge * tests/check/elements/alphacolor.c: * tests/check/elements/aspectratiocrop.c: * tests/check/elements/audioamplify.c: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioecho.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiopanorama.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/avimux.c: * tests/check/elements/avisubtitle.c: * tests/check/elements/capssetter.c: * tests/check/elements/deinterlace.c: * tests/check/elements/dtmf.c: * tests/check/elements/equalizer.c: tests: Fix some more unit tests 2013-05-10 13:10:29 +0200 Sebastian Dröge * tests/check/elements/parser.c: tests: Fix parser tests 2013-05-09 22:20:28 -0400 Nicolas Dufresne * gst/shapewipe/gstshapewipe.c: shapewipe: Can't map twice the same buffer for writing I took the opportunity to simplify that code a bit. We now use gst_buffer_make_writable() to make the buffer writable and map twice the same buffer, with first map being read/write, and second read only. This get rid of the critical: GStreamer-CRITICAL **: gst_structure_set_name: assertion `IS_MUTABLE https://bugzilla.gnome.org/show_bug.cgi?id=700044 2013-05-09 22:15:54 -0400 Nicolas Dufresne * gst/shapewipe/gstshapewipe.c: shapewipe: Ensure caps are writable The exist one case where that we endup with original caps in ret, in which case we are not guaratied to have writable caps. Simply ensure this is the caps are writable before entering the loop. https://bugzilla.gnome.org/show_bug.cgi?id=700044 2013-05-09 22:13:51 -0400 Nicolas Dufresne * gst/shapewipe/gstshapewipe.c: shapewipe: Fix sample pipeline in documentation https://bugzilla.gnome.org/show_bug.cgi?id=700044 2013-05-09 18:05:02 -0400 Nicolas Dufresne * tests/check/elements/jpegenc.c: jpegenc-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 17:49:03 -0400 Nicolas Dufresne * tests/check/elements/vp8enc.c: vp8enc-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 17:20:18 -0400 Nicolas Dufresne * tests/check/elements/vp8dec.c: vp8dec-test: Send inital events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 17:19:53 -0400 Nicolas Dufresne * tests/check/elements/wavpackdec.c: wavpackdec-test: Send initial events https://bugzilla.gnome.org/show_bug.cgi?id=700033 2013-05-09 19:40:49 -0400 Olivier Crête * ext/lame/gstlamemp3enc.c: lamemp3enc: Tell GstAudioEncoder about the number of incoming samples lame does internal resampling, but the base class only cares about the number of raw samples, so tell finish frames about that, not the number of samples in the outgoing frame.: 2013-05-09 16:26:19 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: Revert "videomixer2: Take into account new segments" This reverts commit 84ae670ab40b258a10e1e21471e6dc9d786bf086. Actually this is not how it is supposed to work. videomixer creates a [0,-1] segment and then puts frames of the different streams there based on their running times in their own segments. 2013-05-06 23:43:03 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer2: Take into account new segments Also forward the event downstream on the next opportunity. https://bugzilla.gnome.org/show_bug.cgi?id=699793 2013-05-09 09:07:38 +0100 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: Revert "gstrtspsrc: set buffer-size for multicast buffers" This reverts commit 2481e95d038b42297a016f1d2dc1af26d2175b42. This is already done five lines above, it was added a year ago in commit 561b131e. 2013-05-08 19:54:19 -0400 Nicolas Dufresne * tests/check/elements/videofilter.c: videofilter: Unit test send SEGMENT before CAPS https://bugzilla.gnome.org/show_bug.cgi?id=699966 2013-05-08 19:22:31 -0400 Nicolas Dufresne * tests/check/elements/avimux.c: avimux: Unit test sends SEGMENT before caps https://bugzilla.gnome.org/show_bug.cgi?id=699966 2013-05-08 19:08:24 -0400 Nicolas Dufresne * tests/check/elements/audiowsincband.c: audiowsincband: Test should send segment after CAPS This makes the unit test pass again. https://bugzilla.gnome.org/show_bug.cgi?id=699966 2013-05-08 19:00:28 -0400 Nicolas Dufresne * tests/check/elements/audiowsinclimit.c: audiowsinclimit: Test should send segment after CAPS This makes the unit test pass again. https://bugzilla.gnome.org/show_bug.cgi?id=699966 2013-05-08 18:44:32 -0400 Nicolas Dufresne * gst/audiofx/audiowsinclimit.c: audiowsinclimit: Frequence property renamed cutoff Updating the documentation to reflect this change. See: https://bugzilla.gnome.org/show_bug.cgi?id=699964 2013-05-08 15:25:58 -0300 Aha Unsworth * gst/rtsp/gstrtspsrc.c: gstrtspsrc: set buffer-size for multicast buffers For receiving video data via RTSP when the video is sent via multicast there is no way to specify the udpsrc buffer-size. On windows the native network buffer is not large and with video i-frames being huge the buffer is to small and you get i-frame corruption, it looks terrible, and there is no (easy) way to set the udpsrc buffer-size. https://bugs.freedesktop.org/show_bug.cgi?id=52264 2013-05-08 16:02:05 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer2: Send stream-start before caps event https://bugzilla.gnome.org/show_bug.cgi?id=699895 2013-05-07 19:15:49 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: fix compiler warning on type check 2013-04-18 07:49:54 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: push new caps events when caps change Whenever the demuxer has a new caps on a stream, it should set the new_caps variable to true and a new caps event will be pushed before the next buffer 2013-04-17 16:54:22 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: do not push discont buffers if they aren't discont qtdemux takes its buffers from a GstAdapter. Those buffers are created from the larger buffer that it obtained from upstream and they carry the same flags, including DISCONT if it is set. In these cases, all buffers that qtdemux is going to push would be marked as DISCONT. This scenario can make parsers/decoders flush on every buffer leading to no decoding at all hapenning. This patch prevents this by unsetting the flag if it shouldn't be set. 2013-04-12 09:08:16 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: some code cleanup for mss handling code * Explicitly init variables for fragmented formats at init * Do not use GstClockTime type if the variable isn't a timestamp * Fix a style/readability issue at an if block * Group 2 mss mode conditional blocks together to improve readability Conflicts: gst/isomp4/qtdemux.c 2013-04-12 10:21:11 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: avoid storing non-time newsegments to push later This can confuse downstream when they get a byte segment after receiving the natural time segment from qtdemux that it sends when starting to push buffers. This is specially the case with parsers that try to convert the position from byte to time format and might miss the correct position for playback to start. 2013-04-10 18:02:28 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: avoid setting fields to non-writable caps 2013-03-10 04:15:06 +0100 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: don't send so many segment events Only send one segment event in the beginning of the stream, not after each moov and moof atom. Conflicts: gst/isomp4/qtdemux.c 2013-03-08 16:02:26 +0100 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: place incomming timestamps on output Place the incomming timestamp (if any) directly onto the outgoing buffers and interpollate other timestamps. Conflicts: gst/isomp4/qtdemux.c 2013-05-07 10:16:18 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: improve reset of internal status Reset different variables on state changes to ready and when handling a flush-stop. For handling flush stops we should check if there is an upstream adaptive demuxer driving the pipeline as this means that qtdemux will get a new moov atom. For 'standard' isomedia streams this isn't true and qtdemux should keep the previous moov information around. Conflicts: gst/isomp4/qtdemux.c 2013-02-08 00:29:20 -0300 Thiago Santos * gst/isomp4/qtdemux.c: qtdemux: prepare qtdemux to accept multiple dash moovs in a row Whenever dashdemux switches bitrates it sends a new moov with the new stream configuration. qtdemux should now handle this by splitting the exposing and configuration of streams into separate functions. When the stream is new it is configured and exposed, when it is a new bitrate of an existing stream it is only reconfigured. Conflicts: gst/isomp4/qtdemux.c 2013-02-07 14:12:53 -0200 Andre Moreira Magalhaes (andrunko) * gst/isomp4/qtdemux.c: qtdemux: Move FLUSH_STOP/PAUSED_TO_READY handling to a reset method. Conflicts: gst/isomp4/qtdemux.c 2013-01-23 10:55:33 -0500 Louis-Francis Ratté-Boulianne * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Remove old pads when exposing streams and other general fixes. Conflicts: gst/isomp4/qtdemux.c 2013-04-16 10:41:43 -0300 Thiago Santos * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: handle mss streams smoothstreaming streams should be handled as a special kind of fragmented isomedia. In MSS the fragments will not contain a 'moov' atom with the media descriptions, this has to be extracted from the caps. Additionally, there should be another demuxer upstream that is likely going to be the one to answer/act on queries and events, so qtdemux has to forward those upstream. 2013-05-06 16:54:02 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: request 0 buffers when stopping Without this stopping the pool in *_set_caps() is useless. S_FMT will still fail with EBUSY. https://bugzilla.gnome.org/show_bug.cgi?id=699835 2013-05-07 16:32:03 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: By default assume that we're working on non-packetized input Only detecting this in set_format() does not work because we might not get any caps at all, e.g. from filesrc. 2013-05-07 16:30:59 +0200 Sebastian Dröge * ext/libpng/gstpngdec.c: pngdec: Implement parsing functionality This allows to plug pngdec directly without a parser if that is desired. Parsing code is based on pngparse. 2013-05-07 15:54:24 +0200 Sebastian Dröge * ext/libcaca/gstcacasink.c: cacasink: Fix support for RGB formats and add support for more of them 2013-05-04 13:19:53 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't consider the content size from the HTTP headers as absolutely correct The HTTP server could give wrong information, e.g. if the HTTP stream is chunk-encoded or compressed, or if the server does not know the complete size at the time when the file is requested by the client. Also see https://bugs.webkit.org/show_bug.cgi?id=115354 2012-08-20 09:52:32 +0200 Philipp Zabel * sys/v4l2/gstv4l2bufferpool.c: v4l2: fill out v4l2_buffer.bytesused field for v4l2sink When queuing a buffer for a sink, bytesused must contain the actual amount of data. For a source, the driver must overwrite this, so it doesn't matter what is set here. https://bugzilla.gnome.org/show_bug.cgi?id=699598 2013-05-03 23:43:26 +0200 Sebastian Rasmussen * gst/rtp/gstrtpgstpay.c: rtpgstpay: fix invalid memory access in event handler First process event in payloader, then hand it to the base class which takes ownership of the event. https://bugzilla.gnome.org/show_bug.cgi?id=699637 2013-05-04 09:48:02 +0100 Tim-Philipp Müller * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: ac3parse, dcaparse: check buffer size before trimming and unref old buffer as soon as possible. 2013-05-02 15:00:22 +0200 Andoni Morales Alastruey * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: dcaparse: add support for "audio/x-private1-dts" 2013-05-02 14:56:02 +0200 Andoni Morales Alastruey * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: add support for "audio/x-private1-ac3" 2013-05-03 12:46:37 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: always generate video info from caps In the past gst_video_info_from_caps() only video/x-raw. Now it also supports other video/* and image/* formats. With this patch the format won't be GST_VIDEO_FORMAT_UNKOWN and gst_v4l2_buffer_pool_set_config() handles strides correctly. https://bugzilla.gnome.org/show_bug.cgi?id=699570 2013-05-02 09:41:01 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: try to allocate new buffers with VIDIOC_CREATE_BUFS if needed If max_buffers is 0 then an arbitrary number of buffers (currently 4) is allocated. If this is not enough v4l2src starts copying buffers. With this patch VIDIOC_CREATE_BUFS is used to allocate a new buffer. If this fails v4l2src falls back to copying buffers. https://bugzilla.gnome.org/show_bug.cgi?id=699447 2013-04-15 17:37:01 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: fix setting window handle after transition The destroyed flag was not reset properly and it's also not needed as we can check osxwindow != NULL 2013-05-02 13:45:55 +0200 Andoni Morales Alastruey * gst/rtp/Makefile.am: rtp: fix duplicated symbols with libvpx 2013-04-29 10:58:08 +0200 Andoni Morales Alastruey * gst/goom2k1/Makefile.am: goom2k1: fix duplicated symbols with goom 2013-05-01 15:49:45 +0200 Sebastian Dröge * gst/rtp/gstrtph264pay.c: rtph264pay: If the adapter is empty on EOS don't try to map its content https://bugzilla.gnome.org/show_bug.cgi?id=699314 2013-04-30 14:36:38 +0200 Ognyan Tonchev * gst/matroska/matroska-demux.c: matroskademux: add stream-format=raw to aac caps https://bugzilla.gnome.org/show_bug.cgi?id=699303 2013-04-30 13:07:37 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: fix and cleanup VIDIOC_EXPBUF handling clear the struct, and provide a correct error message https://bugzilla.gnome.org/show_bug.cgi?id=699337 2012-07-05 18:02:27 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2: handle return value -ENOTTY for unimplemented VIDIOC_G_PARM Newer kernels return -ENOTTY, older kernels return -EINVAL if the ioctl is not implemented. With this patch, GStreamer handles both cases. https://bugzilla.gnome.org/show_bug.cgi?id=698825 2013-04-30 09:16:07 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: fix broken boolean expression to detect non-frame buffers https://bugzilla.gnome.org/show_bug.cgi?id=699294 2013-04-29 11:07:56 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Better error message when server version is too old We check for the library version at configure time, but the server version can only really be checked at run-time. https://bugzilla.gnome.org/show_bug.cgi?id=698768 2013-04-27 11:24:38 +0100 Tim-Philipp Müller * gst/udp/gstudp.c: udp: log WARNING debug message if UDP multicast is likely to be broken 2013-04-27 11:16:54 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: add includes to get socklen_t defined on Windows https://bugzilla.gnome.org/show_bug.cgi?id=692400 2013-04-27 09:39:45 +0100 Yury Delendik * gst/isomp4/qtdemux.c: qtdemux: add support for VP6F VP6 flash codec https://bugzilla.gnome.org/show_bug.cgi?id=699010 2012-09-05 16:39:31 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/v4l2_calls.c: v4l2: also poll for output devices Note that the V4L2 API defines that for output devices POLLOUT indicates that a buffer is ready to be dequeued. https://bugzilla.gnome.org/show_bug.cgi?id=698992 2012-08-20 09:52:34 +0200 Philipp Zabel * sys/v4l2/gstv4l2object.c: v4l2: fix copying of encoded buffers The existence of a GstVideoFormatInfo does not guarantee, that the buffer contains video frames, so the format must be checked. Also, for encoded buffers the length is variable and must be set. https://bugzilla.gnome.org/show_bug.cgi?id=698949 2012-07-10 15:29:40 +0200 Michael Olbrich * sys/v4l2/gstv4l2object.c: v4l2: add support for mpeg4 and H.263 https://bugzilla.gnome.org/show_bug.cgi?id=698826 2013-04-26 12:16:49 +0200 Edward Hervey * gst/monoscope/gstmonoscope.c: monoscope: Fix debug statement 2013-04-25 21:50:33 +0200 Alexander Schrab * gst/law/mulaw-decode.c: * gst/law/mulaw-decode.h: * tests/check/Makefile.am: * tests/check/elements/mulawdec.c: mulawdec: change base class to GstAudioDecoder https://bugzilla.gnome.org/show_bug.cgi?id=698894 2013-04-25 20:59:52 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: send stream-start event. 2012-10-18 10:37:35 +0200 Philipp Zabel * sys/v4l2/v4l2_calls.c: v4l2: handle ENODATA return value for VIDIOC_ENUMSTD In kernel v3.7-rc1, VIDIOC_ENUMSTD returns ENODATA if the current input does not support the STD API. https://bugzilla.gnome.org/show_bug.cgi?id=698827 2013-04-25 13:19:35 +0200 Wim Taymans * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: docs: add some pay/depayloaders See https://bugzilla.gnome.org/show_bug.cgi?id=551631 2013-04-25 12:44:15 +0200 Sebastian Dröge * gst/law/mulaw-encode.c: * tests/check/elements/mulawenc.c: mulaw: Some minor memleak fixes and cleanup 2013-04-24 13:56:56 +0200 Alexander Schrab * gst/law/mulaw-encode.c: * gst/law/mulaw-encode.h: * tests/check/Makefile.am: * tests/check/elements/mulawenc.c: mulawenc: change to gstaudioencoder base, added bitrate tags 2012-05-03 16:07:27 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: bufferpool: reset buffer size in release_buffer The buffer might still be in use elsewhere when dequeuing buffers for outputs. https://bugzilla.gnome.org/show_bug.cgi?id=698822 2012-04-20 09:53:35 +0200 Michael Olbrich * sys/v4l2/gstv4l2bufferpool.c: v4l2: bufferpool: remove unused includes The hacks that needed these are long gone. https://bugzilla.gnome.org/show_bug.cgi?id=698821 2013-04-25 12:12:23 +0200 Sebastian Dröge * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: (multi)udpsink: Use separate sockets for IPv4 and IPv6 https://bugzilla.gnome.org/show_bug.cgi?id=534243 2013-04-25 10:44:44 +0200 Sebastian Dröge * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: dynudpsink: Use separate sockets for IPv4 and IPv6 https://bugzilla.gnome.org/show_bug.cgi?id=534243 2013-04-25 10:43:56 +0200 Sebastian Dröge * gst/udp/Makefile.am: udp: Don't include removed gstudp.h in noinst_HEADERS 2013-04-17 16:47:31 -0700 Todd Agulnick * sys/osxaudio/gstosxaudiosink.c: osxaudio: Use gst_audio_channel_positions_to_mask() to create mask https://bugzilla.gnome.org/show_bug.cgi?id=698807 2013-04-17 16:12:26 -0700 Todd Agulnick * sys/osxaudio/gstosxaudiosink.c: osxaudio: Remove unused code 2013-04-25 09:16:14 +0200 Sebastian Dröge * gst/udp/Makefile.am: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudp.h: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.h: udp: Remove unused enum type 2013-04-25 09:13:51 +0200 Sebastian Dröge * gst/udp/Makefile.am: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudp-marshal.list: udp: Use the generic marshaller instead of generating marshallers 2013-04-25 09:07:41 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: Rename instance variable from host to multi_group This is more consistent as it's used for the multicast-group property. 2013-04-25 09:03:56 +0200 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Add bind-address property This is equivalent to multicast-group currently for backwards compatibility. In 2.0 this should be handled separately, the former only being the multicast group and the latter always being the address the socket is bound to, even if a multicast group is given. 2013-04-24 16:24:25 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: return output buffer from process Return the output buffer from the process function instead of pushing it ourselves. This way, the subclass can actually deal with the return value of the push. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693727 2012-10-01 09:29:21 -0300 Diogo Carbonera Luvizon * sys/v4l2/gstv4l2object.c: v4l2: save the format correctly If TRY_FMT is not implemented, gst_v4l2_object_get_nearest_size will use S_FMT and will change the device's operation mode. To save the old device mode we need to set the type field or else it will fail to save the previous format. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685209 2013-04-24 15:38:50 +0200 Wim Taymans * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: rtp: a marker bit should translate to RESYNC A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense of missing data) but it means that the packet is the end of a talkspurt and thus a good opportunity to resync to the clock. Use the RESYNC buffer flag to note this. Real discontinuities are marked with DISCONT still when the seqnum has a GAP or when the input buffer has the DISCONT flag set. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204 2013-04-22 23:51:38 +0100 Tim-Philipp Müller * MAINTAINERS: * README: * README.static-linking: * common: Automatic update of common submodule From 3cb3d3c to 5edcd85 2013-04-22 10:19:29 +0200 Sebastian Dröge * gst/rtp/gstrtpjpegdepay.c: rtpjpegdepay: Drop frame if it's less than 2 bytes large https://bugzilla.gnome.org/show_bug.cgi?id=677560 2013-04-18 12:20:08 +0300 Sreerenj Balachandran * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: autodetect: use _plugin_feature_rank_compare API instead of duplicating the code. 2013-04-18 09:37:30 +0200 Sebastian Dröge * sys/osxaudio/gstosxaudioringbuffer.h: osxaudio: Include gstaudioringbuffer.h to fix compilation in 1.0 2013-04-17 21:05:14 +0200 Philippe Normand * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: channel-mask configuration fixes Set channel-mask according to sink's layout in case of stereo layout. Also initialize and reset the mask when an unrecognized channel is detected. https://bugzilla.gnome.org/show_bug.cgi?id=698224 2013-04-15 19:53:28 -0400 Olivier Crête * sys/v4l2/gstv4l2src.c: v4l2src: Disable renegotiation in the negotiate method This way, we don't block the initial negotiation. Thanks to Jeremy Whiting for doing all the testing. https://bugzilla.gnome.org/show_bug.cgi?id=695981 2013-04-15 19:46:12 -0400 Olivier Crête * sys/v4l2/gstv4l2src.c: Revert "v4l2: disable renegotiation" This reverts commit d1b26e1d594ab2b63324e43a36330475e98cdf18. This causes the initial negotiation to never happen if a reconfigure event is received after gst_base_src_start_complete() but before the loop starts. https://bugzilla.gnome.org/show_bug.cgi?id=695981 2013-04-17 21:12:55 +0200 Stefan Sauer * ext/flac/gstflactag.c: flactag: forward caps event This ensures that the downstream element will get the event and negotiates. Add a FIXME for updating the streamheader field on th caps. 2013-04-17 07:50:27 +0200 Stefan Sauer * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: flac: add more logging 2013-04-17 20:24:48 +0200 Sebastian Dröge * sys/osxaudio/gstosxcoreaudiocommon.h: osxaudio: Fix merge conflicts 2013-04-17 10:10:46 +0200 Sebastian Dröge * configure.ac: osxaudio: Fix configure check for osxaudio plugin 2013-04-17 09:50:43 +0200 Sebastian Dröge * sys/osxaudio/gstosxaudioringbuffer.c: osxaudioringbuffer: First check the type, then cast 2013-04-16 22:46:00 +0900 Takashi Nakajima * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudiosink.h: osxaudio: use GST_IS_OSX_AUDIO_SINK in ring buffer. 2013-04-10 21:06:16 +0900 Takashi Nakajima * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: osxaudio: call set_channel_positions() in osxaudioringbuffer acquire() 2013-04-12 12:18:04 -0700 Todd Agulnick * sys/osxaudio/gstosxaudioringbuffer.c: osxaudio: use GST_AUDIO_INFO_* accessors Changes include the following: * Update classname references * Replace GST_BOILERPLATE_FULL with G_DEFINE_TYPE * Use new GstAudioInfo struct and methods * Use new buffer memory allocation scheme Conflicts: sys/osxaudio/gstosxaudioringbuffer.c 2013-04-12 11:51:46 -0700 Todd Agulnick * sys/osxaudio/gstosxcoreaudiocommon.h: * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: adjust for changes to glib mutex api. 2013-04-10 01:21:49 +0900 Takashi Nakajima * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: osxaudio: try to fix up according to Sebastian's comments 2013-04-05 10:02:38 +0200 Philippe Normand * configure.ac: * sys/osxaudio/gstosxaudioringbuffer.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.h: osxaudio: build fixes Enable the osxaudio plugin build in configure.ac and fix some include directive order issues. 2013-04-02 22:28:09 +0900 ted-n * sys/osxaudio/gstosxaudiosrc.c: osxaudio: fix layout for osxaudiosrc 2013-03-30 22:49:34 +0900 ted-n * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudioringbuffer.c: * sys/osxaudio/gstosxaudioringbuffer.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiocommon.h: osxaudio: port to v.1.0 2013-04-16 19:29:48 -0400 Olivier Crête * gst/videomixer/videomixer2.c: videomixer: Don't unref query, we don't own it Fixes double-unref bug. Bug found by Youness Alaoui 2013-04-16 20:41:10 +0200 Philippe Normand * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix SCHEDULING query support Chain the query up to parent before adding _BANDWIDTH_LIMITED flag, so that all the other flags get set, and push mode gets added as supported activation mode. https://bugzilla.gnome.org/show_bug.cgi?id=693484 https://bugzilla.gnome.org/show_bug.cgi?id=698156 2013-03-31 12:05:49 +0200 Philippe Normand * ext/soup/gstsouphttpsrc.c: souphttpsrc: basic scheduling query support Answer to scheduling queries with default parameters and the new _BANDWIDTH_LIMITED_FLAG so that downstream is advised to minimize seek operations and perform on-disk buffering if possible. Bug 693484 2013-04-15 14:32:46 +0000 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.m: osxvideosink: fix segfault accessing osxwindow when not set yet 2012-10-24 12:15:54 +0200 Sebastian Dröge * ext/twolame/Makefile.am: gst: Add better support for static plugins 2012-10-24 12:15:54 +0200 Sebastian Dröge * ext/lame/Makefile.am: gst: Add better support for static plugins 2012-10-24 12:14:50 +0200 Sebastian Dröge * configure.ac: * ext/aalib/Makefile.am: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/flac/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/jack/Makefile.am: * ext/jpeg/Makefile.am: * ext/libcaca/Makefile.am: * ext/libpng/Makefile.am: * ext/mikmod/Makefile.am: * ext/pulse/Makefile.am: * ext/raw1394/Makefile.am: * ext/shout2/Makefile.am: * ext/soup/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/Makefile.am: * ext/vpx/Makefile.am: * ext/wavpack/Makefile.am: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/audioparsers/Makefile.am: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debugutils/Makefile.am: * gst/deinterlace/Makefile.am: * gst/dtmf/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flv/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/imagefreeze/Makefile.am: * gst/interleave/Makefile.am: * gst/isomp4/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/rtsp/Makefile.am: * gst/shapewipe/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * gst/y4m/Makefile.am: * sys/directsound/Makefile.am: * sys/oss/Makefile.am: * sys/oss4/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxvideo/Makefile.am: * sys/sunaudio/Makefile.am: * sys/v4l2/Makefile.am: * sys/waveform/Makefile.am: * sys/ximage/Makefile.am: gst: Add better support for static plugins 2013-04-12 19:26:11 +0000 Andoni Morales Alastruey * gst/goom2k1/Makefile.am: goom2k1: fix duplicated symbol with goom 2013-03-10 17:17:17 +0000 Josep Torra * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxcoreaudiocommon.h: osxaudio: Fixes error: "GST_LEVEL_DEFAULT" redefined 2013-03-10 17:27:30 +0000 Josep Torra * sys/osxaudio/gstosxcoreaudiohal.c: osxaudio: fixes implicit declaration of function 'getpid' 2013-04-14 17:55:02 +0100 Tim-Philipp Müller * autogen.sh: * common: Automatic update of common submodule From aed87ae to 3cb3d3c 2013-04-14 12:32:06 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: add back "iradio-mode" property to disable sending of icecast request headers In 1.0 we now always send the icecast request headers by default, which makes the server send icecasts metadata inserted into the stream if it supports that. However, there are some use cases where this is not desirable, like when just saving a radio stream to disk, so add back the "iradio-mode" property to allow people to disable this. https://bugzilla.gnome.org/show_bug.cgi?id=697984 2013-04-12 16:16:41 +0100 Wim Taymans * gst/rtp/gstrtp.c: rtp: register tag image types The rtpgstdepay needs the type to be available in order to deserialize the event. 2013-04-12 16:08:58 +0100 Wim Taymans * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: handle event parse failures better 2013-04-11 22:25:05 +0300 Anton Belka * gst/wavenc/gstwavenc.c: wavenc: add TOC setter support 2013-04-12 12:31:30 +0200 Stefan Sauer * gst/wavenc/gstwavenc.c: wavenc: small cleanups for toc handling Don't add empty labl/note chunks. Always pass instance as the first param. Add more logging. 2013-04-12 12:58:50 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Proxy the ntp-sync property of rtpbin 2013-04-12 12:51:05 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Give the manager always the name "manager" This allows to use the GstChildProxy interface to adjust properties on it. 2013-04-11 22:53:28 +0100 Tim-Philipp Müller * tests/check/elements/alphacolor.c: * tests/check/elements/apev2mux.c: * tests/check/elements/id3v2mux.c: * tests/check/pipelines/flacdec.c: tests: fix some printf format issues in debug messages 2013-04-11 19:27:15 +0300 Anton Belka * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: add 'note' chunk support 2013-04-11 20:46:26 +0200 Stefan Sauer * ext/pulse/pulsesink.c: pulsesink: add a little more docs to the audioclock 2013-04-11 15:00:05 +0100 Wim Taymans * gst/rtsp/Makefile.am: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add support for NetClientClock When the server suggests a GstNetTimeProvider in the SDP, set up a GstNetClientClock that slaves to the remote clock and suggest this clock in provide_clock. 2013-04-11 14:57:11 +0100 Wim Taymans * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: udpsink: avoid alloc and free in render function Avoid doing alloc and free in the render function for each buffer. Instead, allocate the needed arrays in _init and use those. 2013-04-10 08:36:00 +0200 Stefan Sauer * gst/wavparse/gstwavparse.c: waveparse: remove superfluous g_list_first() calls The variables already point to the start of the list. 2013-04-09 23:13:18 +0100 Andreas Fenkart * gst/rtp/gstrtpsbcdepay.c: rtpsbcdepay: fix sbc frame length calculation for mono and stereo modes https://bugzilla.gnome.org/show_bug.cgi?id=697463 2013-03-25 14:35:02 +0300 Anton Belka * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: add 'note' chunk support Add 'note' chunk support in TOC as GST_TAG_COMMENT https://bugzilla.gnome.org/show_bug.cgi?id=696549 2013-04-08 17:53:09 -0700 David Schleef * gst/isomp4/qtdemux.c: qtdemux: check value inside enda to set endianness 2013-04-09 21:00:12 +0200 Stefan Sauer * common: Automatic update of common submodule From 04c7a1e to aed87ae 2013-04-09 17:34:12 +0200 Wim Taymans * gst/icydemux/gsticydemux.c: icydemux: avoid copy when we can 2013-04-09 16:52:21 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: gstpay: use bufferlist to avoid memcpy 2013-04-09 16:50:56 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: udpsink: improve debug 2013-04-09 00:28:54 +0100 Tim-Philipp Müller * tests/check/elements/wavparse.c: tests: refactor new wavparse test a little Use fakesrc instead of filesrc with /dev/null. https://bugzilla.gnome.org/show_bug.cgi?id=696684 2013-04-08 11:38:33 +0200 Alexander Schrab * gst/wavparse/gstwavparse.c: * tests/check/Makefile.am: * tests/check/elements/wavparse.c: wavparse: error out if we receive eos before any valid data https://bugzilla.gnome.org/show_bug.cgi?id=696684 2013-04-07 01:47:56 +0200 Matej Knopp * gst/deinterlace/gstdeinterlace.c: deinterlace: force deinterlacing in "interlaced" mode https://bugzilla.gnome.org/show_bug.cgi?id=697467 2013-04-06 12:45:28 -0300 Thibault Saunier * ext/gdk_pixbuf/gstgdkpixbufsink.c: gdkpixbufsink: Add timestamp/running-time/stream-time to the emited message 2013-04-05 14:38:43 +0200 Nicola Murino * gst/rtp/gstrtpsbcdepay.c: rtpsbcdepay: fix printf format compiler warnings https://bugzilla.gnome.org/show_bug.cgi?id=697343 2013-04-05 09:34:23 +0100 Todd Agulnick * sys/osxvideo/osxvideosink.m: osxvideo: include pthread.h to fix compiler warning https://bugzilla.gnome.org/show_bug.cgi?id=697303 2013-04-04 22:48:45 +0200 Stefan Sauer * gst/level/gstlevel.c: * gst/level/gstlevel.h: level: resync on discont Drop pending data on discont and start a new cycle with a new base timestamp. Cleanup some variables. 2013-04-03 23:52:47 +0100 Tom Greenwood * ext/vpx/gstvp8dec.c: vp8dec: Improve logging when vpx_codec_peek_stream_info fails Decode failures and missing keyframes should get different debug output. https://bugzilla.gnome.org/show_bug.cgi?id=697232 2013-04-03 18:24:29 -0400 Olivier Crête * gst/rtp/gstrtpsbcdepay.c: rtpsbcdepay: Rank as secondary This way, it will be selected by decodebin Bug reported by andreas.fenkart@streamunlimited.com https://bugzilla.gnome.org/show_bug.cgi?id=697227 2013-04-03 19:05:38 +0200 Stefan Sauer * gst/level/gstlevel.c: * tests/check/elements/level.c: level: subdivide buffers for sample accurate interval handling Previously we would skip level message when processing buffers > the requested interval. Also the message frequency would contain quite some jitter due to only considering them at the end of buffers. Cleanup the tests while we're at it. 2013-03-19 08:23:25 +0100 Stefan Sauer * ext/flac/gstflacenc.c: flacenc: remove old since comments and update logging Don't pretend that we have a timestamp on a buffer when we never set one. 2013-03-18 20:59:23 +0100 Stefan Sauer * gst/spectrum/gstspectrum.c: spectrum: remove old since comment 2013-04-03 17:53:13 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Proxy the multicast-iface property of udpsrc 2013-04-03 11:09:50 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: free all queued buffers Don't just loop over the first num_queued buffers but loop over all the buffers and check if they need to be freed. It is possible that not all buffers are queued and then the entry in our array will be NULL. Those buffers that are not queued were freed in stop(). Fixes https://bugzilla.gnome.org/show_bug.cgi?id=696651 2013-04-03 11:09:37 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: improve debug 2013-04-02 23:42:23 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock Otherwise we get a race where if the RTCP packet comes in first and while it is added the pads, the segment event arrives on the RTP stream, the event may be lost completely and never forwarded. 2013-04-02 23:35:06 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: No need to explicitely forward the caps They are forwarded with the other events 2013-04-02 22:29:38 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: rtpssrcdemux: Remove unused GstSegment 2013-04-02 22:26:02 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Simplify event forwarding Use the gst_pad_forward() mechanic, this way we won't miss pads that are added while we are pushing 2013-04-02 21:53:10 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Don't cross the internal links We had the wrong condition to check for the internal links, so RTP and RTCP pads got crossed! 2013-03-31 17:54:16 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: fix some debug messages 2013-04-02 23:36:22 +0100 Tim-Philipp Müller * sys/v4l2/v4l2_calls.c: v4l2: fix printf format compiler warning in debug message 2012-08-29 17:24:00 +0200 Arnaud Vrac * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: handle TrueHD audio codec id https://bugzilla.gnome.org/show_bug.cgi?id=697113 2013-03-31 19:14:04 +0200 Wim Taymans * gst/rtp/gstrtptheoradepay.c: theorapay: add delta-unit to output frames 2013-03-23 05:22:23 +0100 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: use timestamp delta as duration if possible https://bugzilla.gnome.org/show_bug.cgi?id=696437 2013-03-30 09:44:41 +0100 Josep Torra * gst/rtp/gstrtpsbcdepay.c: rtp: fixes debug message printf related compiler warnings in SBC depayloader 2013-03-28 16:46:36 +0000 Arun Raghavan * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpsbcdepay.c: * gst/rtp/gstrtpsbcdepay.h: rtp: Add an rtpsbcdepay element Pretty straightforward - takes SBC encapsulated in RTP, depayloads, and pushes out SBC buffers. https://bugzilla.gnome.org/show_bug.cgi?id=690582 2013-03-27 22:18:34 +0000 Tim-Philipp Müller * gst/rtp/gstrtpsbcpay.c: rtp: fix SBC payloader Init RTP buffer on stack correctly, so mapping it works without criticals and the payloader actually works. 2013-03-26 14:44:36 +0100 Sebastian Dröge * sys/directsound/gstdirectsoundsink.c: directsoundsink: Check for a subset instead of non-empty intersection in accept-caps 2013-03-26 14:39:53 +0100 Sebastian Dröge * sys/directsound/gstdirectsoundsink.c: directsoundsink: Properly handle the filter caps in get_caps() 2013-03-26 14:35:38 +0100 Sebastian Dröge * sys/directsound/gstdirectsoundsink.c: directsoundsink: Don't unnecessarily get the parent class in class_init The trampoline generated by G_DEFINE_TYPE does that already. 2013-03-25 18:02:10 -0700 David Schleef * gst/avi/gstavidemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: Use %03u for format in gst_pad_create_stream_id_printf() 2013-03-25 10:12:03 +0100 Sebastian Dröge * gst/debugutils/gstcapssetter.c: capssetter: Prevent unneeded caps copying and allocation 2013-02-01 14:33:41 +0100 Dirk Van Haerenborgh * gst/debugutils/gstcapssetter.c: capssetter: Pass any or filter caps upstream capsetter accepts anything and just forwards different caps, as such it should return ANY caps on the sinkpad. https://bugzilla.gnome.org/show_bug.cgi?id=693005 2013-03-06 13:17:54 +0000 Tom Greenwood * ext/vpx/gstvp8enc.c: vp8enc: Fix for divide by zero when using 0/1 framerate https://bugzilla.gnome.org/show_bug.cgi?id=695709 2013-03-24 17:55:55 +0000 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: expose CUE sheet items as tracks not chapter entries in TOC https://bugzilla.gnome.org/show_bug.cgi?id=677306 2013-03-23 13:11:02 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: add more example pipelines 2013-03-23 12:59:26 +0000 Tim-Philipp Müller * gst/wavenc/gstwavenc.c: wavenc: add some example pipelines 2013-03-20 21:38:40 +0300 Anton Belka * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: add TOC support https://bugzilla.gnome.org/show_bug.cgi?id=680998 2013-03-23 04:56:36 +0100 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: make empty subtitle buffer recognition more robust https://bugzilla.gnome.org/show_bug.cgi?id=696244 2013-03-04 15:49:06 -0800 David Schleef * ext/libpng/gstpngenc.c: pngenc: unmap source frame when done 2013-03-22 15:14:15 -0700 David Schleef * gst/isomp4/gstqtmux.c: qtmux: Fix test regression with one buffer streams 2013-03-05 17:00:17 -0800 David Schleef * gst/isomp4/qtdemux.c: qtdemux: split large raw audio samples In order to deal with a file that has samples that are 24 seconds long. Seeking still doesn't work with such files. 2013-03-22 11:54:08 -0700 David Schleef * gst/isomp4/gstqtmux.c: qtmux: Remove documentation for dts-method 2013-03-22 13:24:33 -0700 David Schleef * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: deprecate dts-method property 2013-03-13 17:08:03 -0700 David Schleef * gst/isomp4/gstqtmux.c: qtmux: Fix problems causing bad durations in file - Fix up out-of-order incoming DTS values. - Fix duration of initial sample. 2013-03-12 19:08:26 -0700 David Schleef * gst/isomp4/gstqtmux.c: qtmux: fix all timestamps once first_ts is determined 2013-02-14 16:34:34 -0800 David Schleef * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Use PTS/DTS from incoming buffers Remove old DTS guessing code. 2013-03-18 12:30:50 +0100 Nicola Murino * gst/isomp4/gstqtmuxmap.c: qtmux: expose mulaw caps https://bugzilla.gnome.org/show_bug.cgi?id=696052 2013-03-22 10:50:34 +0000 Tim-Philipp Müller * configure.ac: Require Orc >= 0.4.17 Orc 0.4.17 fixes a bunch crashes on i386 and RPi when orc functions can't be compiled and the fallback function is supposed to be used. Also fixes some issues on PowerPC. https://bugzilla.gnome.org/show_bug.cgi?id=684399 https://bugzilla.gnome.org/show_bug.cgi?id=693862 2013-03-22 08:47:17 +0000 Rodolfo Schulz de Lima * gst/isomp4/qtdemux.c: qtdemux: fix sample leak when processing private qt tags https://bugzilla.gnome.org/show_bug.cgi?id=696355 2013-03-22 02:24:01 +0100 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: set stream language code from tag https://bugzilla.gnome.org/show_bug.cgi?id=696358 2013-03-21 02:55:06 +0100 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: send GAP events for subtitle streams https://bugzilla.gnome.org/show_bug.cgi?id=696244 2013-03-21 02:53:24 +0100 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: ignore empty subtitle buffers https://bugzilla.gnome.org/show_bug.cgi?id=696244 2013-03-21 02:52:07 +0100 Matej Knopp * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: recognize SBTL subtype for subtitles https://bugzilla.gnome.org/show_bug.cgi?id=696244 2013-03-17 16:27:03 +0300 Anton Belka * gst/audioparsers/gstflacparse.c: flacparse: add support for the toc-select event Select tracks from the CUE sheet by sending a toc-select event based on the uid in the TOC. https://bugzilla.gnome.org/show_bug.cgi?id=540891 2013-03-19 18:09:31 -0700 Michael Smith * gst/isomp4/gstqtmux.c: mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end. 2013-03-20 00:35:17 +0000 Tim-Philipp Müller * gst/audioparsers/gstsbcparse.c: sbcparse: pack multiple frames into one output buffer Don't output a single buffer for every tiny SBC frame 2013-03-18 14:59:35 +0000 Bastien Nocera * sys/v4l2/v4l2_calls.c: v4l2: fix compilation against newer kernel headers as on FC19 2013-03-14 14:12:05 +0100 Kishore Arepalli * gst/deinterlace/gstdeinterlace.c: deinterlace: fix infinite loop on EOS with non-default methods or fields Fixes problem of infinite loop in gst_deinterlace_reset_history. Last field in the history was never deinterlaced because idx becomes negative. Happens e.g. with method=scalerbob fields=bottom or method=greedyl fields=top https://bugzilla.gnome.org/show_bug.cgi?id=695644 https://bugzilla.gnome.org/show_bug.cgi?id=693173 2013-03-12 09:48:31 +0000 Kishore Arepalli * ext/dv/gstdvdemux.c: dvdemux: don't return FALSE when dropping sink events Fixes problem in conjunction with avidemux. https://bugzilla.gnome.org/show_bug.cgi?id=695643 2013-03-12 00:16:18 +0000 Tim-Philipp Müller * gst/avi/gstavimux.c: avimux: change raw video caps order so that GRAY8 is last People like colours. https://bugzilla.gnome.org/show_bug.cgi?id=695543 2013-03-11 14:50:41 +0100 Ognyan Tonchev * gst/rtp/gstrtph264pay.c: rtph264pay: Don't use upstream caps with peer_query_caps () Calling gst_pad_peer_query_caps () on the src pad with the caps upstream can produce as a filter from gst_rtp_h264_pay_getcaps () is wrong and makes caps negotiation fail if upstream caps are not NULL. https://bugzilla.gnome.org/show_bug.cgi?id=695629 2013-03-10 09:10:18 +0100 Dirk Van Haerenborgh * gst/avi/gstavimux.c: avimux: support raw BGR https://bugzilla.gnome.org/show_bug.cgi?id=695543 2013-03-10 09:25:34 +0100 Dirk Van Haerenborgh * gst/avi/gstavidemux.c: avidemux: support raw video with negative height https://bugzilla.gnome.org/show_bug.cgi?id=695541 2013-03-05 14:40:56 +0100 Jonas Holmberg * tests/check/elements/autodetect.c: autodetect checktest: Do not fail without videosink If there is no videosink available autovideosink will contain a fakesink instead which needs special treatment in the unit test. 2013-03-09 01:18:30 +0000 Tim-Philipp Müller * Android.mk: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-dtmf.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * gst-plugins-good.spec.in: * gst/dtmf/gstdtmf.c: * gst/dtmf/gstdtmfcommon.h: * tests/check/Makefile.am: * tests/check/elements/.gitignore: dtmf: move dtmf plugin from -bad to -good https://bugzilla.gnome.org/show_bug.cgi?id=687416 2013-03-09 00:30:38 +0000 Tim-Philipp Müller Merge branch 'dtmf-moved-from-bad' https://bugzilla.gnome.org/show_bug.cgi?id=687416 2013-03-05 21:22:18 +0100 Andoni Morales Alastruey * configure.ac: * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudioelement.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudioremoteio.c: osxaudio: add support for iOS using the RemoteIO AudioUnit 2013-03-05 21:17:52 +0100 Andoni Morales Alastruey * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxcoreaudio.c: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxcoreaudiocommon.c: * sys/osxaudio/gstosxcoreaudiocommon.h: * sys/osxaudio/gstosxcoreaudiohal.c: * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: osxaudio: add a façade for the CoreAudio API 2013-03-07 00:00:41 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 2de221c to 04c7a1e 2013-03-03 11:59:31 +0100 Sebastian Dröge * gst/matroska/lzo.c: matroska: Include config.h, it's needed for _stdint.h 2013-03-03 11:53:04 +0100 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Fix (wrong) use of uninitialized variable compiler warning 2013-03-02 13:59:52 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: add variant field to H.263 caps avdec_h263 won't get plugged otherwise. 2013-02-22 19:06:52 +0100 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: skip disabled tracks ISO/IEC 14496-12 specifies disabled tracks should be completely ignored, so just do it. Avoids deadlock during prerolling for some files. Also prevents 'chapter' subtitle tracks from showing up. https://bugzilla.gnome.org/show_bug.cgi?id=693993 https://bugzilla.gnome.org/show_bug.cgi?id=628790 2013-02-25 09:58:13 +0000 Tim-Philipp Müller * tests/check/elements/level.c: tests: re-add suppression for GValueArray warnings to unit test as well 2013-02-28 13:25:06 +0100 Jonas Holmberg * tests/check/elements/dtmf.c: tests: use relative include for out-of-tree builds in dtmf test 2013-02-28 08:46:59 +0100 Stefan Sauer * gst/spectrum/gstspectrum.c: spectrum: remove the since doc-comment from 0.10 2013-02-28 08:44:18 +0100 Stefan Sauer * gst/level/gstlevel.c: * gst/level/gstlevel.h: * tests/examples/level/level-example.c: level: add a "post-messages" property and deprecate "message" In spectrum this was changed from 0.10 to 1.0, lets do this here too. 2013-02-27 18:56:50 -0500 Olivier Crête * tests/check/elements/dtmf.c: tests: Add tests for dtmfsrc 2013-02-27 16:15:27 -0500 Olivier Crête * tests/check/elements/dtmf.c: tests: Fix ref leak in dtmf test 2013-02-26 14:18:20 -0500 Olivier Crête * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional Specific case here is Wowza 3.5.0 2013-02-25 00:35:58 +0100 Thomas Vander Stichele * gst/level/gstlevel.c: level: put back deprecation warnings 2013-02-24 17:00:14 +0100 Thomas Vander Stichele * gst/level/gstlevel.c: * tests/check/elements/level.c: level: send last message on EOS 2013-02-23 14:34:35 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode: handle some more 0-size buffer cases Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684944 2013-02-23 18:50:52 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: fix up example pipeline in docs 2012-11-20 12:14:07 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Update segdone periodically This makes sure that we update segdone based on the read index received during latency updates. As the comment notes, we make some compromises to deal with the fact that segdone is a segment multiple, while the read index offers finer granularity. The updates are also not very often (100ms since that is how often automatic timing updates are provided). All this is required for the baseaudiosink sample alignment code to work at all. https://bugzilla.gnome.org/show_bug.cgi?id=694257 2013-02-13 10:46:54 +0100 Paul HENRYS * gst/rtpmanager/rtpsession.c: rtpsession: Fix wrong code organisation in case of collision change_ssrc field of RTPSession should be set before calling rtp_session_schedule_bye_locked () as this function will call reconsider function that will wake up rtcp_thread which will call rtp_session_on_timeout () that will check change_ssrc to change the ssrc. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184 2013-02-21 11:15:23 -0500 Jean-François Fortin Tam * gst/alpha/gstalpha.c: alpha: improve descriptions of chroma keying-related properties and enums https://bugzilla.gnome.org/show_bug.cgi?id=694374 2013-02-21 15:01:15 -0500 Youness Alaoui * gst/alpha/gstalpha.c: alpha: Do not override the method with custom r/g/b values Depending on the order g_object_set() calls aare made, the target r/g/b settings will override the method if set to green/blue. Change that so we do not use the target-r/g/b values unless the method is set to custom. https://bugzilla.gnome.org/show_bug.cgi?id=694374 2013-02-20 15:46:43 +0100 Ognyan Tonchev * gst/auparse/gstauparse.c: auparse: do not leak src_caps https://bugzilla.gnome.org/show_bug.cgi?id=694275 2013-02-20 21:03:27 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: only delay RTCP when we are a sender Only delay the RTCP thread when we are a sender, which we can know because we have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we are only a receiver and then there is no code path that wakes up the RTCP thread and we end up without RTCP packets. 2013-02-19 11:47:20 +0100 Benjamin Gaignard * configure.ac: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: v4l2: Add support of dmabuf v4l has add a new IOCTL to export a buffer by using dmabuf. This patch allow to use this new IOTCL if it has been defined in videodev2.h I introduce a new IO mode (GST_V4L2_IO_DMABUF) to enable this way of working. https://bugzilla.gnome.org/show_bug.cgi?id=693826 2013-02-18 20:04:05 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: fix up dodgy code that tries to fix up a broken moov atom After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely append to the already-existing memory instead of filling it. 2013-02-18 16:32:13 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: fix potential crash on short MOOV atom Don't unmap short MOOV atom buffer twice, which happened in the case where we don't fix up the MOOV atom. Fixes crashes when thumbnailing partial mp4 file where the MOOV atom is still incomplete. https://bugzilla.gnome.org/show_bug.cgi?id=694010 2013-02-16 16:49:22 +0000 Tim-Philipp Müller * ext/soup/Makefile.am: souphttpsrc: set SOUP_VERSION_{MIN_REQUIRED,MAX_ALLOWED} to suppress deprecations with newer versions https://bugzilla.gnome.org/show_bug.cgi?id=693911 2013-02-16 15:47:02 +0000 Tim-Philipp Müller * configure.ac: * ext/soup/gstsouphttpsrc.c: soup: use default proxy resolver instead of deprecated GNOME proxy resolver Apparently there's no reason to use it any longer. Drop libsoup-gnome dependency while at it, now that we don't need anything from it any more (it only consists entirely of deprecated API now anyways). https://bugzilla.gnome.org/show_bug.cgi?id=693911 2013-02-15 15:43:43 +0000 Tim-Philipp Müller * tests/check/pipelines/tagschecking.c: tests: fix some h264 caps Doesn't fix anything in particular, but is still needed here for correctness. 2013-02-15 08:19:24 +0100 Stefan Sauer * gst/audiofx/audiopanorama.c: audiopanorama: remove channel-mask from caps The channel-mask is only needed for channels>2 which we don't do. 2013-02-15 16:21:21 +0100 Benjamin Gaignard * sys/v4l2/gstv4l2bufferpool.c: v4l2: don't check stride for encoded formats Don't try to check the stride for encoded formats. Some drivers output something != 0 and then we don't want to fail on that. 2013-02-15 14:11:36 +0000 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions So we have to worry less about portability. https://bugzilla.gnome.org/show_bug.cgi?id=692400 2013-02-14 14:13:27 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: remove sof-marker from template caps for now Now that the subset check actually works, this breaks things with demuxers that don't put a "sof-marker" in their jpeg caps, and we don't have a good parser to plug either yet. 2013-02-13 12:32:10 +0100 Sebastian Dröge * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: Put the SOF marker into the caps 2013-02-13 12:02:46 +0100 Sebastian Dröge * gst/rtp/gstrtpamrdepay.c: * tests/check/elements/rtp-payloading.c: rtp-payloading: Fix unit test caps and AMR depayloader sink template caps Fields were missing from the actual caps, or too many fields existed in the template caps. 2013-02-13 11:53:01 +0100 Sebastian Dröge * tests/check/elements/aacparse.c: aacparse: Fix caps used in the unit test The AAC caps passed were incomplete. 2013-02-13 11:49:40 +0100 Sebastian Dröge * tests/check/elements/wavpackenc.c: * tests/check/elements/wavpackparse.c: wavpack: Fix unit tests, width is now called depth in the caps in 1.0 2013-02-12 23:31:22 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: make souphttpsrc unit test work even if http_proxy is set We're testing with an http server on localhost, but don't support an exception list for the http_proxy, so just unset the environment variable to make sure we can run this test properly even if the environment has http_proxy set. Also, don't skip all tests if there is an issue with the SSL server, just run the non-SSL tests then. https://jenkins.qa.ubuntu.com/view/Raring/view/JHBuild%20Gnome/job/jhbuild-amd64-gst-plugins-good/ 2013-02-12 12:53:52 -0800 Michael Smith * gst/isomp4/qtdemux.c: qtdemux: extract codec_data for ProRes 2013-02-08 01:02:10 +1100 Tim 'mithro' Ansell * gst/avi/gstavimux.c: avimux: Fixing buffer leak in gst_avi_mux_do_buffer gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop. 2013-02-10 15:10:32 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: correct duration for audio VBR buffers in pull mode 2013-02-08 21:28:02 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: proper position reporting and push mode timestamping ... and align current_total semantics in push and pull mode, which tracks bytes for CBR and blocks for VBR. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481 2013-02-08 17:05:27 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: delay RTCP until first RTP packet Delay sending the first RTCP packet until we have sent the first RTP packet. Otherwise we will send out a Receiver Report instead of a sender report. See https://bugzilla.gnome.org/show_bug.cgi?id=691400 2013-02-07 15:06:40 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: remove dead code Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355 2013-01-29 10:48:17 +0100 Paul HENRYS * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: forward sticky events and then set caps When a new src pad is added, first forward the sticky events and then set the caps on the src pad Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786 2013-02-07 14:32:26 +0100 Markovtsev Vadim * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: improve debug output Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935 2011-09-26 14:42:51 -0700 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: rework cleanup of streams Move the work of cleaning up the client streams in the free_stream function. This allows us to properly clean up the client streams when we remove an RTP stream as well. Based on patch by Sujay Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156 2013-02-07 11:40:35 +0100 Tim 'mithro' Ansell * gst/videomixer/videomixer2.c: videomixer2: avoid caps leak Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307 2013-02-06 17:15:11 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: do skew estimation only for new timestamps Only run the skew estimation code when we have a new RTP timestamp. If we have the same RTP timestamp, we simply use the previous estimation. This works because the new observation with the same RTP timestamp has to have a bigger receiver time and is thus not going to influence the estimation except for causing more jitter. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023 2013-02-06 13:52:26 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: only EOS when our source sends BYE Only EOS when we receive a BYE event from the SSRC of our stream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453 2013-02-06 13:47:51 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: save the stream SSRC Conflicts: gst/rtsp/gstrtspsrc.c 2013-02-06 13:18:18 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: flush connection when stopping When we stop, we can flush all pending commands so that we can stop and join the task. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924 2013-02-05 22:02:13 +0100 Stefan Sauer * gst/spectrum/README: spectrum: remove outdates readme Lets remove the readme from pre-0.1.0 that is completely irrelevant now. 2013-02-05 07:32:29 +0100 Stefan Sauer * gst/audiofx/audiopanorama.c: audiopanorama: add more debug logging 2013-02-05 08:26:14 +0100 Stefan Sauer * tests/examples/level/level-example.c: level-example. avoid taking the arrays again for each channel for clarity Also introduce some blank lines for better readability and update the comments. 2013-02-04 18:38:41 +0000 Rico Tzschichholz * gst/audioparsers/Makefile.am: audioparsers: fix typo in noinst_headers 2013-02-04 11:08:23 +0100 Stefan Sauer * gst/audiofx/audiopanorama.c: audiopanorama: further port to 1.0 Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though. 2013-02-03 22:45:52 +0100 Stefan Sauer * gst/audiofx/audiopanorama.c: audiopanorama: fix caps We don't turn float into 32bit pcm. Looks like a typo from updating the caps. 2013-02-03 13:14:50 +0100 Olivier Crête * gst/level/gstlevel.c: level: Add missing coma between formats 2013-01-31 22:55:18 +1100 Matthew Waters * gst/videomixer/videomixer2.c: videomixer: fix eos timestamp check fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935 2013-01-31 11:35:09 +0100 Dirk Van Haerenborgh * gst/avi/gstavimux.c: avimux: add support for raw monochrome 8-bit video Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932 2013-01-18 21:08:12 +0400 Alexey Chernov * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: osxvideosink: Make GstNavigation key input events in osxvideosink compatible with x(v)imagesink ones 2013-01-29 10:30:32 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: avoid '...is used uninitialized' 2013-01-09 13:24:49 -0500 Youness Alaoui * gst/isomp4/qtdemux.c: qtdemux: set interleaved layout correctly for LPCM audio https://bugzilla.gnome.org/show_bug.cgi?id=663458 2013-01-08 20:45:21 -0500 Youness Alaoui * gst/isomp4/qtdemux.c: qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7) https://bugzilla.gnome.org/show_bug.cgi?id=663458 2013-01-08 20:42:35 -0500 Youness Alaoui * gst/isomp4/qtdemux.c: qtdemux: print all debug for sound sample description v2 https://bugzilla.gnome.org/show_bug.cgi?id=663458 2013-01-08 20:14:17 -0500 Youness Alaoui * gst/isomp4/qtdemux.c: qtdemux: sound sample description v2 doesn't override samples_per_packet https://bugzilla.gnome.org/show_bug.cgi?id=663458 2013-01-08 19:57:50 -0500 Youness Alaoui * gst/isomp4/qtdemux.c: qtdemux: pass stsd data to qtdemux_audio_caps() We will need that later for LPCM format support. Disable QDM2 parsing of stsd data which dead code before as well because data was always NULL. https://bugzilla.gnome.org/show_bug.cgi?id=663458 2013-01-08 19:56:46 -0500 Youness Alaoui * gst/isomp4/qtdemux.c: qtdemux: add len check for sound sample descriptions v1 and v2 https://bugzilla.gnome.org/show_bug.cgi?id=663458 2013-01-28 22:42:25 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpssrcdemux.c: rtpmanager: use C89-style comments 2013-01-28 18:06:15 -0500 Olivier Crête * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Fix double-declared variable 2013-01-28 17:58:20 -0500 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: rtp: Fix compilation errors in previous patches 2011-04-28 22:59:28 +0200 Haakon Sporsheim * gst/rtpmanager/gstrtpsession.c: rtpsession: Ensure MT safe event handling and plug event leak. https://bugzilla.gnome.org/show_bug.cgi?id=667826 2011-10-17 23:45:37 +0200 Idar Tollefsen * gst/rtpmanager/gstrtpsession.c: rtpsession: mt-safe event-push By taking a ref of the sink-pad under lock, it won't dissappear while the push is taking place https://bugzilla.gnome.org/show_bug.cgi?id=667816 2012-01-04 10:29:45 +0100 Pascal Buhler * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE https://bugzilla.gnome.org/show_bug.cgi?id=667815 2013-01-28 20:42:26 +0100 Stefan Sauer * common: Automatic update of common submodule From a942293 to 2de221c 2013-01-28 11:54:54 +0000 Tim-Philipp Müller * gst/audioparsers/gstsbcparse.c: sbcparse: init some variables to avoid bogus compiler warnings 2013-01-28 12:41:04 +0100 Wim Taymans * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvrawdepay.c: rtpdepay: remove payload type restrictions Remove the pt restrictions for all the depayloaders that have an encoding-name. We can use this to autoplug decoders. Remove the encoding-name for all the payloaders with a fixed payload type. We now either have an encoding-name or a pt in the sinkpad caps of a depayloader. See https://bugzilla.gnome.org/show_bug.cgi?id=639292 2013-01-28 12:23:41 +0100 Marc Leeman * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpmp4vdepay.c: rtp: remove payload requirements from selected depayloaders encoding name is required in the caps and is a better fit for autoplugging than the pt value. Hardware manufacturers have a bad habit of skimming through RFCs and in this case; use unassigned numbers for encoders instead of dynamic numbers. In essence, this patch will add support for a lot of Bosch hardware encoders without breaking autoplugging. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292 2013-01-27 10:17:59 +0530 B.Prathibha * tests/examples/jack/jack_client.c: * tests/examples/rtp/server-alsasrc-PCMA.c: * tests/icles/ximagesrc-test.c: tests: use g_timeout_add_seconds instead of g_timeout_add https://bugzilla.gnome.org/show_bug.cgi?id=692615 2013-01-27 12:54:15 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: push mode: only parse moov 1 once Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570 2013-01-26 22:58:29 +0000 Tim-Philipp Müller * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: fix compiler warning gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1': gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function 2013-01-25 21:06:05 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfdepay.c: rtpdtmfdepay: Fix missing work in doc 2013-01-24 21:00:08 -0500 Olivier Crête * tests/check/elements/dtmf.c: tests: Add test for rtpdtmfdepay and rtpdtmfsrc 2013-01-25 20:39:33 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Post the messages after the clock wait This way, the messages will be closer in time to when the packets are sent out 2013-01-25 20:37:53 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Only set the duration when starting to send The duration depends on the clock rate, which could change due to renegotiation 2013-01-25 20:37:09 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: remove "ssrc" from caps ssrc is uint and we don't have a uint range type 2013-01-24 21:08:51 +0000 Tim-Philipp Müller * gst/isomp4/atoms.h: qtmux: set language to 'undefined' instead of English by default 2013-01-23 21:35:25 -0500 Olivier Crête * sys/ximage/gstximagesrc.c: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: ximagesrc: Set the pixel aspect ratio correctly in the caps 2013-01-08 08:56:45 +0100 Sjoerd Simons * sys/v4l2/gstv4l2src.c: v4l2: Re-enable prepare-format emission With the port to gstreamer 1.0 the prepare-format signal stopped being emitted. Start emitting this again for use in uvch264src. While there change the emission to include the caps for extra flexibility instead of fource, width, height. https://bugzilla.gnome.org/show_bug.cgi?id=692042 2013-01-22 18:12:10 +0100 Benjamin Gaignard * autogen.sh: autogen.sh: allow calling from out-of-tree Signed-off-by: Benjamin Gaignard https://bugzilla.gnome.org/show_bug.cgi?id=692309 2013-01-22 19:26:09 +0100 Mark Nauwelaerts * gst/audioparsers/gstsbcparse.c: audioparsers: sbc: fix bogus compiler warning gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame': gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i 2013-01-19 13:27:48 +0000 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: don't error out if pa_stream_proplist_update() with new tags fails Shouldn't really happen these days, but if it does, it's not really a problem either. https://bugzilla.gnome.org/show_bug.cgi?id=656068 2013-01-16 18:01:23 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: skip souphttpsrc tests if there is no local http server to use Skip tests if the server couldn't be started or we can't connect to it for some reason (e.g. draconic build bot environments). 2013-01-16 14:32:56 +0100 Thijs Vermeir * gst/audioparsers/gstsbcparse.c: autoparsers: use appropriate printf format for gsize 2013-01-15 15:05:43 +0100 Martin Pitt * tests/check/Makefile.am: tests: use _1_0 variants for the various registry variables These override the variants without version suffix. Makes 'make check' work properly in environments that set the suffixed variant for 1.0, such as jhbuild. 2013-01-11 19:24:43 +0400 Alexey Chernov * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideosink: Fix crash in osxvideosink with external window output 2013-01-16 12:04:59 +0400 Alexey Chernov * sys/osxvideo/cocoawindow.m: osxvideosink: Make GstGLView propagate input events to its parent view Fixes bug #691832 2013-01-16 10:19:36 +0000 Tim-Philipp Müller * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: update some fields in the caps to their new name and to match the parser. "mode" got renamed to "channel-mode" and "allocation" to "allocation-method". 2013-01-15 17:44:33 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-rtp.xml: docs: add sbcparse and rtpsbcpay to plugin docs 2013-01-15 17:38:24 +0000 Tim-Philipp Müller * gst/audioparsers/Makefile.am: * gst/audioparsers/gstsbcparse.c: * gst/audioparsers/gstsbcparse.h: * gst/audioparsers/plugin.c: audioparsers: add SBC audio parser From-scratch rewrite, the bluez one was useless and broken. https://bugzilla.gnome.org/show_bug.cgi?id=690582 2013-01-15 15:05:04 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From a72faea to a942293 2013-01-10 12:38:13 +0000 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtp: import rtpsbcpay from bluez and port to 1.0 Compiles, but not tested yet (sbc elements still need to be ported). https://bugzilla.gnome.org/show_bug.cgi?id=690582 2013-01-09 19:59:16 -0500 Olivier Crête * gst/dtmf/Makefile.am: * gst/dtmf/gstdtmf.c: * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfdetect.h: * gst/dtmf/tone_detect.c: * gst/dtmf/tone_detect.h: dtmf/spandsp: Move dtmfdetect to use libspandsp Remove our copy of the tone_detect.c file and use the original from libspandsp. Also move the element to the spandsp plugin. 2011-02-13 17:51:45 -0800 Marcel Holtmann * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Remove workaround for compiler warnings 2010-05-19 16:59:30 +0200 Marcel Holtmann * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Add pragma based workaround for GStreamer warnings 2010-01-01 17:08:17 -0800 Marcel Holtmann * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Update copyright information 2009-01-30 00:31:15 +0100 Marcel Holtmann * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2009-01-01 19:33:20 +0100 Marcel Holtmann * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Update copyright information 2008-12-23 05:25:50 +0100 Marcel Holtmann * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2008-12-20 21:42:49 +0200 Johan Hedberg * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: More coding style fixes 2008-02-29 19:37:15 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2008-02-28 19:38:53 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2008-02-20 13:37:00 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2008-02-19 19:49:24 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Update gstreamer plugin to use new sbc API. 2008-02-02 03:37:05 +0000 Marcel Holtmann * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Update copyright information 2008-01-30 14:21:43 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Fixes gstreamer caps and code cleanup. 2008-01-24 14:25:29 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2008-01-23 19:17:33 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2008-01-23 13:14:02 +0000 Luiz Augusto von Dentz * gst/rtp/gstrtpsbcpay.c: * gst/rtp/gstrtpsbcpay.h: rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-08 16:27:42 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtp: small improvements 2013-01-07 15:50:33 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor handle sync code Move the code that combines the last SR packet and the current jitterbuffer sync values into a sync structure, into its own function. We want to reuse this bit later. 2013-01-07 15:45:10 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtp: include downstream latency in SR calculations When we make a mapping between an RTP timestamp and an NTP timestamp, include the downstream latency applied to the sinks. This makes it possible to have both sinks run with different latencies and still have correct sync on the client. It also is more correct because the RTP timestamp in the SR report will actually correspond more closely to the NTP time it was sent on the server. For pipelines with high latency on the sender side, this actually allows a GStreamer receiver to perform synchronisation instead of dropping the RTCP packets. 2013-01-07 14:25:14 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: don't cast event functions There is no need to cast the event functions and only causes problems later when we change the signature later and things silently compiles wrong code. 2013-01-07 14:23:34 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtp: more debug 2013-01-07 14:22:48 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: improve debug 2013-01-02 00:03:27 +0000 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: sanity check size of available packet data for reading to avoid memory waste On Windows and OS/X, _get_available_bytes() may not return the size of the next pending packet, but the size of all pending packets in the kernel-side buffer, which might be rather large depending on configuration. Sanity-check the size returned by _get_available_bytes() to make sure we never allocate more memory than the max. size for a packet, if it's an IPv4 socket. https://bugzilla.gnome.org/show_bug.cgi?id=610364 2013-01-04 10:03:32 +0100 Robert Krakora * sys/v4l2/v4l2_calls.c: v4l2: Also handle the new ENOENT return value of VIDIOC_QUERYCTRL https://bugzilla.gnome.org/show_bug.cgi?id=691098 2013-01-01 19:14:36 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: add test for souphttpsrc error handling with data https://bugzilla.gnome.org/show_bug.cgi?id=678429 2012-06-22 21:56:52 +0000 Norbert Waschbuesch * ext/soup/gstsouphttpsrc.c: souphttpsrc: error out properly when receiving data along with an error status When receiving an error code from the http server, such as 404, data might be sent along with it, like a web page. We don't want to output that data in this case, and we also want to pass the FLOW_ERROR return back to the base class, so it can stop properly. https://bugzilla.gnome.org/show_bug.cgi?id=678429 2013-01-01 12:20:20 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: docs: update for new rtspsrc proxy-id and proxy-pw properties 2013-01-01 12:19:23 +0000 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-cairo.xml: docs: fix docs build and update after removal of old cairo elements 2013-01-01 12:12:02 +0000 Tim-Philipp Müller * ext/cairo/Makefile.am: * ext/cairo/gstcairo.c: * ext/cairo/gstcairorender.c: * ext/cairo/gstcairorender.h: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: * ext/cairo/gsttimeoverlay.c: * ext/cairo/gsttimeoverlay.h: cairo: remove old cairo-based text renderering element They haven't worked well or at all in a very long time and were rather bit-rotten, and there's no need for them any more. 2013-01-01 11:52:09 +0000 Tim-Philipp Müller * configure.ac: * ext/cairo/.gitignore: * ext/cairo/Makefile.am: * ext/cairo/gstcairo-marshal.list: * ext/cairo/gstcairo.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: * tests/examples/Makefile.am: * tests/examples/cairo/Makefile.am: * tests/examples/cairo/cairo_overlay.c: cairo: port cairooverlay to 0.11 The other elements are not that interesting now that we're using pangocairo in the pango plugin, and should probably just be removed. 2012-12-31 18:59:18 +0000 Tim-Philipp Müller * tests/examples/rtp/server-decodebin-H263p-AMR.sh: examples: check for uri argument in decodebin-h264p-amr server example Otherwise people get a rather confusing error message. 2012-12-31 00:22:27 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add "proxy-id" and "proxy-pw" properties to match souphttpsrc. user/password passed via the URI will still take precedence though. https://bugzilla.gnome.org/show_bug.cgi?id=395427 2012-12-25 16:48:43 +0000 Tim-Philipp Müller * sys/oss4/oss4-sink.c: oss4sink: notify "volume" property on open to make apps query initial volume The initial volume might not be the property default, so emit a notify on the volume property to make apps get an up-to-date reading of the current volume. https://bugzilla.gnome.org/show_bug.cgi?id=631053 2012-12-20 17:12:30 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix cmd comparison Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476 2012-12-20 17:12:20 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add some more debug 2012-12-20 16:44:24 +0100 Wim Taymans * ext/raw1394/gst1394clock.c: 1394clock: mark our clock type as OTHER 2012-12-20 16:15:13 +0100 Jonas Holmberg * tests/check/elements/rtp-payloading.c: tests: add jpegpay unit test See also https://bugzilla.gnome.org/show_bug.cgi?id=684955 2012-12-20 15:55:02 +0100 Wim Taymans * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: pass flowreturn upstream 2012-09-27 15:42:56 +0200 Jonas Holmberg * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: handle width and height > 2040 If width or height is greater than 2040 set width and height to zero in the rtp header and add x-dimensions to outcaps. Solves #684955 2012-12-20 13:03:41 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: cleanup in flag define 2012-12-20 13:02:57 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: improve debug 2012-12-18 15:56:59 +0100 Thijs Vermeir * ext/wavpack/gstwavpackenc.c: wavpack: use appropriate printf format for gsize 2012-12-18 15:55:43 +0100 Thijs Vermeir * ext/taglib/gstid3v2mux.cc: taglib: use appropriate printf format for gsize 2012-12-18 15:54:08 +0100 Thijs Vermeir * ext/gdk_pixbuf/gstgdkpixbufdec.c: gdkpixbuf: use appropriate printf format for gsize 2012-12-18 15:51:46 +0100 Thijs Vermeir * gst/rtp/gstrtpgstdepay.c: rtp: use appropriate printf format for gsize 2012-12-18 15:46:56 +0100 Thijs Vermeir * gst/deinterlace/gstdeinterlace.c: deinterlace: use appropriate printf format for gsize 2012-12-17 16:35:56 +0100 Philippe Normand * gst/interleave/interleave.c: * gst/interleave/interleave.h: interleave: set src pad caps upon last sink pad CAPS event Gather caps on all sink pads before setting the src pad caps. This is specially needed when the audio channel mapping is set on the sink pads and the element needs to preserve it on its src pad. https://bugzilla.gnome.org/show_bug.cgi?id=690267 2012-12-17 22:55:12 +0000 Tim-Philipp Müller * gst/matroska/matroska-read-common.c: matroskademux: skip empty tags instead of trying to add tags with empty strings, which causes criticals at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=690358 2012-12-17 15:17:12 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: Make sure the caps are actually writable before changing them 2012-12-17 15:01:02 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: Use the peer caps for restrictions instead of the srcpad allowed caps Otherwise we will intersect with the srcpad template caps and add all the caps fields that the parser will ever set, no matter if downstream restricts this field or not. This requires upstream to set this field on the caps to successfully negotiate. https://bugzilla.gnome.org/show_bug.cgi?id=690184 2012-12-14 22:25:08 +0000 Koop Mast * configure.ac: * sys/v4l2/gstv4l2object.h: v4l2: Teach where the videodev2.h header lives on freebsd. https://bugzilla.gnome.org/show_bug.cgi?id=690233 2012-12-16 23:27:41 +0000 Alexey Fisher * gst/matroska/matroska-mux.c: matroskamux: set appropriate block header flag for VP8 invisible frames Useful for debugging mostly. https://bugzilla.gnome.org/show_bug.cgi?id=654259 2012-12-16 15:25:03 +0000 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-rtpmanager.xml: * gst/rtpmanager/gstrtpdtmfmux.c: docs: add rtpmux and rtpdtmfmux to plugin docs https://bugzilla.gnome.org/show_bug.cgi?id=629117 2012-12-16 15:13:38 +0000 Tim-Philipp Müller * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpmuxer.c: * tests/check/Makefile.am: * tests/check/elements/.gitignore: rtpmanager: move rtpmux and rtpdtmfmux elements from -bad https://bugzilla.gnome.org/show_bug.cgi?id=629117 2012-11-03 20:38:00 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * gst/rtpmanager/gstrtpmuxer.c: * tests/check/elements/rtpmux.c: rtpmux: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-10-17 17:34:26 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there. 2012-09-10 20:38:14 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * tests/check/elements/rtpmux.c: rtpmux: Misc fix for 0.11 Convert the incoming caps before proxying them Clear the last_pad when going to ready tests: Implement accept_caps, don't leak event 2012-07-17 16:39:02 +0200 Wim Taymans * gst/rtpmanager/gstrtpmux.c: rtpmux: update for RTP buffer api changes 2012-04-05 18:02:56 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpmuxer.c: rtpmux: Update for GST_PLUGIN_DEFINE() API changes 2012-04-02 11:07:18 +0200 Wim Taymans * gst/rtpmanager/gstrtpmux.c: rtpmux: fix compilation 2012-03-11 19:06:59 +0100 Wim Taymans * gst/rtpmanager/gstrtpmux.c: rtpmux: fix for caps api changes 2012-01-26 06:58:46 -0500 Matej Knopp * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix compiler warnings 2012-01-29 18:01:05 +0000 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Unref non-forwarded events Also, don't unref forwarded ones 2012-01-28 16:57:03 +0000 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: resync iterator on resync 2012-01-27 12:08:52 +0100 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Re-push sticky events on input pad change 2012-01-25 15:43:01 +0100 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Don't leak gvalue from iterator 2012-01-25 16:46:44 +0100 Wim Taymans * gst/rtpmanager/gstrtpmux.c: rtpmux: more porting 2012-01-24 14:20:52 +0100 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * tests/check/elements/rtpmux.c: rtpmux: port to 0.11 2011-11-04 12:22:37 +0100 Wim Taymans * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: make request pads take _%u 2011-04-14 14:34:26 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpdtmfmux: Add last-stop to dtmf-event upstream events Add the running time of the last outputted buffer to the upstream "dtmf-event" events so that the dtmf source does not leave a gap. 2010-11-25 19:21:11 +0100 Edward Hervey * gst/rtpmanager/gstrtpmux.c: rtpmux: Remove dead assignments 2010-10-19 13:43:14 +0300 Stefan Kost * gst/rtpmanager/gstrtpmux.c: rtpmux: add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. 2010-09-30 16:07:29 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: Improve documentation Add an example pipeline, and try to explain a bit more what it does. 2010-09-24 13:29:55 +0300 Stefan Kost * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: remove unused variable 2010-09-24 13:25:22 +0300 Stefan Kost * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: remove unused signal boilerplate 2010-09-24 13:24:48 +0300 Stefan Kost * gst/rtpmanager/gstrtpmux.c: rtpmux: no need to ref pad in _chain() 2010-08-25 22:56:03 -0400 Youness Alaoui * gst/rtpmanager/gstrtpmux.c: rtpmux: Unlock the right mutex The mutex locked is for the 'mux' object, but we unlock the pad, which means that if the rtpmux gets a flush, then the object lock will stay locked forever, causing it to freeze the next time it tries to take it. Fixes bug #627991 2010-07-01 15:19:12 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Add support for GstBufferList Factor out most of the buffer handling and implement a chain_list function. Also, the DTMF muxer has been modified to just have a function to accept or reject a buffer instead of having to subclass both chain and chain_list. 2010-07-01 15:15:49 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Don't leak invalid buffers 2010-06-03 10:43:20 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: fix missing debug log message argument 2010-05-10 18:37:55 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Add some debug messages 2010-05-07 18:56:57 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpdtmfmux: Remove stream-lock event handling 2010-05-07 18:54:49 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Update doc for simplification 2010-05-07 18:40:30 -0400 Olivier Crête * tests/check/elements/rtpmux.c: tests: Change tests to not use the priority pads instead of the events 2010-05-06 19:51:59 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink 2010-05-06 18:11:40 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpdtmfmux: Add priority sink pads 2010-05-07 17:15:47 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Cleanup event function 2010-05-07 16:42:22 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * tests/check/elements/rtpmux.c: rtpmux: Aggregate incoming segments 2010-05-06 19:09:48 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Update documentation 2010-05-06 18:10:45 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Simplify request pad creation 2010-03-21 21:39:18 +0100 Benjamin Otte * tests/check/elements/rtpmux.c: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple 2009-11-18 16:38:33 +0100 unknown * gst/rtpmanager/gstrtpmux.c: rtpmux: update the current_ssrc from the caps Fixes #604101 2009-12-09 14:42:21 +0100 Håvard Graff * gst/rtpmanager/gstrtpmux.c: rtpmux: release pads when disposing Because of an allocated priv (GstRTPMuxPadPrivate), the element will leak memory if not gst_rtp_mux_release_pad() is called. This would previously only happen if release_request_pad() was called explicitly, somthing that should not be neccesary. Fixes #604099 2009-12-09 13:40:43 +0100 Wim Taymans * gst/rtpmanager/gstrtpdtmfmux.c: dtmfmux: method name cleanups 2009-10-08 19:06:26 -0400 Olivier Crête * tests/check/elements/rtpmux.c: tests: Add test for rtpdtmfmux locking 2009-09-28 19:54:53 -0400 Olivier Crête * tests/check/elements/rtpmux.c: tests: Add unit test for rtpmux 2009-09-28 13:36:44 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Don't ignore requested pad name 2009-07-29 17:23:31 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Remove empty finalize 2009-07-21 15:31:33 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Free the pad private data on pad release Free the pad private data on pad release instead of using a weak ref, which is not thread safe. Also, lock the content of the pad private using the element's object lock. 2009-04-28 16:10:21 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Reject wrong caps 2009-04-28 16:03:19 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal 2009-04-28 15:58:41 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal 2009-04-22 18:01:07 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix warning 2009-04-20 20:00:15 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Set different caps depending on the input 2009-04-22 16:25:07 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Only free pad private when pad is disposed 2009-04-20 18:41:39 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Remove useless caps mangling 2009-04-20 18:36:42 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Rename variable for more clarity 2009-04-20 17:43:39 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: Use GST_BOILERPLATE 2009-04-20 17:42:40 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: * gst/rtpmanager/gstrtpmux.c: rtpmux: Do the includes locally 2009-04-15 13:23:01 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: Add GST_DEBUG_FUNCPTRs 2009-04-15 13:15:55 -0400 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Release locked pad on release_pad Release the special pad if the pad is removed from the muxer. 2009-04-15 13:09:27 -0400 Laurent Glayal * gst/rtpmanager/gstrtpdtmfmux.c: rtpdtmfmux: Release special on pad dispose Fixes #577690 2009-02-25 11:45:05 +0200 Stefan Kost * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: docs: various doc fixes No short-desc as we have them in the element details. Also keep things (Makefile.am and sections.txt) sorted. Reword ambigous returns. No text after since please. 2009-02-10 17:02:24 +0000 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmuxer.c: rtpmux: Move rtpmux from gst-plugins-farsight to -bad 2009-02-20 17:45:50 -0500 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * gst/rtpmanager/gstrtpmuxer.c: rtpmux: Re-indent to Gst style 2009-02-10 19:11:15 +0000 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Document rtp muxer a bit 2009-02-20 13:30:49 -0500 Laurent Glayal * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: rtpmux: Add signals before stream lock and after unlocking 2009-02-18 20:18:46 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Let ssrc through getcaps 2009-02-18 19:58:58 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Rename have_base to have_ts_base 2009-02-18 18:14:52 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Protect the seqnum with object lock in rtpmux 2009-02-18 18:07:44 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.h: rtpmux: Remove unused sink_ts_base 2009-02-18 15:20:58 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Have getcaps to force the same clockrate on all pads 2009-02-18 17:05:13 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Validate RTP data in RTP Mux 2009-02-18 14:16:00 -0500 Olivier Crête * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Remove unused clock-rate property 2009-02-18 13:56:36 -0500 Olivier Crête * gst/rtpmanager/gstrtpdtmfmux.h: rtpmux: Clarify locking in rtpdtmfmux 2009-02-18 13:32:56 -0500 Laurent Glayal * gst/rtpmanager/gstrtpmux.c: rtpmux: Missing format parameter 2008-12-01 17:55:22 -0500 Håvard Graff * gst/rtpmanager/gstrtpmux.c: rtpmux: Update seqnum base in rtp muxer With help from Wim 2008-12-01 17:54:58 -0500 Håvard Graff * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix some more leaks 2008-12-01 17:48:29 -0500 Håvard Graff * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: Fix leak 2008-09-29 15:03:05 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Don't unref caps we don't know (thanks Wim) 2008-08-12 12:48:02 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Put per-buffer debug at level LOG 2008-08-12 12:47:14 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Make debug print accurate 2008-08-12 12:46:23 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Set our caps on the buffers 2008-08-12 12:46:07 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Take the clock-base stored from the last setcaps 2008-08-12 12:41:59 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Store the clock-base on setcaps 2008-08-12 12:30:52 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Add padprivate to the request pads 2008-08-11 21:20:06 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Make indentation more correct 2008-08-11 21:05:34 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix typo 2008-08-11 21:03:22 -0400 Olivier Crête * gst/rtpmanager/gstrtpmux.c: rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer 2007-08-15 13:50:38 +0000 Zeeshan Ali * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: more debug 20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz 2007-08-20 18:50:32 +0000 Youness Alaoui * gst/rtpmanager/gstrtpmux.c: rtpmux: missing comment 20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz 2007-07-12 19:53:36 +0000 Olivier Crete * gst/rtpmanager/gstrtpmux.c: rtpmux: Make buffer writable before writing into it 20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz 2007-07-06 20:24:59 +0000 Olivier Crete * gst/rtpmanager/gstrtpmux.c: rtpmux: Set pads active when adding them to a potentially running element 20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz 2007-06-07 12:01:21 +0000 Olivier Crete * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix multiple ref leaks (patches by SP GLE) 20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz 2007-05-28 15:25:05 +0000 Zeeshan Ali * gst/rtpmanager/gstrtpmux.c: rtpmux: send event to all src pads 20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz 2007-05-28 12:37:49 +0000 Zeeshan Ali * gst/rtpmanager/gstrtpmux.c: rtpmux: print a warning if receive an error iterating sinkpads 20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz 2007-05-28 12:28:08 +0000 Zeeshan Ali * gst/rtpmanager/gstrtpmux.c: rtpmux: deal with all the gst_iterator_next() return values 20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz 2007-05-25 12:31:16 +0000 Zeeshan Ali * gst/rtpmanager/gstrtpmux.c: rtpmux: Return correct value from the event handler 20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz 2007-05-25 10:27:09 +0000 Zeeshan Ali * gst/rtpmanager/gstrtpmux.c: rtpmux: Ville's original patch to fix the traversal of dtmf event 20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz 2007-03-29 13:52:50 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Set the correct ts-offset on the get_prop value 20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz 2007-03-29 13:52:23 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Refactorize state_change 20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz 2007-03-29 13:36:22 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: set SSRC on the packets 20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz 2007-03-29 13:19:36 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Code clean-up and more debug output 20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz 2007-03-28 11:22:19 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Use own clock-base 20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz 2007-03-23 16:31:39 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Only accept RTP streams that have the same clock-rate 20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz 2007-03-22 16:15:52 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: Some more code-cleanups 20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz 2007-03-22 15:42:51 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: return newpad instead of NULL and warn if failed to create a pad 20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz 2007-03-22 12:41:32 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Refactorize the RTPMux code 20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz 2007-03-22 12:14:53 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: Some more doc fixing 20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz 2007-03-22 11:32:28 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: More Refactoring 20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz 2007-03-22 11:31:54 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: More documentation 20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz 2007-03-21 16:33:11 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpdtmfmux.c: rtpmux: Refactor the event handler function 20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz 2007-03-21 14:52:44 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpdtmfmux.c: * gst/rtpmanager/gstrtpdtmfmux.h: * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: * gst/rtpmanager/gstrtpmuxer.c: rtpmux: Add RTPDTMFMux element 20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz 2007-03-21 12:31:49 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: * gst/rtpmanager/gstrtpmux.h: rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz 2007-03-20 12:05:24 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Put more helpful description 20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz 2007-03-16 15:16:41 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: remove the (commented-out) code for blocking the pads 20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz 2007-03-16 13:14:44 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Drop buffers instead of blocking the sinkpads 20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz 2007-03-14 17:16:18 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Implement stream locking, needed for DTMF 20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz 2007-03-14 10:20:58 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: use GST_*_OBJECT instead of g_* 20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz 2007-03-14 10:18:54 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: No need to manage pads, parent does that for us 20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz 2007-03-14 09:03:58 +0000 zeenix@gmail.com * gst/rtpmanager/gstrtpmux.c: rtpmux: Fix copyright header 20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz 2007-03-07 08:53:07 +0000 zeeshan.ali@nokia.com * gst/rtpmanager/gstrtpmux.c: rtpmux: The first implementation of RTP muxer 20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz 2012-12-15 21:27:01 +0000 Tim-Philipp Müller * gst/audiofx/gstscaletempo.c: * gst/audiofx/gstscaletempo.h: scaletempo: no need for a private struct 2012-12-14 15:13:31 +0000 Tim-Philipp Müller * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: docs: update plugin docs 2012-12-14 15:13:19 +0000 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: docs: add scaletempo to docs 2012-11-06 13:36:39 +0000 Tim-Philipp Müller * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: audiofx: move scaletempo element from -bad https://bugzilla.gnome.org/show_bug.cgi?id=687262 2012-10-23 14:33:21 +0200 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: Fix event leak 2012-10-23 14:32:24 +0200 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: Fix timestamp tracking 2012-10-23 14:06:37 +0200 Sebastian Dröge * gst/audiofx/gstscaletempo.c: scaletempo: Implement LATENCY query 2012-10-23 13:39:17 +0200 Sebastian Dröge * gst/audiofx/gstscaletempo.c: * gst/audiofx/gstscaletempo.h: scaletempo: Store instance private data in the instance struct Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE() is really slow. 2012-10-17 17:34:26 +0100 Tim-Philipp Müller * gst/audiofx/gstscaletempo.c: scaletempo: use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there. 2012-09-14 17:08:49 +0200 Mark Nauwelaerts * gst/audiofx/gstscaletempo.c: scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 16:45:34 +0200 Wim Taymans * gst/audiofx/gstscaletempo.c: scaletempo: ffmpegcolorspace is no more 2012-04-05 18:02:56 +0200 Sebastian Dröge * gst/audiofx/gstscaletempoplugin.c: scaletempo: Update for GST_PLUGIN_DEFINE() API changes 2012-03-18 18:32:55 +0100 Mark Nauwelaerts * gst/audiofx/gstscaletempo.c: scaletempo: port to 0.11 2011-07-07 10:52:50 -0700 Stefan Kost * gst/audiofx/gstscaletempo.c: scaletempo: improve the docs Fix the syntax, add more explanation and xref the properties. 2011-03-22 13:46:42 +0100 Chris E Jones * gst/audiofx/gstscaletempo.c: scaletempo: Correctly handle newsegment events with stop==-1 Fixes bug #645420. 2010-10-19 13:43:14 +0300 Stefan Kost * gst/audiofx/gstscaletempo.c: scaletempo: add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/audiofx/gstscaletempo.c: scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple 2009-11-05 13:40:38 -0300 Thiago Santos * gst/audiofx/gstscaletempo.c: scaletempo: properly update new segments Scaletempo was missing an update of 'stop' in new segment parameters when pushing it downstream, which caused files to end earlier when rate < 1. Fixes #599903 Based on patch by: Bastian Hecht 2009-06-14 20:00:51 +0200 Maximilian Högner * gst/audiofx/gstscaletempo.c: scaletempo: Explicitely cast to signed integers to fix a segfault Fixes bug #585660. 2009-02-13 12:18:48 -0800 Michael Smith * gst/audiofx/gstscaletempo.c: scaletempo: Do not use void pointer arithmetic. 2008-10-30 12:13:18 +0000 Stefan Kost scaletempo: Return the result of parent_class->event() Original commit message from CVS: * gst/audiofx/gstscaletempo.c: Return the result of parent_class->event(). 2008-08-31 12:20:33 +0000 Rov Juvano Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r... Original commit message from CVS: Patch by: Rov Juvano * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-scaletempo.xml: * examples/scaletempo/Makefile.am: * examples/scaletempo/demo-gui.c: (pop_status_bar), (status_bar_printf), (demo_gui_seek_bar_format), (update_position), (demo_gui_seek_bar_change), (demo_gui_do_change_rate), (demo_gui_do_set_rate), (demo_gui_do_rate_entered), (demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled), (demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause), (demo_gui_do_play_pause), (demo_gui_do_open_file), (demo_gui_do_playlist_prev), (demo_gui_do_playlist_next), (demo_gui_do_about_dialog), (demo_gui_do_quit), (demo_gui_request_set_stride), (demo_gui_request_set_overlap), (demo_gui_request_set_search), (demo_gui_rate_changed), (demo_gui_playing_started), (demo_gui_playing_paused), (demo_gui_playing_ended), (demo_gui_player_errored), (demo_gui_stride_changed), (demo_gui_overlap_changed), (demo_gui_search_changed), (demo_gui_set_player_func), (demo_gui_set_playlist_func), (build_gvalue_array), (create_action), (demo_gui_show_func), (demo_gui_set_player), (demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property), (demo_gui_set_property), (demo_gui_init), (demo_gui_class_init), (demo_gui_get_type): * examples/scaletempo/demo-gui.h: * examples/scaletempo/demo-main.c: (handle_error_message), (handle_quit), (main): * examples/scaletempo/demo-player.c: (no_pipeline), (demo_player_event_listener), (demo_player_state_changed_cb), (demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate), (demo_player_scale_rate_func), (demo_player_set_rate_func), (_set_state_and_wait), (demo_player_load_uri_func), (demo_player_play_func), (demo_player_pause_func), (_seek_to), (demo_player_seek_by_func), (demo_player_seek_to_func), (demo_player_get_position_func), (demo_player_get_duration_func), (demo_player_scale_rate), (demo_player_set_rate), (demo_player_load_uri), (demo_player_play), (demo_player_pause), (demo_player_seek_by), (demo_player_seek_to), (demo_player_get_position), (demo_player_get_duration), (demo_player_get_property), (demo_player_set_property), (demo_player_init), (demo_player_class_init), (demo_player_get_type): * examples/scaletempo/demo-player.h: * gst/audiofx/Makefile.am: * gst/audiofx/gstscaletempo.c: (best_overlap_offset_float), (best_overlap_offset_s16), (output_overlap_float), (output_overlap_s16), (fill_queue), (reinit_buffers), (gst_scaletempo_transform), (gst_scaletempo_transform_size), (gst_scaletempo_sink_event), (gst_scaletempo_set_caps), (gst_scaletempo_get_property), (gst_scaletempo_set_property), (gst_scaletempo_base_init), (gst_scaletempo_class_init), (gst_scaletempo_init): * gst/audiofx/gstscaletempo.h: * gst/audiofx/gstscaletempoplugin.c: (plugin_init): Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a rate!=1.0. Integrate it into the docs and add the example application for it. Fixes bug #537700. 2012-12-13 12:36:20 +0100 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: check: add (but disable) more rtp jitterbuffer tests Tests need to be ported to 1.0 before they can be enabled but added here so they don't get forgotten. See https://bugzilla.gnome.org/show_bug.cgi?id=667838 2012-01-13 01:11:31 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: bundle together late lost-events The scenario where you have a gap in a steady flow of packets of say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer will idle up until it receives the first buffer after the gap, but will then go on to produce 499 lost-events, to "cover up" the gap. Now this is obviously wrong, since the last possible time for the earliest lost-events to be played out has obviously expired, but the fact that the jitterbuffer has a "length", represented with its own latency combined with the total latency downstream, allows for covering up at least some of this gap. So in the case of the "length" being 200ms, while having received packet 500, the jitterbuffer should still create a timeout for packet 491, which will have its time expire at 10,02 seconds, specially since it might actually arrive in time! But obviously, waiting for packet 100, that had its time expire at 2 seconds, (remembering that the current time is 10) is useless... The patch will create one "big" lost-event for the first 490 packets, and then go on to create single ones if they can reach their playout deadline. See https://bugzilla.gnome.org/show_bug.cgi?id=667838 2012-12-13 09:27:14 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix TCP reconnect Ignore other commands when reconnecting, otherwise the loop function would pause and the reconnection would not happen. Continue looping after doing a reconnect so that we have a chance to actually read the new data. 2012-12-13 01:02:34 +0400 Руслан Ижбулатов * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: * sys/waveform/gstwaveformsink.h: directsound, waveform: fix compilation errors caused by circular includes https://bugzilla.gnome.org/show_bug.cgi?id=690124 2012-12-12 17:35:04 +0000 Sebastian Dröge * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackutil.h: * ext/libpng/gstpngenc.c: * ext/pulse/pulseprobe.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * ext/vpx/gstvp8enc.c: * sys/oss/common.h: * sys/oss/gstossaudio.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-audio.h: ext/sys: Fix some compilation errors caused by circular includes 2012-12-12 12:07:34 +0100 Philippe Normand * gst/interleave/deinterleave.c: deinterleave: properly set srcpad channel position The src pad caps always describe a single audio channel so only the first position matters if deinterleave is configured to keep channel positions in its src pads. 2012-12-12 11:09:42 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: timeout on udpsrc is in nanoseconds 2012-12-12 11:08:13 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: improve timeouts Make it possible to set the timeout after we went to the READY state by using the timeout when checking the condition. This also makes it possible to set the timeout with a higher granularity than seconds. 2012-12-11 13:00:46 +0100 Wim Taymans * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: add support for strides Implement stride support correctly by taking it from the GstVideoFrame. Propose a bufferpool upstream when not operating in passthrough. 2012-09-27 12:17:58 -0700 Aleix Conchillo Flaque rtspsrc: do not change state to PLAYING if currently chaning state * gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be happening in the application thread, so we don't change the state to PLAYING in the gstrtspsrc thread unless it is safe. A specific case is when chaning the state to NULL from the application thread. This will synchronously try to stop the task (with the element state lock acquired), but we will try a gst_element_set_state from gstrtspsrc thread which will block on the element state lock causing a deadlock. https://bugzilla.gnome.org/show_bug.cgi?id=684312 2012-12-10 11:44:26 +0000 Alexey Chernov <4ernov@gmail.com> * sys/osxvideo/osxvideosink.m: osxvideosink: Fix resizing the Cocoa window on receiving new caps Fixes bug #689732. 2012-11-30 20:37:47 +0000 Tim-Philipp Müller * configure.ac: * sys/v4l2/Makefile.am: v4l2src: link against -lrt for clock_gettime() Need to explicitly link against -lrt for clock_gettime(), which we don't get in the libs any more, because core moved the gmodule-no-export-2.0 bit into Requires.Private. Not required for newer glibc, but for older ones, so check for that. 2012-11-30 17:22:59 +0000 Tim-Philipp Müller * ext/shout2/gstshout2.c: shout2send: accept audio/webm as well as video/webm https://bugzilla.gnome.org/show_bug.cgi?id=689336 2012-11-30 17:20:18 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: * tests/check/elements/matroskamux.c: webmux: fix linking with shout2send element Shout2send only accepts webm format, not matroska, but due to a bug in matroskamux, webmmux's source pad is also created with the matroska source pad template as pad template, which makes the link function think it can't link webmmux to shout2send. Also add unit test. https://bugzilla.gnome.org/show_bug.cgi?id=689336 2012-11-27 11:13:37 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use new option parser function 2012-11-26 15:17:13 +0000 Tim-Philipp Müller * gst/law/mulaw-conversion.c: law: fix accidental file permissions change https://bugzilla.gnome.org/show_bug.cgi?id=687469 2012-11-25 16:05:11 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2: remove unused define 2012-11-25 14:16:09 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: avoid criticals if unknown fourcc has space at beginning or end https://bugzilla.gnome.org/show_bug.cgi?id=682936 2012-11-24 19:32:51 +0000 Tim-Philipp Müller * gst/videobox/gstvideobox.c: videobox: fix border filling for planar YUV formats We would get a green border instead of a black one, for example. https://bugzilla.gnome.org/show_bug.cgi?id=684991 2012-11-24 14:27:33 +0000 Tim-Philipp Müller * gst/law/mulaw-conversion.c: mulaw: const-ify some arrays 2012-11-02 12:38:44 -0400 Roland Krikava * gst/law/mulaw-conversion.c: mulawdec: fix integer overrun There might be more than 65535 samples in a chunk of data. https://bugzilla.gnome.org/show_bug.cgi?id=687469 2012-11-22 11:34:31 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: pause the task instead of spinning Actually pause the loop task instead of spinning forever. 2012-11-19 03:31:37 -0500 Joshua M. Doe * gst/videofilter/gstvideoflip.c: videoflip: Add gray 8/16 support 2012-11-19 11:25:14 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From b497c4f to a72faea 2012-11-16 15:38:29 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle segment event Make a segment event when we send a new range header to a client (first PLAY request or after a seek). Send the segment event in interleaved mode. Clean the segment event on cleanup Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382 2012-11-16 15:18:07 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix check for active streams A stream can be active without a srcpad yet and we want to send events on those streams as well. 2012-11-16 13:31:04 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: create and add pads outside of lock Create and add the ghostpad for the new stream outside of the lock because it is not needed and causes deadlocks. 2012-09-12 22:11:20 -0700 Aleix Conchillo Flaque rtspsrc: allow client to disable reconnection * gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before, rtspsrc always tried to reconnect to the server when the RTSP connection was closed by the server. This property lets the user decide whether it wants rtspsrc to reconnect or not. https://bugzilla.gnome.org/show_bug.cgi?id=683912 2012-11-16 12:16:05 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: clear variables before retrying Else we might unref an old udpsrc twice in cleanup. 2012-11-16 12:00:14 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: propose ports in multicast When the user configured a port-range, propose ports from this range as the multicast ports. The server is free to ignore this request but if it honours it, increment our ports so that we suggest the next port pair for the next stream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420 2012-11-16 11:58:53 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add more debug 2012-11-16 09:09:38 +0000 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: post messages in max-size mode as well No reason not to really. 2012-11-15 14:37:44 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: post error before stopping 2012-11-14 00:13:36 +0000 Tim-Philipp Müller * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmparobustdepay.c: gst_adapter_prev_timestamp -> gst_adapter_prev_pts https://bugzilla.gnome.org/show_bug.cgi?id=675598 2012-11-12 19:23:41 +0100 Nicolas Dufresne * gst/videofilter/gstvideoflip.c: videoflip: Add NV12/NV21 support https://bugzilla.gnome.org/show_bug.cgi?id=688225 2012-11-12 13:01:23 +0100 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Don't leak GstVideoCodecFrames that cause the creation of invisible frames Fixes bug #682714. 2012-11-12 11:47:17 +0100 Sebastian Dröge * ext/pulse/pulsesink.c: pulse: Use new GType for GThread instead of just G_TYPE_POINTER 2012-11-12 11:14:34 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: protect against invalid RTP packets 2012-11-12 10:44:01 +0100 Sebastian Dröge * ext/libpng/gstpngdec.c: pngdec: Actually use the stop() vfunc implementation 2012-11-12 10:31:59 +0100 Sebastian Dröge * ext/vpx/gstvp8dec.c: vp8dec: Fix last commit 2012-11-12 10:10:15 +0100 Sebastian Dröge * ext/libpng/gstpngdec.c: pngdec: Keep the input state in reset() It's still valid after a flush and we might not get a new one. 2012-11-12 10:08:57 +0100 Sebastian Dröge * ext/vpx/gstvp8dec.c: vp8dec: Also destroy decoder in set_format() if it was created already Fixes a memory leak. 2012-11-12 09:48:45 +0100 Sebastian Dröge * ext/vpx/gstvp8dec.c: vp8dec: Don't clear input state in reset() The input state is still valid after flushing until new caps arrive. Fixes bug #688092. 2012-11-10 18:21:28 +0000 Tim-Philipp Müller * gst/videocrop/gstvideocrop.c: videocrop: add support for YV12 We can do I420, so we can do YV12 as well. 2012-11-10 12:39:08 +0100 Alessandro Decina * gst/multifile/gstmultifilesink.c: multifilesink: don't write stream headers with key-unit-event Don't write stream headers, let upstream elements insert them in the stream if all_headers=true is set in key unit events. 2012-11-09 13:27:16 +0100 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: Add NV12/NV21 support https://bugzilla.gnome.org/show_bug.cgi?id=687964 2012-11-09 16:31:05 +0100 Debarshi Ray * ext/vpx/gstvp8dec.c: vp8dec: Don't give up so easily if failed to decode a frame https://bugzilla.gnome.org/show_bug.cgi?id=687436 2012-11-09 11:22:30 +0100 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Also clear GError 2012-11-09 11:20:27 +0100 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets See bug #529454 and #687782 and commit 751f2bb3646f2beff3698c9f09900dbd0ea08abb 2012-11-07 20:35:50 +0000 Tim-Philipp Müller * configure.ac: configure.ac: update courtesy of autoupdate 2012-11-07 18:48:49 +0000 Tim-Philipp Müller * common: * configure.ac: configure: let AG_GST_PLUGIN_DOCS check for python And update common for move from AS_PATH_PYTHON to AM_PATH_PYTHON, which as a side-effect should pick up newer python versions as well. https://bugzilla.gnome.org/show_bug.cgi?id=563903 2012-11-07 13:36:33 +0100 Christian Fredrik Kalager Schaller * gst/rtp/Makefile.am: Fix vp8rtp header names in Makefile 2012-11-06 15:03:55 +0100 Nicolas Dufresne * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: * tests/check/elements/videocrop.c: videocrop: Add support for automatic cropping This change enable automatic cropping using -1 set to left, top, right or bottom property. In the case both side are set to automatic cropping, the croping will be done equally on both side (in the odd case, right and bottom cropping will be 1 pixel more). https://bugzilla.gnome.org/show_bug.cgi?id=687761 2012-11-02 16:39:28 +0100 Debarshi Ray * ext/speex/gstspeexdec.c: speexdec: Don't unmap or finish_frame an invalid GstBuffer https://bugzilla.gnome.org/show_bug.cgi?id=687464 2012-11-06 13:22:58 +0100 Marc Leeman * gst/rtsp/gstrtspsrc.c: rtsp: the RTCP port number is inclusive The configured port number pair has its upper bound set to the maximum allowed RTCP port, inclusive. See https://bugzilla.gnome.org/show_bug.cgi?id=639420 2012-11-03 20:38:00 +0000 Tim-Philipp Müller * tests/check/elements/mpg123audiodec.c: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-11-03 20:38:00 +0000 Tim-Philipp Müller * gst/audiofx/gststereo.c: * gst/audiofx/gststereo.h: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-11-03 20:38:00 +0000 Tim-Philipp Müller * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfdetect.h: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-11-04 00:07:18 +0000 Tim-Philipp Müller * ext/aalib/gstaasink.c: * ext/aalib/gstaasink.h: * ext/cairo/gstcairo.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: * ext/cairo/gstcairorender.c: * ext/cairo/gstcairorender.h: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cairo/gsttimeoverlay.h: * ext/dv/gstdv.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: * ext/dv/gstsmptetimecode.c: * ext/dv/gstsmptetimecode.h: * ext/flac/gstflac.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: * ext/flac/gstflactag.c: * ext/flac/gstflactag.h: * ext/gdk_pixbuf/gstgdkanimation.c: * ext/gdk_pixbuf/gstgdkanimation.h: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gdk_pixbuf/gstgdkpixbufdec.h: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: * ext/gdk_pixbuf/gstgdkpixbufplugin.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/gstgdkpixbufsink.h: * ext/gdk_pixbuf/pixbufscale.c: * ext/gdk_pixbuf/pixbufscale.h: * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudioclient.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: * ext/jack/gstjackringbuffer.h: * ext/jack/gstjackutil.c: * ext/jack/gstjackutil.h: * ext/jpeg/gstjpeg.c: * ext/jpeg/gstjpeg.h: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokedec.h: * ext/jpeg/gstsmokeenc.c: * ext/jpeg/gstsmokeenc.h: * ext/jpeg/smokecodec.c: * ext/jpeg/smokecodec.h: * ext/jpeg/smokeformat.h: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacasink.h: * ext/libpng/gstpng.c: * ext/libpng/gstpng.h: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: * ext/mikmod/README: * ext/mikmod/gstmikmod.c: * ext/mikmod/gstmikmod.h: * ext/mikmod/mikmod_types.c: * ext/mikmod/mikmod_types.h: * ext/pulse/plugin.c: * ext/pulse/pulseprobe.c: * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: * ext/raw1394/gst1394.c: * ext/raw1394/gst1394clock.c: * ext/raw1394/gst1394clock.h: * ext/raw1394/gst1394probe.c: * ext/raw1394/gst1394probe.h: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gstdv1394src.h: * ext/raw1394/gsthdv1394src.c: * ext/raw1394/gsthdv1394src.h: * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: * ext/soup/gstsouphttpclientsink.h: * ext/speex/gstspeex.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexdec.h: * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstapev2mux.h: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gstid3v2mux.h: * ext/taglib/gsttaglibplugin.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp8utils.c: * ext/vpx/gstvp8utils.h: * ext/vpx/plugin.c: * ext/wavpack/gstwavpack.c: * ext/wavpack/gstwavpackcommon.c: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackstreamreader.c: * ext/wavpack/gstwavpackstreamreader.h: * gst-libs/gst/gettext.h: * gst-libs/gst/glib-compat-private.h: * gst-libs/gst/gst-i18n-plugin.h: * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: * gst/apetag/gstapedemux.c: * gst/apetag/gstapedemux.h: * gst/audiofx/audioamplify.c: * gst/audiofx/audioamplify.h: * gst/audiofx/audiochebband.c: * gst/audiofx/audiochebband.h: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiocheblimit.h: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiodynamic.h: * gst/audiofx/audioecho.c: * gst/audiofx/audioecho.h: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofirfilter.h: * gst/audiofx/audiofx.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audiofxbaseiirfilter.h: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioiirfilter.h: * gst/audiofx/audioinvert.c: * gst/audiofx/audioinvert.h: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiokaraoke.h: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiopanorama.h: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: * gst/audiofx/audiowsinclimit.h: * gst/audiofx/math_compat.h: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstamrparse.h: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: * gst/audioparsers/gstwavpackparse.c: * gst/audioparsers/gstwavpackparse.h: * gst/audioparsers/plugin.c: * gst/auparse/gstauparse.c: * gst/auparse/gstauparse.h: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautoaudiosrc.h: * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautodetect.h: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosink.h: * gst/autodetect/gstautovideosrc.c: * gst/autodetect/gstautovideosrc.h: * gst/avi/avi-ids.h: * gst/avi/gstavi.c: * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/cutter/gstcutter.h: * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/cpureport.h: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapsdebug.h: * gst/debugutils/gstdebug.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavigationtest.h: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstnavseek.h: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gstpushfilesrc.h: * gst/debugutils/gsttaginject.c: * gst/debugutils/gsttaginject.h: * gst/debugutils/progressreport.c: * gst/debugutils/progressreport.h: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/debugutils/tests.c: * gst/debugutils/tests.h: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/greedyhmacros.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/plugins.h: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: * gst/deinterlace/tvtime/x86-64_macros.inc: * gst/effectv/gstaging.c: * gst/effectv/gstaging.h: * gst/effectv/gstdice.c: * gst/effectv/gstdice.h: * gst/effectv/gstedge.c: * gst/effectv/gstedge.h: * gst/effectv/gsteffectv.c: * gst/effectv/gsteffectv.h: * gst/effectv/gstop.c: * gst/effectv/gstop.h: * gst/effectv/gstquark.c: * gst/effectv/gstquark.h: * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: * gst/effectv/gstrev.c: * gst/effectv/gstrev.h: * gst/effectv/gstripple.c: * gst/effectv/gstripple.h: * gst/effectv/gstshagadelic.c: * gst/effectv/gstshagadelic.h: * gst/effectv/gststreak.c: * gst/effectv/gststreak.h: * gst/effectv/gstvertigo.c: * gst/effectv/gstvertigo.h: * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer.h: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer10bands.h: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizer3bands.h: * gst/equalizer/gstiirequalizernbands.c: * gst/equalizer/gstiirequalizernbands.h: * gst/flv/amfdefs.h: * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: * gst/flv/gstindex.c: * gst/flv/gstindex.h: * gst/flv/gstmemindex.c: * gst/flx/flx_color.c: * gst/flx/flx_color.h: * gst/flx/flx_fmt.h: * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: * gst/goom/config_param.c: * gst/goom/convolve_fx.c: * gst/goom/drawmethods.c: * gst/goom/drawmethods.h: * gst/goom/filters.c: * gst/goom/filters_mmx.s: * gst/goom/flying_stars_fx.c: * gst/goom/goom.h: * gst/goom/goom_config.h: * gst/goom/goom_config_param.h: * gst/goom/goom_core.c: * gst/goom/goom_filters.h: * gst/goom/goom_fx.h: * gst/goom/goom_graphic.h: * gst/goom/goom_plugin_info.h: * gst/goom/goom_tools.c: * gst/goom/goom_tools.h: * gst/goom/goom_typedefs.h: * gst/goom/goom_visual_fx.h: * gst/goom/graphic.c: * gst/goom/gstgoom.c: * gst/goom/gstgoom.h: * gst/goom/lines.c: * gst/goom/lines.h: * gst/goom/mathtools.c: * gst/goom/mathtools.h: * gst/goom/motif_goom1.h: * gst/goom/motif_goom2.h: * gst/goom/plugin_info.c: * gst/goom/ppc_drawings.h: * gst/goom/ppc_drawings.s: * gst/goom/ppc_zoom_ultimate.h: * gst/goom/ppc_zoom_ultimate.s: * gst/goom/sound_tester.c: * gst/goom/sound_tester.h: * gst/goom/surf3d.c: * gst/goom/surf3d.h: * gst/goom/tentacle3d.c: * gst/goom/tentacle3d.h: * gst/goom/v3d.c: * gst/goom/v3d.h: * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: * gst/icydemux/gsticydemux.c: * gst/icydemux/gsticydemux.h: * gst/id3demux/gstid3demux.c: * gst/id3demux/gstid3demux.h: * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: * gst/interleave/deinterleave.c: * gst/interleave/deinterleave.h: * gst/interleave/interleave.c: * gst/interleave/interleave.h: * gst/interleave/plugin.c: * gst/interleave/plugin.h: * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/atomsrecovery.c: * gst/isomp4/atomsrecovery.h: * gst/isomp4/descriptors.c: * gst/isomp4/descriptors.h: * gst/isomp4/fourcc.h: * gst/isomp4/ftypcc.h: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmoovrecover.h: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux-doc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/gstqtmuxmap.h: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/gstrtpxqtdepay.h: * gst/isomp4/isomp4-plugin.c: * gst/isomp4/properties.c: * gst/isomp4/properties.h: * gst/isomp4/qtatomparser.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_lang.c: * gst/isomp4/qtdemux_lang.h: * gst/isomp4/qtdemux_types.c: * gst/isomp4/qtdemux_types.h: * gst/isomp4/qtpalette.h: * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: * gst/law/alaw.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.c: * gst/law/mulaw-encode.h: * gst/law/mulaw.c: * gst/level/gstlevel.c: * gst/level/gstlevel.h: * gst/matroska/ebml-ids.h: * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: * gst/matroska/matroska.c: * gst/matroska/webm-mux.c: * gst/matroska/webm-mux.h: * gst/monoscope/convolve.c: * gst/monoscope/convolve.h: * gst/monoscope/gstmonoscope.c: * gst/monoscope/gstmonoscope.h: * gst/multifile/gstmultifile.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitfilesrc.h: * gst/multifile/patternspec.c: * gst/multifile/patternspec.h: * gst/multipart/multipart.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: * gst/rtp/fnv1hash.c: * gst/rtp/fnv1hash.h: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3depay.h: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpac3pay.h: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvdepay.h: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpbvpay.h: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpchannels.c: * gst/rtp/gstrtpchannels.h: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvdepay.h: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpdvpay.h: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722depay.h: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg722pay.h: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723depay.h: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpg729pay.h: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263depay.h: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcdepay.h: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegdepay.h: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpjpegpay.h: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp1sdepay.h: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tdepay.h: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp2tpay.h: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4adepay.h: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4apay.h: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmparobustdepay.h: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvdepay.h: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpmpvpay.h: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqcelpdepay.h: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpqdmdepay.h: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirendepay.h: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsirenpay.h: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtpsv3vdepay.h: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtspext.c: * gst/rtsp/gstrtspext.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: * gst/smpte/barboxwipes.c: * gst/smpte/gstmask.c: * gst/smpte/gstmask.h: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: * gst/smpte/gstsmptealpha.c: * gst/smpte/gstsmptealpha.h: * gst/smpte/paint.c: * gst/smpte/paint.h: * gst/smpte/plugin.c: * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudp.c: * gst/udp/gstudp.h: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstaspectratiocrop.h: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: * gst/videofilter/gstgamma.c: * gst/videofilter/gstgamma.h: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: * gst/videofilter/gstvideomedian.c: * gst/videofilter/gstvideomedian.h: * gst/videofilter/gstvideotemplate.c: * gst/videofilter/plugin.c: * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixer2pad.h: * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: * sys/directsound/gstdirectsoundplugin.c: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: * sys/oss/common.h: * sys/oss/gstossaudio.c: * sys/oss/gstossdmabuffer.c: * sys/oss/gstossdmabuffer.h: * sys/oss/gstosshelper.c: * sys/oss/gstosshelper.h: * sys/oss/gstosssink.c: * sys/oss/gstosssink.h: * sys/oss/gstosssrc.c: * sys/oss/gstosssrc.h: * sys/oss4/oss4-audio.c: * sys/oss4/oss4-audio.h: * sys/oss4/oss4-property-probe.c: * sys/oss4/oss4-property-probe.h: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-sink.h: * sys/oss4/oss4-source.c: * sys/oss4/oss4-source.h: * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudioelement.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudio.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixeroptions.c: * sys/sunaudio/gstsunaudiomixeroptions.h: * sys/sunaudio/gstsunaudiomixertrack.c: * sys/sunaudio/gstsunaudiomixertrack.h: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: * sys/sunaudio/gstsunaudiosrc.c: * sys/sunaudio/gstsunaudiosrc.h: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2radio.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2videooverlay.c: * sys/v4l2/gstv4l2videooverlay.h: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/gstv4l2vidorient.h: * sys/v4l2/tuner.c: * sys/v4l2/tuner.h: * sys/v4l2/tunerchannel.c: * sys/v4l2/tunerchannel.h: * sys/v4l2/tunernorm.c: * sys/v4l2/tunernorm.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: * sys/waveform/gstwaveformplugin.c: * sys/waveform/gstwaveformsink.c: * sys/waveform/gstwaveformsink.h: * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: * tests/check/elements/aacparse.c: * tests/check/elements/ac3parse.c: * tests/check/elements/alphacolor.c: * tests/check/elements/amrparse.c: * tests/check/elements/apev2mux.c: * tests/check/elements/aspectratiocrop.c: * tests/check/elements/audioamplify.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioecho.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiopanorama.c: * tests/check/elements/autodetect.c: * tests/check/elements/avimux.c: * tests/check/elements/avisubtitle.c: * tests/check/elements/capssetter.c: * tests/check/elements/deinterlace.c: * tests/check/elements/deinterleave.c: * tests/check/elements/flacparse.c: * tests/check/elements/flvdemux.c: * tests/check/elements/flvmux.c: * tests/check/elements/gdkpixbufsink.c: * tests/check/elements/icydemux.c: * tests/check/elements/id3demux.c: * tests/check/elements/id3v2mux.c: * tests/check/elements/imagefreeze.c: * tests/check/elements/interleave.c: * tests/check/elements/jpegdec.c: * tests/check/elements/jpegenc.c: * tests/check/elements/level.c: * tests/check/elements/matroskamux.c: * tests/check/elements/matroskaparse.c: * tests/check/elements/mpegaudioparse.c: * tests/check/elements/multifile.c: * tests/check/elements/parser.c: * tests/check/elements/parser.h: * tests/check/elements/qtmux.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpbin_buffer_list.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/shapewipe.c: * tests/check/elements/souphttpsrc.c: * tests/check/elements/spectrum.c: * tests/check/elements/sunaudio.c: * tests/check/elements/udpsink.c: * tests/check/elements/udpsrc.c: * tests/check/elements/videocrop.c: * tests/check/elements/videofilter.c: * tests/check/elements/vp8dec.c: * tests/check/elements/vp8enc.c: * tests/check/elements/wavpackdec.c: * tests/check/elements/wavpackenc.c: * tests/check/elements/wavpackparse.c: * tests/check/elements/y4menc.c: * tests/check/generic/states.c: * tests/check/pipelines/effectv.c: * tests/check/pipelines/flacdec.c: * tests/check/pipelines/simple-launch-lines.c: * tests/check/pipelines/tagschecking.c: * tests/check/pipelines/wavenc.c: * tests/check/pipelines/wavpack.c: * tests/examples/audiofx/firfilter-example.c: * tests/examples/audiofx/iirfilter-example.c: * tests/examples/cairo/cairo_overlay.c: * tests/examples/level/level-example.c: * tests/examples/pulse/pulse.c: * tests/examples/rtp/client-PCMA.c: * tests/examples/rtp/server-alsasrc-PCMA.c: * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: * tests/examples/spectrum/spectrum-example.c: * tests/examples/v4l2/camctrl.c: * tests/icles/equalizer-test.c: * tests/icles/gdkpixbufsink-test.c: * tests/icles/test-oss4.c: * tests/icles/v4l2src-test.c: * tests/icles/videobox-test.c: * tests/icles/videocrop-test.c: * tests/icles/videocrop2-test.c: * tests/icles/ximagesrc-test.c: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-11-03 20:40:37 +0000 Tim-Philipp Müller * ext/twolame/gsttwolamemp2enc.c: * ext/twolame/gsttwolamemp2enc.h: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-11-03 20:40:37 +0000 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: * ext/lame/gstlamemp3enc.h: * ext/lame/plugin.c: * tests/check/pipelines/lame.c: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520 2012-11-02 18:47:26 +0000 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: don't access rtp buffer after unmap Read the marker bit before we unmap the rtp packet. 2012-11-02 09:34:25 +0100 Sebastian Dröge * ext/vpx/gstvp8dec.c: vp8dec: Immediately return if opening the decoder failed Instead of ignoring any errors. 2012-11-01 22:02:39 +0100 Debarshi Ray * ext/vpx/gstvp8dec.c: vp8dec: Short circuit gst_vp8_dec_handle_frame if keyframe is missing https://bugzilla.gnome.org/show_bug.cgi?id=687376 2012-11-02 10:53:57 +1300 Douglas Bagnall * gst/videomixer/blend.c: videoconvert: Compare y offset with height, not width, when testing for overlap This could have prevented images showing that should have when the source height is greater than its width. When width exceeds height, as is common, it probably only caused a miniscule amount of unnecessary work. I haven't tested. 2012-11-01 21:09:56 +0000 Tim-Philipp Müller * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: rtpvp8: include config.h and minor style fixes 2012-11-01 20:13:43 +0000 Tim-Philipp Müller * gst/rtp/Makefile.am: rtp: fix tabs/space mess in Makefile.am 2012-11-01 20:05:49 +0000 Tim-Philipp Müller * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpvp8.c: rtp: move VP8 payloader and depayloader from -bad Spec is still in draft state, but should hopefully not change much now. Besides, we announce things as VP8-DRAFT-IETF-01 in our caps, so even if things change in incompatible ways it should not break anything. https://bugzilla.gnome.org/show_bug.cgi?id=687263 2012-10-17 17:34:26 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: rtpvp8: use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there. 2012-09-14 17:08:49 +0200 Mark Nauwelaerts * gst/rtp/gstrtpvp8pay.c: rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-04-05 18:02:56 +0200 Sebastian Dröge * gst/rtp/gstrtpvp8.c: rtpvp8: update for GST_PLUGIN_DEFINE() API changes 2012-03-28 12:49:54 +0200 Wim Taymans * gst/rtp/gstrtpvp8pay.c: rtpvp8: update for buffer changes 2012-03-01 14:59:55 -0300 Danilo Cesar Lemes de Paula * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: rtpvp8; fix compatibility with the third draft https://bugzilla.gnome.org/show_bug.cgi?id=671073 2012-01-25 16:20:41 +0100 Mark Nauwelaerts * gst/rtp/gstrtpvp8pay.c: rtpvp8: port some more to new memory API 2012-01-25 10:45:51 +0100 Olivier Crête * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: rtpvp8: port to 0.11 2011-10-03 12:06:27 +0200 Sebastian Dröge * gst/rtp/gstrtpvp8pay.c: rtpvp8pay: Fix typo 2011-09-23 22:58:30 +0000 Youness Alaoui * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: rtpvp8: Update the pay/depay to the ietf-draft-01 spec 2011-09-10 11:31:20 +0100 Vincent Penquerc'h * gst/rtp/dboolhuff.c: * gst/rtp/dboolhuff.h: * gst/rtp/gstrtpvp8pay.c: rtpvp8: fix bitstream parsing using the wrong kind of bitreader VP8 uses a probabilistic bool coder, not a straight bit coder. This fixes parsing when error-resilient is set. This commit includes a copy of libvpx's bool coder, BSD licensed. https://bugzilla.gnome.org/show_bug.cgi?id=652694 2011-07-12 18:03:53 -0400 Olivier Crête * gst/rtp/gstrtpvp8pay.c: rtpvp8: Reject unknown bitstream versions 2011-03-04 11:59:44 +0100 Edward Hervey * gst/rtp/gstrtpvp8pay.c: rtpvp8: Fix unitialized variable Makes macosx compiler happy. 2011-01-23 17:02:38 +0000 Sjoerd Simons * gst/rtp/gstrtpvp8depay.c: rtpvp8depay: Accept packets with only one byte of data When fragmenting partions it can happen that an RTP packet only caries 1 byte of RTP data. 2011-01-23 16:42:17 +0000 Sjoerd Simons * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: rtpvp8pay: Treat the frame header just like any other partition When setting up the initial mapping just act as if the global frame information is another partition. This saves special-casing it later in the actual packetizing code. 2010-05-16 17:23:17 +0100 Sjoerd Simons * gst/rtp/dboolhuff.LICENSE: * gst/rtp/gstrtpvp8.c: * gst/rtp/gstrtpvp8depay.c: * gst/rtp/gstrtpvp8depay.h: * gst/rtp/gstrtpvp8pay.c: * gst/rtp/gstrtpvp8pay.h: rtpvp8: Add simple payloaders and depayloaders for VP8 Minimal implementation of http://www.webmproject.org/code/specs/rtp/, version 0.3.2 2012-11-01 18:42:39 +0000 Wim Taymans * gst/rtp/gstrtpgstpay.c: gstpay: fix for 1.0 events Caps events are sometimes not followed by a buffer but by an event. Flush any pending caps before we make a packet with the event. Chain up to the parent event handler before we attempt to push RTP packets, it might be a segment event. 2012-11-01 18:42:24 +0000 Wim Taymans * gst/rtp/gstrtpgstdepay.c: gstdepay: fix small leak 2012-11-01 17:44:11 +0000 Wim Taymans * gst/rtp/gstrtpgstdepay.c: gstdepay: add support for events Conflicts: gst/rtp/gstrtpgstdepay.c 2012-11-01 17:40:31 +0000 Wim Taymans * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: rtpgstpay: add support for sending events We currently only send tags and custom events. The other events might interfere with the receiver timings or are otherwise handled by RTP. Conflicts: gst/rtp/gstrtpgstpay.c 2012-11-01 15:54:58 +0000 Wim Taymans * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: gstpay: rewrite payloader Use adapter to assemble the payload and make a flush function to turn this payload into (fragmented) packets. Conflicts: gst/rtp/gstrtpgstpay.c gst/rtp/gstrtpgstpay.h 2012-11-01 13:03:44 +0000 Douglas Bagnall * gst/videomixer/blend.c: videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH https://bugzilla.gnome.org/show_bug.cgi?id=687330 2012-11-01 13:02:16 +0000 Douglas Bagnall * gst/videobox/gstvideobox.c: videbox: fix border filling for gray formats Get the height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH. https://bugzilla.gnome.org/show_bug.cgi?id=687330 2012-11-01 11:58:57 +0000 Wim Taymans * gst/rtp/gstrtpgstdepay.c: gstdepay: check for correct fragment offset Make sure we only insert the rtp packet in the adapter when the frag_offset matches. When the first packet of a fragment is dropped, it avoids putting the remaining packets in the adapter and processing the partial fragment. Conflicts: gst/rtp/gstrtpgstdepay.c 2012-11-01 11:54:50 +0000 Wim Taymans * gst/rtp/gstrtpgstpay.c: gstpay: set C flag on all buffers of the fragment Set the C flags on all the fragments instead of only those with caps in them. This makes it easier in the receiver to check if there is a caps in the assembled fragments just by looking at the last RTP packet flags. 2012-11-01 10:55:03 +0000 Wim Taymans * gst/rtp/gstrtpgstdepay.c: gstdepay: use the capsversion Take the caps from the input caps and store it in the slot given by capsversion. 2012-11-01 10:52:25 +0000 Wim Taymans * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: gstpay: send caps inline Place the capsversion on the outgoing caps so that they end up in an SDP as well. Receivers need to know what capsversion a particular caps is for to be able to match the caps to the CV in the RTP packets. Place the caps inside the RTP packet whenever the caps change. Based on patch by Andrzej Bieniek Conflicts: gst/rtp/gstrtpgstpay.c gst/rtp/gstrtpgstpay.h 2012-10-31 16:17:48 +0000 Andrzej Bieniek * gst/rtp/gstrtpgstpay.c: gstpay: add debug Conflicts: gst/rtp/gstrtpgstpay.c 2012-10-31 16:09:26 +0000 Andrzej Bieniek * gst/rtp/gstrtpgstdepay.c: depay: correctly skip caps header size Conflicts: gst/rtp/gstrtpgstdepay.c 2012-09-28 00:43:38 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: matroskademux: put streamheaders on vorbis/speex/flac/theora caps to make remuxing work https://bugzilla.gnome.org/show_bug.cgi?id=640589 2012-10-28 00:07:46 +0100 Tim-Philipp Müller * ext/pulse/pulsesrc.c: pulsesrc: don't assert in get_time() when called after shutdown Which might happen if the source gets set to NULL state before the rest of the pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=686985 2012-10-30 11:10:49 +0000 Wim Taymans * tests/examples/level/level-example.c: tests: fix level example Use the GValueArray in the message. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=687154 2012-10-30 09:27:24 +0100 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: mpg123: removed unnecessary finalize function https://bugzilla.gnome.org/show_bug.cgi?id=687176 2012-10-30 10:20:09 +1100 Jan Schmidt * ext/mpg123/gstmpg123audiodec.c: mpg123: Fix leaks from not chaining up in the finalize function 2012-10-27 23:22:36 +0100 Tim-Philipp Müller * gst/auparse/Makefile.am: * gst/level/Makefile.am: * gst/y4m/Makefile.am: gst: fix variable order in some Makefile.am https://bugzilla.gnome.org/show_bug.cgi?id=687013 2012-10-27 17:27:16 -0400 Antoine Tremblay * ext/libcaca/Makefile.am: * gst/auparse/Makefile.am: * gst/level/Makefile.am: * gst/videocrop/Makefile.am: * gst/y4m/Makefile.am: gst: add various missing GST_PLUGINS_BASE_LIBS in Makefile.am Those plugins depend on either libgstaudio or libgstvideo, which are in gst-plugins-base. https://bugzilla.gnome.org/show_bug.cgi?id=687013 2012-10-27 13:24:24 +0100 Alexey Fisher * gst/matroska/matroska-demux.c: matroskademux: mark invisible VP8 frames with the DECODE_ONLY flag https://bugzilla.gnome.org/show_bug.cgi?id=654259 2012-10-26 10:55:28 +0100 Tim-Philipp Müller * tests/check/elements/multifile.c: tests: add multifilesrc test for fix in previous commit Make sure the stop-index set is honoured. https://bugzilla.gnome.org/show_bug.cgi?id=654853 2012-10-26 10:33:03 +0100 Stas Sergeev * gst/multifile/gstmultifilesrc.c: multifilesrc: fix stop index handling Make sure the stop index is always honoured. Avoids endless loop if one wants to read and output the same file N times, for example. https://bugzilla.gnome.org/show_bug.cgi?id=654853 2012-08-25 02:26:29 +0400 Руслан Ижбулатов * gst/matroska/matroska-read-common.c: matroskademux: Support recursive SimpleTags Fixes #682644 Depends on #682615 2012-08-24 13:55:41 +0400 Руслан Ижбулатов * gst/matroska/matroska-ids.h: * gst/matroska/matroska-read-common.c: matroskademux: Expand the tag mapping. * Also expose unknown tags as key=value pairs. * Arrange tag map in the same order tags are listed in Matroska spec, leaving unmapped tags as comments. * More specific TODOs. * Remove duplicate DATE define. Fixes #682615 Depends on #682524 2012-10-26 10:09:39 +0200 Sebastian Dröge * gst/matroska/matroska-read-common.c: matroskademux: Fix uninitialized variable compiler warning 2012-08-23 15:07:22 +0400 Руслан Ижбулатов * gst/matroska/matroska-ids.h: * gst/matroska/matroska-read-common.c: matroskademux: Matroska tag TargetType support * Reads TargetType and TargetTypeValue from a Tag. * After Tag is completely read, processes taglist, substituting some of the tags depending on target type value and the presence of video/subtitle streams. * Supports reading two new simpletags - PART_NUMBER and TOTAL_PARTS Depends on #682448 Fixes #682524 2012-08-22 15:32:41 +0400 Руслан Ижбулатов * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-read-common.c: matroskademux: Per-track tags for Matroska Requires Matroska file to have sane layout (track info before tag info). Uses replace-merge. Makes track UIDs 64-bit. Fixes #682448 2012-10-25 20:18:36 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesrc.c: multifilesrc: fix typo in property description 2012-10-25 12:18:03 -0700 Michael Smith * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: read video format header fully (so we can find 'pasp' atoms) for more fourccs. Fixes aspect ratio of prores files. 2012-10-25 00:44:34 -0300 Thiago Santos * gst/imagefreeze/gstimagefreeze.c: imagefreeze: the new get_caps already does the filter intersection It should be faster to pass the caps to intersect as the filter caps, rather than using NULL and intersecting 'manually' later. https://bugzilla.gnome.org/show_bug.cgi?id=686837 2012-10-25 00:43:51 -0300 Thiago Santos * gst/imagefreeze/gstimagefreeze.c: imagefreeze: avoid assertion when using accept caps query This query must receive a fixed caps, so imagefreeze should fixate its framerate before sending the query downstream. https://bugzilla.gnome.org/show_bug.cgi?id=686837 2012-10-25 12:33:24 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to feature development === release 1.0.2 === 2012-10-25 01:01:09 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.0.2 2012-10-24 13:41:00 +0100 Tim-Philipp Müller * tests/check/elements/mpg123audiodec.c: tests: fix up mpg123 test a little - dist input files - fix sample leak - simplify check for elements - only run mpg123 test if mpg123 is available and selected - fix build in uninstalled setup https://bugzilla.gnome.org/show_bug.cgi?id=686595 2012-10-24 12:30:10 +0200 Carlos Rafael Giani * tests/check/elements/mpg123audiodec.c: tets: add unit test for mpg123audiodec https://bugzilla.gnome.org/show_bug.cgi?id=686595 2012-10-24 00:36:42 +0200 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: mpg123: added gtkdoc section https://bugzilla.gnome.org/show_bug.cgi?id=686595 2012-10-24 00:22:05 +0200 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: mpg123: fixed bug with last frame, disabled internal resampler & chatter * The last MP3 frame wasn't being pushed when base class was draining * Made sure mpg123 cannot ever use its (crude) internal resampler * Disabled mpg123 stderr output https://bugzilla.gnome.org/show_bug.cgi?id=686595 2012-10-24 13:50:00 +0200 Arnaud Vrac * gst/isomp4/qtdemux.c: qtdemux: use correct type for channel-mask bitmask Fixes crash on 32-bit systems. 2012-10-24 00:21:45 +0200 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: mpg123: cleaned up comments, formatting, and logging lines also replaced mpg123decoder->handle != NULL checks with asserts https://bugzilla.gnome.org/show_bug.cgi?id=686595 2012-10-24 11:17:55 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Flush the ringbuffer on GAP events without duration This is required to properly start the ringbuffer and clock. 2012-10-02 20:51:29 +0200 Oleksij Rempel * ext/vpx/gstvp8enc.c: vp8enc: set DECODE_ONLY flag on invisible AltRef frames https://bugzilla.gnome.org/show_bug.cgi?id=654216 2012-10-23 16:02:05 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: fix coverart extraction if vorbis comments come after picture header See sample file for bug #684701. 2012-10-23 13:45:17 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: ignore bad headers if we have a valid STREAMINFO header If we run into any header parsing issues and we have a valid STREAMINFO header already, don't error out, but just stop header parsing and try to find some audio frames. https://bugzilla.gnome.org/show_bug.cgi?id=684701 2012-10-23 13:43:10 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: post proper error message and fix buffer leak on header parsing error https://bugzilla.gnome.org/show_bug.cgi?id=684701 2012-10-22 22:32:49 -0700 Michael Smith * gst/isomp4/qtdemux.c: qtdemux: with raw audio, set a default channel-mask for multichannel audio. This doesn't actually parse 'chan' because it's absurdly complex. 2012-10-22 15:54:17 +0200 Sebastian Rasmussen * gst/udp/gstudpsrc.c: updsrc: fix typo causing compilation error gstudpsrc.c: In function 'gst_udpsrc_create': gstudpsrc.c:365: error: 'ret' may be used uninitialized in this function https://bugzilla.gnome.org/show_bug.cgi?id=686642 2012-10-22 11:55:59 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi_ fix invert function Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686550 2012-10-22 11:55:22 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: fix debug 2012-10-22 11:39:37 +0200 Wim Taymans * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: add support for 'generic' samples Add support for stuffing a complete stream into 1 sample. See https://bugzilla.gnome.org/show_bug.cgi?id=686550 2012-10-20 13:01:41 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: remove superfluous g_type_init() call It's deprecated in newer GLib and not needed here. https://bugzilla.gnome.org/show_bug.cgi?id=686456 2012-10-20 11:32:27 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: fix caps leak in acceptcaps function 2012-10-19 19:24:23 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: don't leak gst_riff_strf_auds in case of MS/RIFF audio https://bugzilla.gnome.org/show_bug.cgi?id=681192 2012-10-18 22:20:39 +0200 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: unsigned subtitle template 2012-10-18 11:32:10 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: in accept_caps() check if ring buffer is NULL before de-referencing And sprinkle some thread-safety (take object lock for accessing ring buffer, and pa main loop lock for the context). https://bugzilla.gnome.org/show_bug.cgi?id=683782 2012-09-13 00:10:00 +0000 Youness Alaoui * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer2: Fix race condition where a src setcaps is ignored If both pads receive data at the same time, they will both get their sink_setcaps called which will call the src_setcaps, but there is a race condition where the second one might not be called. Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=683842 2011-10-31 15:43:25 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: do not use unoffical V_MJPEG codec id Since it's not spec'ed, consider it a VfW compatibility case. Many applications (e.g. avidemux) don't understand the unofficial V_MJPEG id. Fixes #659837. Conflicts: gst/matroska/matroska-mux.c 2012-10-17 17:34:26 +0100 Tim-Philipp Müller * gst/audiofx/gststereo.c: Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there. 2012-10-17 17:34:26 +0100 Tim-Philipp Müller * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there. 2012-10-17 17:03:39 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8enc.c: jpeg, png, vpx: use gst_element_class_set_static_metadata() Avoids some string copies. 2012-10-17 14:23:01 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: jpegdepay: store quant tables in zigzag order 2012-10-17 13:55:45 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtsession: fix compiler warning 2012-10-17 13:35:07 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: clarify the ntp-sync option 2012-10-17 13:15:48 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: update caps in the source Inform the source when caps changed. This was removed in the port to 1.0 leaving the source unaware of the clock-rate and unable to interpollate rtp timestamps for SR packets. 2012-10-17 12:46:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: rtpbin: set PTS and DTS in jitterbufffer 2012-10-17 12:24:22 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: disable check for ntp-sync Disable the check for the ntp-sync method. It is expected that a rather larger offset needs to be applied with this method. 2012-10-17 12:17:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: rtpbin: use running-time for NTP time When use-pipeline-clock is set, use the running-time of the pipeline to calculate the NTP timestamps. This method would previously only work when the base-time is set to 0 but with this change it can also work with different offsets and we can also implement pause/resume of the sender and receiver now. 2012-10-17 10:20:12 +0200 Wim Taymans * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: port to videofilter 2012-10-17 09:36:50 +0200 Wim Taymans * gst/videobox/gstvideobox.c: videobox: use out_info for out properties 2012-10-16 14:40:19 +0200 Wim Taymans * gst/videofilter/gstvideomedian.c: * gst/videofilter/gstvideomedian.h: median: small cleanups 2012-10-16 13:56:19 +0200 Wim Taymans * Makefile.am: * gst/median/.gitignore: * gst/median/Makefile.am: * gst/median/gstmedian.c: * gst/median/gstmedian.h: * gst/median/median.vcproj: median: remove now that it is in videofilter 2012-10-16 13:49:11 +0200 Wim Taymans * configure.ac: configure: remove median from build 2012-10-16 13:47:24 +0200 Wim Taymans * gst/videofilter/Makefile.am: * gst/videofilter/gstvideomedian.c: * gst/videofilter/gstvideomedian.h: * gst/videofilter/plugin.c: videomedian: copy media to videomedian Copy the median video filter to videofilters and rename to videomedian. 2012-10-16 13:12:21 +0200 Wim Taymans * configure.ac: * gst/median/Makefile.am: * gst/median/gstmedian.c: * gst/median/gstmedian.h: media: port to 1.0 2012-10-16 01:02:11 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: append palette data to paletted 8-bit RGB frames Fixes playback of 8-bit indexed RGB videos, with fixes in -base. https://bugzilla.gnome.org/show_bug.cgi?id=686046 2012-10-15 15:36:04 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: And this time fix the default target-bitrate value for real 2012-10-15 15:30:33 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Fix default target-bitrate value 2012-10-13 00:03:29 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: don't assert if upstream size is not available when guessing bitrates Fixes abort in push mode where the source is not seekable and the size of the file is not available, as with cat foo.mp4 | gst-launch-1.0 playbin uri=fd://0 Less noticable with releases, since we disable all g_assert() there. https://bugzilla.gnome.org/show_bug.cgi?id=686008 2012-10-12 14:38:33 -0700 Michael Smith * gst/isomp4/qtdemux.h: qtdemux: allow more streams. Bump this constant to 32, which should be enough for real-world files. 2012-10-12 14:35:24 -0700 Michael Smith * gst/isomp4/qtdemux.c: qtdemux: support more different fourcc values for other ProRes variants. 2012-10-11 22:36:21 +0100 Tim-Philipp Müller * tests/examples/rtp/client-H263p-AMR.sh: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H263p.sh: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-H264.sh: * tests/examples/rtp/client-PCMA.c: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh: * tests/examples/rtp/server-VTS-H263p.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-decodebin-H263p-AMR.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: examples: update some element names for 1.0 in RTP examples gstrtpbin -> rtpbin ffdec_* -> avdec_* ffenc_* -> avenc_* 2012-10-10 12:05:34 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: remove unused include 2012-10-10 10:55:28 +0200 Rasmus Rohde * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: add multicast-iface property udpsrc already has support for setting the multicast interface, which is useful for multi-homed machines. This patch adds the same code to the multiudpsink. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685864 2012-10-10 11:32:17 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: multiudpsink: don't error on send errors but only warn Don't error on send errors but simply post a warning, it's possible that the next packet will be fine. 2012-10-10 10:28:24 +0200 Rasmus Rohde * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: add force-ipv4 option Add an option to the multiudpsink that makes it possible to force the use of an IPv4 socket. This can e.g. be used to handle the issue described in https://bugzilla.gnome.org/show_bug.cgi?id=682481 2012-10-10 10:18:52 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: remove unused field 2012-10-10 10:10:26 +0200 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: use negotiated allocator or pool Use the base class to allocate a buffer for us because it knows how to use the negotiated allocator or bufferpool. 2012-10-10 10:09:37 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: multiudpsink: post error when something goes wrong 2012-10-10 10:09:10 +0200 Wim Taymans * gst/spectrum/gstspectrum.c: spectrum: elements post element messages 2012-10-07 16:56:38 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development (bug fixing) === release 1.0.1 === 2012-10-07 15:31:12 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.0.1 2012-10-06 14:57:10 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 6c0b52c to 6bb6951 2012-10-05 15:12:27 -0700 Michael Smith * gst/interleave/deinterleave.c: deinterleave: output channels should be marked as MONO, not FRONT_LEFT, if we're not preserving input channel positions. 2012-10-04 15:13:20 -0700 Michael Smith * gst/interleave/interleave.c: interleave: use gst_audio_channel_positions_to_mask instead of a local copy of half of it. Handles some values more correctly. 2012-10-04 20:32:45 +0200 Rasmus Rohde * gst/rtp/gstrtpgstdepay.c: gstrtpdepay: don't leak input buffer The rtp buffer is never unmapped in the normal code exit path of gst_rtp_gst_depay_process(..) resulting in a memory leak. https://bugzilla.gnome.org/show_bug.cgi?id=685512 2012-10-04 18:37:18 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Add support for NV12 and NV21 2012-10-01 15:11:05 +0200 Patricia Muscalu * gst/rtp/gstrtph264pay.c: * tests/check/elements/rtp-payloading.c: rtph264pay: do not push unmapped data Also do not use a GstBuffer after it has been pushed into the adapter. https://bugzilla.gnome.org/show_bug.cgi?id=685213 2012-10-03 10:51:45 -0700 Michael Smith * gst/interleave/deinterleave.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/ximageutil.c: meta info: threadsafe registration using g_once 2012-10-01 15:44:01 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode; handle some initial junk before hdrl list Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685059 2012-10-01 14:03:19 +0100 Tim-Philipp Müller * tests/icles/gdkpixbufsink-test.c: tests: port gdkpixbufsink test 2012-09-29 11:59:31 +0100 Tim-Philipp Müller * gst/level/gstlevel.c: * tests/check/elements/videocrop.c: Purge references to liboil https://bugzilla.gnome.org/show_bug.cgi?id=673285 2012-09-28 16:51:01 +0200 Mark Nauwelaerts * gst/avi/avi-ids.h: * gst/avi/gstavidemux.c: avidemux: recognize all xsub frames as keyframes Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977 2012-09-28 16:50:25 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode: find the correct chunk for segment following seek Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977 2012-09-27 22:17:49 +0100 Arnaud Vrac * gst/isomp4/qtdemux.h: qtdemux: fix parsing in push mode when moov atom is at the end When playing an mp4 file with the MOOV atom at the end of the file, playback fails with the error message "no 'moov' atom within the first 10 MB". This is due to a mistake in the upstream_size typing, making the seek to the end of file never happening. https://bugzilla.gnome.org/show_bug.cgi?id=684972 2012-09-27 15:50:49 -0300 Andre Moreira Magalhaes (andrunko) * gst/videofilter/gstgamma.c: gamma: remove duplicate entries at format at caps Avoids extra caps/structures processing 2012-09-27 14:13:42 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: negotiate pool with srcpad caps 2012-09-27 11:02:51 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: The convert and duration queries are not supposed to change the format 2012-09-26 09:28:59 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: videomixer: clear video frame more correctly Make sure not to touch memory that doesn't belong to our frame, we might be one part of a side-by-side 3D frame, or in a picture-in-picture scenario. 2012-09-26 00:44:59 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: minor clean-up Use GstByteWriter, because we can, and g_value_take_boxed. 2012-09-10 10:27:28 +0400 Dmitriy Samonenko * gst/flv/gstflvdemux.c: flvdemux: fix speex audio decoding by creating fake stream header https://bugzilla.gnome.org/show_bug.cgi?id=683622 2012-09-25 21:21:15 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: * tests/check/pipelines/simple-launch-lines.c: videomixer: fix warnings when using transparent background gst_video_frame_map() increases the refcount, which makes the buffer not writable any more technically, so calling gst_buffer_memset() on it will cause nasty warnings. Unit test disabled because it very rarely (for me) fails, possibly negotiation-related. https://bugzilla.gnome.org/show_bug.cgi?id=684398 2012-09-25 10:43:28 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Add some useful debug logging 2012-09-25 10:41:44 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix telecine This only affects behaviour in telecine cases with pattern locking enabled. The default case should be untouched. This works with the output from fieldanalysis at least, but the field order looks swapped for telecine mixed buffers with the David_slides_Schleef clip. 2012-09-25 14:43:15 +0200 Edward Hervey * ext/vpx/gstvp8enc.c: vp8enc: Disable GLIB deprecation warnings GValueArray has been deprecated since 2.32 ... but there's no usable replacement for it. See https://bugzilla.gnome.org/show_bug.cgi?id=667228 2012-09-25 14:18:35 +0200 Edward Hervey * gst/videomixer/videomixer2.c: videomixer: Fix leak 2012-09-24 16:46:18 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development (bug fixing) === release 1.0.0 === 2012-09-24 14:06:42 +0100 Tim-Philipp Müller * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 1.0.0 2012-09-24 11:56:56 +0100 Tim-Philipp Müller * tests/check/elements/rganalysis.c: tests: remove g_printerr() that's not needed any longer now that tcase_skip_broken_test() prints it as well. 2012-09-23 19:50:42 +0100 Tim-Philipp Müller * tests/check/elements/rganalysis.c: tests: disable failing replaygain tests 2012-09-23 16:31:37 +0100 Tim-Philipp Müller * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: smpte: send stream-start event 2012-09-23 16:10:36 +0100 Tim-Philipp Müller * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: multipartmux: send stream-start event 2012-09-23 16:02:19 +0100 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: send stream-start 2012-09-23 15:57:35 +0100 Tim-Philipp Müller * gst/isomp4/gstqtmux.c: qtmux: send stream-start event 2012-09-23 15:48:54 +0100 Tim-Philipp Müller * gst/interleave/interleave.c: * gst/interleave/interleave.h: interleave: add a bunch of FIXMEs Needs some more work, so stream-start, caps and tags are sent in the right order. 2012-09-23 15:18:54 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: send stream-start event 2012-09-23 15:16:14 +0100 Tim-Philipp Müller * gst/avi/gstavimux.c: avimux: send stream-start event 2012-09-22 15:00:27 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfdepay.c: rtpdtmfdepay: Use 1.0-style caps negotiation and audio/x-raw 2012-09-22 16:08:05 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 4f962f7 to 6c0b52c 2012-09-21 21:54:36 +0100 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: answer URI query Without this, something also answered the query with TRUE but without setting a uri, not sure what that was.. 2012-09-20 17:28:47 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: Make sure the caps don't have duplicated sps/pps 2012-09-20 19:58:12 +0200 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Mute stream post-connection if required A bug in PulseAudio causes PA_STREAM_START_MUTED to be rejected on record streams. Until this is fixed upstream, we mute the stream manually at startup. Based on a patch by Alban Browaeys . https://bugzilla.gnome.org/show_bug.cgi?id=684469 2012-09-20 18:00:59 -0700 Michael Smith * gst/isomp4/qtdemux.c: qtdemux: 24 bit audio here is S24LE, not S24_3LE. 2012-09-20 10:07:24 +0200 Sjoerd Simons * sys/v4l2/gstv4l2src.c: v4l2src: handle latency query before setting up the bufferpool Fixes crash if no bufferpool is set up yet. https://bugzilla.gnome.org/show_bug.cgi?id=684430 2012-09-19 09:17:03 +0530 Arun Raghavan * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: Specify endianness in IEC 61937 payloading Corresponds to an API change in gst-plugins-base. This needs to be fixed to query the expected byte order using appropriate API. https://bugzilla.gnome.org/show_bug.cgi?id=678021 2012-09-19 09:15:53 +0530 Arun Raghavan * sys/directsound/gstdirectsoundsink.c: directsoundsink: Specify endianness in IEC 61937 payloading DirectSound expects native endian byte order. https://bugzilla.gnome.org/show_bug.cgi?id=678021 2012-09-19 09:13:11 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Specify endianness in IEC 61937 payloading Corresponds to an API change in gst-plugins-base. https://bugzilla.gnome.org/show_bug.cgi?id=678021 2012-09-19 00:39:01 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Remove incorrect logic I don't understand why these lines were added, they don't make sense to me now and both David and I agree that removing them moves closer to related logic being correct, therefore, they're being removed. I've tested a few progressive, interlaced and telecine clips and they all behave properly timestamp-wise and visually after these changes. 2012-09-19 00:17:49 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix field duration The frame rate fraction is correctly adjusted in the cases preceding the field duration calculation and so the factor of 2 is incorrect. 2012-09-18 10:34:03 -0700 Michael Smith * gst/videobox/gstvideobox.c: videobox: Fix U/V strides for a number of cases. 2012-09-18 12:13:21 +0200 Mark Nauwelaerts * gst/videomixer/videomixer2.c: videomixer: init videoinfo ... to prevent random bogus caps fields. 2012-09-18 12:12:39 +0200 Mark Nauwelaerts * gst/videomixer/videomixer2.c: videomixer: chain up to collectpads query function 2012-09-17 13:17:00 -0400 Nicolas Dufresne * gst/videomixer/videomixer2.c: videomixer: Don't let GstCollectPad shadow custom sink pad query func In the current implementation, the custom pad query function is not called. This patch, set that query function on the GstCollectPads to avoid this shadowing. See https://bugzilla.gnome.org/show_bug.cgi?id=684237 2012-09-17 18:23:11 +0100 Tim-Philipp Müller * tests/files/Makefile.am: tests: dist image.jpg for jpeg test === release 0.11.99 === 2012-09-17 17:57:58 +0100 Tim-Philipp Müller * configure.ac: * gst-plugins-good.doap: * win32/common/config.h: Release 0.11.99 2012-09-17 16:57:30 +0100 Tim-Philipp Müller * ext/twolame/Makefile.am: Remove -DGST_USE_UNSTABLE_API 2012-09-17 16:57:30 +0100 Tim-Philipp Müller * ext/lame/Makefile.am: Remove -DGST_USE_UNSTABLE_API 2012-09-17 16:53:04 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.types: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update 2012-09-17 13:30:15 +0200 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Fix spec file for vp8 move 2012-09-17 13:23:36 +0200 Sebastian Dröge * Makefile.am: annodex: Add to the CRUFT_DIRS 2012-09-17 12:14:07 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-monoscope.xml: docs: update 2012-09-17 09:48:56 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Correctly finish frames Previously we would always get the same frame if multiple frames are pending, leaking memory of the previous frames and breaking timestamps. 2012-09-17 09:40:41 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Allow changing bitrate and other parameters during playback Fixes bug #648276. 2012-09-17 09:16:39 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: vp8enc: Store configuration in the vpx_codec_enc_cfg_t struct instead of duplicating all variables Also protect encoder with a mutex. 2012-09-16 16:03:06 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Update documentation to reflect new property names ...and also link to the WebM encoder parameters website. 2012-09-16 15:57:58 +0200 Sebastian Dröge * ext/vpx/gstvp8enc.c: vp8enc: Make some property names more readable 2012-09-16 15:47:16 +0200 Sebastian Dröge * tests/check/elements/.gitignore: vp8: Add tests to .gitignore 2012-09-16 15:46:31 +0200 Sebastian Dröge * tests/check/elements/vp8enc.c: vp8enc: Update patch to the new property names 2012-09-16 15:46:22 +0200 Sebastian Dröge * tests/check/Makefile.am: vpx: Integrate test into the build system too 2012-02-07 17:00:26 +0100 Wim Taymans * tests/check/elements/vp8dec.c: * tests/check/elements/vp8enc.c: [MOVED FROM BAD 6/6] tests: fix more unit tests 2011-11-24 21:42:39 +0100 René Stadler * tests/check/elements/vp8dec.c: * tests/check/elements/vp8enc.c: [MOVED FROM BAD 5/6] tests: update for gstcheck API change 2010-07-10 15:46:51 +0200 Sebastian Dröge * tests/check/elements/vp8dec.c: [MOVED FROM BAD 4/6] vp8dec: Add simple unit test for vp8dec 2010-07-10 15:46:43 +0200 Sebastian Dröge * tests/check/elements/vp8enc.c: [MOVED FROM BAD 3/6] vp8enc: Improve unit test a bit 2010-07-10 15:32:29 +0200 Sebastian Dröge * tests/check/elements/vp8enc.c: [MOVED FROM BAD 2/6] vp8enc: Also check the output caps in the unit test 2010-07-10 15:29:46 +0200 Sebastian Dröge * tests/check/elements/vp8enc.c: [MOVED FROM BAD 1/6] vp8enc: Add simple unit test 2012-09-16 15:43:39 +0200 Sebastian Dröge * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-vpx.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * ext/Makefile.am: vpx: Integrate into the build system 2012-09-16 15:33:57 +0200 Sebastian Dröge * ext/vpx/GstVP8Enc.prs: * ext/vpx/Makefile.am: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp8enc.c: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp8utils.c: * ext/vpx/gstvp8utils.h: * ext/vpx/plugin.c: vpx: Rename vp8 plugin to vpx This is using libvpx, which can support more codecs than just VP8 and will likely support future codecs. 2012-09-16 15:32:24 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: vp8: Apply remaining changes that got lost while moving the plugin via git am thanks to merges 2012-09-16 15:25:08 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 134/134] vp8dec: Unref input/output states when stopping the decoder 2012-09-16 15:18:20 +0200 Sebastian Dröge * ext/vp8/GstVP8Enc.prs: [MOVED FROM BAD 133/134] vp8enc: Update realtime profile to the new properties 2012-09-16 10:56:07 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 132/134] vp8: Require latest libvpx release (1.1.0 from May 2012) Fixes bug #684116 and simplifies configure checks. 2012-09-15 20:23:13 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 131/134] vp8enc: Use a string field for the profile in the caps Just for consistency with all the other codecs. 2012-09-15 00:04:07 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 130/134] vp8enc: Correctly set profile in caps 2012-09-14 23:41:48 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 129/134] vp8: Update copyright and authors 2012-09-08 15:38:40 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 128/134] vp8enc: Rework encoder properties to be more in line with the libvpx tools and API Also add all available properties. 2012-09-14 17:08:49 +0200 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 127/134] replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-07-19 09:05:28 +0200 Edward Hervey * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 126/134] vp8dec: Call gst_video_decoder_negotiate() 2012-08-14 11:17:25 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8dec.h: [MOVED FROM BAD 125/134] vp8dec: Add support for multiple decoding threads 2012-08-14 11:09:46 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 124/134] vp8dec: Add support for the MFQE postprocessing flag Which is enabled by default if postprocessing is enabled. 2012-08-09 13:37:22 +0200 Sebastian Dröge * ext/vp8/Makefile.am: [MOVED FROM BAD 123/134] vp8: Use pkg-config file for getting the LIBS and CFLAGS 2012-08-08 17:06:20 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 122/134] vp8enc: Update the per-component strides for every frame too This is necessary because of GstVideoAlignment 2012-07-26 19:31:14 +0200 Oleksij Rempel * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 121/134] vp8enc: initiate encoder to fix a crash. Without this patch vp8enc send header before and after first key frame. On second keyframe vp8dec will crash without getting decoded frame. With this pipe it is easy to reproduce this issue: gst-launch-1.0 videotestsrc ! vp8enc ! vp8dec ! fakesink https://bugzilla.gnome.org/show_bug.cgi?id=680667 2012-07-28 00:32:58 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 120/134] tag: Update for taglist/tag event API changes 2012-07-23 10:35:03 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 119/134] ext: Update for video base classes API changes 2012-07-21 19:59:21 +0200 Oleksij Rempel * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 118/134] vp8enc: fix memory leak unref frame. i hope it is correct place to do it. Signed-off-by: Oleksij Rempel 2012-07-06 11:50:53 +0200 Wim Taymans * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 117/134] update for query api changes 2012-07-06 11:26:55 +0200 Wim Taymans * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 116/134] update for query api changes 2012-07-06 11:03:04 +0200 Wim Taymans * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 115/134] update for allocation query changes 2012-06-07 12:33:31 +0100 Vincent Penquerc'h * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 114/134] vp8: fix codec state leaks I only tested that vp8enc ! vp8dec does not crash, as valgrind does not grok at least one of the instructions used by vp8enc, preventing me from checking a leak, and the lack of one after the patch. 2012-06-06 13:02:40 +0200 Wim Taymans * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 113/134] update for tag event change 2012-05-28 16:05:21 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 112/134] vp8: Port to 0.11 again 2012-05-18 12:46:55 +0100 Vincent Penquerc'h * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 111/134] vp8enc: fix target bitrate config with libvpx 1.1.0 libvpx 1.1.0 disallows a bitrate of 0, which was used by vp8enc as a default value. Instead, we use the default libvpx bitrate, scaled to our video size, if no bitrate was specified. This fixes encoding VP8 video with libvpx 1.1.0. https://bugzilla.gnome.org/show_bug.cgi?id=676245 2012-05-16 14:04:28 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 110/134] vp8enc: Update for GstVideoCodecFrame API changes 2012-04-27 18:22:42 -0300 Thiago Santos * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8dec.h: [MOVED FROM BAD 109/134] vp8dec: Improve output_state handling Avoid getting output_state for every buffer as that requires getting the objectlock and doing reference counting. Store it locally when it is created and use it. 2012-04-27 09:05:57 -0300 Thiago Santos * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 108/134] vp8dec: Use outputstate when copying output buffer data Using the input state was causing a crash because the strides/offsets would be wrong. Fix it by using the output as we are dealing with the decoded frame. 2012-04-24 11:08:41 +0200 Edward Hervey * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 107/134] vp8: Port to -base video base classes Conflicts: ext/vp8/Makefile.am ext/vp8/gstvp8dec.c ext/vp8/gstvp8enc.c Back to 0.10 state for now, need to be ported again. 2012-05-18 12:46:55 +0100 Vincent Penquerc'h * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 106/134] vp8enc: fix target bitrate config with libvpx 1.1.0 libvpx 1.1.0 disallows a bitrate of 0, which was used by vp8enc as a default value. Instead, we use the default libvpx bitrate, scaled to our video size, if no bitrate was specified. This fixes encoding VP8 video with libvpx 1.1.0. https://bugzilla.gnome.org/show_bug.cgi?id=676245 2012-04-05 18:02:56 +0200 Sebastian Dröge * ext/vp8/plugin.c: [MOVED FROM BAD 105/134] gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-04 14:41:22 +0200 Sebastian Dröge * ext/vp8/Makefile.am: [MOVED FROM BAD 104/134] gst: Update versioning 2012-03-06 15:21:17 +0100 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 103/134] vp8enc: Fix 'argument to 'sizeof' in 'memset' call is the same expression as the destination' compiler warning 2012-01-30 17:17:16 +0100 Wim Taymans * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 102/134] update for HEADER flag 2012-01-25 18:49:58 +0100 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 101/134] port some more to new memory API Fixes #668677. 2012-01-24 11:22:46 +0100 Oleksij Rempel (Alexey Fisher) * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 100/134] vp8enc: trace outgoing timestamps add info level prints for outgoing timestamps. Signed-off-by: Oleksij Rempel (Alexey Fisher) 2012-01-04 11:05:48 +0100 Oleksij Rempel (Alexey Fisher) * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 099/134] vp8dec: use is_alt_data option to prevent timestamp collisions altref/invisible frames usually stored in container with same timestamp as dependet frame. This make basevideodecoder to update timestamp for dependet frame and couse TS colision on next frame: ^- here is altref time : 1 2 3 4 5 6 7 8 9 webm ts : 1 3 5 5 7 9 vp8dec ts: 1 3 7 7 9 Fix bug: https://bugzilla.gnome.org/show_bug.cgi?id=655245 Signed-off-by: Oleksij Rempel (Alexey Fisher) 2012-01-02 08:28:13 +0100 Oleksij Rempel (Alexey Fisher) * ext/vp8/GstVP8Enc.prs: * ext/vp8/Makefile.am: [MOVED FROM BAD 098/134] vp8: add initial preset file This is initial preset file, currently with only one profile for realtime encoding. Signed-off-by: Oleksij Rempel (Alexey Fisher) 2011-11-28 13:08:27 +0000 Vincent Penquerc'h * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 097/134] various: fix pad template ref leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 2011-11-25 11:36:14 +0000 Tim-Philipp Müller * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 096/134] vp8dec: use new basevideodecoder API to drop frames and get QoS messages posted 2011-11-10 15:13:34 +0200 Mart Raudsepp * ext/vp8/Makefile.am: [MOVED FROM BAD 095/134] mimic, opencv, vp8, acmmp3dec, linsys: Don't build static plugins Pass --tag=disable-static to libtool everywhere where it's been forgotten https://bugzilla.gnome.org/show_bug.cgi?id=663768 2011-11-03 14:01:41 +0100 Edward Hervey * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 094/134] vp8: Port to 0.11 2011-08-21 20:15:25 -0700 David Schleef * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 093/134] vp8enc: fix drop-frame property Fixes #656929. 2011-08-19 19:17:15 +0100 Vincent Penquerc'h * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 092/134] vp8: probe for the new tuning API to keep building with older libvpx https://bugzilla.gnome.org/show_bug.cgi?id=656928 2011-08-18 10:39:26 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 091/134] vp8enc: Remove unused and useless variable in tags handling 2011-08-12 12:08:08 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 090/134] vp8enc: Update for basevideoencoder ::get_caps() removal 2011-07-09 18:53:24 -0700 David Schleef * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 089/134] vp8enc: Add more properties 2011-06-19 16:06:46 +0200 Alexey Fisher * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 088/134] vp8enc: add min/maxsection-pct option This options should be good to redeuce decode CPU load. for lowend hardware: minsection-pct=15 maxsection-pct=400 for hiend hw: minsection-pct=5 maxsection-pct=800 see example: http://www.webmproject.org/tools/encoder-parameters/#2-pass_vbr_encoding_for_smooth_playback_on_low-end_hardware Signed-off-by: Alexey Fisher Signed-off-by: David Schleef 2011-06-19 11:05:36 +0200 Alexey Fisher * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 087/134] vp8enc: add lag-in-frames option. This option set maximum of frames codec should remember, to make better prediktion for alt-ref frames. See example: http://www.webmproject.org/tools/encoder-parameters/#2-pass_best_quality_vbr_encoding Signed-off-by: Alexey Fisher Signed-off-by: David Schleef 2011-06-19 07:16:57 +0200 Alexey Fisher * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 086/134] vp8enc: use multipass.cache file name as default for multipass mode. Signed-off-by: Alexey Fisher Signed-off-by: David Schleef 2011-07-21 08:03:51 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 085/134] vp8enc: Update for GstBaseVideoEncoder::finish() signature change 2011-07-12 18:05:25 -0400 Olivier Crête * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 084/134] vp8: Fix set-but-unused warnings 2011-07-09 11:31:02 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 083/134] vp8enc: Use destroy notify to free the coder hook 2011-06-18 15:56:49 -0700 David Schleef * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 082/134] vp8enc: update for new libvpx api 2011-06-26 15:15:54 +0200 Alexey Fisher * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 081/134] vp8enc: generate a timestamp for alt-ref frames. It will fix handling of altref/invisible frames since matroska-mux drop any fram with no timestamp. see also: http://www.webmproject.org/code/specs/container/ The encoder will currently set the AR's timestamp as close as possible to the previous frame while attempting to provide a timestamp that is strictly increasing. In cases where the time base given to the encoder at configure time is not granular enough to allow for this the AR will share the same timestamp as D, but should be treated as having no duration. Fixes bug #652951 Signed-off-by: Alexey Fisher 2011-06-18 17:47:36 +0200 Alexey Fisher * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 080/134] vp8dec: add check if we have legal aspect-ratio before reset it. the commit f9b552f0494e (vp8dec: set par to 1/1) will fix situation where no aspect-ratio is set, but it brake stream with available aspect-ratio. This patch fix it. Fixes: #652902. Signed-off-by: Alexey Fisher 2011-06-03 19:36:59 -0700 David Schleef * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 079/134] vp8dec: set par to 1/1 2011-05-18 13:27:20 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 078/134] vp8enc: Name max/min quantizer properties {max,min}-quantizer Also improve quality property description. 2011-05-18 13:26:23 +0200 Alexey Fisher * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 077/134] vp8enc: Add properties to select a maximum and minimum quantizer Fixes bug #641405. 2011-05-18 13:18:58 +0200 Alexey Fisher * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 076/134] vp8enc: Fix quality to (constant) quantizer mapping This now allows to select all possible quantizers between 0 and 63. See bug #641405. 2011-04-01 22:13:55 +0200 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 075/134] vp8dec: debug code style fixes 2011-04-01 22:13:00 +0200 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 074/134] vp8dec: propagate downstream flow return to upstream 2011-03-30 10:18:23 +0200 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 073/134] basevideodecoder: really and only set src pad caps whenever requested ... since subclass is expected to be wise enough to know when to do so. 2011-03-29 10:41:54 +0200 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 072/134] basevideodecoder: invoke subclass start method at state change and use set_format While this changes API slightly (e.g. actually uses set_format now), which is OK for unstable API, it has following merits: * symmetric w.r.t. stop at state change * in line with other base class practice * otherwise no subclass method at state change (global activation time) Moreover, subclassese are either unaffected or trivially adjusted accordingly. 2011-03-28 08:59:20 +0200 Mark Nauwelaerts * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 071/134] basevideodecoder: subsume skip_frame into finish_frame 2011-03-24 14:10:07 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 070/134] basevideoencoder: provide proper upstream flow return handling 2011-03-24 13:59:35 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 069/134] vp8enc: minor optimization in setting up image buffer 2011-03-24 12:50:23 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 068/134] vp8enc: refactor frame processing 2011-03-24 11:55:41 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 067/134] vp8enc: do init at set_format time 2011-03-24 10:15:55 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 066/134] vp8enc: fix keyframe forcing 2011-03-23 09:45:20 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 065/134] basevideocodec: remove redundant caps field ... as it is already at hand as the src pad's negotiated caps. 2011-03-23 08:50:31 +0100 Mark Nauwelaerts * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 064/134] vp8enc: use baseclass event virtual handler 2011-02-20 14:16:18 -0800 David Schleef * ext/vp8/gstvp8dec.h: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 063/134] basevideo: merge utils header into basevideocodec 2011-03-17 16:34:02 +0000 Tim-Philipp Müller * ext/vp8/Makefile.am: [MOVED FROM BAD 062/134] vp8: fix LIBADD order in Makefile.am 2011-02-04 09:08:26 +0100 Alexey Fisher * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 061/134] vp8enc: Add description for bitrate units. 2010-11-30 18:43:24 -0800 David Schleef * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 060/134] vp8enc: Readd setting of granulepos Revert parts of last patch that removed setting of granulepos. oggmux still requires correct granulepos in incoming packet. 2010-11-29 20:21:31 -0800 David Schleef * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 059/134] vp8enc: Don't override timestamps set by base class Because the base class does it correctly. Fixes: #635720, #625558. 2010-11-25 18:52:47 +0100 Edward Hervey * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 058/134] vp8: Remove dead assignments 2010-10-09 17:36:07 -0700 David Schleef * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 057/134] basevideo: Move common fields/functions to basecodec 2010-09-18 17:28:48 -0700 David Schleef * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 056/134] basevideo: Move deadline to frame structure 2010-08-13 14:34:21 +0200 Philip Jägenstedt * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 055/134] vp8dec: Set GstBaseVideoDecoder::packetized to TRUE as soon as possible This fixes an infinite loop if an EOS event is received before GstBaseVideoDecoder::start() is called, e.g. immediately when the pads are activated. Fixes bug #626815. 2010-07-10 16:52:10 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 054/134] vp8enc: Add support for enabling automatic insertion of alt-ref frames by the encoder 2010-07-10 16:51:53 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 053/134] vp8enc: Fix handling of invisible/alt ref frames 2010-07-03 17:47:29 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8dec.h: * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: [MOVED FROM BAD 052/134] vp8: Add initial documentation, based on the theoradec/theoraenc documentation 2010-07-03 17:34:58 +0200 Sebastian Dröge * ext/vp8/Makefile.am: * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8dec.h: * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8enc.h: * ext/vp8/plugin.c: [MOVED FROM BAD 051/134] vp8: Move structure definitions, etc to public header files for gtk-doc 2010-06-12 09:02:29 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 050/134] vp8enc: Implement multipass encoding Fixes bug #621348. 2010-06-14 15:56:24 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 049/134] vp8enc: Set VP8E_SET_CPUUSED to 0 This setting controls how much CPU can be used by the encoder, specified in fractions of 16. Negative values mean strict enforcement of this while positive values are adaptive. The default value is -4, which means that we're not running as fast as possible and probably are wasting some quality. 0 is the recommended default by libvpx upstream. 2010-06-14 15:51:30 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 048/134] vp8enc: Use VPX defines for REALTIME, GOOD/BEST quality deadlines instead of our own These are the values used for the speed property. 2010-06-03 10:49:40 +0100 Tim-Philipp Müller * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 047/134] vp8enc: fix printf format warning in log message gstvp8enc.c:564: error: format ‘%d’ expects type ‘int’, but argument 8 has type ‘size_t’ gstvp8enc.c:744: error: format ‘%d’ expects type ‘int’, but argument 8 has type ‘size_t’ 2009-07-03 16:08:38 +0100 Tim-Philipp Müller * ext/vp8/Makefile.am: [MOVED FROM BAD 046/134] basevideo, vp8: guard unstable API with GST_USE_UNSTABLE_API Add some guards and fat warnings to the header files with still unstable API, so people who just look at the installed headers know that it actually is unstable API. Merging previous commit into current codebase. 2010-06-01 15:54:51 -0700 David Schleef * ext/vp8/Makefile.am: * ext/vp8/gst/video/gstbasevideocodec.c: * ext/vp8/gst/video/gstbasevideocodec.h: * ext/vp8/gst/video/gstbasevideodecoder.c: * ext/vp8/gst/video/gstbasevideodecoder.h: * ext/vp8/gst/video/gstbasevideoencoder.c: * ext/vp8/gst/video/gstbasevideoencoder.h: * ext/vp8/gst/video/gstbasevideoparse.c: * ext/vp8/gst/video/gstbasevideoparse.h: * ext/vp8/gst/video/gstbasevideoutils.c: * ext/vp8/gst/video/gstbasevideoutils.h: * ext/vp8/gst/video/gstvideocompat.c: * ext/vp8/gst/video/gstvideocompat.h: [MOVED FROM BAD 045/134] basevideo: Move base video from vp8 to gst-libs 2010-05-26 06:52:15 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8utils.h: [MOVED FROM BAD 044/134] vp8: Use VPX_PLANE_* instead of PLANE_* 2010-05-24 11:04:02 +0200 Sebastian Dröge * ext/vp8/gstvp8utils.h: [MOVED FROM BAD 043/134] vp8: Add compatilibity defines to work with older versions of libvpx too 2010-05-23 09:28:13 +0200 Philip Jägenstedt * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 042/134] vp8dec: s/IMG_FMT_I420/VPX_IMG_FMT_I420/ This corresponds to upstream libvpx commit 6cd4a10e167203d1deb79abf60ee72599e97891b 2010-05-22 12:55:45 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 041/134] vp8enc: Allow a maximum keyframe distance of 0, i.e. all frames are keyframes 2010-05-22 08:45:35 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 040/134] vp8dec: Set decoder deadline from the QoS information 2010-05-28 16:35:12 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 039/134] vp8enc: Move debug output one line above where the packet is still valid 2010-05-28 15:53:30 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 038/134] vp8enc: Correctly ignore non-frame packets from the encoder Fixes bug #619916. 2010-05-22 07:44:27 +0200 Sebastian Dröge * ext/vp8/gst/video/gstbasevideodecoder.c: [MOVED FROM BAD 037/134] basevideodecoder: Take the frame duration into account when calculating the earliest time This formula is used in many other elements too. Fixes bug #619318. 2010-05-22 07:35:01 +0200 Sebastian Dröge * ext/vp8/gst/video/gstbasevideodecoder.c: [MOVED FROM BAD 036/134] basevideodecoder: Reset QoS values when necessary 2010-05-22 09:35:24 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 035/134] vp8enc: Use GST_VIDEO_CAPS_YUV(I420) instead of handwritten I420 caps for the pad template Fixes bug #619344. 2010-05-21 20:53:36 +0200 Philip Jägenstedt * ext/vp8/gst/video/gstbasevideodecoder.c: * ext/vp8/gst/video/gstbasevideodecoder.h: * ext/vp8/gst/video/gstbasevideoutils.h: * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 034/134] vp8dec: drop late frames after decoding them This saves a memcpy, which is always something. 2010-05-21 21:28:29 +0200 Philip Jägenstedt * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 033/134] vp8enc: threads property Increasing from 1 to 2 threads on an Thinkpad X60s decreased encode time in a test from ~24 s to ~19 s, so this is quite useful. Ideally we should let 0 be the default and automatically match the number of CPU cores (or something). 2010-05-21 15:17:46 +0200 Philip Jägenstedt * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 032/134] vp8enc: add mode property to switch between CBR/VBR Always using CBR when bitrate is used isn't that great, VBR mode can produce meaningful results too. 2010-05-21 10:54:57 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 031/134] vp8dec: Only enable postprocessing if the decoder supports it 2010-05-21 08:23:58 +0200 Philip Jägenstedt * ext/vp8/plugin.c: [MOVED FROM BAD 030/134] vp8: typo: s/HAVE_VP8_DECODER/HAVE_VP8_ENCODER/ Fixup for bug #619172. 2010-05-21 08:13:06 +0200 Philip Jägenstedt * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 029/134] vp8: move #ifdef HAVE_VP8_ENCODER/DECODER Otherwise we'll try including e.g. which doesn't exist. 2010-05-20 20:06:09 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 028/134] vp8enc: Write GStreamer element and version in the vorbiscomment vendor string 2010-05-20 16:49:03 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/plugin.c: [MOVED FROM BAD 027/134] vp8: Only enable the encoder or decoder if it's available in libvpx Fixes bug #619172. 2010-05-20 10:19:54 +0200 Philip Jägenstedt * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/plugin.c: [MOVED FROM BAD 026/134] vp8: exlcude dec/enc based on CONFIG_VP8_DECODER/ENCODER This may not be very autotoolish, but works with libvpx in the state that libvpx is actually in. Moved the debug init to the elements themselves to minimize amount of #ifdefs 2010-05-20 09:24:53 +0200 Philip Jägenstedt * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 025/134] vp8enc: Limit max-latency to 25 to match libvpx From libvpx/vp8/encoder/onyx_int.h: #define MAX_LAG_BUFFERS (CONFIG_REALTIME_ONLY? 1 : 25) While we don't need to be tied to what libvpx does internally, it doesn't make sense to pretend to support longer frame lags than are actually possible. 2010-05-20 09:56:25 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8utils.c: [MOVED FROM BAD 024/134] vp8: Undef HAVE_CONFIG_H before including libvpx headers A public libvpx header includes private headers if this is defined, causing compilation failures because the private headers are not installed of course. 2010-05-20 08:53:12 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 023/134] vp8enc: Some more minor adjustments for the Ogg mapping 2010-05-19 23:02:19 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 022/134] vp8dec: Fix memory leak 2010-05-19 21:34:42 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 021/134] vp8enc: Adjust Ogg mapping for the changes 2010-05-19 18:12:18 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 020/134] vp8dec: Add properties to control the VP8 decoder post processing feature This is disabled by default for now. 2010-05-19 17:16:54 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 019/134] vp8enc: Rename keyframe-interval to max-keyframe-distance And use default settings for buffer sizes until we expose this somehow. 2010-05-19 17:13:17 +0200 Sebastian Dröge * ext/vp8/Makefile.am: * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/gstvp8utils.c: * ext/vp8/gstvp8utils.h: [MOVED FROM BAD 018/134] vp8: Improve error handling and debug output 2010-05-19 14:46:48 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 017/134] vp8: Use correct strides and plane offsets for GStreamer 2010-05-18 14:47:54 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 016/134] vp8enc: Implement GstTagSetter interface 2010-05-18 14:33:49 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 015/134] vp8enc: Fix setting of the keyframe flag on encoded frames 2010-05-18 14:30:15 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 014/134] vp8enc: Post an error message on the bus if encoder initialization fails 2010-05-18 14:28:55 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 013/134] vp8dec: Fix memory leaks and fail if initializing the decoder fails 2010-05-18 02:44:54 -0700 David Schleef * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 012/134] vp8enc: Set timebase Also misc cleanup. 2010-05-16 10:36:12 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 011/134] vp8dec: Fix decoding of invisible frames 2010-05-14 14:26:34 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 010/134] vp8enc: Update the latency when initializing the encoder 2010-05-14 14:02:53 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 009/134] vp8dec: Correctly initialize stream info before peeking at the stream Otherwise peeking will fail and we'll get invalid values 2010-05-14 11:01:29 +0200 Sebastian Dröge * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 008/134] vp8dec: Make sure to pass a keyframe as first frame to the decoder, copy output frames only once and require width/height/etc on the input caps 2010-05-14 10:30:18 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 007/134] vp8enc: Add support for invisible frames and the Ogg mapping 2010-05-14 01:14:46 -0700 David Schleef * ext/vp8/gstvp8dec.c: [MOVED FROM BAD 006/134] vp8dec: Fix reset after seeking Also remove some unused code. 2010-05-13 21:19:32 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 005/134] vp8enc: Set frame numbers as buffer offsets 2010-05-13 21:18:08 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 004/134] vp8enc: Always get as many frames as possible from the encoder 2010-05-13 21:08:03 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 003/134] vp8enc: Fill the oldest pending frame instead of the newest 2010-05-13 20:20:32 +0200 Sebastian Dröge * ext/vp8/gstvp8enc.c: [MOVED FROM BAD 002/134] vp8enc: Correctly set delta unit flag for non-keyframes 2010-05-13 01:04:04 -0700 David Schleef * ext/vp8/Makefile.am: * ext/vp8/gst/video/gstbasevideocodec.c: * ext/vp8/gst/video/gstbasevideocodec.h: * ext/vp8/gst/video/gstbasevideodecoder.c: * ext/vp8/gst/video/gstbasevideodecoder.h: * ext/vp8/gst/video/gstbasevideoencoder.c: * ext/vp8/gst/video/gstbasevideoencoder.h: * ext/vp8/gst/video/gstbasevideoparse.c: * ext/vp8/gst/video/gstbasevideoparse.h: * ext/vp8/gst/video/gstbasevideoutils.c: * ext/vp8/gst/video/gstbasevideoutils.h: * ext/vp8/gst/video/gstvideocompat.c: * ext/vp8/gst/video/gstvideocompat.h: * ext/vp8/gstvp8dec.c: * ext/vp8/gstvp8enc.c: * ext/vp8/plugin.c: [MOVED FROM BAD 001/134] vp8: Add encoder/decoder 2012-09-15 22:16:52 +0200 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Update spec file with F18 name change and add deinterlacer 2012-09-15 19:06:06 +0200 Mark Nauwelaerts * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: use gst_element_factory_get_metadata to replace obsolete API 2012-09-14 17:55:16 +0200 Mark Nauwelaerts * sys/osxaudio/gstosxaudiosink.c: replace _get_caps_reffed with _get_caps 2012-09-14 17:08:49 +0200 Mark Nauwelaerts * gst/audiofx/gststereo.c: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 17:08:49 +0200 Mark Nauwelaerts * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 17:07:26 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * tests/check/elements/qtmux.c: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 13:30:37 +0200 Wim Taymans * ext/jpeg/gstjpegenc.c: * gst/multipart/multipartmux.c: * gst/rtp/README: * gst/videocrop/gstaspectratiocrop.c: * gst/y4m/gsty4mencode.c: * tests/examples/equalizer/demo.c: * tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh: * tests/examples/rtp/server-VTS-H263p.sh: * tests/examples/rtp/server-decodebin-H263p-AMR.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: * tests/icles/gdkpixbufsink-test.c: fix more caps 2012-09-14 02:57:44 +0100 Tim-Philipp Müller * configure.ac: Back to development === release 0.11.94 === 2012-09-14 02:48:43 +0100 Tim-Philipp Müller * ChangeLog: * configure.ac: * gst-plugins-good.doap: * win32/common/config.h: Release 0.11.94 2012-09-14 01:50:44 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2012-09-14 01:46:14 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update docs 2012-09-14 00:47:38 +0100 Tim-Philipp Müller * tests/check/elements/wavpackenc.c: tests: push stream-start and segment events in wavpackenc test 2012-09-13 10:56:27 +0200 Wim Taymans * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: remove unused properties 2012-09-13 10:15:54 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: disable reconfigure See https://bugzilla.gnome.org/show_bug.cgi?id=683902 2012-09-10 22:09:59 -0700 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: deinterlace: Don't treat every custom-downstream event as EOS Don't fall through to the EOS handling after receiving a custom-downstream event. 2012-09-12 21:05:44 +0200 Stefan Sauer * ext/cairo/gsttextoverlay.c: * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-mux.c: * gst/multipart/multipartmux.c: * gst/smpte/gstsmpte.c: * gst/videomixer/videomixer2.c: collectpads: remove gst_collect_pads_add_pad_full Rename gst_collect_pads_add_pad_full() to gst_collect_pads_add_pad() and fix all invocations. 2012-09-12 17:14:46 +0200 Mark Nauwelaerts * gst/udp/gstmultiudpsink.c: udp: add include for IPPROTO_* 2012-09-12 16:39:08 +0200 Mark Nauwelaerts * gst/udp/gstmultiudpsink.c: udp: properly match braces and cpp directives Fixes compilation where IPV6_TCLASS not defined. 2012-09-12 14:42:07 +0200 Edward Hervey * gst/shapewipe/gstshapewipe.c: shapewipe: Use default query handler where needed And clean up get_caps code while I'm at it 2012-09-12 13:28:07 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: improve framerate transform Handle G_MAXINT in the framerates better. If we cannot double or divide the framerate, clamp to the smallest/largest possible value we can express instead of failing. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683861 2012-09-12 13:17:54 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: small cleanup 2012-09-07 17:20:57 -0400 Youness Alaoui * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/videomixer2.c: videomixer2: Adding nv12 and nv21 support Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683841 2012-09-12 10:18:53 +0200 Michael Smith * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: add support for prores Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683839 2012-09-12 00:16:31 +0100 Tim-Philipp Müller * tests/check/elements/rganalysis.c: tests: fix most of the rganalysis unit tests Before the element would post messages on the bus itself, now the sinks do that based on the tag events they receive. But since we don't have proper sink elements in these unit tests, but just dangling pads, we have to post the tag messages the test checks for ourselves. Down from 52/55 failing to 7/52 failing. 2012-09-11 17:36:51 +0200 Mark Nauwelaerts * ext/dv/gstdvdemux.c: * gst/avi/gstavidemux.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/wavparse/gstwavparse.c: ext, gst: only activate in pull mode if upstream is seekable 2012-09-11 15:38:23 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2: disable renegotiation We can't yet wait for the bufferpool to DRAIN before starting renegotiation so disable it for now. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682770 2012-09-11 12:48:39 +0200 Mark Nauwelaerts * tests/check/elements/rtpbin.c: tests: rtpbin: port to the new GLib thread API 2012-09-11 12:36:56 +0200 Mark Nauwelaerts * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: directsoundsink: port to the new GLib thread API 2012-09-11 11:59:54 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: don't reset segment Don't reset the segment because we need the values for accumulation. the segment is reset at start and after a flushing seek. Fixes some problems with files with quicktime segments. 2012-09-10 17:14:37 +0200 Wim Taymans * tests/check/elements/id3demux.c: tests: fix id3demux test 2012-09-10 14:31:02 +0200 Mark Nauwelaerts * gst/flv/amfdefs.h: * gst/flv/gstflvdemux.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsv3vdepay.c: gst: adjust comment style 2012-09-10 14:30:42 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: remove defunct commented code 2012-09-10 13:35:15 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: consider stream alive when not connected yet When we start and renegotiate, there is a moment where the stream is created but not yet connected. Make sure all functions deal with this situation correctly instead of erroring out. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681247 2012-09-10 12:15:25 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: don't fail when not negotiated yet When get_time is called but we are not yet negotiated, return 0 instead of posting an error. It's possible that the base class is still negotiating when our get_time is called. 2012-09-10 11:32:25 +0200 Wim Taymans * ext/pulse/pulsesrc.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-source.c: update for audio base src api change 2012-09-10 00:42:52 +0100 Tim-Philipp Müller * gst/avi/gstavimux.c: * gst/isomp4/qtdemux.c: video/x-3ivx and video/x-xvid -> video/mpeg,mpegversion=4 If it ever turns out that we really must use thoe specific fourccs and not the generic one, we can still add a flavor field to the caps later. 2012-09-07 16:15:42 +0200 Daniela * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid leak When setup fails, make sure to cleanup afterwards. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673509 2012-09-07 15:23:44 +0200 Mark Nauwelaerts * gst/rtp/gstrtpamrdepay.c: rtpamrdepay: unmap rtp buffer ... thereby plugging a memleak. 2012-09-07 14:13:17 +0200 Mark Nauwelaerts * tests/check/elements/rtp-payloading.c: tests: rtp-payloading: adjust to modified bufferlist semantics ... now implemented by buffer memory blocks. 2012-09-07 14:11:39 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: avoid crashing on NULL access in debug message 2012-09-07 14:11:02 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263ppay.c: rtph263ppay: plug caps leak 2012-09-06 17:09:20 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: remove redundant _set_allocation call 2012-09-06 17:05:00 +0200 Mark Nauwelaerts * tests/check/elements/deinterlace.c: tests: deinterlace: do not leak deinterlace pads 2012-09-06 17:04:39 +0200 Mark Nauwelaerts * gst/deinterlace/gstdeinterlace.c: deinterlace: plug some leaks 2012-09-06 16:49:02 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: reuse core function for GCD 2012-09-06 16:31:00 +0200 Mark Nauwelaerts * gst/deinterlace/gstdeinterlace.c: deinterlace: support filter in getcaps 2012-09-06 16:30:44 +0200 Mark Nauwelaerts * gst/deinterlace/gstdeinterlace.c: deinterlace: do not leak getcaps result 2012-09-06 16:23:28 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: add support for bufferpool Add bufferpool support to avoid a memcpy in the videosink when actively interlacing. Remove some commented obsolete code. 2012-09-06 13:38:52 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: proxy allocation query in passthrough We can let the allocation query pass when we are operating in passthrough mode. 2012-09-06 13:23:46 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: use default event functions instead of blindly forwarding unknown events. 2012-09-06 13:23:30 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: small cleanups 2012-09-06 12:56:30 +0200 Wim Taymans * gst/deinterlace/gstdeinterlace.c: deinterlace: call default query handlers Call the default query handler instead of forwarding the query blindly. Fixes issues of strides because of proxying the allocation query wrongly. 2012-09-06 10:42:21 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: remove unused code. 2012-09-06 10:42:06 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulse: improve debug 2012-09-05 11:50:05 +0200 Mark Nauwelaerts * ext/dv/gstdvdemux.c: dvdemux: remove obsolete update newsegment handling code 2012-09-04 12:35:53 +0200 Wim Taymans * gst/videofilter/gstvideobalance.c: videobalance: avoid deadlock _update_properties takes the object lock and should not be called when the object lock is already taken. 2012-09-03 12:46:03 +0100 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: extract interlaced-ness of video track from interlace-mode field instead of the old boolean "interlaced" field. 2012-09-03 02:51:24 +0100 Tim-Philipp Müller * gst/avi/gstavimux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/rtp/gstrtpmp4vpay.c: * tests/check/elements/avimux.c: video/x-xvid -> video/mpeg,mpegversion=4 2012-09-02 02:50:50 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: text/plain + text/x-pango-markup -> text/x-raw 2012-09-02 01:31:53 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * gst/matroska/matroska-demux.c: gst_message_new_duration -> gst_message_new_duration_changed 2012-08-30 22:07:24 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: also stop probatation on existing sources Receiving an RTCP packet should also stop probation on sources we have seen before. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065 2012-08-22 16:36:21 -0700 Aleix Conchillo Flaque * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 12:21:01 +0200 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbuf: adjust to modified video overlay composition API 2012-08-30 11:30:01 +0200 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: fixup 0.11 port of suspect frame checking Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682959 2012-08-28 18:56:19 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: avoid invalid H264 bytestream codec_data Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681369 2012-08-28 19:00:44 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: port segment event creation to 0.11 2012-08-28 16:28:13 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: release extra event ref when replacing pending newsegment event 2012-07-03 17:50:24 +0200 David Corvoysier * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_types.c: isomp4: add DASH tfdt box support MPEG DASH has defined a set of new boxes to specify duration, indexes and offsets of ISOBMFF fragments. The Track Fragment Base Media Decode Time (tfdt) Box can in particular be included inside a traf box to specify the absolute decode time, measured on the media timeline, of the first sample in decode order in the track fragment. This information can be used by the isomp4 demux to find out the current position of an MP4 fragment in the timeline. This patch adds code to isomp4 to: - parse the tfdt box - adjust the time/position member of the new segment sent when playback starts Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677535 2012-08-26 22:39:55 +0100 Tim-Philipp Müller * ext/aalib/gstaasink.c: * ext/cairo/gstcairorender.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/libcaca/gstcacasink.c: * ext/libpng/gstpngdec.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/atomsrecovery.c: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstsplitfilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/README: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/wavparse/gstwavparse.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: * tests/examples/cairo/cairo_overlay.c: * tests/examples/rtp/client-H263p-AMR.sh: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H263p.sh: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-H264.sh: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh: * tests/examples/rtp/server-VTS-H263p.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-decodebin-H263p-AMR.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: * tests/examples/shapewipe/shapewipe-example.c: * tests/icles/gdkpixbufsink-test.c: * tests/icles/videocrop-test.c: docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-26 22:32:54 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: * gst/videomixer/videomixer2.c: docs: gst-launch-0.11 -> gst-launch-1.0 2012-08-26 22:08:54 +0100 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: * tests/check/elements/deinterlace.c: deinterlace: the field in caps is "interlace-mode" not "interlace-method" Fix deinterlace unit test. Need to set right field on output caps. Also remove right field (not old 0.10 "interlaced" boolean field) from caps in unit test before comparing old and new. 2012-08-26 21:45:44 +0100 Tim-Philipp Müller * tests/check/elements/icydemux.c: tests: fix icydemux unit test Was waiting for a tag message on the bus, which would never come, because elements don't post those themselves any more but let sinks post them from tag events. Only that there are no sinks in this unit test. 2012-08-26 21:27:00 +0100 Tim-Philipp Müller * tests/check/elements/videocrop.c: tests: fix videocrop crop_to_1x1 unit test for GRAY8 format Update table with pixel values with the value actually produced by videotestsrc. 2012-08-27 09:00:45 +0200 Sjoerd Simons * ext/pulse/pulsesrc.c: pulsesrc: Only print caps if they're provided 2012-08-24 19:43:08 +0100 Michael Rubinstein * gst/videomixer/blend.c: videomixer: fix endianness check on systems where non-glib endianness defines are not set On Windows LITTLE_ENDIAN without the G_ in was not defined, so the test comes out wrong. 2012-08-22 17:23:25 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: udpsink: don't crash on NULL error Check if there is an error before retrieving its message. See https://bugzilla.gnome.org/show_bug.cgi?id=682481 2012-08-22 13:30:19 +0200 Stefan Sauer * common: Automatic update of common submodule From 668acee to 4f962f7 2012-08-22 13:18:00 +0200 Stefan Sauer * configure.ac: configure: bump gtk-doc req to 1.12 (mar-2009) This allows us to e.g. unconditionally use gtkdoc-rebase. 2012-08-22 11:21:38 +0200 Martin Ertsaas * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: Make osxvideosink use the non-deprecated threading api from glib. https://bugzilla.gnome.org/show_bug.cgi?id=682446 2012-08-14 15:40:31 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Handle negotiation events This makes sure that we: a) Destroy an existing stream if a negotiate() request comes in: this is required when receiving a downstream renegotiation request after a stream has been created. b) Create a new stream on prepare(): this is required since we do a setcaps() in negotiate(), which causes the stream to be dropped by a ringbuffer release() call (this does not happen during first negotiation since the release is only done on a running ringbuffer). The subsequent call to ringbuffer acquire() fails because the stream was lost on release(). https://bugzilla.gnome.org/show_bug.cgi?id=681247 2012-08-14 15:38:27 +0530 Arun Raghavan * ext/pulse/pulseutil.c: pulse: Clear unpositioned flag when setting positions If converting a PA channel map to gst channel positions results in a valid set of channel positions, we clear the unpositioned flag from the ringbuffer spec. 2012-08-14 09:37:45 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Remove redundant channel-mask setting for stereo case The gstaudio helper libraries already take care of this case for us. 2012-08-14 09:36:30 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Don't use memset to set invalid channel positions This itereates over the GstAudioInfo to set invalid channel positions rather than use memset() which works right now because it assumes that GST_AUDIO_CHANNEL_POSITION_INVALID is -1. 2012-08-22 10:30:04 +0100 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufsink.c: gdkpixbufsink: minor docs improvement 2012-08-22 10:23:24 +0100 Tim-Philipp Müller * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbufplugin.c: gdkpixbuf: re-enable already-ported gdkpixbufsink 2012-08-22 10:08:08 +0100 Tim-Philipp Müller * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: * ext/gdk_pixbuf/gstgdkpixbufplugin.c: gdkpixbuf: port gdkpixbufoverlay element to 0.11 2012-08-22 00:00:46 +0100 Tim-Philipp Müller * configure.ac: * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbufdec.c: * ext/gdk_pixbuf/gstgdkpixbufdec.h: * ext/gdk_pixbuf/gstgdkpixbufplugin.c: gdkpixbuf: re-enable already-ported gdkpixbuf element as gdkpixbufdec Not sure why it as disabled exactly given that it had already been ported (though without metas or baseclass). Move plugin_init bits into separate source file, and rename decoder element to gdkpixbufdec. 2012-08-21 23:25:47 +0100 Tim-Philipp Müller * ext/gdk_pixbuf/gst_loader.c: gdkpixbuf: remove old and unused gst_loader source file Once upon a time used to load GStreamer vids via GdkPixbuf API. 2012-08-16 16:51:16 -0700 Aleix Conchillo Flaque * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: make jitterbuffer drop-on-latency available (fix #682055) Conflicts: gst/rtsp/gstrtspsrc.h 2012-08-21 19:47:45 +0800 Huacai Chen * sys/v4l2/v4l2_calls.c: v4l2: make gst_v4l2_fill_lists() adapt to kernel 3.3+ When do v4l2_ioctl() with VIDIOC_ENUMINPUT fails on some devices, kernels before 3.3.0 return EINVAL, but newer kernels return ENOTTY. This patch make those devices work well on kernel 3.3+. Related kernel commit: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=commit;h=07d106d0a33d6063d2061305903deb02489eba20 Signed-off-by: Huacai Chen Signed-off-by: Rui Wang Signed-off-by: Jie Chen 2012-08-20 23:30:38 +0100 Tim-Philipp Müller * docs/plugins/inspect/plugin-matroska.xml: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: video/x-dvd-subpicture -> subpicture/x-dvd 2012-08-17 20:52:42 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesrc.c: multifilesrc: fix example pipeline in docs 2012-08-17 14:59:57 +0200 Stefan Sauer * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * tests/check/elements/equalizer.c: equalizer: enable presets for the n-band equalizer Add a test for saving and restoring the preset. 2012-08-14 01:20:19 +0100 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix not-negotiated errors on variable or missing framerate in input caps Remove some bogus code I added during porting that would error out on missing or variable framerates in input caps. Handle this like we do in 0.10 Fixes test_mode_disabled_passthrough unit test check. 2012-08-12 13:16:32 +0200 Sjoerd Simons * gst/law/alaw-decode.c: * gst/law/mulaw-decode.c: law: Filter layout caps field The layout caps field shouldn't be passed through to the sink pad of {mu,a}lawdec. https://bugzilla.gnome.org/show_bug.cgi?id=681677 2012-08-09 19:41:34 +0300 Anton Belka * ext/flac/gstflacenc.c: flacenc: allow a TOC with single alternative top-level entry Allow a TOC that has a single alternative top-level entry with multiple sequence sub-entries https://bugzilla.gnome.org/show_bug.cgi?id=540891 2012-08-09 11:48:39 +0200 Sebastian Dröge * ext/mpg123/gstmpg123audiodec.c: mpg123: Give MARGINAL rank to the mpg123 decoder element 2012-08-09 10:31:39 +0200 Sebastian Dröge * configure.ac: configure: And fix the GTK check to use the correct pkg-config package name 2012-08-09 10:25:38 +0200 Sebastian Dröge * configure.ac: configure: Fix GTK required version variable name 2012-08-09 08:35:23 +0100 Matthias Clasen * sys/v4l2/gstv4l2bufferpool.c: v4l2: fix build with recent kernels, the v4l2_buffer input field was removed This was unused apparently and removed in the kernel in commit: From 2b719d7baf490e24ce7d817c6337b7c87fda84c1 Mon Sep 17 00:00:00 2001 From: Sakari Ailus Date: Wed, 2 May 2012 09:40:03 -0300 Subject: [PATCH] [media] v4l: drop v4l2_buffer.input and V4L2_BUF_FLAG_INPUT Remove input field in struct v4l2_buffer and flag V4L2_BUF_FLAG_INPUT which tells the former is valid. The flag is used by no driver currently. https://bugzilla.gnome.org/show_bug.cgi?id=681491 Conflicts: sys/v4l2/gstv4l2bufferpool.c 2012-08-08 17:25:36 -0700 Olivier Crête * gst/rtp/gstrtph264pay.c: * tests/check/elements/rtp-payloading.c: rtph264pay: Make it actually work after cleanups 2012-08-08 17:40:34 +0200 Sebastian Dröge * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: gst: Set alignment at the correct place of GstAllocationParams 2012-08-08 17:39:07 +0200 Sebastian Dröge * ext/jpeg/gstjpegenc.c: * gst/matroska/matroska-demux.c: * gst/multipart/multipartmux.c: * gst/videomixer/videomixer2.c: gst: Set alignment at the correct place of GstAllocationParams 2012-08-08 16:25:58 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: Back to development === release 0.11.93 === 2012-08-08 15:22:04 +0100 Tim-Philipp Müller * configure.ac: * gst-plugins-good.doap: * win32/common/config.h: Release 0.11.93 2012-08-08 15:17:22 +0100 Tim-Philipp Müller * Makefile.am: * win32/MANIFEST: * win32/common/tuner-enumtypes.c: * win32/common/tuner-enumtypes.h: * win32/common/tuner-marshal.c: * win32/common/tuner-marshal.h: win32: add generated tuner-marshal/enumtypes files for v4l2src and update And gst-indent the right rtp marshal files; add missing files to MANIFEST. 2012-08-08 15:10:37 +0100 Tim-Philipp Müller * gst/deinterlace/tvtime-dist.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videomixer/blendorc-dist.c: gst: update disted orc files 2012-08-08 12:58:50 +0100 Tim-Philipp Müller * ext/mpg123/Makefile.am: mpg123: dist header file 2012-08-08 11:31:59 +0100 Tim-Philipp Müller * ext/wavpack/gstwavpackdec.c: * gst/rtpmanager/gstrtpssrcdemux.c: * sys/oss4/oss4-audio.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: Silence some 'variable may be used uninitialized' compiler warnings When compiling with -DG_DISABLE_ASSERT 2012-08-08 10:56:51 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * gst/isomp4/gstqtmoovrecover.c: * tests/icles/ximagesrc-test.c: No code with side-effects inside g_assert() please 2012-08-07 11:14:21 -0700 Olivier Crête * gst/udp/gstmultiudpsink.c: multiudpsink: Return FLUSHING instead of ERROR on unlock If the base class asks multiudpsink to unlock, then it should return FLUSHING, not ERROR 2012-07-26 16:19:57 +0300 Anton Belka * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: add TOC support Add TOC as embedded cuesheets in flac files. https://bugzilla.gnome.org/show_bug.cgi?id=54089 2012-08-07 12:12:09 +0200 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: generate empty vorbiscomment for complete streamheaders if needed Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681335 2012-08-06 18:02:50 -0700 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Block pad while it is announced. Block the RTP pad and associated RTCP pads while they are being announced. This it to prevent a race where one is announced and before the callback has connected it, the other one gets a buffer. We can't use the "padlock" of ssrcdemux because it causes deadlocks. 2012-08-06 15:00:57 +0100 Tim-Philipp Müller * common: common: un-do accidental common update revert in commit 7b5925b5 2012-08-06 14:50:53 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: set correct data_size for generated dummy frame ... which prevents getting stuck in a loop if such one is needed. 2012-08-06 14:50:03 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: improve and fix debug statement ... so it really informs about next rather than past frame. 2012-08-06 12:34:55 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: update available bytewriter space when repositioning ... and add some more assert to catch potential surprises early on. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558 2012-08-04 12:47:44 +0200 Sebastian Dröge * common: * ext/dv/gstdvdemux.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: gst: Add stream-id to stream-start events 2012-08-04 12:54:32 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Chain up to the parent class' query handler if no pad is provided 2012-08-02 01:48:29 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: add a better detection for the main run loop 2012-07-27 16:13:49 +0200 Xavi Artigas * sys/directsound/gstdirectsoundsink.c: directsoundsink: Do not overwrite the DS buffer when testing for AC3 support https://bugzilla.gnome.org/show_bug.cgi?id=680706 Conflicts: sys/directsound/gstdirectsoundsink.c 2012-08-05 16:39:23 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 94ccf4c to 668acee 2012-08-03 16:13:52 +0100 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Release lock before signalling new pad This prevents a deadlock where something would try to push an event through the SSRC demux from the callback, causing the pads to be iterated and the lock taken. 2012-08-04 16:13:36 +0100 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: gst_tag_list_free -> gst_tag_list_unref 2012-08-04 16:10:16 +0100 Tim-Philipp Müller * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/audioparsers/gstflacparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/debugutils/gsttaginject.c: * gst/flv/gstflvdemux.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/wavparse/gstwavparse.c: * tests/check/elements/apev2mux.c: * tests/check/elements/icydemux.c: * tests/check/elements/id3demux.c: * tests/check/elements/id3v2mux.c: * tests/check/elements/qtmux.c: * tests/check/elements/rganalysis.c: * tests/check/pipelines/tagschecking.c: gst_tag_list_free -> gst_tag_list_unref 2012-08-03 13:43:31 +0100 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.c: mpg123: map input buffer in READ mode, not WRITE mode Makes things actually work. 2012-08-03 11:50:10 +0100 Tim-Philipp Müller * ext/mpg123/gstmpg123audiodec.c: mpg123: query supported output formats at run-time Fixes stuff. We use a string here since we can't be bothered with GValue. 2012-08-03 14:10:32 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: manage race between connection closing and flushing ... where the former can happen in task thread and the latter in mainloop upon downward state change. 2012-08-03 14:02:23 +0200 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: improve and relax audio frame parsing ... so as to properly recognize first audio frame. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681077 Conflicts: ext/flac/gstflacdec.c 2012-08-03 11:48:02 +0100 Tim-Philipp Müller * ext/mpg123/Makefile.am: mpg123: hook up to build system 2012-08-03 11:13:48 +0100 Carlos Rafael Giani * ext/mpg123/gstmpg123audiodec.c: * ext/mpg123/gstmpg123audiodec.h: mpg123: add new libmpg123-based mp3 decoder plugin Needs a bit of cleaning up. https://bugzilla.gnome.org/show_bug.cgi?id=681003 2012-08-01 12:16:41 +0200 René Stadler * gst/isomp4/qtdemux.c: qtdemux: fix double unref of private tag buffer 2012-07-30 17:54:51 +0300 Anton Belka * gst/wavparse/gstwavparse.c: wavparse: create TOC as needed Avoid creating the toc if the wav has no or empty cue chunk. Also a small code cleanup. 2012-07-28 11:26:01 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: update for TOC API changes 2012-07-28 11:22:43 +0100 Tim-Philipp Müller * gst/matroska/matroska-read-common.c: matroska: update for TOC API changes 2012-07-28 11:20:08 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: update for TOC API changes 2012-07-28 00:19:51 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: * ext/flac/gstflactag.c: * ext/soup/gstsouphttpsrc.c: * ext/wavpack/gstwavpackdec.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavisubtitle.c: * gst/debugutils/gsttaginject.c: * gst/flv/gstflvdemux.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-read-common.c: * gst/multipart/multipartdemux.c: * gst/replaygain/gstrganalysis.c: * gst/wavparse/gstwavparse.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rgvolume.c: tag: Update for taglist/tag event API changes 2012-07-27 12:05:44 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: * gst/isomp4/isomp4-plugin.c: * gst/isomp4/qtdemux.c: qt(de)mux: pass private blob tags in a sample ... rather than a buffer, and the detailed info in the sample info rather than caps. 2012-07-27 11:31:13 +0200 Robert Swain * gst/videocrop/gstvideocrop.c: videocrop: Don't return NULL from _transform_caps If _transform_caps () returns NULL, the basetransform _transform_caps tries to call gst_caps_is_subset () with a NULL subset which hits an assertion. 2012-07-27 11:26:18 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: obtain image type from the sample info 2012-07-27 11:25:49 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: remove extraneous _unref ... since we did not obtain a buffer ref from the GstSample. 2012-07-27 10:14:23 +0200 Robert Swain * ext/flac/gstflacenc.c: flacenc: Update to use GstSample tag setting API 2012-07-26 16:34:21 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: modify buffer data rather than buffer itself 2012-07-26 16:28:33 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: avoid leaking bytewriter instance 2012-07-26 16:04:23 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix timestamp adjustment and caps 2012-07-26 16:03:57 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix/simplify telecine state checks 2012-07-26 12:08:58 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Improve debug output 2012-07-26 12:08:36 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix low-latency pattern locking 2012-07-24 16:19:53 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: RFF should be ignored in deinterlace RFF only occurs on progressive frames in telecine sequences. For deinterlace, we don't want these repeated fields as we will simply be pushing the progressive frame and then moving on. However, we need to consider RFF in order to correctly identify patterns and adjust the timestamps. 2012-07-24 14:59:47 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Improve process logic The logic now works better if we filter orphans, then progressive, then telecine interlaced fields which need to be woven and fall through to interlace. Telecine interlaced fields will be regularly deinterlaced if there is no pattern lock for us to be sure that we have a telecine pattern. Telecine sequences that aren't 24fps progressive with RFF flags can't really be tested until fieldanalysis is ported. 2012-07-25 16:02:34 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: only set complete output caps once ... so as to avoid downstream complaints about missing streamheaders. 2012-07-25 15:29:04 +0200 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: also support S24_32 output 2012-07-25 15:28:14 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: pass correct parameters to encoder lib 2012-07-25 14:57:13 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: adjust to modified audioencoder getcaps helper API 2012-07-25 12:50:01 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: go and stay in the loop function on PLAY When we have a PLAY request, go into the LOOP function next. When we are looping, keep on looping until we are told otherwise. This fixed rtsp and TCP connections. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680551 2012-07-25 12:49:35 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: set caps after activating the pad 2012-07-25 12:49:07 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: h264depay: small cleanups 2012-07-25 10:08:52 +0200 Wim Taymans * gst/isomp4/gstrtpxqtdepay.c: xqtdepay: fix buffer refcount error After pushing the buffer into the adapter, we should not let the baseclass push it out anymore. This error was introduced while porting to 0.11. See https://bugzilla.gnome.org/show_bug.cgi?id=680540 2012-07-24 21:41:53 +0200 Stefan Sauer * gst/level/gstlevel.c: level: remove obsolete liboil comment 2012-07-24 21:11:18 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: push mode: increase segment accuracy following seek Conflicts: gst/matroska/matroska-demux.c 2012-07-24 16:41:51 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: perform proper KEY_UNIT seek also in push mode Conflicts: gst/matroska/matroska-demux.c 2012-07-24 19:04:39 +0100 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: don't crash dereferencing NULL error when leaving multicast group on shutdown Strangely enough, if we do pass an error variable to be filled, we no longer get an error on leaving. 2012-07-24 15:55:12 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: rearrange some checks to avoid NULL use 2012-07-24 15:38:24 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: use same fourcc to determine caps in determining uncompressed-ness Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673898 Conflicts: gst/avi/gstavidemux.c 2012-07-24 15:36:54 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: Revert "avidemux: Don't consider 0 fcc_handler as uncompressed." This reverts commit c6b9f5b25ab435669816a07049b0e5a8f01e09ca. fourcc GST_RIFF_rgb = 0 still leads to raw uncompressed rgb caps. See also https://bugzilla.gnome.org/show_bug.cgi?id=673898 2012-07-24 12:10:46 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix up example pipeline some more No more ffmpegcolorspace 2012-07-20 16:30:00 +0300 Sreerenj Balachandran * ext/jpeg/gstjpegdec.c: jpegdec: Fix the example gst-launch pipeline. 2012-07-24 12:33:33 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: avoid NULL access when checking subtitle Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680388 2012-07-24 12:22:08 +0200 Edward Hervey * gst/audioparsers/gstaacparse.c: aacparse: Reset parser when we have caps without codec_data This ensures the detection (and proper downstream caps settings) will actually happen when we have new incoming caps without codec_data. This was easily triggered by streams from matroskademux which initially provided caps with a constructed codec_data, but then pushed new caps without the codec_data once it detected the stream was adts. 2012-07-24 09:17:09 +0200 Wim Taymans * gst/videomixer/blend.c: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: * gst/videomixer/blendorc.orc: videomixer: prefix orc functions with video_mixer_orc_ 2012-07-24 09:13:48 +0200 Wim Taymans * gst/videobox/gstvideobox.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videobox/gstvideoboxorc.orc: videobox: prefix orc functions with video_box_orc_ 2012-07-23 18:51:00 +0200 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Update spec file with latest changes 2012-07-23 17:37:58 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: generate correct segment stream time Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680275 2012-07-23 16:42:56 +0200 Wim Taymans * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: rtp: always use buffer lists 2012-07-23 15:24:17 +0200 Patricia Muscalu * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: rtpmp4vpay: always enable buffer-lists 2012-07-23 15:22:24 +0200 Patricia Muscalu * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpjpegpay.h: rtpjpegpay: always enable buffer-lists 2012-07-23 15:49:04 +0200 Wim Taymans * configure.ac: * gst/deinterlace/gstdeinterlace.c: deinterlace: get frame flags correctly Also move the deinterlace plugin to ported status 2012-07-23 15:33:54 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: proper parse recovery after seek Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680427 2012-07-23 12:39:05 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: clear old segment event when requesting new one Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680283 2012-07-23 10:32:36 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: ext: Update for video base classes API changes 2012-07-23 08:49:07 +0200 Alban Browaeys * gst/wavparse/gstwavparse.c: wavparse: convert all non GST_FORMAT_BYTES to format bytes. Convert all non GST_FORMAT_BYTES to format bytes: fixes: GStreamer-CRITICAL **: gst_query_set_duration: assertion `format == g_value_get_enum (gst_structure_id_get_value (s, GST_QUARK (FORMAT)))' failed when playing more than one wav stream. gst-plugins-base/tests/icles/playback/test7 uri1.wav uri2.wav 2012-07-23 09:25:23 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Don't fail if more data then needed is available when parsing cue chunks Fixes bug #680328. 2012-07-23 09:22:20 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Some minor cleanup to the cue/labl parsing 2012-07-23 08:45:28 +0200 Sebastian Dröge * common: Automatic update of common submodule From 98e386f to 94ccf4c 2012-07-19 14:55:45 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: deinterlace: Port to 1.0 This requires the additional INTERLACED buffer flag recently added to -base 2012-07-20 15:18:46 +0200 Wim Taymans * gst/interleave/interleave.c: interleave: convert the output segment to time Convert the stored input segment to time before pushing it out. Conflicts: gst/interleave/interleave.c 2012-07-20 13:12:44 +0200 Wim Taymans * gst/interleave/interleave.c: * gst/interleave/interleave.h: interleave: try to fix segment handling Conflicts: gst/interleave/interleave.c 2012-07-20 15:28:21 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Non-update seeks should still make sure that reverse playback status is reset Conflicts: gst/matroska/matroska-demux.c 2012-07-20 15:18:21 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Properly initialize from_offset and from_time 2012-07-20 14:25:43 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: We need an index and index entry for reverse playback Reverse playback does not work with index-less files yet. 2012-07-20 14:10:41 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: clean up push mode segment handling Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680277 2012-07-20 13:35:29 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: properly transform incoming segment event ... which is really useful for proper push mode seeking. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680278 2012-07-20 11:07:58 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: Fix reverse playback for seeks without stop position Conflicts: gst/matroska/matroska-demux.c gst/matroska/matroska-demux.h 2012-07-20 10:48:34 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Only take the stream_start_time into account for SET seeks For other seeks the stream_start_time is already added to the segment values. Conflicts: gst/matroska/matroska-demux.c 2012-07-08 20:36:22 +0300 Anton Belka * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: Add TOC support Add support for: * Cue Chunk * Associated Data List Chunk * Label Chunk https://bugzilla.gnome.org/show_bug.cgi?id=677306 2012-05-09 15:58:16 +0200 Maria Giovanna Chiossa * gst/rtsp/gstrtspsrc.c: rtspsrc: also set UDP buffer size in multicast Also set the UDP buffer size in multicast mode. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675448 2012-07-18 23:43:59 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: fix header parsing in push mode Fix 'break' that got warped to the wrong place, probably as part of a merge. Fixes GST_IS_BUFFER criticals in parse_idit() when being accidentally passed a NULL buffer because of the missing break. gst-launch-1.0 playbin uri=http://docs.gstreamer.com/media/sintel_trailer-480i.avi 2012-07-18 22:47:22 +0200 Alban Browaeys * configure.ac: * ext/soup/gstsouphttpsrc.c: soup: deprecated soup_message_headers _get -> _get_one https://bugzilla.gnome.org/show_bug.cgi?id=680206 2012-07-18 18:27:40 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: jpeg/png: Call video_decoder_negotiate() 2012-07-18 17:57:59 +0200 Wim Taymans * gst/debugutils/gstpushfilesrc.c: update for ghostpad changes 2012-07-18 11:36:27 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Pass seek rate to upstream seek events in push mode Fixes bug #679435. Conflicts: gst/matroska/matroska-demux.c 2012-07-17 16:39:02 +0200 Wim Taymans * gst/dtmf/gstrtpdtmfdepay.c: update for RTP buffer api changes 2012-07-17 16:38:27 +0200 Wim Taymans * gst/isomp4/gstrtpxqtdepay.c: * gst/rtp/gstasteriskh263.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpsession.c: * gst/rtsp/gstrtpdec.c: update for RTP buffer api changes 2012-07-16 11:07:44 +0200 Patricia Muscalu * gst/rtp/gstrtph264pay.c: rtph264pay: use buffer lists Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994 2012-07-17 10:01:54 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Fix parsing of ISRC from the cuesheets 2012-07-05 14:15:25 +0300 Anton Belka * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: add TOC support Add support embedded cuesheets in flac files. Parsing METADATA_BLOCK_CUESHEET as TOC. https://bugzilla.gnome.org/show_bug.cgi?id=540891 2012-07-13 14:43:31 +0200 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: avoid some more frame misparsing by additional header sanity check ... using a required constant blocking_strategy bit. https://bugzilla.gnome.org/show_bug.cgi?id=679807 2012-07-13 13:51:48 +0200 Edward Hervey * ext/dv/gstdvdemux.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: demux: Push STREAM_START event when needed 2012-07-11 13:10:07 +0200 Stefan Sauer * gst/isomp4/gstqtmux.c: qtmux: avoid warning if both ts are equal 2012-07-11 12:28:23 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: check the right size when warning about too large udp packets What matters is the total size, not the size of any of the individual memory chunks that make up the packet. 2012-07-10 14:38:21 +0200 Wim Taymans * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosink.h: autodetect: proxy ts-offset properties Proxy the ts-offset property in the audio*sink elements. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679343 2012-07-09 16:27:10 +0200 Wim Taymans * gst/isomp4/qtdemux.c: * sys/v4l2/gstv4l2bufferpool.c: fix for allocator API changes 2012-07-09 12:22:02 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: * gst/matroska/matroska-demux.c: * gst/wavparse/gstwavparse.c: update for riff field rename 2012-05-21 13:54:51 +0200 Mathias Hasselmann * tests/check/Makefile.am: tests: drop redundant elements_level_LDADD line https://bugzilla.gnome.org/show_bug.cgi?id=676302 2012-07-08 13:30:34 +0100 Tim-Philipp Müller * tests/check/elements/jpegdec.c: tests: minor jpegdec clean-ups and fixes Fix race condition in eos checking and a leak. And build pipeline without parse_launch. 2012-05-21 13:53:54 +0200 Mathias Hasselmann * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/jpegdec.c: * tests/files/image.jpg: tests: Add some basic tests for jpegdec https://bugzilla.gnome.org/show_bug.cgi?id=676302 2012-07-08 00:08:55 +0100 Tim-Philipp Müller * gst/dtmf/gstdtmfsrc.c: dtmfsrc: pass unhandled non-custom events to the base class https://bugzilla.gnome.org/show_bug.cgi?id=666626 2012-07-06 19:11:02 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: rtph264pay: avoid some relocations 2012-07-06 14:49:18 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmp4vpay.c: rtpmp4vpay: remove deprecated send-config property Use config-interval instead. 2012-07-06 14:42:19 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: remove deprecated "byte-stream" and "access-unit" properties These will be picked automatically based on downstream caps now, so if you want the depayloader to output a specific format, make sure the element downstream advertises that preference or use a capsfilter after the depayloader to force it. 2012-07-06 14:13:54 +0100 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: rtph264pay: remove deprecated and non-functional "profile-level-id" property This is now optionally taken from downstream caps, so can be specified via a capsfilter after the payloader. 2012-07-06 15:07:51 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: aacparse: perform additional sanity check before confirming ADTS format ... and tweak confusing debug message. 2012-07-06 15:29:14 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: aacparse: remove unhelpful stray debug message 2012-07-06 13:16:00 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpsession.c: rtpsession: remove deprecated and unused "ntp-ns-base" property 2012-07-06 12:57:20 +0100 Tim-Philipp Müller * gst/isomp4/gstqtmux-doc.c: docs: update isomp4 docs for gppmux -> 3gppmux change as well 2012-07-06 12:54:02 +0100 Tim-Philipp Müller * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * tests/check/pipelines/tagschecking.c: isomp4: remove gppmux, which was deprecated in favour of 3gppmux 2012-07-06 12:49:54 +0100 Tim-Philipp Müller * gst/smpte/gstsmpte.c: smtp: remove deprecated "fps" property 2012-07-06 12:46:30 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: remove deprecated and unused "autoscan" property Replaced by boundary=NULL. 2012-07-06 09:07:41 +0100 Tim-Philipp Müller * gst/rtp/gstrtph263ppay.c: * tests/check/elements/rtp-payloading.c: rtph263ppay: accept any h263 input unless downstream forces specific requirements rtph263ppay should accept any input compatible with its sink template caps if it just outputs to e.g. udpsink or fakesink. rtph263ppay ! rtph263pdepay should also work with any compatible input. This would fail before with not-negotiated errors because the get_caps function would see the encoding-name in the depayloader's template caps and default to baseline H.263 because there's no profile/level information in those caps, which is the right thing to do if downstream has filtercaps from an SDP, but not if those fields are absent because they can be anything like with the depayloader's template caps. Makes videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink work. 2012-07-05 22:57:05 +0100 Tim-Philipp Müller * tests/check/elements/rtp-payloading.c: tests: fix h263p payload ! depayload unit test Need to add h263version field to input caps since the payloader sink get_caps function will contain it in the the caps, and the stricter caps subset check requires this to be present in the input caps as well then. 2012-07-06 11:50:50 +0200 Wim Taymans * ext/aalib/gstaasink.c: * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngenc.c: * sys/v4l2/gstv4l2sink.c: update for query api changes 2012-07-06 11:26:46 +0200 Wim Taymans * ext/dv/gstdvdec.c: * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * gst/rtp/gstrtpvrawdepay.c: * sys/v4l2/gstv4l2src.c: update for query api changes 2012-07-06 11:02:24 +0200 Wim Taymans * ext/aalib/gstaasink.c: * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngenc.c: * sys/v4l2/gstv4l2sink.c: update for allocation query changes 2012-07-05 15:14:33 +0100 Tim-Philipp Müller * tests/check/elements/rgvolume.c: tests: fix rgvolume unit test event handling Must flush after EOS before sending more buffers or another EOS event, or the event or buffer will be rejected. Also send a SEGMENT event at the start of each stream for good measure. 2012-07-05 13:13:09 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/rtsp/gstrtspsrc.c: * gst/wavparse/gstwavparse.c: gst: Implement segment-done event 2012-07-05 12:35:49 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Remove the TOC query handling 2012-07-04 19:52:22 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-read-common.c: matroska: Update for new GstToc API TOC support in matroskamux is disabled for now as it was broken anyway. 2012-07-04 23:57:18 +0100 Tim-Philipp Müller * tests/check/elements/rganalysis.c: tests: fix rganalysis unit test event handling Must flush after EOS before sending more buffers or another EOS event, or the event or buffer will be rejected. Also send a SEGMENT event at the start of each stream for good measure. 2012-07-04 18:58:46 +0100 Tim-Philipp Müller * gst/imagefreeze/gstimagefreeze.c: imagefreeze: clear 0 DTS on buffers output, as sinks will prefer DTS over PTS for syncing Since the initial decoded still image buffer will have dts=pts=0, and we only set PTS on buffers we push out, all buffers pushed out would have a DTS of 0. Sinks, however, will prefer DTS over PTS if both are set, and will therefore always see a timestamp of 0 no matter what the PTS is set to. Fixes unit test too. 2012-07-04 20:59:03 +0400 Руслан Ижбулатов * sys/directsound/gstdirectsoundsink.c: directsoundsink: Fix query function implementation; more debugging 2012-07-04 19:41:52 +0400 Руслан Ижбулатов * sys/directsound/gstdirectsoundsink.c: directsoundsink: Fix spec stuff in directsoundsink 2012-05-31 19:22:47 +0200 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: directsoundsink: fix access to invalid pointer in set_volume 2012-06-13 12:12:39 +0200 Sebastian Dr=C3=B6ge * sys/directsound/gstdirectsoundsink.c: directsoundsink: Fix caps leaks 2012-05-29 11:37:59 +0000 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: directsoundsink: fix acceptcaps check 2012-05-25 10:14:57 +0000 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: directsoundsink: use helper function to check for spdif formats 2012-05-25 10:19:09 +0000 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: directsoundsink: add support for DTS 2012-05-08 16:23:42 +0200 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: directsoundsink: force 48000 kHz force AC-3 over spdif 2012-07-04 17:42:49 +0400 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: directsoundsink: add support for ac-3 over spdif 2012-07-04 12:37:40 +0100 Tim-Philipp Müller * tests/check/elements/deinterlace.c: tests: disable deinterlace test for now, element still needs to be ported But leave it active and print a FIXME. Porting is in progress. 2012-07-03 19:38:39 +0100 Tim-Philipp Müller * gst/interleave/deinterleave.c: deinterleave; downgrade caps change failure debug message Add some more info and downgrade to warning, so it doesn't look like the unit test failed. 2012-07-03 17:52:11 +0100 Tim-Philipp Müller * gst/audiofx/audiopanorama.c: audiopanorama: fix negotiation and unit test Must remove a possibly-fixed channel-mask field if we're going to set unfixed channels on the structure, or a different channel count. 2012-07-03 17:26:26 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Only push the TOC event, the message is handled by the sinks 2012-07-03 12:47:58 +0900 Javier Jardón * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/icles/gdkpixbufsink-test.c: tests: do not use deprecated gtk+ symbols https://bugzilla.gnome.org/show_bug.cgi?id=679301 2012-07-03 09:27:17 +0100 Tim-Philipp Müller * configure.ac: configure: require Gtk+ 3.0 for tests/examples 2012-07-03 12:57:18 +0900 Javier Jardón * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: rtp: remove some outdated comments https://bugzilla.gnome.org/show_bug.cgi?id=679301 2012-06-29 11:51:30 +0100 Tim-Philipp Müller * sys/osxvideo/osxvideosink.m: osxvideosink: default to force-aspect-ratio=true 2012-06-28 20:03:05 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: add push mode support https://bugzilla.gnome.org/show_bug.cgi?id=656317 2012-06-28 11:29:55 +0200 David Corvoysier * gst/isomp4/qtdemux.c: isomp4: Try to seek upstream before processing seek push event When it receives a seek in push mode, the qtdemux should first try to push the event upstream, and only if upstream fails fall back to its own seek logic. 2012-06-28 11:47:20 +0200 David Corvoysier * gst/isomp4/qtdemux.c: isomp4: Allow duration queries to be forwarded upstream When receiving a duration query for TIME format, try to query upstream, and only if upstream fails fall back to qtdemux duration handling. 2012-06-28 11:59:11 +0200 Wim Taymans * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: cleanups Use the caps properties for alignment and format. Remove some old properties, we always want to use bufferlists when we can now. 2012-06-28 11:32:03 +0200 Wim Taymans * gst/rtp/gstrtph264pay.c: h264pay: prefer AVC, it's easier to parse etc 2012-06-27 09:09:32 +0200 Wim Taymans * ext/jpeg/gstjpegenc.c: jpegenc: mark all output frames as keyframes 2012-06-26 18:48:11 +0100 Tim-Philipp Müller * gst/matroska/matroska-read-common.c: matroska: update for GstToc API additions 2012-06-26 17:04:41 +0200 Wim Taymans * gst/matroska/matroska-demux.c: matroska: set interlace-mode 2012-06-26 13:19:02 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: improve debug 2012-06-26 13:02:13 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: Revert "v4l2: free kernel buffers before allocating new ones" This reverts commit 1b09bc609a578e731f0dbc8f6e698e25d8f4c5f8. Seems to make libv4l2 complain, maybe because we call REQBUFS with 0 buffers before we allocated buffers. 2012-06-26 12:07:47 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: free kernel buffers before allocating new ones See https://bugzilla.gnome.org/show_bug.cgi?id=670257 2012-06-26 12:07:29 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2src: improve debug 2012-06-26 11:14:59 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: setup strides and offsets for all planes 2012-06-25 20:11:53 +0100 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroska-mux: update for GstTocSetter changes 2012-06-25 13:31:16 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Return FALSE from queries if we can't answer POSITION/DURATION queries 2012-06-21 17:15:11 +0300 Anton Belka * gst/matroska/matroska-demux.c: matroskademux: Return FALSE from TOC query if no TOC exists instead of an empty TOC 2012-06-24 22:51:16 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-read-common.c: matroska: update for GstToc API changes 2012-06-23 14:57:28 +0100 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: update for gst_element_make_from_uri() changes 2012-06-20 12:31:01 +0200 Wim Taymans * tests/check/elements/flvdemux.c: * tests/check/elements/flvmux.c: * tests/check/elements/id3demux.c: update for bus api changes 2012-06-20 10:33:42 +0200 Wim Taymans * ext/dv/gstdvdemux.c: * gst/avi/gstavidemux.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: * gst/wavparse/gstwavparse.c: update for task api change 2012-06-20 09:59:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: update for clock api changes 2012-06-19 12:15:33 +0200 Josep Torra * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: osxaudiosink: respect the prefered channel layout In OSX is allowed to configure the default audio output device, prefered channel layout and speaker positions through the tool "Audio MIDI Setup". 2012-04-30 22:59:58 +0200 Matej Knopp * gst/matroska/matroska-demux.c: matroska-demux: Send gap events for subtitle streams 2012-06-17 01:00:40 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: fix up docs for 0.11 2012-06-16 23:29:41 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: small uri handler fixup and some more docs Get URI location using gst_uri_get_location(), so any escaped bits get unescaped. https://bugzilla.gnome.org/show_bug.cgi?id=609049 2012-06-17 00:59:21 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: re-port to 0.11 2012-06-16 19:06:25 +0100 Bastien Nocera * gst/multifile/gstsplitfilesrc.c: splitfilesrc: Implement splitfile:// URI scheme https://bugzilla.gnome.org/show_bug.cgi?id=609049 Conflicts: gst/multifile/gstsplitfilesrc.c 2012-06-14 10:43:56 +0200 Wim Taymans * gst/rtp/gstrtptheoradepay.c: theoradepay: fix buffer memory The memory was added to the input buffer instead of the output buffer. 2012-06-13 13:36:45 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't reset time in flush-stop Don't reset the time in flush-stop. Live sources can do this flush in the playing state and so the pipeline will never have a chance to update the base_time of the elements, which only happens when going from paused to playing. 2012-06-12 12:42:31 +0200 Josep Torra * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxcoreaudio.h: * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: osxaudiosink: Add support for SPDIF output A big refactoring to allow passthrough AC3/DTS over SPDIF. Several random cleanups and minor fixes. 2011-09-01 15:41:26 +0100 Vincent Penquerc'h * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: send QoS messages when dropping a frame https://bugzilla.gnome.org/show_bug.cgi?id=657941 2012-06-12 16:05:40 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Rework the async state handling Always send the flushing events to the udp elements now that basesrc supports this. This makes sure a segment event is sent correctly after a flush. Keep track of the currently executing command and make it possible to specify what command you want to cancel when starting a new async command. See https://bugzilla.gnome.org/show_bug.cgi?id=677905 2012-06-11 18:24:20 +0200 Stefan Sauer * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/videomixer/videomixer2.c: childproxy: update api use 2012-06-11 12:54:27 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: always perform full seek if seek is flushing Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677838 2012-06-11 11:20:18 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: printf format fix for long -> int change 2012-06-08 20:38:34 +0200 Hans de Goede * sys/v4l2/gstv4l2object.c: v4l2object: Don't probe UVC devices for being interlaced UVC devices are never interlaced, and doing VIDIOC_TRY_FMT on them causes expensive and slow USB IO, so don't probe them for interlaced. This shaves 2 seconds of the startup time of cheese with a Logitech Webcam Pro 9000. Signed-off-by: Hans de Goede Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677722 2012-06-09 16:53:54 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: debug: change rndbuffersize properties from long to int These should all be int instead of long, to avoid bugs when passing these as varargs with g_object_set(), and there was no reason to use long in the first place here. Fixes FIXME. 2012-06-08 15:54:42 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/monoscope/gstmonoscope.c: * gst/rtsp/gstrtpdec.c: elements: Use gst_pad_set_caps() instead of manual event fiddling 2012-06-08 15:04:59 +0200 Edward Hervey * common: Automatic update of common submodule From 03a0e57 to 98e386f 2012-06-08 10:11:12 +0200 Wim Taymans * ext/flac/gstflacenc.c: * ext/wavpack/gstwavpackenc.c: * gst/audioparsers/gstwavpackparse.c: * sys/oss4/oss4-audio.c: * tests/check/elements/interleave.c: update for audio api change 2012-06-07 16:12:34 +0200 Sebastian Dröge * configure.ac: Back to development === release 0.11.92 === 2012-06-07 16:12:22 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.11.92 2012-06-07 16:11:17 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2012-06-07 15:03:02 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: improve clock handling Post the notify outside of the pa_lock to avoid a deadlock caused by basesrc calling get_time with the object lock. Reset the clock on connect. Post clock-lost and clock-provide messages. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673977 Conflicts: ext/pulse/pulsesrc.c 2012-04-12 13:21:17 +0300 Mohammed Sameer * ext/pulse/pulsesrc.c: Better GstClock for pulsesrc This clock uses the actual stream time (pa_stream_get_time) to get a more accurate timestamp. Conflicts: ext/pulse/pulsesrc.c 2012-06-07 11:16:50 +0100 Vincent Penquerc'h * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: png: fix video state leaks 2012-06-07 11:16:37 +0100 Vincent Penquerc'h * ext/jpeg/gstjpegdec.c: jpegdec: fix video state leak 2012-06-07 12:11:14 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: only reset the manager object when we did a seek Only reset the manager object when we used a Range header, ie. when we did a seek. Otherwise we just paused and we can resume just fine. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677475 2012-06-06 16:13:29 +0200 Wim Taymans * tests/check/elements/rtpbin.c: tests: add test for rtpsession cleanup 2012-06-06 18:18:41 +0200 Edward Hervey * common: Automatic update of common submodule From 1fab359 to 03a0e57 2012-06-06 14:17:08 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Update for TOC event API change 2012-06-06 13:02:12 +0200 Wim Taymans * ext/dv/gstdvdemux.c: * ext/flac/gstflactag.c: * ext/soup/gstsouphttpsrc.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavisubtitle.c: * gst/debugutils/gsttaginject.c: * gst/flv/gstflvdemux.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-read-common.c: * gst/multipart/multipartdemux.c: * gst/replaygain/gstrganalysis.c: * gst/wavparse/gstwavparse.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rgvolume.c: update for tag event change 2012-06-06 13:00:58 +0200 Wim Taymans * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * tests/check/elements/aspectratiocrop.c: * tests/check/elements/videocrop.c: fix Y800 format 2012-06-01 01:19:35 -0300 Thiago Santos * configure.ac: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/osxvideosink.m: osxvideo: straightforward port to 0.11 2012-05-31 18:39:25 -0300 Thiago Santos * ext/libpng/gstpngdec.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpmp2tpay.c: Some printf variable format fixes The osx compiler complains about those 2012-06-05 09:18:12 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: Fix GstBaseParse::get_sink_caps() implementations They should take the filter caps into account and always return the template caps appended to the actual caps. Otherwise the parsers stop to accept unparsed streams where upstream does not know about channels, rate, etc. Fixes bug #677401. 2012-06-04 16:17:51 +0200 Wim Taymans * ext/jpeg/gstjpegdec.c: jpegdec: set colorimetry on output info 2012-06-04 08:10:15 +0200 Josep Torra * sys/osxaudio/gstosxringbuffer.c: osxaudiosink: Handle endianness correctly 2012-06-01 16:37:00 +0200 Josep Torra * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxringbuffer.c: osxaudiosink: Add support for int audio 2012-06-01 10:28:53 +0200 Edward Hervey * common: Automatic update of common submodule From f1b5a96 to 1fab359 2012-05-31 13:36:32 +0200 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: set the palette size correctly 2012-05-31 10:15:43 +0200 Michael Jones * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2vidorient.h: v4l2: add missing G_END_DECLS G_BEGIN_DECLS didn't have matching G_END_DECLS https://bugzilla.gnome.org/show_bug.cgi?id=677165 2012-05-31 13:08:19 +0200 Sebastian Dröge * common: Automatic update of common submodule From 92b7266 to f1b5a96 2012-05-31 10:26:27 +0200 Josep Torra * sys/osxvideo/osxvideosink.h: osxvideosink: Really fix the build on 10.5 The API that we use to run the Cocoa loop in another thread does not exist in 10.5 or earlier. 2012-05-26 12:21:18 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: fix race in starting the runloop thread Block gst_osx_video_sink_run_cocoa_loop until the loop thread has started and finished initializing NSApp. Fixes occasional warnings/crashes due to two threads going inside NSApp before finishLaunching had completed. 2012-05-30 16:03:55 +0200 Josep Torra * sys/osxvideo/osxvideosink.h: osxvideosink: Fix last commit to actually work MAC_OS_X_VERSION_10_6 is obviously not defined on 10.5. 2012-05-30 13:51:35 +0200 Sebastian Dröge * sys/osxvideo/Makefile.am: osxvideosink: Put the right flags in the right variable 2012-05-30 13:24:03 +0200 Sebastian Dröge * configure.ac: configure: Fix GST_OBJCFLAGS 2012-05-30 12:45:23 +0200 Sebastian Dröge * common: Automatic update of common submodule From ec1c4a8 to 92b7266 2012-05-30 12:43:37 +0200 Sebastian Dröge * sys/osxvideo/osxvideosink.h: osxvideosink: NSWindowDelegate is available in all OSX versions newer than 10.6 2012-05-30 12:40:57 +0200 Josep Torra * sys/osxvideo/osxvideosink.h: osxvideosink: Fix build with older OSX versions 2012-05-30 11:09:25 +0200 Sebastian Dröge * configure.ac: * sys/osxvideo/Makefile.am: configure: Add OBJC specific compiler flags See bug #643939. 2012-05-30 11:23:36 +0200 Sebastian Dröge * common: Automatic update of common submodule From 3429ba6 to ec1c4a8 2012-05-29 17:50:21 +0200 Wim Taymans * gst/videocrop/gstvideocrop.c: video: remove duplicate format 2012-05-29 16:52:02 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Post error message if EOS before pads were created Happens with some files with only headers 2012-05-28 15:22:26 +0200 Sebastian Dröge * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: png: Port to 0.11 again 2012-05-14 12:46:57 +0200 Jens Georg * ext/soup/gstsouphttpsrc.c: soup: Drop transferMode.dlna.org header Leave it to the application to decide on the header. No header at all is better than having the wrong header as DLNA mandates that a missing header has to be tolerated while a wrong header is an error. https://bugzilla.gnome.org/show_bug.cgi?id=676020 2012-04-07 09:52:09 +0200 Edward Hervey * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: png: Port to base video classes Conflicts: ext/libpng/gstpngdec.c ext/libpng/gstpngdec.h ext/libpng/gstpngenc.c ext/libpng/gstpngenc.h Reverted to 0.10, needs to be ported again. 2012-05-27 00:02:08 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: * gst/matroska/matroska-read-common.c: flv, matroska: don't use GstStructure API on tag lists 2012-05-26 11:57:16 +0200 Edward Hervey * gst/rtp/gstrtpmp2tdepay.c: rtpmp2tdepay: Only output integral mpeg-ts packets From RFC 2250 2. Encapsulation of MPEG System and Transport Streams ... For MPEG2 Transport Streams the RTP payload will contain an integral number of MPEG transport packets. To avoid end system inefficiencies, data from multiple small MTS packets (normally fixed in size at 188 bytes) are aggregated into a single RTP packet. The number of transport packets contained is computed by dividing RTP payload length by the length of an MTS packet (188). .... Since it needs to contain "an integral number of MPEG transport packets", a simple fix is to check that's the case, and strip off any leftover data. Fixes #676799 Conflicts: gst/rtp/gstrtpmp2tdepay.c 2012-05-24 20:43:16 +0200 Andoni Morales Alastruey * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: make sure all selectors are performed on the same thread When we are using a dedicated thread to run the main run loop we must make sure that all selectors are performed on this same thread. For instance if performSelectorOnMainThread is called from the real main thread, it will not go through the message queue and will be executed from the real main thread. By forcing the target thread, we ensure that all functions will be called either from the real main thread when the main run loop is running or from our thread spinning the main loop. 2012-05-24 16:09:54 +0200 Mathias Hasselmann * ext/jpeg/gstjpegdec.c: jpegdec: remove framerate The jpeg decoder doesn't need/care about the framerate to so it should not be in the caps. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676302 2012-05-24 13:08:35 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.m: osxvideosink: start the loop before calling [gstview haveSuperview] ...as haveSuperview requires the mainloop to be running 2012-05-24 13:08:13 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.m: osxvideosink: fix indentation 2012-05-22 16:47:36 +0200 Alessandro Decina * sys/osxvideo/Makefile.am: osxvideosink: enable running the cocoa main runloop in a thread 2012-05-22 16:45:28 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: add code to optionally run the cocoa main runloop in a separate thread Add a little hack to run the cocoa main runloop from a separate thread _when_ the main runloop is not being run (which means that the app doesn't use cocoa). Runloops are thread specific, so the hack boils down to getting the runloop for the main thread and setting it as the runloop for our dedicated thread. 2012-05-22 16:32:53 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.m: osxvideosink: reset app_started to FALSE when shutting down 2012-05-22 14:49:17 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.m: osxvideosink: rename cocoa runloop helper funcs 2012-05-22 14:26:13 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.m: osxvideosink: don't create application menus 2012-05-16 21:52:45 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: reset the embed property for backward compatilibity 2012-05-16 21:12:22 +0200 Andoni Morales Alastruey * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideosink: fix navigation when force-aspect-ratio is activated 2012-05-16 18:52:45 +0200 Andoni Morales Alastruey * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: add force-aspect-ratio property 2012-05-14 18:01:02 +0200 Andoni Morales Alastruey * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: start internal window if no view is provided 2012-05-14 14:27:58 +0200 Andoni Morales Alastruey * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.m: osxvideosink: implement the navigation interface 2012-05-11 18:24:08 +0200 Andoni Morales Alastruey * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osvideosink: create, destroy, resize and draw from the main thread 2012-04-19 08:37:28 +0200 Alessandro Decina * gst/matroska/matroska-demux.c: matroskademux: increase NEWSEGMENT accuracy after seeking demux->common.segment is populated during seek handling with the target start/stop positions. Don't override them when sending out a NEWSEGMENT. Conflicts: gst/matroska/matroska-demux.c 2012-04-19 08:31:00 +0200 Alessandro Decina * gst/matroska/matroska-demux.c: matroskademux: don't discard the incoming seek segment on push based seeking The incoming seek segment was being discarded leading to push based seeking being potentially inaccurate. 2012-05-23 18:12:24 +0200 Sebastian Rasmussen * common: common: Update so the plugin scanner changes are included Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676674 2012-05-23 18:07:35 +0200 Sebastian Rasmussen * configure.ac: configure: suppress some warnings when debug is disabled Warnings about unused variables should be suppressed if core has the debug system disabled. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676671 2012-05-24 09:29:25 +0100 Luis de Bethencourt * gst/rtp/gstrtph264pay.c: rtp: fix build issue in gstrtph264pay.c 2012-05-21 12:17:35 +0200 Jonas Holmberg * gst/rtp/gstrtph264pay.c: rtph264pay: Add unrestricted caps If there are no profile restrictions downstream, return caps with profile=constrained-baseline in the first structure and append unrestricted caps as the last structure. Fixes bug #672019 2012-05-24 09:57:31 +0200 Maria Giovanna Chiossa * gst/rtsp/gstrtspsrc.c: rtsp: add the Scale header when needed Setting GST_SEEK_FLAG_SKIP when sending a seek event in rtspsrc should set the "Scale" field in the rtsp PLAY header. Because the boolean "src->skip" is set after the call, "Speed" instead of "Scale" is always set. Move the assignment before issuing the _play request. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676618 2012-05-17 16:23:59 +0300 Sreerenj Balachandran * gst/videobox/gstvideobox.c: videobox: Fix the sample pipeline. 2012-05-22 12:35:04 +0400 Anton Novikov * gst/icydemux/gsticydemux.c: icydemux: warning if setting srcpad caps fails 2012-05-22 12:35:29 +0400 Anton Novikov * gst/icydemux/gsticydemux.c: icydemux: activate srcpad before setting caps Before gst_pad_set_active() is called, the pad has FLUSHING flag set, so setting the caps fails 2012-05-22 13:46:27 +0100 Luis de Bethencourt * ext/Makefile.am: * ext/libmng/Makefile.am: * ext/libmng/gstmng.c: * ext/libmng/gstmng.h: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.c: * ext/libmng/gstmngenc.h: mng: remove ext/libmng Port to 0.10 was never finished. Interest was lost. https://bugzilla.gnome.org/show_bug.cgi?id=324364 2012-05-18 16:37:04 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: fix assertion when handling a date tag as a string Date tags are GDate, not strings. Add a special case to convert it to the exif date format representation in string to avoid the assertion 2012-05-21 11:47:07 +0200 Sjoerd Simons * ext/pulse/pulsesrc.c: pulsesrc: Listen to source output events, not sink input 2012-05-18 12:53:44 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp2tpay.c: rtpmp2tpay: respect mtu and packet boundaries See #659915. 2012-05-18 11:10:46 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: jpeg: Remove dead code Conflicts: ext/jpeg/gstjpegdec.c 2012-05-18 11:05:35 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Fix compilation 2012-05-18 11:02:52 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: jpegdec: When dropping frames on EOS, flush out data Cleaner way of handling stray data 2012-05-17 09:34:03 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: Remove unused variable Conflicts: ext/jpeg/gstjpegdec.c 2012-05-17 09:33:18 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: jpegdec: Only parse for SOI when we didn't see it before 2012-05-17 09:31:41 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: jpegdec: Remember if we saw SOI and handle stray data on EOS 2012-05-15 20:58:25 +0000 Youness Alaoui * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Allow U and V components to use different quant tables if they contain the same data This allows some cameras (Logitech C920) that specify different quant tables but both with the same data, to work. Bug reported by Robert Krakora 2012-05-14 15:51:29 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix possible data corruption after seeking Consider a downstream element that may issue seeks in very short succession (e.g. queue2), depending on the access pattern of the downstream element (e.g. qtdemux with audio/video chunks interleaved so that there's always a sizeable gap between the current chunks for each stream). In this case, queue2 will maintain two ranges, and even when it serves a chunk from memory, it will switch ranges and make souphttpsrc seek to the end of the available data for that range, assuming that that's where we'll want to continue reading from next. This may lead to the following seek request pattern: - source reading position A - seek to B - now reading position still A, requested_postion is B - streaming thread to be restarted to continue from B - seek to A, before streaming thread had time to do the seek - do_seek() now sees reading position == seek position and returns early. - however, requested position is still B from the earlier seek request - streaming thread starts up, sees that a seek to B is pending and requests data from B from the server, while the GstBaseSrc segment has of course been updated/reset to position A, which was the last seek request. - we will now send data for position B and pretend that's the data from position A (via the newsegment event, etc.) - this causes data corruption Reproducible doing seek-emulated fast-forward/backward on 006648. 2012-05-16 09:12:55 +0200 Sebastian Dröge * configure.ac: configure: Require core/base 0.11.91 2012-01-13 18:09:50 -0500 Matej Knopp * .gitignore: .gitignore: add visual studio IDE files and OS X .DS_Store files https://bugzilla.gnome.org/show_bug.cgi?id=667899 2012-05-03 09:32:50 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpeg: Port to 0.11 again 2012-04-06 12:13:24 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpeg: Port jpegdec/jpegenc to base video classes Conflicts: ext/jpeg/gstjpegdec.c ext/jpeg/gstjpegdec.h ext/jpeg/gstjpegenc.c ext/jpeg/gstjpegenc.h Reverted to 0.10 versions for now, next port again. 2012-05-13 19:21:19 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-annodex.xml: * ext/Makefile.am: * ext/annodex/Makefile.am: * ext/annodex/gstannodex.c: * ext/annodex/gstannodex.h: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmldec.h: * ext/annodex/gstcmmlenc.c: * ext/annodex/gstcmmlenc.h: * ext/annodex/gstcmmlparser.c: * ext/annodex/gstcmmlparser.h: * ext/annodex/gstcmmltag.c: * ext/annodex/gstcmmltag.h: * ext/annodex/gstcmmlutils.c: * ext/annodex/gstcmmlutils.h: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: annodex: remove annodex plugin and CMML elements This never really took off and is most likely completely unused. If there is still a need for this, it should probably be done differently, perhaps inside oggdemux/mux. 2012-05-13 16:59:58 +0200 Sebastian Dröge * configure.ac: Back to development === release 0.11.91 === 2012-05-13 16:31:03 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * common: * configure.ac: * gst-plugins-good.doap: * win32/common/config.h: Release 0.11.91 2012-05-13 16:30:03 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2012-05-13 15:56:05 +0200 Sebastian Dröge * common: Automatic update of common submodule From dc70203 to 3429ba6 2012-05-09 15:14:55 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: only send flush-stop if it was a flushing seek 2012-05-09 12:54:11 +0200 Peter Seiderer * sys/v4l2/v4l2_calls.c: v4l2src: fix v4l2_std_id logging input.std is of type v4l2_std_id which is defined as 64-bit unsigned integer. Casting to uint means the higher bits, wich are used for the private video standards of the TI video capture/display driver for example, are lost. 2012-05-09 12:24:37 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: must send flush-stop after acquiring the stream lock Otherwise the streaming thread might just keep on going and we might never get the stream lock. 2012-05-09 11:15:21 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: port seeking code to 0.11 2012-05-08 19:07:04 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: add support for seeks Useful for e.g. filesrc ! rndbuffersize ! queue2 ! ... 2012-05-08 18:45:34 +0100 Tim-Philipp Müller * gst/debugutils/rndbuffersize.c: rndbuffersize: send SEGMENT event before pushing buffers Conflicts: gst/debugutils/rndbuffersize.c 2012-05-09 11:15:57 +0200 Wim Taymans * gst/interleave/interleave.c: interleave: fix compilation again 2012-01-13 10:49:43 +0100 Pascal Buhler * gst/rtpmanager/rtpsession.c: rtpsession: creation should be signaled before validation https://bugzilla.gnome.org/show_bug.cgi?id=667850 2012-05-04 15:20:47 -0300 Thiago Santos * ext/jpeg/gstjpegenc.c: jpegenc: do not proxy our filter caps downstream on caps queries Downstream likely won't accept video/x-raw and the caps query will return EMPTY caps. Instead, create a copy of the caps that has all structure names replaced by 'image/jpeg' Simple pipeline that shows the problem: gst-launch-1.0 videotestsrc num-buffers=1 ! "video/x-raw, \ width=(int)640, height=(int)480" ! videoscale ! jpegenc ! \ "image/jpeg, width=(int)800, height=(int)600" ! filesink \ location=/tmp/image.jpg 2012-05-02 21:17:43 +0200 Alban Browaeys * gst/isomp4/qtdemux.c: isomp4: set layout=interleaved on raw audio caps This fixes a not-negotiated error at least on mov files with twos audio with two channels and video dvcp. As playbin and gst-launch sample coming from the qtdemux.c file uses audioconvert and the latter require format interleaved. https://bugzilla.gnome.org/show_bug.cgi?id=675326 2012-05-02 21:49:56 +0400 Руслан Ижбулатов * sys/waveform/Makefile.am: waveform: No more gstinterfaces Fixes #675319 2012-05-02 20:14:24 +0400 Руслан Ижбулатов * sys/directsound/Makefile.am: directsound: No more gstinterfaces Fixes #675319 2012-05-01 18:58:03 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: change sink pad template name from sink_%d to sink_%u 2012-04-30 11:00:19 +0200 Wim Taymans * gst/interleave/interleave.c: interleave: handle EOS on all pads When all pads go to EOS immediately, we are not negotiated and our collected function is called (without any available data). Handle this case gracefully. Conflicts: gst/interleave/interleave.c 2012-04-30 10:59:31 +0200 Wim Taymans * gst/interleave/interleave.c: interleave: improve debugging 2012-05-01 13:31:51 +0200 Sebastian Dröge * sys/v4l2/gstv4l2src.c: v4l2src: Update for basesrc API changes 2012-04-30 23:57:28 +0100 Tim-Philipp Müller * gst/alpha/gstalpha.c: alpha: don't set up stuff before the input and output formats are known Fixes crash on startup. 2012-04-30 14:09:23 +0200 Peter Seiderer * gst/multifile/gstmultifilesink.c: multifilesink: don't write stream header twice for first file 2012-04-30 13:32:41 +0200 Peter Seiderer * gst/multifile/gstmultifilesink.c: multifilesink: fix buffer list size calculation in render_list Fix uninitialized 'size' variable in call to gst_buffer_list_foreach(). 2012-04-30 21:58:00 +0100 Luis de Bethencourt * gst/multifile/gstmultifilesrc.c: multifile: unnecessary size check 2012-04-30 21:30:56 +0100 Luis de Bethencourt * gst/avi/gstavidemux.c: avi: fix build errors fix redundant declarations and also style/indent issues 2012-04-26 12:47:27 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: implement forward snapping keyframe seeking Requires an index. 2012-04-26 12:46:11 +0100 Vincent Penquerc'h * gst/avi/gstavidemux.c: avi: implement forward snapping keyframe seeking In pull mode with an index. 2012-04-28 23:14:24 +0100 Tim-Philipp Müller * tests/check/elements/matroskamux.c: tests: fix matroskamux unit test after media type changes 2012-04-28 19:57:51 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: matroska: update for media type changes 2012-04-24 16:08:47 +0200 idc-dragon * gst/rtp/gstrtpceltdepay.c: celtdepay: calculate size correctly The summation was done wrong, causing the de-payloader to exit its loop too early, before all frames are processed. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674472 2012-04-24 15:57:19 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: improve debug 2012-04-24 15:34:57 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: start unmuted when requested When we explicitely set the mute property to FALSE, connect to pulseaudio with the PA_STREAM_START_UNMUTED flag set, otherwise pulseaudio will use its previously used value (which might start the stream muted). Fixes https://bugzilla.gnome.org/show_bug.cgi?id=672401 2012-04-25 09:41:46 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2: improve timestamp code Sample the pipeline clock and device clock closer to eachother to reduce jitter. Don't subtract the frame duration from the timestamp when we can use the device timestamps. Assume a delay of 1 frame in read-write mode. 2012-04-24 12:37:33 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2src.c: v4l2: use driver timestamps Use the drive timestamps for timestamping outgoing buffers. 2012-04-23 18:01:31 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2src.c: v4l2: Improve buffer management Query the amount of available buffers when doing set_config(). This allows us to configure the parent bufferpool with the number of buffers to preallocate. Keep track of the provided allocator and use it when we need to allocate a buffer in RW mode. When we are can not allocate the requested max_buffers amount of buffers, make sure we keep 2 buffers around in the pool and copy them into an output buffer. This makes sure that we always have a buffer to capture into. We also need to detect those copied buffers and unref them when they return to the pool. 2012-04-23 16:51:28 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: free the queued buffers Only free the queued buffers that we keep track of in our buffer array. for rw io-mode, we do allocate buffers but we don't keep track of them in the buffer array. 2012-04-23 16:10:17 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: mark memory as no-share We don't support sharing our mmapped memory so mark it as NO_SHARE. 2012-04-23 16:09:55 +0200 Wim Taymans * sys/v4l2/v4l2src_calls.c: v4l2: remove old unused file 2012-04-23 13:32:48 +0200 Wim Taymans * sys/v4l2/v4l2src_calls.c: v4l2: remove unused function 2012-04-11 12:42:17 +0100 Bastien Nocera * ext/soup/gstsouphttpsrc.c: soup: Handle icy and icyx URI schemes As handled by QuickTime (for icy), and Orban/Coding Technologies AAC/aacPlus Player (for icyx). See also: https://bugzilla.gnome.org/show_bug.cgi?id=394207 https://bugzilla.gnome.org/show_bug.cgi?id=403285 https://bugzilla.gnome.org/show_bug.cgi?id=673899 2012-04-23 10:03:19 +0300 Mart Raudsepp * sys/v4l2/gstv4l2src.c: docs: Add Since tag for new GstV4l2Src::prepare-format signal 2012-04-23 10:07:12 +0200 Chris Pankow * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Fix time-domain convolution for multichannel input Fixes bug #674025. 2012-04-21 11:08:51 +0200 Mark Nauwelaerts * po/POTFILES.in: po: remove some more non-existent files from the list 2012-04-21 10:05:45 +0400 Руслан Ижбулатов * po/POTFILES.in: po: Remove non-existent potfiles from the list Fixes #674518 2012-04-20 18:13:15 +0200 Mark Nauwelaerts * tests/icles/test-oss4.c: tests: oss4: limit test scope 2012-04-20 18:13:01 +0200 Mark Nauwelaerts * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * sys/oss4/Makefile.am: * sys/oss4/oss4-audio.c: * sys/oss4/oss4-audio.h: * sys/oss4/oss4-mixer-enum.c: * sys/oss4/oss4-mixer-enum.h: * sys/oss4/oss4-mixer-slider.c: * sys/oss4/oss4-mixer-slider.h: * sys/oss4/oss4-mixer-switch.c: * sys/oss4/oss4-mixer-switch.h: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-mixer.h: * sys/oss4/oss4-property-probe.c: * sys/oss4/oss4-property-probe.h: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-sink.h: * sys/oss4/oss4-source.c: * sys/oss4/oss4-source.h: oss4: port to 0.11 2012-04-20 18:12:54 +0200 Mark Nauwelaerts * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * sys/oss/Makefile.am: * sys/oss/gstossaudio.c: * sys/oss/gstosshelper.c: * sys/oss/gstosshelper.h: * sys/oss/gstossmixer.c: * sys/oss/gstossmixer.h: * sys/oss/gstossmixerelement.c: * sys/oss/gstossmixerelement.h: * sys/oss/gstossmixertrack.c: * sys/oss/gstossmixertrack.h: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss/gstosssrc.h: oss: port to 0.11 2012-04-20 16:49:56 +0200 Wim Taymans * gst/multipart/multipartdemux.c: multipartdemux: first activate pad then set caps 2012-04-20 13:35:15 +0200 Wim Taymans * gst/matroska/matroska-mux.c: matroskamux: set caps on srcpad Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674219 2012-04-19 14:16:01 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: update for video api change 2012-04-19 12:38:58 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: fix compilation on older v4l2 Fix compilation on systems where the H264 format is not defined. 2012-04-19 12:20:59 +0200 Sebastian Dröge * ext/dv/gstdvdec.c: * ext/raw1394/Makefile.am: * gst/rtp/gstrtpvrawpay.c: * gst/y4m/gsty4mencode.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: video: Update for libgstvideo API changes 2012-04-19 08:27:01 +0000 Youness Alaoui * sys/v4l2/gstv4l2object.c: * sys/v4l2/v4l2src_calls.c: v4l2src: Allow mpeg-ts cameras to negociate format This removes an ugly hack until the reason for the hack can be documented 2012-04-19 09:50:25 +0200 Sebastian Dröge * sys/v4l2/gstv4l2object.c: v4l2src: Fix merge 2012-04-19 09:40:53 +0200 Sebastian Dröge * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: v4l2src: Rename pre-set-format signal to prepare-format 2012-04-16 22:08:21 +0000 Youness Alaoui * sys/v4l2/gstv4l2object.c: v4l2src: Add H264 encoded stream support to the caps This is not enough to properly support H264 cameras, but it will allow an H264 stream to be generated by v4l2src using the default settings of the camera. If used with the pre-set-format signal, the H264 encoder can be fully configured. Conflicts: sys/v4l2/gstv4l2object.c 2012-04-16 22:06:21 +0000 Youness Alaoui * sys/v4l2/.gitignore: * sys/v4l2/gstv4l2-marshal.list: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: v4l2src: Adding a pre-set-format signal In order to support UVC H264 encoding cameras, an H264 Probe&Commit must happen before the normal v4l2 set-format. This new signal is meant to allow an external application or bin to do it. It also serves to expose the file descriptor used by v4l2src in case some custom ioctls need to be called. Conflicts: sys/v4l2/Makefile.am sys/v4l2/gstv4l2src.c sys/v4l2/v4l2src_calls.c 2012-04-18 17:09:45 +0200 Mark Nauwelaerts * configure.ac: * ext/raw1394/gst1394probe.c: * ext/raw1394/gst1394probe.h: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: dv1394: port to 0.11 2012-04-17 15:14:27 +0200 Mark Nauwelaerts * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: * gst/interleave/interleave.c: * gst/interleave/interleave.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixer2pad.h: collectpads2: rename to collectpads 2012-04-16 16:37:49 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-mux.c: * gst/smpte/gstsmpte.c: * gst/videomixer/videomixer2.c: misc: chain up to collectpads event handler 2012-04-16 09:09:11 +0200 Sebastian Dröge * common: Automatic update of common submodule From 6db25be to dc70203 2012-04-15 22:49:47 +0100 Tim-Philipp Müller * ext/shout2/gstshout2.c: shout2: update for ogg media type changes 2012-04-13 16:54:53 +0200 Mark Nauwelaerts * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: smpte: use some more boilerplate 2012-04-13 16:54:50 +0200 Mark Nauwelaerts * gst/flx/gstflxdec.c: flxdec: improve segment handling ... to send a proper TIME segment downstream. 2012-04-13 16:54:46 +0200 Mark Nauwelaerts * configure.ac: * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: flxdec: port to 0.11 2012-04-13 16:54:42 +0200 Mark Nauwelaerts * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: adjust to deprecated GMutex setup 2012-04-13 16:54:38 +0200 Mark Nauwelaerts * configure.ac: * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: port to 0.11 2012-04-13 16:54:31 +0200 Mark Nauwelaerts * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/smpte/gstsmptealpha.c: alpha, smpte: adjust to removed color-matrix caps field 2012-04-13 16:27:34 +0200 Víctor Manuel Jáquez Leal * sys/v4l2/Makefile.am: v4l2: ensure autogenerated files are created The tuner marshal and enumtypes are autogenerated, and they need to be created before the compilation of gstv4l2tuner.c This patch adds the automake instruction for ensuring the autogeneration of those files previous the compilation. 2012-04-13 13:41:44 +0200 Sebastian Dröge * autogen.sh: * configure.ac: configure: Modernize autotools setup a bit Also we now only create tar.bz2 and tar.xz tarballs. 2012-04-13 13:37:10 +0200 Sebastian Dröge * common: Automatic update of common submodule From 464fe15 to 6db25be 2012-04-13 13:04:12 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * ext/pulse/Makefile.am: * ext/pulse/plugin.c: * ext/pulse/pulsemixer.c: * ext/pulse/pulsemixer.h: * ext/pulse/pulsemixerctrl.c: * ext/pulse/pulsemixerctrl.h: * ext/pulse/pulsemixertrack.c: * ext/pulse/pulsemixertrack.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * gst/rtsp/Makefile.am: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2videooverlay.c: * sys/v4l2/gstv4l2videooverlay.h: * sys/v4l2/tuner-marshal.list: * sys/v4l2/tuner.c: * sys/v4l2/tuner.h: * sys/v4l2/tunerchannel.c: * sys/v4l2/tunerchannel.h: * sys/v4l2/tunernorm.c: * sys/v4l2/tunernorm.h: * tests/check/Makefile.am: * tests/examples/pulse/Makefile.am: * tests/icles/Makefile.am: * tests/icles/v4l2src-test.c: Update everything for the removal of the interface library and mixer/tuner interfaces 2012-04-12 15:50:16 +0200 Edward Hervey * gst/rtp/gstrtpmparobustdepay.c: rtp: Use unchecked variant of GstByteWriter where applicable The size was checked before 2012-04-12 15:49:44 +0200 Edward Hervey * gst/matroska/ebml-read.c: * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: matroska: Check return value of GstByteReader/Writer 2012-04-12 15:48:57 +0200 Edward Hervey * gst/isomp4/atoms.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_dump.c: isomp4: Check return value of GstByteWriter And use unchecked variant of GstByteReader where applicable 2012-04-12 15:48:00 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Use unchecked variant of GstByteReader We know there's at least 7 bytes (checked above) 2012-04-12 15:47:49 +0200 Edward Hervey * gst/avi/gstavimux.c: avi: Check return value of GstByteWriter 2012-04-12 15:47:24 +0200 Edward Hervey * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: Check return value of GstBitReader/GstByteReader 2012-04-12 11:57:59 +0100 uraeus * gst-plugins-good.spec.in: Add interleave plugin to spec file 2012-04-12 11:19:01 +0200 Sebastian Dröge * configure.ac: Back to development === release 0.11.90 === 2012-04-12 10:27:31 +0200 Sebastian Dröge * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * gst/deinterlace/tvtime-dist.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videomixer/blendorc-dist.c: * win32/common/config.h: Release 0.11.90 2012-04-12 10:26:52 +0200 Sebastian Dröge * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2012-04-11 00:19:30 +0400 Руслан Ижбулатов * ext/jpeg/gstjpegenc.c: Fix format string Fixes #673859 2012-04-11 00:19:16 +0400 Руслан Ижбулатов * sys/waveform/gstwaveformsink.c: Remove unused variable Fixes #673859 2012-04-10 11:57:53 +0200 Mark Nauwelaerts Merge remote-tracking branch 'origin/0.10' Conflicts: gst/flv/gstflvdemux.c gst/matroska/matroska-demux.c 2012-04-10 11:37:48 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: some more segment handling tweaking 2012-04-10 00:51:41 +0100 Tim-Philipp Müller * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairorender.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/pulse/pulsemixer.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstsplitfilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: Use new gst_element_class_set_static_metadata() 2012-04-10 00:47:44 +0100 Tim-Philipp Müller * ext/twolame/gsttwolamemp2enc.c: Use new gst_element_class_set_static_metadata() 2012-04-10 00:47:44 +0100 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: Use new gst_element_class_set_static_metadata() 2012-04-09 12:55:34 +0100 Tim-Philipp Müller * tests/check/pipelines/simple-launch-lines.c: tests: disable simple smokeenc/dec launch lines test Disable test for smoke elements, which aren't ported yet (and maybe shouldn't be ported). 2012-04-09 00:14:48 +0100 Tim-Philipp Müller * gst/interleave/interleave.c: * gst/interleave/interleave.h: * tests/check/elements/interleave.c: interleave: make channel-poisitions property a GValueArray again Or perhaps it should just be a guint64 channel mask, which would be nicer in C, but more awkward for bindings (even more so since we can't add a flags type for it, since that only supports guint size flags). Fixes wavenc unit test. https://bugzilla.gnome.org/show_bug.cgi?id=669643 2012-04-06 16:03:47 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: cleanly initialize and set needed segment Fixes #673165. 2012-04-05 17:17:22 -0400 Nicolas Dufresne * gst/flv/gstflvdemux.c: flvdemux: Fix threading issue in index handling 2012-04-06 09:13:31 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Don't use static variables to hold index associations This not really threadsafe in any way. 2012-04-05 19:17:48 +0200 Mark Nauwelaerts * tests/check/elements/flvmux.c: * tests/check/elements/interleave.c: tests: make few tests more valgrind-friendly 2012-04-05 19:17:42 +0200 Mark Nauwelaerts * configure.ac: * tests/check/elements/deinterleave.c: (de)interleave: fix ported unit test and enable as ported 2012-04-05 19:17:38 +0200 Mark Nauwelaerts * tests/check/elements/cmmldec.c: tests: cmmldec: adjust to tag events no longer posted on bus by element 2012-04-05 19:17:29 +0200 Mark Nauwelaerts * gst/udp/gstudpsrc.c: updsrc: clear error 2012-04-05 18:42:53 +0200 Sebastian Dröge * common: Automatic update of common submodule From 7fda524 to 464fe15 2012-04-05 18:02:56 +0200 Sebastian Dröge * gst/audiofx/gststereo.c: gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 18:02:56 +0200 Sebastian Dröge * gst/dtmf/gstdtmf.c: gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:40:12 +0200 Sebastian Dröge * ext/twolame/gsttwolamemp2enc.c: gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:40:12 +0200 Sebastian Dröge * ext/lame/plugin.c: gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +0200 Sebastian Dröge * ext/aalib/gstaasink.c: * ext/annodex/gstannodex.c: * ext/cairo/gstcairo.c: * ext/dv/gstdv.c: * ext/flac/gstflac.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/jack/gstjack.c: * ext/jpeg/gstjpeg.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmng.c: * ext/libpng/gstpng.c: * ext/mikmod/gstmikmod.c: * ext/pulse/plugin.c: * ext/raw1394/gst1394.c: * ext/shout2/gstshout2.c: * ext/soup/gstsoup.c: * ext/speex/gstspeex.c: * ext/taglib/gsttaglibplugin.c: * ext/wavpack/gstwavpack.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audiofx.c: * gst/audioparsers/plugin.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautodetect.c: * gst/avi/gstavi.c: * gst/cutter/gstcutter.c: * gst/debugutils/gstdebug.c: * gst/debugutils/gstnavigationtest.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gsteffectv.c: * gst/equalizer/gstiirequalizer.c: * gst/flv/gstflvdemux.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/plugin.c: * gst/isomp4/isomp4-plugin.c: * gst/law/alaw.c: * gst/law/mulaw.c: * gst/level/gstlevel.c: * gst/matroska/matroska.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifile.c: * gst/multipart/multipart.c: * gst/replaygain/replaygain.c: * gst/rtp/gstrtp.c: * gst/rtpmanager/gstrtpmanager.c: * gst/rtsp/gstrtsp.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/plugin.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstudp.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstvideotemplate.c: * gst/videofilter/plugin.c: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundplugin.c: * sys/oss/gstossaudio.c: * sys/oss4/oss4-audio.c: * sys/osxaudio/gstosxaudio.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudio.c: * sys/v4l2/gstv4l2.c: * sys/waveform/gstwaveformplugin.c: * sys/ximage/gstximagesrc.c: gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 13:26:19 +0200 Sebastian Dröge * configure.ac: configure: Update version to 0.11.89.1 2012-04-04 20:06:58 +0200 Mark Nauwelaerts * tests/check/elements/qtmux.c: tests: qtmux: ensure initialized test buffer memory 2012-04-04 14:41:22 +0200 Sebastian Dröge * gst/dtmf/Makefile.am: gst: Update versioning 2012-04-04 14:38:53 +0200 Sebastian Dröge * ext/twolame/Makefile.am: gst: Update versioning 2012-04-04 14:38:53 +0200 Sebastian Dröge * ext/lame/Makefile.am: gst: Update versioning 2012-04-04 14:33:23 +0200 Sebastian Dröge * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/version.entities.in: * ext/aalib/Makefile.am: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/flac/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/jack/Makefile.am: * ext/jpeg/Makefile.am: * ext/libcaca/Makefile.am: * ext/libpng/Makefile.am: * ext/pulse/Makefile.am: * ext/raw1394/Makefile.am: * ext/soup/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/Makefile.am: * ext/wavpack/Makefile.am: * gst-plugins-good.spec.in: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/audioparsers/Makefile.am: * gst/auparse/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debugutils/Makefile.am: * gst/deinterlace/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flv/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/interleave/Makefile.am: * gst/isomp4/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/multifile/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/rtsp/Makefile.am: * gst/shapewipe/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * gst/y4m/Makefile.am: * pkgconfig/Makefile.am: * pkgconfig/gstreamer-plugins-good-uninstalled.pc.in: * sys/directsound/Makefile.am: * sys/oss/Makefile.am: * sys/oss4/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxvideo/Makefile.am: * sys/sunaudio/Makefile.am: * sys/v4l2/Makefile.am: * sys/waveform/Makefile.am: * sys/ximage/Makefile.am: * tests/check/Makefile.am: * tests/examples/audiofx/Makefile.am: * tests/examples/cairo/Makefile.am: * tests/examples/pulse/Makefile.am: * tests/examples/spectrum/Makefile.am: * tests/icles/Makefile.am: gst: Update versioning 2012-04-04 12:10:45 +0200 Sebastian Dröge Merge remote-tracking branch 'origin/0.10' Conflicts: gst/matroska/matroska-demux.c gst/matroska/matroska-mux.c gst/matroska/matroska-read-common.c gst/matroska/matroska-read-common.h 2012-04-03 18:36:50 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegenc.c: jpegenc: plug template caps leak 2012-04-03 11:50:00 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: avi only knows about DTS Only set DTS on outgoing buffers unless we have a keyframe and then we can set the PTS to DTS as well. 2012-04-02 23:35:43 +0200 Stefan Sauer * gst/matroska/matroska-read-common.c: mkv: port toc changes to 0.11 2012-04-02 23:18:00 +0200 Stefan Sauer Merge branch '0.10' Conflicts: gst/matroska/matroska-demux.c gst/matroska/matroska-mux.c gst/matroska/matroska-read-common.c gst/matroska/matroska-read-common.h 2012-03-29 23:22:28 +0400 Alexander Saprykin * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroska: add GstToc support for muxer 2012-03-29 23:12:13 +0400 Alexander Saprykin * gst/matroska/matroska-demux.c: matroska: add support for GstToc in demuxer 2012-03-29 23:05:14 +0400 Alexander Saprykin * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: add chapter support in GstMatroskaReadCommon 2012-04-02 13:00:19 +0200 Sebastian Dröge * gst/goom2k1/lines.c: goom2k1: Fix 'may be used uninitialized in this function' compiler warning 2012-04-02 11:13:09 +0200 Wim Taymans * gst/alpha/gstalphacolor.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: use transform_ip_on_passthrough 2012-03-31 15:43:49 +0200 Wim Taymans * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/videomixer/videomixer2.c: * tests/check/elements/equalizer.c: * tests/examples/equalizer/demo.c: * tests/icles/equalizer-test.c: update for child proxy api change 2012-03-30 18:13:08 +0200 Wim Taymans * ext/jpeg/gstjpegenc.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/flv/gstflvmux.c: * gst/isomp4/atoms.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsrc.c: * gst/y4m/gsty4mencode.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/ximageutil.c: * tests/check/elements/deinterleave.c: * tests/check/elements/interleave.c: update for buffer api change 2012-03-30 12:53:44 +0200 Sebastian Dröge * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: speexenc: Use new gst_audio_encoder_set_headers() API 2012-03-30 12:18:45 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: * ext/speex/gstspeexenc.c: * ext/wavpack/gstwavpackenc.c: ext: Update for GstAudioEncoder API changes 2012-03-29 23:22:28 +0400 Alexander Saprykin * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroska: add GstToc support for muxer 2012-03-29 23:12:13 +0400 Alexander Saprykin * gst/matroska/matroska-demux.c: matroska: add support for GstToc in demuxer 2012-03-29 23:05:14 +0400 Alexander Saprykin * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: add chapter support in GstMatroskaReadCommon 2012-03-29 17:22:23 +0200 Mark Nauwelaerts * tests/check/pipelines/wavpack.c: tests: wavpack: fewer buffers are also adequate and more convenient 2012-03-29 17:22:19 +0200 Mark Nauwelaerts * tests/check/elements/videocrop.c: tests: videocrop: unmap video frame and unref caps 2012-03-29 17:22:04 +0200 Mark Nauwelaerts * tests/check/elements/audiowsincband.c: tests: audiowsincband: unmap examined output buffers 2012-03-29 17:21:53 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: plug ref leak 2012-03-29 17:21:50 +0200 Mark Nauwelaerts * gst/audiofx/audiopanorama.c: audiopanorama: fix supported template caps and sample processing 2012-03-29 17:21:43 +0200 Mark Nauwelaerts * gst/alpha/gstalphacolor.c: alphacolor: plug structure leak 2012-03-29 16:04:26 +0100 uraeus * gst-plugins-good.spec.in: Update spec file with latest ported plugins 2012-03-29 15:03:09 +0200 Sebastian Dröge Merge remote-tracking branch 'origin/0.10' Conflicts: configure.ac 2012-03-28 16:26:56 +0200 Mark Nauwelaerts * tests/check/pipelines/tagschecking.c: tests: tagschecking: muxers need TIME format 2012-03-28 16:26:15 +0200 Mark Nauwelaerts * tests/check/pipelines/flacdec.c: tests: flacdec: needs flacparse nowadays 2012-03-28 14:49:03 +0200 Mark Nauwelaerts * ext/wavpack/gstwavpackenc.c: wavpackenc: query downstream for BYTE seeking support 2012-03-28 14:48:46 +0200 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: query downstream for BYTE seeking support 2012-03-28 14:46:03 +0200 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: clean up obsolete log statement 2012-03-28 12:49:19 +0200 Wim Taymans * ext/mikmod/gstmikmod.c: * ext/wavpack/gstwavpackenc.c: * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/y4m/gsty4mencode.c: * tests/check/elements/parser.c: update for buffer changes 2012-03-28 12:16:45 +0200 Mark Nauwelaerts * tests/check/elements/audiodynamic.c: tests: audiodynamic: correctly port original test to mind in place transform 2012-03-28 11:05:43 +0200 Mark Nauwelaerts * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: audiofx: more adjustment to changed semantics of audiofilter _setup method 2012-03-28 11:10:24 +0200 Mark Nauwelaerts * tests/check/elements/audiofirfilter.c: tests: audiofirfilter: negotiate the intended raw audio format 2012-03-27 18:41:45 +0200 Stefan Sauer * gst/audioparsers/gstwavpackparse.c: wavpackparse: init datastructure 2012-03-27 17:18:40 +0200 Wim Taymans * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstrev.c: * gst/effectv/gstwarp.c: effectv: fix strides 2012-03-27 16:41:06 +0200 Wim Taymans * gst/avi/gstavimux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/law/alaw-encode.c: * gst/law/mulaw-encode.c: * gst/matroska/matroska-demux.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpspeexpay.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * sys/oss/gstosssink.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/ximage/gstximagesrc.c: * tests/check/elements/qtmux.c: caps: improve caps handling Avoid caps copy and leaks 2012-03-27 14:04:48 +0200 Mark Nauwelaerts * tests/check/elements/icydemux.c: tests: icydemux: activate internal test helper src pad 2012-03-27 12:44:46 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: v4l2: update for get_param Remove const from the GstCaps. Plug some GstStructure leaks 2012-03-27 00:02:08 +0300 Raimo Järvi * configure.ac: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: udp: Fix compiling with mingw. https://bugzilla.gnome.org/show_bug.cgi?id=672880 2012-03-26 18:31:41 +0200 Mark Nauwelaerts * tests/check/elements/rganalysis.c: * tests/check/elements/rgvolume.c: tests: replaygain: misc compatibility fixes Discard caps event when checking for and counting various tag events, and remove all testing of 8 bits depth in 16 bits width format since it no longer exists. 2012-03-26 18:28:26 +0200 Mark Nauwelaerts * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtpbin.c: tests: rtp: misc compatibiliy fixes ... such as always setting pad caps and providing needed caps fields. 2012-03-26 18:26:40 +0200 Mark Nauwelaerts * tests/check/elements/videofilter.c: tests: videofilter: ensure initial segment event 2012-03-26 18:25:28 +0200 Mark Nauwelaerts * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: shapewipe: proper video info and frame management ... particularly since each incoming pad has a distinct format. 2012-03-26 18:24:08 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: ensure output caps are set when pushing output data ... even if some SPS/PPS has not passed by yet. 2012-03-26 18:22:53 +0200 Mark Nauwelaerts * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: videofilter: avoid holding object lock when calling basetransform function 2012-03-26 18:22:03 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: rtpbin: fix some lock management ... to avoid trying to take a non-recursive lock twice. 2012-03-26 18:21:11 +0200 Mark Nauwelaerts * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: rtpL16(de)pay: fix raw audio format in template caps 2012-03-26 18:20:40 +0200 Mark Nauwelaerts * gst/replaygain/gstrganalysis.c: replaygain: also still post the results of the analysis 2012-03-26 15:59:01 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2src: don't error in shutdown Don't log with the ERROR category when we are stopping because we are shutting down. Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=672824 2012-03-26 15:51:28 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2: fix latency 2012-03-26 15:30:00 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: called base class start Chain up to the base class start method so that metadata is properly tagged. Remove an unused variable. fixes: https://bugzilla.gnome.org/show_bug.cgi?id=672813 2012-03-26 12:12:45 +0200 Wim Taymans Replace master with 0.11 2012-03-25 00:00:59 +0000 Tim-Philipp Müller * configure.ac: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: gdkpixbufoverlay: add "alpha" property to set alpha of overlay image .. or turn the overlay off by setting alpha to 0.0 2012-03-24 09:51:06 +0100 Mark Nauwelaerts * gst/imagefreeze/gstimagefreeze.c: imagefreeze: plug caps leak 2012-03-23 18:47:45 +0100 Mark Nauwelaerts * tests/check/elements/imagefreeze.c: tests: imagefreeze: remove extraneous _unref 2012-03-23 18:47:03 +0100 Mark Nauwelaerts * tests/check/elements/avimux.c: tests: avimux: adjust to modified sink pad template name 2012-03-23 18:46:36 +0100 Mark Nauwelaerts * tests/check/elements/qtmux.c: tests: qtmux: cleanup element sooner ... to avoid stray refs in sticky caps events. 2012-03-23 18:45:56 +0100 Mark Nauwelaerts * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/avimux.c: * tests/check/elements/qtmux.c: tests: arrange for sending an initial segment event ... which is needed nowadays since various gst_segment_to_... no longer automatically set the format to the specified one (from _UNDEFINED). 2012-03-23 18:44:15 +0100 Mark Nauwelaerts * gst/imagefreeze/gstimagefreeze.c: imagefreeze: immediately return GST_FLOW_EOS ... rather than _OK since we will not be caring about subsequent buffer anyway. 2012-03-23 18:43:36 +0100 Mark Nauwelaerts * gst/imagefreeze/gstimagefreeze.c: imagefreeze: fix query and _getcaps handling 2012-03-23 18:42:48 +0100 Mark Nauwelaerts * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: audiofx: adjust to changed semantics of audiofilter _setup method ... in that it will now call subclass with info on proposed audio format without having set that info already in base class. As such, subclass can not rely on audio format info being available there. 2011-07-14 16:23:49 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set This allows outputting streams in AVC format even if the SPS/PPS are sent inside the RTP stream. https://bugzilla.gnome.org/show_bug.cgi?id=654850 Ported from master 2012-01-29 18:39:54 +0000 Olivier Crête * gst/udp/gstmultiudpsink.c: udpsink: Unlock on error 2012-03-22 18:27:30 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: use sink pad template caps rather than src 2012-03-22 18:23:22 +0100 Mark Nauwelaerts Merge branch 'master' into 0.11 2012-03-22 18:21:52 +0100 Mark Nauwelaerts * configure.ac: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: * gst/smpte/gstsmptealpha.c: * gst/smpte/gstsmptealpha.h: smpte: port to 0.11 2012-03-22 16:10:33 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: audioparsers: intersect downstream allowed peer caps with sink pad template 2012-03-22 15:55:28 +0100 Wim Taymans * configure.ac: back to development === release 0.11.2 === 2012-03-22 15:51:13 +0100 Wim Taymans * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: * win32/common/gstudp-marshal.c: Release 0.11.2 2012-03-22 11:55:28 +0100 Wim Taymans Merge branch 'master' into 0.11 2012-03-22 11:53:24 +0100 Wim Taymans Merge branch 'master' into 0.11 unport gdkpixbuf not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850 Conflicts: docs/plugins/Makefile.am docs/plugins/gst-plugins-good-plugins-docs.sgml docs/plugins/gst-plugins-good-plugins-sections.txt docs/plugins/gst-plugins-good-plugins.hierarchy docs/plugins/inspect/plugin-avi.xml docs/plugins/inspect/plugin-png.xml ext/flac/gstflacdec.c ext/flac/gstflacdec.h ext/libpng/gstpngdec.c ext/libpng/gstpngenc.c ext/speex/gstspeexdec.c gst/audioparsers/gstflacparse.c gst/flv/gstflvmux.c gst/rtp/gstrtpdvdepay.c gst/rtp/gstrtph264depay.c 2012-03-22 11:45:11 +0100 Mark Nauwelaerts * gst/smpte/gstsmpte.c: smpte: only start collectpads2 at state change rather than init 2012-03-21 13:22:43 +0100 Wim Taymans * tests/check/elements/audioamplify.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioecho.c: * tests/check/elements/audiopanorama.c: * tests/check/elements/rtp-payloading.c: tests: update for memory api changes 2012-03-20 10:24:05 +0100 Wim Taymans * gst/matroska/matroska-demux.c: update for memory api changes 2012-03-19 12:01:40 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: perform additional frame crc check if applicable ... such as a frame header parsing throwing some suspicious warnings. So we can be a bit more convinced we determine the right frame end. 2012-03-19 11:58:15 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: avoid indefinite extended search for frame end if possible ... which is particularly useful if locked on to the wrong frame start and/or corrupt frame being crc checked. 2012-03-16 18:23:29 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: improve error handling and resilience ... by noting that one occurred in the first place, and then appropriately ignoring some transient ones. 2012-03-19 10:33:48 +0100 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: negotiate an allocator on the srcpads We do an ALLOCATION query to find out an allocator and parameters on the srcpads. This way decoders (and sinks) can specify the memory and parameters they want us to write into. 2012-03-17 20:53:31 +0000 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-wavpack.xml: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: docs: update docs for new properties and add gdkpixbufoverlay element Somewhat at least. No idea why it doesn't pick up the description or example pipeline. 2012-03-18 00:11:19 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: make most properties controllable and flag them as mutable-playing 2012-03-17 23:41:38 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: gdkpixbufoverlay: add properties for positioning and sizing 2012-03-17 20:18:19 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: * ext/gdk_pixbuf/gstgdkpixbufoverlay.h: gdkpixbuf: add gdkpixbufoverlay element Still lacks features such as positioning or resizing, or animations, but it's usable already, and supports lots of formats. 2012-03-16 22:52:02 +0100 Wim Taymans * gst/alpha/gstalphacolor.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: don't poke into basetransform internals But use the methods 2012-03-16 21:47:21 +0100 Wim Taymans * ext/libpng/gstpngdec.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-parse.c: * gst/wavparse/gstwavparse.c: don't pass random pointers to pull_range 2012-03-15 22:15:47 +0100 Wim Taymans * gst/monoscope/gstmonoscope.c: updarte for bufferpool changes 2012-03-15 22:11:17 +0100 Wim Taymans * ext/dv/gstdvdec.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/rtp/gstrtpvrawdepay.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: update for bufferpool changes 2012-03-15 20:37:56 +0100 Wim Taymans * ext/aalib/gstaasink.c: * ext/dv/gstdvdec.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/monoscope/gstmonoscope.c: * gst/rtp/gstrtpvrawdepay.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: update for allocation query changes 2011-07-14 16:23:49 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set This allows outputting streams in AVC format even if the SPS/PPS are sent inside the RTP stream. https://bugzilla.gnome.org/show_bug.cgi?id=654850 2012-03-15 14:06:40 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: update for bufferpool api change 2012-03-15 13:38:16 +0100 Wim Taymans * ext/lame/gstlamemp3enc.c: update for memory api changes 2012-03-15 13:37:36 +0100 Wim Taymans * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: update for memory api changes 2012-03-15 13:36:17 +0100 Wim Taymans * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/flac/gstflacdec.c: * ext/jpeg/gstjpegenc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/matroska/matroska-demux.c: * gst/multifile/gstsplitfilesrc.c: * gst/multipart/multipartmux.c: * gst/shapewipe/gstshapewipe.c: * gst/videomixer/videomixer2.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: update for memory api changes 2012-03-14 21:36:03 +0100 Wim Taymans * ext/jpeg/gstjpegenc.c: update for memory api changes 2012-03-14 19:55:32 +0100 Wim Taymans * ext/aalib/gstaasink.c: * ext/dv/gstdvdec.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/rtp/gstrtpvrawdepay.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: take padding into account 2012-03-14 17:07:50 +0100 Mark Nauwelaerts * configure.ac: * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: port to 0.11 2012-03-14 15:45:38 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reply FALSe on serialized queries 2012-03-13 23:08:38 +0100 Andrej Gelenberg * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: pngenc: add support for 8- and 16-bit gray images Add support for direct encoding of 8- and 16-bit big endian gray images. https://bugzilla.gnome.org/show_bug.cgi?id=672025 2012-03-14 11:21:32 +0100 Wim Taymans * gst/rtp/gstrtpmp4vpay.c: mp4vpay: we can also handle x-divx 2012-03-14 10:39:53 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackenc.c: wavpackenc: do not set output caps directly ... but use base class function instead. 2012-03-13 21:31:48 +0100 Wim Taymans * gst/rtp/gstrtpmp4vdepay.c: mp4vdepay: fix buffer handling Don't always output the payload subbuffer, use a separate variable to make things clearer and without the error. 2012-03-13 20:49:43 +0100 Wim Taymans * gst/udp/gstmultiudpsink.c: udpsink: make buffer-size work again 2012-03-13 20:36:56 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: fix SO_RCVBUF handling 2012-03-13 19:26:47 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: don't leak the address 2012-03-13 19:26:23 +0100 Wim Taymans * gst/rtp/gstrtph264depay.c: h264depay: unmap on empty packet 2012-03-13 18:07:18 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: do DTS and PTS correctly 2012-03-13 17:54:50 +0100 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: set DTS and PTS on output buffers Set PTS and DTS on output buffers instead of just the PTS. In streaming cases you want to synchronized encoded data based on the DTS because that is monotonically increasing. 2012-03-13 17:54:28 +0100 Wim Taymans * gst/isomp4/qtdemux_dump.c: qtdemux: debug additional sdtp flag 2012-03-13 17:27:32 +0100 Wim Taymans * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpmp4gdepay.c: rtp: fix unmap calls 2012-03-13 13:25:09 +0100 Wim Taymans * ext/pulse/pulsesink.h: pulse: fix formats, we can not handle S8 but only U8 2012-03-13 12:40:37 +0100 Wim Taymans * ext/flac/gstflacenc.c: flacenc: fix streamheaders Fix the caps of flacenc, the reference encoder only support 24 bits in 32 bits. Set streamheader on output caps. 2012-03-12 17:17:01 +0100 Wim Taymans * gst/monoscope/gstmonoscope.c: update for caps api changes 2012-03-12 16:43:27 +0200 Sreerenj Balachandran * configure.ac: configure.ac : bump GLib requirement to 2.31.14 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=671911 2012-03-12 15:27:27 +0100 Ross Burton * ext/flac/gstflacenc.c: flacenc: generate seektables every 10 sec by default Since this is what the command line tool does as well, it seems like a better default. 2012-03-10 13:44:08 +0000 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: only unlock pad when it was locked This fixes the mutex being unlocked too much and ending up allowing other threads when they should not. https://bugzilla.gnome.org/show_bug.cgi?id=671776 2012-03-07 13:39:50 +0100 Andrej Gelenberg * ext/libpng/gstpngdec.c: pngdec: add support for video/x-raw-gray formats pngdec can now decode gray 8- and 16-bit images without alpha channel direct to video/x-raw-gray format. 16-bit gray images have big-endian format, because it's native PNG endianness. Gray images with alpha channel still converted to RGBA. Signed-off-by: Andrej Gelenberg 2012-03-08 17:07:51 +0100 Marc Leeman * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: gstrtspsrc: disable RTSP keep-alive on request 2012-03-12 14:48:47 +0100 Wim Taymans * gst/smpte/gstsmpte.c: smpte: fix stride handling 2012-03-12 12:23:15 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: * tests/check/elements/videocrop.c: * tests/check/elements/videofilter.c: fix for caps _normalize changes 2012-03-12 11:47:35 +0100 Wim Taymans * gst/alpha/gstalphacolor.c: * gst/matroska/matroska-demux.c: fix for caps api change 2012-03-12 10:43:57 +0100 Wim Taymans * gst/alpha/gstalphacolor.c: * gst/matroska/matroska-demux.c: * sys/oss4/oss4-audio.c: fix for _do_simplify changes 2012-03-12 08:48:32 +0100 Nicola Murino * gst/flv/gstflvmux.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-mux.c: gst: Fix some query leaks 2012-03-11 19:06:59 +0100 Wim Taymans * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: fix for caps api changes 2012-03-11 19:06:37 +0100 Wim Taymans * ext/aalib/gstaasink.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/pulse/pulsesrc.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/videomixer/videomixer2.c: * sys/v4l2/gstv4l2src.c: * sys/ximage/gstximagesrc.c: fix for caps api changes 2012-03-10 10:51:44 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: * gst/alpha/gstalphacolor.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstwavpackparse.c: * gst/auparse/gstauparse.c: * gst/goom2k1/gstgoom.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: fix template caps refcount 2012-03-09 15:53:32 +0000 Tim-Philipp Müller * configure.ac: configure: fix use of AC_LANG_PROGRAM No need to include the int main () { } bits, the body is enough. 2012-03-09 15:25:02 +0000 Tim-Philipp Müller * configure.ac: configure: fix autogen.sh warnings configure.ac:410: warning: AC_LANG_CONFTEST: no AC_LANG_SOURCE call detected in body 2012-03-08 13:06:13 +0100 Wim Taymans * ext/aalib/gstaasink.c: * ext/aalib/gstaasink.h: aasink: propose videometa uptream subclass from videosink. Propose videometa upstream because we can handle it with the video api. 2012-03-08 01:53:50 -0500 Matej Knopp * gst/isomp4/gstqtmux.c: qtmux: do not unref sample caps https://bugzilla.gnome.org/show_bug.cgi?id=671534 2012-03-08 11:36:01 +0100 Wim Taymans * tests/check/elements/autodetect.c: * tests/check/elements/videocrop.c: tests: improve more tests 2012-03-08 11:20:43 +0100 Wim Taymans * tests/check/elements/capssetter.c: * tests/check/elements/gdkpixbufsink.c: tests: fix some more tests 2012-03-07 15:22:36 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: improve cleanup Reuse cleanup methods to make sure we remove all pads correctly 2012-03-07 15:00:26 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: set caps without the lock Release the lock before setting the caps on the srcpad, which triggers an event, which could eventually call back into us and cause a deadlock. 2012-03-07 14:55:08 +0100 Wim Taymans * gst/rtpmanager/gstrtpptdemux.c: ptdemux: set caps after activating the pad Set the caps after we activated the pad or else it will just fail. 2012-03-07 14:54:15 +0100 Wim Taymans * gst/law/alaw.c: * gst/law/mulaw.c: law: add layout to audio caps 2012-03-07 14:51:09 +0100 Wim Taymans * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: * gst/law/mulaw-decode.c: * gst/law/mulaw-decode.h: law: use GstAudioInfo Use GstAudioInfo to generate output caps. 2012-03-07 04:20:00 -0500 Matej Knopp * gst/isomp4/gstqtmux.c: qtdemux: covert art tag type is GstSample not GstBuffer now https://bugzilla.gnome.org/show_bug.cgi?id=671534 2012-03-07 10:28:58 +0000 Tim-Philipp Müller * po/POTFILES.in: po: fix POTFILES.in for new wavpackparse location in source tree 2012-03-06 21:44:36 -0800 David Schleef * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: udp: Change the default port to 5004 udpsrc/udpsink are almost always used with RTP, so let's use an RTP port as the default port. It's unclear why 4951 was used, it goes back to early commits in CVS. 2012-03-06 21:36:02 -0800 David Schleef Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-03-06 15:58:20 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: use base class tag handling helper ... so as to ensure these to be handled and sent at proper time. 2012-03-06 14:25:27 +0100 Sebastian Dröge * ext/wavpack/gstwavpackstreamreader.c: wavpack: Fix possible underflow of unsigned integer variable 2012-03-06 14:22:43 +0100 Sebastian Dröge * sys/ximage/gstximagesrc.c: ximagesrc: Fix 'comparison of unsigned expression >= 0 is always true' This variable can never be below zero anyway. 2012-03-06 14:18:33 +0100 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Use correct enum for return values 2012-03-06 14:16:21 +0100 Sebastian Dröge * gst/rtp/gstrtpdvdepay.c: dvdepay: Fix 'comparison of unsigned expression >= 0 is always true' compiler warning This was an actual bug as it could've caused reading from invalid memory areas when the input is broken. 2012-03-06 13:21:12 +0100 Sebastian Dröge * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopTop.inc: deinterlace: Fix 'variable 'oldbx' is uninitialized when used here' compiler warnings 2012-03-06 13:19:24 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix 'implicit conversion from enumeration type 'GstDeinterlaceFields' to different enumeration type 'GstDeinterlaceMode'' compiler warning 2012-03-05 15:29:56 +0100 Wim Taymans * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbuf.h: gdk: cleanups and fix rowstride Fix the output rowstride, we need to take the stride of the output video frame. Since we are also dealing with planes, take the plane data and stride. Don't store the same info twice in different variables. 2012-03-05 13:31:44 +0100 Wim Taymans * ext/gdk_pixbuf/gstgdkpixbuf.c: gdkpixbuf: fix event handling 2012-03-05 12:20:07 +0100 Mark Nauwelaerts * tests/check/Makefile.am: * tests/check/elements/wavpackdec.c: * tests/check/elements/wavpackenc.c: * tests/check/elements/wavpackparse.c: * tests/check/pipelines/wavpack.c: tests: port wavpack tests to 0.11 2012-03-05 13:36:39 +0100 Mark Nauwelaerts * configure.ac: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: wavpackdec: port to 0.11 2012-03-05 12:17:39 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackcommon.c: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.c: wavpackenc: port to 0.11 2012-03-05 13:34:36 +0100 Mark Nauwelaerts * docs/plugins/Makefile.am: * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpack.c: * ext/wavpack/gstwavpackparse.c: * ext/wavpack/gstwavpackparse.h: wavpack: remove legacy wavpackparse 2012-03-05 12:15:44 +0100 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/gstwavpackparse.c: * gst/audioparsers/gstwavpackparse.h: * gst/audioparsers/plugin.c: audioparsers: port wavpackparse to 0.11 2012-03-05 13:29:59 +0100 Mark Nauwelaerts Merge branch 'master' into 0.11 Conflicts: ext/wavpack/gstwavpackparse.c sys/v4l2/gstv4l2bufferpool.c sys/v4l2/gstv4l2bufferpool.h sys/v4l2/gstv4l2videooverlay.c 2012-03-05 12:43:17 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: x-raw-bayer -> x-bayer 2012-03-05 11:17:30 +0100 Oleksij Rempel (Alexey Fisher) * sys/v4l2/gstv4l2xoverlay.c: v4l2sink: don't use deprecated XKeycodeToKeysym https://bugzilla.gnome.org/show_bug.cgi?id=671299 Signed-off-by: Oleksij Rempel (Alexey Fisher) 2012-03-05 12:03:01 +0100 Wim Taymans * sys/ximage/Makefile.am: * sys/ximage/gstximagesrc.c: ximage: use new style caps 2012-03-05 10:49:33 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackdec.c: wavpackdec: allow some timestamp tolerance to arrange for perfect timestamping ... which also happens to make some more unit tests pass. 2012-03-05 10:47:44 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackdec.c: wavpackdec: fix copying output data 2012-03-05 10:46:51 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackenc.c: wavpackenc: restore legacy buffer offset decorating somewhat ... at least sufficiently to aid in recognizing rewritten header buffer making unit test pass. 2012-03-05 10:51:33 +0100 Stefan Sauer * gst/audioparsers/gstwavpackparse.c: wavpackparse: initialize header to silence older gcc versions 2012-03-05 10:45:46 +0100 Stefan Sauer * ext/wavpack/gstwavpackparse.c: wavpackparse: remove empty lines in varable declarations caused by old indent 2012-03-05 10:44:54 +0100 Stefan Sauer * ext/jack/gstjack.h: jack: fix obvious wrong definition for the master flag 2012-03-04 19:55:26 +0100 Stefan Sauer * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: change the transport-mode enum into flags One can use (or not use) master and slave mode independently. 2012-03-02 11:49:02 -0500 Antoine Tremblay * gst/avi/gstavimux.c: avimux: support up to 6 channels of AC-3 https://bugzilla.gnome.org/show_bug.cgi?id=671220 2012-03-03 13:04:48 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2bufferpool.c: v4l2: clear DISCONT flag when recycling buffers into the buffer pool The base class may have set the DISCONT flag on the first buffer pushed out. We need to clear that when recycling buffers back into the buffer pool, otherwise we constantly push out buffers with the discont flag set, which might upset downstream elements, esp. for compressed formats like mpeg-ts. 2012-03-01 14:15:29 +0100 Oleksij Rempel (Alexey Fisher) * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2src: fix v4l2_munmap() for compressed formats Make sure we always call munmap() with the same size we called mmap() with before. Current v4l2src uses the same structure for VIDIOC_QUERYBUF, VIDIOC_QBUF and v4l2_munmap calls. The problem is that the video buffer size (length) may vary for compressed or emulated bufs. VIDIOC_QBUF will change it if we pass the pointer of a v4l2_buffer. This is why we should avoid using same variable for mmap and video buffers. https://bugzilla.gnome.org/show_bug.cgi?id=671126 2012-03-02 11:17:33 +0100 Sebastian Dröge * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/flv/gstindex.c: gst: Update for the gstmarshal.[ch] removal 2012-03-02 10:13:08 +0100 Sebastian Dröge * ext/pulse/pulsemixerctrl.h: * gst/videofilter/gstvideobalance.c: * sys/v4l2/gstv4l2colorbalance.h: mixer/colorbalance: Update for API changes 2012-03-01 17:15:57 +0100 Wim Taymans * ext/aalib/gstaasink.c: aasink: fix stride 2012-03-01 11:36:34 +0100 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/plugin.c: audioparsers: disable non-ported wavpackparse 2012-03-01 11:29:50 +0100 Mark Nauwelaerts Merge branch 'master' into 0.11 Conflicts: ext/wavpack/gstwavpackenc.c tests/check/elements/audioiirfilter.c tests/examples/v4l2/probe.c 2012-02-29 22:31:21 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbufsink.c: gdkpixbufsink: remove deprecated property 2012-02-29 22:30:56 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbuf.c: gdkpixbufscale: remove deprecated property 2012-02-29 22:28:01 +0100 Mark Nauwelaerts * configure.ac: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/gstgdkpixbufsink.h: gdkpixbufsink: port to 0.11 2012-02-29 22:25:23 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/pixbufscale.c: * ext/gdk_pixbuf/pixbufscale.h: gdkpixbufscale: port to 0.11 2012-02-29 22:24:46 +0100 Mark Nauwelaerts * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbuf.h: gdkpixbufdec: port to 0.11 2012-02-29 17:26:01 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: update for metadata API changes 2012-02-28 13:51:10 +0100 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/gstwavpackparse.c: * gst/audioparsers/gstwavpackparse.h: * gst/audioparsers/plugin.c: audioparsers: add baseparse based wavpackparse 2012-02-28 11:38:59 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/ximageutil.c: update for metadata tags 2012-02-27 23:46:15 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: * tests/check/elements/wavpackdec.c: wavpackdec: adjust to audio format limitations ... which does not allow expressing arbitrary depth in a GstAudioFormat. Also adjust unit test to modified behaviour. 2012-02-27 23:46:08 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: wavpackdec: determine depth from bytes per sample ... rather than from bits per sample, since spec states values are already left justified w.r.t. bits per sample but not w.r.t. bytes per sample (and so the latter determines the normalization, or indicated depth). 2012-02-27 23:46:03 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: wavpackdec: port to audiodecoder 2012-02-27 23:45:54 +0100 Mark Nauwelaerts * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackenc.h: * tests/check/elements/wavpackenc.c: wavpackenc: port to audioencoder Also adjust unit test to slightly modified behaviour. 2012-02-27 14:47:25 +0100 Edward Hervey * ext/annodex/gstannodex.c: * ext/annodex/gstcmmlparser.c: * ext/annodex/gstcmmltag.c: * ext/pulse/pulseprobe.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/interleave/interleave.c: * gst/rtpmanager/rtpsession.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * sys/oss4/oss4-audio.c: * sys/oss4/oss4-property-probe.c: * sys/v4l2/gstv4l2object.c: * tests/check/elements/audiofirfilter.c: * tests/check/elements/audioiirfilter.c: * tests/check/elements/cmmldec.c: * tests/check/elements/interleave.c: * tests/check/pipelines/wavenc.c: * tests/examples/audiofx/firfilter-example.c: * tests/examples/audiofx/iirfilter-example.c: * tests/examples/pulse/pulse.c: * tests/examples/rtp/server-alsasrc-PCMA.c: * tests/examples/v4l2/probe.c: * tests/icles/test-oss4.c: Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 13:09:31 +0100 Wim Taymans * ext/speex/gstspeexenc.c: speexenc: chain up to parent event handler 2012-02-27 13:05:33 +0100 Wim Taymans * ext/flac/gstflacenc.c: flacenc: fix event handling Fix dodgy segment event handling Chain up to parent event handler 2012-02-27 09:14:04 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: use public api instead of poking into the private structures of the base class 2012-02-27 06:35:01 +0100 Alessandro Decina * ext/lame/Makefile.am: amrwbdec, lame, mad: link to libgstbase 2012-02-27 01:09:11 +0000 Tim-Philipp Müller * gst/flv/gstflvmux.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-mux.c: flvmux, matroskamux, qtmux: if in doubt about downstream seekability default to streaming=true If downstream didn't answer our SEEKING query and told us it's seekable, default to streaming=true. We couldn't do this in 0.10 for backwards compatibility reasons, but we can in 0.11. Play it safe. 2012-02-27 01:00:03 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 Conflicts: gst/audioparsers/gstmpegaudioparse.c 2012-02-27 00:56:37 +0000 Tim-Philipp Müller Merge commit 'f9207722ca8fd8dcc1e7215d8af85efe4debfdf4' into 0.11 2012-02-27 00:55:38 +0000 Tim-Philipp Müller * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: fix up after merge 2012-02-27 00:48:57 +0000 Tim-Philipp Müller Merge commit '38516ad367128d83f9e156529018adb4433cd328' into 0.11 Conflicts: ext/pulse/pulseaudiosink.c gst/audioparsers/gstmpegaudioparse.c 2012-02-26 20:39:52 +0100 Alessandro Decina * gst/goom2k1/gstgoom.c: goom2k1: fix compiler warning 2012-02-26 20:30:24 +0100 Alessandro Decina * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: fix compiler warning 2012-02-25 15:55:15 +0000 Tim-Philipp Müller * gst/isomp4/gstqtmux.c: qtmux: create streamable output if downstream is not seekable Ignore the "streamable" property setting and create streamable output if downstream is known not to be seekable (as queried via a SEEKABLE query). Fixes pipelines like qtmux ! appsink possibly creating seemingly corrupted output if streamable has not been set to true. 2012-02-25 15:48:44 +0000 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: create streamable output if downstream is not seekable Ignore the "streamable" property setting and create streamable output if downstream is known not to be seekable (as queried via a SEEKABLE query). Fixes pipelines like flvmux ! appsink possibly creating seemingly corrupted output if streamable has not been set to true. 2012-02-25 15:40:39 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: create streamable output if downstream is not seekable Ignore the "streamable" property setting and create streamable output if downstream is known not to be seekable (as queried via a SEEKABLE query). Fixes pipelines like webmmux ! appsink creating seemingly corrupted output if streamable has not been set to true. 2012-02-24 11:03:48 +0100 Wim Taymans * gst/alpha/gstalpha.c: * gst/debugutils/gstcapssetter.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstvideoflip.c: update for basetransform change 2012-02-24 10:26:26 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/ximageutil.c: update for metadata change 2012-02-23 08:42:25 -0800 David Schleef * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect/plugin-efence.xml: * gst/debugutils/Makefile.am: * gst/debugutils/efence.c: * gst/debugutils/efence.h: * gst/debugutils/efence.vcproj: efence: remove plugin Valgrind is much more useful these days. 2012-02-23 12:05:20 +0000 Tim-Philipp Müller * NEWS: * RELEASE: Update NEWS and RELEASE as well 2012-02-23 11:07:35 +0000 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Bump version after release 2012-02-23 12:03:24 +0100 Wim Taymans * gst/audiofx/audioecho.c: * gst/audiofx/audioecho.h: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audiofxbaseiirfilter.h: audiofx: remove transform lock usage 2012-02-23 11:16:21 +0100 Wim Taymans * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: * gst/videofilter/gstvideobalance.c: update for basetransform lock removal 2012-02-22 23:36:54 +0000 Tim-Philipp Müller * gst/debugutils/Makefile.am: debugutils: disable efence plugin properly We don't want it built if mmap isn't available either.. 2012-02-22 17:39:16 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: fix get_caps function some more so that all structures have channel info Set channels and channel-layout on the right structure; that is, the structure we are going to append to the caps we are building, and not the structure we are using as a template for all the structures. Fixes first structure of the returned caps not having any channel info set on it. 2012-02-22 17:09:25 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: microoptimisation: avoid unnecessary list and string copies 2012-02-22 17:03:42 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: audio caps have a *list* of formats, not an array of formats A list of things in caps is something where one is picked in the course of negotiation. An array is always something that only makes sense as a whole in that order. 2012-02-22 18:02:27 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: remove post-port bogus _unref 2012-02-22 17:00:19 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: remove bogus pad locking that causes deadlocks It's not clear why the pad object lock is taken here. But gst_pad_{has,get}_current_caps() will try to take the lock as well and deadlock, since it's not recursive. 2012-02-22 16:59:42 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: set right number of channels on caps in get_caps function 2012-02-21 17:16:32 -0800 David Schleef * autogen.sh: autogen: avoid touching .po files during 'make' A simple workaround to deal with GNU gettext automake integration failing to deal with git. Fixes: #669207 2012-02-22 02:06:17 +0100 Wim Taymans * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/flv/gstflvmux.c: * gst/isomp4/atoms.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/multifile/gstmultifilesrc.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsrc.c: * gst/y4m/gsty4mencode.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/ximageutil.c: * tests/check/elements/deinterleave.c: * tests/check/elements/interleave.c: update for new memory api 2012-02-21 17:57:44 +0100 Vincent Untz * ext/pulse/pulseaudiosink.c: pulse: Fix a build warning when compiling with asserts disabled Return a value even if the code will never be reached, to make compilers happy. https://bugzilla.gnome.org/show_bug.cgi?id=670561 2012-02-21 18:42:31 +0100 Mark Nauwelaerts * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: mpegaudioparse: support parsing freeform bitrate stream 2012-02-21 18:39:18 +0100 Mark Nauwelaerts * configure.ac: * gst/monoscope/gstmonoscope.c: * gst/monoscope/gstmonoscope.h: monoscope: port to 0.11 2012-02-21 10:53:56 +0100 Wim Taymans Merge branch 'master' into 0.11 2012-02-20 12:22:12 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-21 10:40:00 +0100 Edward Hervey * ext/taglib/gstid3v2mux.cc: id3v2mux: Fix merge error 2012-02-20 12:22:12 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-20 16:35:18 +0100 Wim Taymans * gst/udp/gstmultiudpsink.c: fix compiler warnings 2012-01-26 03:29:28 -0500 Matej Knopp * gst/udp/gstudpsrc.c: fix compiler warnings 2012-01-26 06:58:46 -0500 Matej Knopp * gst/dtmf/gstdtmfsrc.c: Fix compiler warnings 2012-02-18 11:38:36 +0000 Tim-Philipp Müller * tests/check/elements/level.c: tests: fix up level test for GstValueList -> GValueArray change https://bugzilla.gnome.org/show_bug.cgi?id=670303 2012-02-16 18:01:29 +0200 Peteris Krisjanis * gst/level/gstlevel.c: level: use GValueArray instead of GstValueList in messages Updated GstLevel element to use GValueArray instead of GstValueList for rms/peak/decay keys attached to element message. https://bugzilla.gnome.org/show_bug.cgi?id=670303 2012-02-18 00:00:54 +0100 Wim Taymans * win32/common/config.h: win32: back to development 2012-02-17 23:54:29 +0100 Dominique Leuenberger * docs/plugins/Makefile.am: No longer reference deprecated header files while building docs. 2012-02-17 23:49:21 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/equalizer/gstiirequalizer.c 2012-02-17 17:21:53 +0000 Tim-Philipp Müller * gst/equalizer/gstiirequalizer.c: equalizer: fix switching from passthrough to non-passthrough when parameters change commit b5bf0294 moved the if(need_new_coefficients) set_passthrough(equ) after the if(is_passthrough) return FLOW_OK shortcut, so the passthrough mode would never get updated even if the coefficients change. Fixes equalizer-test doing .. nothing. 2012-02-17 17:57:03 +0100 Mark Nauwelaerts * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: goom*: fix leaked caps event 2012-02-17 13:26:53 +0100 Mark Nauwelaerts * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: parse either Xing or VBRI data ... and avoid confusing debug message claiming neither present. 2012-02-17 14:38:03 +0100 Wim Taymans * gst/matroska/matroska-demux.c: matrosk: fix segment update 2012-02-17 11:05:27 +0100 Wim Taymans * configure.ac: back to development === release 0.11.1 === 2012-02-17 11:04:47 +0100 Wim Taymans * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtpmanager.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/eo.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: * win32/common/gstrtpbin-marshal.c: * win32/common/gstrtpbin-marshal.h: RELEASE 0.11.1 2012-02-16 23:33:15 +0100 Mark Nauwelaerts * gst/goom/gstgoom.c: goom: fix buffer leak 2012-02-16 23:40:58 +0100 Mark Nauwelaerts * gst/goom2k1/gstgoom.c: goom2k1: use some more boilerplate 2012-02-16 23:33:01 +0100 Mark Nauwelaerts * configure.ac: * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom2k1: port to 0.11 2012-02-16 15:31:53 +0100 Mark Nauwelaerts * ext/shout2/gstshout2.c: shout2: use some more boilerplate 2012-02-16 15:29:34 +0100 Mark Nauwelaerts * configure.ac: * ext/shout2/gstshout2.c: shout2: port to 0.11 2012-02-14 11:56:00 +0100 Philippe Normand * gst/interleave/Makefile.am: * gst/interleave/interleave.c: * gst/interleave/interleave.h: * gst/interleave/plugin.c: * gst/interleave/plugin.h: * tests/check/elements/interleave.c: interleave: port to 0.11 Port of the interleave element and its unittests. https://bugzilla.gnome.org/show_bug.cgi?id=669643 2012-02-16 14:23:50 +0100 Wim Taymans Merge branch 'master' into 0.11 2012-02-16 17:14:20 +0800 Gary Ching-Pang Lin * sys/v4l2/v4l2_calls.c: v4l2src: failure to query some optional controls is not a fatal error Don't post a (fatal) error message on the bus just because we failed to query some control. Fixes issue with built-in Suyin Corp webcam for HP notebook (usbid 064e:e28a) on OpenSuse 12.1, where querying red/blue balance fails. https://bugzilla.gnome.org/show_bug.cgi?id=670197 2012-02-16 12:59:10 +0000 Tuukka Pasanen * sys/v4l2/v4l2_calls.c: v4l2src: fix for webcamstudio vloopback Because vlooback emits 25 - ENOTTY and no EINVAL v4l2src thought it can't handle this and does not work. https://bugzilla.gnome.org/show_bug.cgi?id=669455 2012-02-16 11:21:28 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: declare variables at the beginning of the block It's how we roll. Fixes 'ISO C90 forbids mixed declarations and code' compiler warning. 2012-02-15 23:55:44 +0000 Tim-Philipp Müller * tests/examples/spectrum/Makefile.am: examples: fix spectrum example build issues Find fft headers in uninstalled setup, fix LIBS order. 2012-02-15 12:41:43 +0100 Wim Taymans * gst/audioparsers/gstaacparse.c: aacparse: remove some unused declarations 2012-02-15 11:25:45 +0100 Stefan Sauer * tests/examples/spectrum/Makefile.am: * tests/examples/spectrum/demo-audiotest.c: spectrum-demo: show the effect of fast-mode 2012-02-14 12:26:37 +0100 Wim Taymans * gst/videocrop/gstaspectratiocrop.c: aspectratiocrop: fix caps refcount 2012-02-14 11:22:46 +0100 Wim Taymans * tests/check/pipelines/effectv.c: tests: fix test, use videoconvert 2012-02-14 10:51:38 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: tests/check/elements/flacparse.c 2012-02-09 13:41:53 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparsers: adjust to modified baseparse API 2012-02-13 17:13:17 +0100 Wim Taymans * gst/multifile/gstmultifilesink.c: * gst/udp/gstmultiudpsink.c: update for memory api change 2012-02-13 12:06:37 +0100 Mark Nauwelaerts * tests/check/elements/flacparse.c: tests: flacparse: check and compare intended data 2012-02-12 17:03:37 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 Conflicts: ext/taglib/gstapev2mux.cc ext/taglib/gstid3v2mux.cc ext/taglib/gsttaglibmux.c ext/taglib/gsttaglibmux.h 2012-02-12 16:22:21 +0000 Tim-Philipp Müller * ext/taglib/Makefile.am: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstapev2mux.h: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gstid3v2mux.h: * ext/taglib/gsttaglibmux.c: * ext/taglib/gsttaglibmux.h: * ext/taglib/gsttaglibplugin.c: taglib: port to GstTagMux base class 2012-02-12 12:24:50 +0000 Vincent Penquerc'h * ext/taglib/gsttaglibmux.c: taglib: finish off a few missed variable changes Local variables are now unused, and the values from the segment copy are used instead, so remove the now useless local variables and write to the segment where appropriate. 2012-02-10 16:23:14 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacenc.c ext/jack/gstjackaudioclient.c ext/jack/gstjackaudiosink.c ext/jack/gstjackaudiosrc.c ext/pulse/plugin.c ext/shout2/gstshout2.c gst/matroska/matroska-mux.c gst/rtp/gstrtph264pay.c 2012-02-08 23:03:28 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: rtph264pay: add stream-format and alignment to h264 sink caps We're happy to accept both byte-stream and avc, advertise that on the sink caps and fix up _get_caps() function to not just return "video/x-h264". https://bugzilla.gnome.org/show_bug.cgi?id=606662 2012-02-08 20:58:04 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: rtph264depay: add stream-format and alignment fields to src template caps Because we can. And so we get a warning if we try to output avc with nal alignment or somesuch. https://bugzilla.gnome.org/show_bug.cgi?id=606662 2012-02-10 13:44:43 +0000 Tim-Philipp Müller * tests/check/elements/rtp-payloading.c: tests: clean up rtp-payloading test a little Feed data into the pipeline using appsrc instead of fdsrc and a pipe. Store unsigned byte values in guint8 instead of char. Getting rid of the capsfilter also helps to avoid 'format is not fully specified' warnings when pushing "video/x-h264" data into rtph264pay with fully specified h264 caps in the sink template. 2012-02-10 10:07:34 +0100 Wim Taymans * gst/flv/gstflvdemux.c: flv: use default pad query We need to chain up unknown queries to the default query handler instead of blindly forwarding them. In this case it caused the caps query to be forwarded to the upstream typefind and return the wrong type for the audio/video pad. 2012-02-09 22:12:14 +0100 Mark Nauwelaerts * tests/check/elements/mpegaudioparse.c: tests: mpegaudioparse: remove stray declaration 2012-02-09 22:07:48 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: aacparse: correctly set ADIF src caps 2012-02-09 22:10:07 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: prevent a few direct exits without cleanup 2012-02-09 22:07:18 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: shift in proper direction for audio sample conversion 2012-02-09 18:09:45 +0100 Wim Taymans * tests/check/elements/deinterleave.c: tests: fix compilation 2012-02-09 10:11:48 +0100 Marc Leeman * gst/udp/gstmultiudpsink.c: multiudpsink: typo fix (bytes send -> bytes sent) 2012-02-08 16:34:00 +0100 Wim Taymans * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/wavpack/gstwavpackenc.c: * gst/effectv/gstquark.c: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/qtdemux.c: * gst/multifile/gstsplitfilesrc.c: * gst/replaygain/gstrganalysis.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/udp/gstudpsrc.c: * gst/wavenc/gstwavenc.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/ximage/gstximagesrc.c: GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-08 16:37:13 +0100 Wim Taymans * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-07 14:10:44 -0800 Ralph Giles * ext/shout2/gstshout2.c: shout2send: send video/webm through libshout. This requires SHOUT_FORMAT_WEBM, added in libshout 2.3.0, so video/webm support is contingent on that symbol being defined. Also an indentation change required by the pre-commit hook. https://bugzilla.gnome.org/show_bug.cgi?id=669590 2012-01-30 16:40:19 +0100 Philippe Normand * configure.ac: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.c: * gst/interleave/deinterleave.h: * gst/interleave/plugin.c: * gst/interleave/plugin.h: * tests/check/elements/deinterleave.c: deinterleave: port to 0.11 Port of the deinterleave element and its unittests. The interleave element will be ported as part of another patch, hence disabling it for now. https://bugzilla.gnome.org/show_bug.cgi?id=668847 2012-02-07 23:41:13 +0200 Raimo Järvi * sys/directsound/gstdirectsoundsink.h: directsoundsink: Fix compiling https://bugzilla.gnome.org/show_bug.cgi?id=669607 2012-02-08 00:08:49 +0200 Raimo Järvi * sys/waveform/gstwaveformsink.c: waveformsink: Port to 0.11 https://bugzilla.gnome.org/show_bug.cgi?id=669612 2012-02-07 21:57:47 +0100 Stefan Sauer * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: rework transport support Move common code to jackclient. There we can also handle the request state message in a better way, as the element callbacks are only run if the element is active. 2012-02-07 10:47:19 +0100 Wim Taymans * tests/check/elements/apev2mux.c: * tests/check/elements/id3v2mux.c: tests: improve tagmux tests 2012-02-07 10:29:11 +0100 Wim Taymans * ext/taglib/gsttaglibmux.c: taglib: fix object registration We can't use G_DEFINE_TYPE because the class is not set in the class_init and we need it to get the srcpad template. Fix a caps leak 2012-02-07 10:16:32 +0100 Wim Taymans * tests/check/elements/jpegenc.c: tests: fix jpeg test 2012-02-07 10:15:51 +0100 Wim Taymans * ext/soup/gstsouphttpsrc.c: soup: fix caps 2012-02-07 09:54:00 +0100 Wim Taymans * gst/effectv/gstdice.c: * gst/effectv/gstshagadelic.c: effecttv: fix initialisation 2012-02-07 09:42:04 +0100 Wim Taymans * gst/y4m/gsty4mencode.c: y4m: fix negotiation 2012-02-07 09:41:51 +0100 Wim Taymans * tests/check/elements/videofilter.c: * tests/check/elements/y4menc.c: tests: fix more tests 2012-02-06 22:13:53 +0100 Mark Nauwelaerts * configure.ac: * ext/dv/Makefile.am: * ext/dv/gstdvdec.c: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dv: port to 0.11 2012-02-06 18:35:01 +0100 Wim Taymans * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: * tests/check/elements/spectrum.c: * tests/check/elements/videocrop.c: test: fix more tests 2012-02-06 15:52:49 +0100 Wim Taymans * tests/check/elements/id3demux.c: * tests/check/elements/level.c: * tests/check/elements/multifile.c: tests: fix more tests 2012-02-06 15:52:36 +0100 Wim Taymans * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: flv: fix caps 2012-02-06 15:20:55 +0100 Wim Taymans * gst/equalizer/gstiirequalizer.c: * tests/check/elements/equalizer.c: iirequalizer: fix equalizer and unit test 2012-02-06 13:44:20 +0100 Wim Taymans * tests/check/elements/audiopanorama.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: tests: fix some more tests 2012-02-06 13:43:49 +0100 Wim Taymans * gst/avi/gstavimux.c: avimux: take the pad from collectpads2 correctly 2012-02-06 13:29:24 +0100 Wim Taymans * tests/check/elements/audioiirfilter.c: * tests/check/elements/audioinvert.c: tests: fix more unit tests 2012-02-06 13:28:55 +0100 Wim Taymans * gst/audiofx/audiodynamic.c: audiodynamic: fix negotiation 2012-01-28 11:13:16 +0100 Nicola Murino * gst/matroska/matroska-demux.c: matroskademux: avoid posting invalid duration for each frame https://bugzilla.gnome.org/show_bug.cgi?id=666583 2012-02-06 10:07:06 +0100 Wim Taymans * tests/check/elements/audioamplify.c: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioecho.c: tests: fix more tests 2012-02-06 09:49:38 +0100 Wim Taymans * tests/check/elements/aspectratiocrop.c: * tests/check/elements/rganalysis.c: tests: improve some tests 2012-02-06 09:23:49 +0100 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: tests: fix jitterbuffer test 2012-02-06 09:23:07 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix caps after pt change 2012-02-06 09:18:17 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix caps leak 2012-02-03 22:05:59 +0530 Arun Raghavan * ext/pulse/plugin.c: pulseaudiosink: Lower rank to prevent autoplugging pulseaudiosink breaks visualisations in its current form, so let's prevent it from being autoplugged for the time being. The best we can hope to do in the 0.10 series is query the list of available sinks and their formats, and expose these as the bin's sinkpad caps. While this is not a comprehensive solution, it will make sure that we're only trying to support compressed formats if we're certain that one exists. The long-term fix for this will be in the form of proper upstream renegotiation support in the 0.11/1.0 series. https://bugzilla.gnome.org/show_bug.cgi?id=666361 2012-02-03 17:23:48 +0100 Wim Taymans * tests/check/elements/cmmldec.c: tests: fix more tests 2012-02-03 16:13:51 +0100 Wim Taymans * tests/check/elements/apev2mux.c: * tests/check/elements/audiofirfilter.c: * tests/check/elements/audioiirfilter.c: * tests/check/elements/cmmldec.c: * tests/check/elements/id3v2mux.c: * tests/check/elements/interleave.c: * tests/check/elements/parser.c: * tests/check/pipelines/wavenc.c: tests: fix some more tests 2012-02-03 16:12:59 +0100 Wim Taymans * gst/audioparsers/gstaacparse.c: aacparse: fix srcpad caps handling 2012-02-03 16:12:24 +0100 Wim Taymans * ext/annodex/gstcmmlenc.c: cmmlenc: fix caps handling 2012-02-03 14:53:31 +0000 Vincent Penquerc'h * ext/flac/gstflacenc.c: flacenc: fix event leak when there is no peer on the src pad 2012-02-02 16:21:29 +0000 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Update spec file 2012-02-02 12:27:09 +0000 Vincent Penquerc'h * gst/flv/gstflvmux.c: flvmux: specify we only accept raw AAC in template caps No header seems to be added, and the codec ID is the same as used for raw by flvdemux, so raw seems the only supported case. https://bugzilla.gnome.org/show_bug.cgi?id=665394 2012-02-02 12:25:21 +0000 Vincent Penquerc'h * gst/flv/gstflvdemux.c: flvdemux: specify we only output raw AAC in template caps https://bugzilla.gnome.org/show_bug.cgi?id=665394 2012-02-01 18:01:27 +0100 Mark Nauwelaerts * configure.ac: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gsttaglibmux.c: * ext/taglib/gsttaglibmux.h: taglib: port to 0.11 2012-02-01 16:40:51 +0000 Tim-Philipp Müller * ext/annodex/Makefile.am: * gst/audiofx/Makefile.am: * gst/rtpmanager/Makefile.am: * tests/examples/audiofx/Makefile.am: * tests/examples/rtp/Makefile.am: build: ignore GValueArray deprecation warnings for the time being until this gets sorted out with the GLib folks and we have a viable alternative. https://bugzilla.gnome.org/show_bug.cgi?id=667228 2012-02-01 16:36:53 +0000 Tim-Philipp Müller * ext/pulse/pulseprobe.c: * ext/pulse/pulseprobe.h: pulse: disable some unused property probe code which was using GValueArray 2012-02-01 16:20:46 +0100 Sebastian Dröge * ext/twolame/gsttwolamemp2enc.c: twolame: Use new audio encoder/decoder base class API for srcpad caps 2012-02-01 16:20:26 +0100 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lame: Use new audio encoder/decoder base class API for srcpad caps 2012-02-01 16:11:14 +0100 Sebastian Dröge * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: Use new audio encoder/decoder base class API for srcpad caps 2012-02-01 16:05:51 +0100 Sebastian Dröge * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: flac: Use new audio encoder/decoder base class API for srcpad caps 2012-01-31 15:39:09 +0100 Wim Taymans * tests/check/elements/equalizer.c: * tests/check/elements/id3demux.c: * tests/check/elements/interleave.c: * tests/check/elements/level.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/shapewipe.c: * tests/check/elements/spectrum.c: * tests/check/elements/udpsrc.c: * tests/check/elements/y4menc.c: * tests/check/pipelines/flacdec.c: * tests/check/pipelines/wavenc.c: tests: fix more tests 2012-01-30 14:52:37 +0000 Vincent Penquerc'h * gst/rtp/gstrtpmp2tpay.c: rtpmp2tpay: do not try to flush a packet when no data is available https://bugzilla.gnome.org/show_bug.cgi?id=668874 2012-01-31 13:41:45 +0100 Wim Taymans * tests/check/elements/alphacolor.c: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiofirfilter.c: * tests/check/elements/audioiirfilter.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/avimux.c: * tests/check/elements/deinterlace.c: * tests/check/elements/deinterleave.c: tests: update some tests for new memory api 2012-01-31 12:22:19 +0100 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: adapt to control-source type changes 2012-01-30 21:39:34 +0100 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: rename control-bindings gst_control_binding_xxx -> gst_xxx_control_binding for consistency. 2012-01-30 17:16:51 +0100 Wim Taymans * ext/annodex/gstcmmlenc.c: * ext/flac/gstflacenc.c: * ext/soup/gstsouphttpclientsink.c: * ext/speex/gstspeexenc.c: * gst/audioparsers/gstflacparse.c: * gst/flv/gstflvmux.c: * gst/isomp4/gstqtmux.c: * gst/matroska/ebml-write.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: update for HEADER flag 2010-06-11 08:36:33 +0200 Pascal Buhler * gst/rtp/gstrtph264depay.c: rtph264depay: Exclude NALu size from payload length on truncated packets. https://bugzilla.gnome.org/show_bug.cgi?id=667846 2012-01-28 23:35:50 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: remove obsolete variable, set but not used Reported by andredieb on #gstreamer. 2012-01-28 13:05:09 +0000 Vincent Penquerc'h * gst/videobox/gstvideobox.c: videobox: avoid wrapping opaque to transparent 2012-01-28 12:35:13 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: do not free memory twice A recent change to fix leaking codec ID string accidentally caused one of the very few places that weren't leaking to now free twice. 2012-01-27 16:27:49 +0100 Olivier Crête * gst/law/alaw-decode.c: alawdec: Each output sample is 2 bytes 2012-01-27 12:14:49 +0100 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Don't leak caps event when not pushing 2012-01-27 12:04:53 +0100 Olivier Crête * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: Forward sticky events 2012-01-27 12:04:05 +0100 Olivier Crête * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: Protect all uses pad list with OBJECT LOCK Actually protect the entire pad list and use it in a thread safe way. 2012-01-27 12:02:25 +0100 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Forward sticky events to new pads 2012-01-27 12:01:40 +0100 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Add ssrc to forwarded CAPS events Also iterate the list of GstRtpSsrcDemuxPad safely 2012-01-27 11:59:08 +0100 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrccdemux: Factor out getting dpad by pad 2012-01-26 18:35:48 +0100 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Keep the buffer mapped while it is being modified 2012-01-26 18:35:27 +0100 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.h: rtpsession: Initialise the address pointer to NULL 2012-01-27 12:07:43 +0100 Olivier Crête * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: dtmf: Use new-style caps 2012-01-27 16:37:19 +0100 Andoni Morales Alastruey * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: direcsoundsink: Port element to 0.11 2012-01-26 19:48:14 +0100 Wim Taymans * gst/videomixer/videomixer2.c: videomixer2: remove pad event function We use the one from collectpads 2012-01-26 18:26:02 +0000 Vincent Penquerc'h * gst/isomp4/qtdemux.c: Revert "qtdemux: fix GstDateTime/GDateTime mixup" This reverts commit 53261261120b4c008de61691c70e94354b28004a. The GstDateTime->GDateTime change in core was apparently accidental, and is now reverted. 2012-01-26 18:25:21 +0000 Vincent Penquerc'h * gst/avi/gstavidemux.c: Revert "avidemux: fix GstDateTime/GDateTime mixup" This reverts commit acc9f150968b25c5ae5a6940b34ad2d51b174fd2. The GstDateTime->GDateTime change in core was apparently accidental, and is now reverted. 2012-01-26 17:50:30 +0000 Vincent Penquerc'h * gst/avi/gstavidemux.c: avidemux: fix GstDateTime/GDateTime mixup This is a blind fix to match the one I just made to qtdemux, as I do not have an AVI file where the code gets executed. 2012-01-26 17:47:29 +0000 Vincent Penquerc'h * gst/isomp4/qtdemux.c: qtdemux: fix GstDateTime/GDateTime mixup 2012-01-26 18:51:30 +0100 Wim Taymans * gst/videomixer/videomixer2.c: videomixer: more fixes 2012-01-26 18:43:06 +0100 Wim Taymans * gst/videomixer/videomixer2.c: videomixer: make videomixer work somewhat 2012-01-26 18:15:51 +0100 Wim Taymans * configure.ac: * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: port to 0.11 It builds and gst-inspect-0.11 works.. otherwise untested 2012-01-26 15:48:01 +0000 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: dynudpsink: fix get-stats signal registration some more 2012-01-26 15:46:13 +0000 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: Revert "udp: mark action signals as RUN_FIRST" This reverts commit 5c8308599129d9e1606eedb2d3543617658dc306. 2012-01-26 15:39:33 +0000 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: udp: mark action signals as RUN_FIRST 2012-01-26 15:37:23 +0000 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: udp: mark "get-stats" as action signal 2012-01-26 15:30:41 +0000 Tim-Philipp Müller * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.c: udp: fix get-stats action signal registration It returns a GstStructure now, not a GValueArray 2012-01-26 16:05:34 +0100 Andoni Morales Alastruey * gst/udp/gstudpsrc.c: udpsrc: fix print format 2012-01-26 11:50:19 +0100 Sebastian Dröge * gst/matroska/ebml-write.c: matroskamux: Fix size of output buffers 2012-01-26 11:33:07 +0100 Wim Taymans * gst/isomp4/gstqtmux.c: qtmux: include right collectpads version 2012-01-26 11:29:11 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Properly use the alignment parameter of gst_buffer_new_allocate() It's a bitmask for the alignment, not the alignment itself. 2012-01-26 11:18:40 +0100 Sebastian Dröge * gst/matroska/ebml-write.c: matroskamux: Properly unmap WRITE maps of the output buffers 2012-01-26 10:44:28 +0100 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer2: Update for the new collectpads2 event handling API 2012-01-26 10:40:06 +0100 Sebastian Dröge * gst/isomp4/gstqtmux.c: qtmux: Update for the new collectpads2 event handling API 2012-01-26 10:37:52 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Update for the new collectpads2 event handling API 2012-01-26 10:28:51 +0100 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Update for new collectpads2 event handling API 2012-01-26 10:27:40 +0100 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Update for new collectpads2 event handling API 2012-01-25 18:41:38 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Only forward the event when we didn't handle it ourselves 2012-01-25 18:40:03 +0100 Wim Taymans * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixer2pad.h: videomixer: some more porting 2012-01-25 18:00:52 +0100 Wim Taymans * gst/videomixer/blend.c: * gst/videomixer/blend.h: videomixer: port blend function 2012-01-25 16:58:12 +0100 Edward Hervey * gst/flv/gstflvdemux.c: flv: Fix unitialized variables (or rather circumvent issues with naive compilers ...) 2012-01-25 15:21:44 +0000 Jayakrishnan M * ext/cairo/Makefile.am: cairo: fix build, make sure libgstvideo can be found https://bugzilla.gnome.org/show_bug.cgi?id=668648 2012-01-25 14:50:50 +0100 Wim Taymans * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: port to new memory API 2012-01-25 13:19:12 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpsession.c: rtpmanager: don't pretend our random hostnames are fully-qualified domain names 2012-01-25 13:47:30 +0100 Thomas Vander Stichele * common: Automatic update of common submodule From c463bc0 to 7fda524 2012-01-25 12:49:34 +0100 Sebastian Dröge Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:11 +0100 Sebastian Dröge Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacdec.c ext/jpeg/gstjpegenc.c ext/pulse/pulsesink.c sys/v4l2/gstv4l2src.c 2012-01-25 12:41:42 +0100 Wim Taymans * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: png: port to new memory API 2012-01-25 12:41:30 +0100 Wim Taymans * gst/matroska/matroska-demux.c: matroska: port to new memory API 2012-01-24 14:38:58 +0100 Wim Taymans * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: * ext/pulse/pulsesink.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtpmanager/rtpsession.c: * gst/rtsp/gstrtspsrc.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/videocrop/gstvideocrop.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/ximage/gstximagesrc.c: * tests/check/elements/parser.c: more memory API porting 2012-01-23 17:25:37 +0100 Wim Taymans * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/tests.c: * gst/equalizer/gstiirequalizer.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/id3demux/gstid3demux.c: * gst/isomp4/atomsrecovery.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstsplitfilesrc.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: update for new memory API 2012-01-25 07:24:59 +0100 Wim Taymans * ext/twolame/gsttwolamemp2enc.c: port to new memory API 2012-01-25 07:24:59 +0100 Wim Taymans * ext/lame/gstlamemp3enc.c: port to new memory API 2012-01-25 11:21:50 +0100 Olivier Crête * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: port to 0.11 2012-01-25 11:38:11 +0100 Sebastian Dröge * common: Automatic update of common submodule From 2a59016 to c463bc0 2012-01-24 18:24:13 +0100 Mark Nauwelaerts * ext/libpng/gstpngenc.c: pngenc: disably snapshot behaviour by default ... since such behaviour is not consistent, if allowable at all. 2012-01-24 18:23:22 +0100 Mark Nauwelaerts * configure.ac: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: pngdec: port to 0.11 2012-01-24 18:21:08 +0100 Mark Nauwelaerts * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: pngenc: port to 0.11 2012-01-24 14:53:38 +0000 Vincent Penquerc'h * gst/udp/gstudpsrc.c: udpsrc: fix string leak 2012-01-24 14:52:09 +0000 Vincent Penquerc'h * gst/udp/gstudpsrc.c: udpsrc: fix use of freed memory 2011-12-01 15:49:40 +0100 Matej Knopp * gst/matroska/matroska-demux.c: Don't crash on empty laces https://bugzilla.gnome.org/show_bug.cgi?id=665224 2012-01-23 13:15:46 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpsession.c: rtpmanager: don't reveal the user's username, hostname or real name by default Send a randomly made-up user@hostname as CNAME and don't send a NAME at all by default. https://bugzilla.gnome.org/show_bug.cgi?id=668320 2012-01-21 20:07:56 +0100 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: move from control-binding to control-binding-direct 2012-01-22 23:31:19 +0000 Tim-Philipp Müller * gst-libs/gst/glib-compat-private.h: * gst/audiofx/audiochebband.c: * gst/audiofx/audiochebband.h: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiocheblimit.h: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofirfilter.h: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioiirfilter.h: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: * gst/audiofx/audiowsinclimit.h: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstaspectratiocrop.h: Don't use deprecated GLib API 2012-01-22 23:15:19 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpclientsink.c: * gst-libs/gst/glib-compat-private.h: * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: * gst/interleave/interleave.c: * gst/rtpmanager/gstrtpsession.c: * sys/oss4/oss4-mixer.c: * tests/check/elements/multifile.c: * tests/check/elements/souphttpsrc.c: * tests/icles/equalizer-test.c: * tests/icles/gdkpixbufsink-test.c: * tests/icles/test-oss4.c: * tests/icles/v4l2src-test.c: * tests/icles/videocrop-test.c: Use new GLib API unconditionally 2012-01-20 17:06:42 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: simplify internal src event debug logging ... which avoids almost superfluous obtaining of rtsp element. 2012-01-20 17:03:50 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid NULL string comparison 2012-01-20 17:03:21 +0100 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: rtpbin: arrange for initialized variables 2012-01-20 17:02:15 +0100 Mark Nauwelaerts * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: prevent out-of-bound array access 2012-01-20 17:01:37 +0100 Mark Nauwelaerts * gst/isomp4/atomsrecovery.c: isomp4: recovery: add sanity check ... on possibly bogus/corrupt input data. 2012-01-20 17:00:51 +0100 Mark Nauwelaerts * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: remove dead code 2012-01-20 16:58:28 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroska-demux: remove redundant variable 2012-01-20 16:57:52 +0100 Mark Nauwelaerts * gst/deinterlace/gstdeinterlace.c: deinterlace: fix arithmetic for unsigned comparison 2012-01-20 16:55:06 +0100 Mark Nauwelaerts * gst/imagefreeze/gstimagefreeze.c: imagefreeze: add various missing break 2012-01-20 16:54:06 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: tweak DEFAULT format duration query response 2012-01-20 16:49:14 +0100 Mark Nauwelaerts * gst/alpha/gstalphacolor.c: alphacolor: remove redundant statement 2012-01-20 16:48:49 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: improve upstream peer duration querying ... to avoid accepting unhandled duration query result. 2012-01-20 16:47:36 +0100 Mark Nauwelaerts * ext/pulse/pulsesrc.c: pulsesrc: additional error condition checking 2012-01-20 16:46:21 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: additional error condition checking 2012-01-20 16:44:21 +0100 Mark Nauwelaerts * ext/jpeg/gstjpegenc.c: jpegenc: check _alloc_buffer result and perform fallback alloc if needed ... rather than carrying on with NULL buffer. 2012-01-20 14:45:01 +0100 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: adapt to control binding changes 2012-01-20 11:37:38 +0100 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: adapt to controller api changes Don't use the convenience api for control sources. 2012-01-19 14:24:04 +0000 Tim-Philipp Müller * common: * configure.ac: Add --disable-fatal-warnings configure option 2012-01-19 12:44:39 +0100 Wim Taymans * ext/jpeg/gstjpegenc.c: * gst/udp/gstmultiudpsink.c: update for memory API 2012-01-19 11:33:53 +0100 Wim Taymans * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/jack/gstjackaudioclient.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpclientsink.h: * ext/wavpack/gstwavpackparse.c: * gst/avi/gstavidemux.c: * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer.h: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmoovrecover.h: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/videomixer/videomixer2.c: * gst/wavparse/gstwavparse.c: * sys/v4l2/gstv4l2videooverlay.c: * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: * tests/check/elements/deinterleave.c: port to new gthread API 2012-01-18 16:58:12 +0100 Sebastian Dröge * configure.ac: configure.ac: Remove GIO check, this is in gst-glib2.m4 now 2012-01-18 16:46:17 +0100 Sebastian Dröge * common: Automatic update of common submodule From 0807187 to 2a59016 2012-01-18 16:15:59 +0100 Sebastian Dröge * configure.ac: configure.ac: Require GLib 2.31.10 and improve GIO check 2012-01-17 16:58:07 +0100 Sebastian Dröge * gst/udp/gstudpsrc.c: udpsrc: Remove unneeded socket.h include 2012-01-17 16:53:31 +0100 Sebastian Dröge * configure.ac: * gst/rtp/Makefile.am: * gst/rtp/gstasteriskh263.c: configure: Remove socket/winsock specific checks Not necessary anymore. 2012-01-17 16:49:10 +0100 Sebastian Dröge * gst/rtsp/Makefile.am: * gst/rtsp/gstrtspsrc.c: rtspsrc: Update for the new GIO versions of the udp elements 2012-01-17 13:08:42 +0100 Sebastian Dröge * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpmanager: Port to GIO 2012-01-17 11:19:33 +0100 Sebastian Dröge * configure.ac: * gst/udp/Makefile.am: configure: Require GIO 2.31.10 2012-01-17 11:18:33 +0100 Sebastian Dröge * gst/udp/gstudp.c: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: udp: Remove now unecessary code 2012-01-17 11:18:15 +0100 Sebastian Dröge * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsink.h: udpsink/multiudpsink: Port to GIO 2012-01-17 09:38:33 +0100 Sebastian Dröge * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: * gst/udp/gstudpsrc.c: dynudpsink: Port to GIO 2012-01-17 09:32:27 +0100 Sebastian Dröge * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: dynudpsink: Port to GIO 2012-01-17 09:03:38 +0100 Sebastian Dröge * gst/udp/Makefile.am: * gst/udp/gstdynudpsink.c: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: Port to GIO 2012-01-16 17:51:18 +0000 Vincent Penquerc'h * gst/cutter/gstcutter.c: cutter: fix leak of unused GValue 2012-01-16 16:10:08 +0000 Vincent Penquerc'h * tests/check/elements/autodetect.c: tests: fix autodetect test not testing correctly for state change success State change to PAUSED can be done async, so if this happens, we need to wait for the change to be done (or failed). 2012-01-16 15:42:46 +0000 Vincent Penquerc'h * gst/rtp/gstrtph263ppay.c: rtph263ppay: fix caps leak 2012-01-16 12:13:50 +0000 Vincent Penquerc'h * gst/deinterlace/gstdeinterlace.c: deinterlace: make interlacedness test deterministic If the interlaced flag is not present in the caps, we assume the data is not interlaced, instead of leaving the boolean uninitialized. 2012-01-13 18:12:05 -0500 Matej Knopp * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/multifile/gstmultifilesink.c: matroska: fix printf format compiler warnings https://bugzilla.gnome.org/show_bug.cgi?id=662615 2012-01-13 18:11:36 +0000 Vincent Penquerc'h * ext/pulse/pulsesrc.c: pulsesrc: fix wrong error check pa_stream_* functions return negative on error, despite the defines for error codes being positive. I only got to repro the error twice, so I'm not sure 100% sure this fixes the issue (the negative var being uninitialized after returning from pa_stream_get_latency). 2012-01-13 17:43:49 +0000 Vincent Penquerc'h * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: oss4: fix caps leaks 2012-01-13 17:25:59 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2src.c: v4l2src: fix caps leak 2012-01-13 15:57:20 +0000 Vincent Penquerc'h * tests/check/elements/videocrop.c: tests: fix caps leak in videotestsrc test 2012-01-13 12:50:06 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: clean up obsolete closing segment handling 2012-01-13 10:32:59 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: plug pad leak in error code path Based on patch by: Stig Sandnes Don't leak srcpad if there are no caps. https://bugzilla.gnome.org/show_bug.cgi?id=667820 2011-10-04 10:00:02 +0200 Stig Sandnes * sys/osxvideo/cocoawindow.m: osxvideo: Fix leak of NSOpenGLPixelFormat object https://bugzilla.gnome.org/show_bug.cgi?id=667818 2011-09-05 10:43:19 +0200 Havard Graff * sys/v4l2/gstv4l2src.c: v4l2src: Don't assert when the interface is not implemented. Simply return FALSE instead. https://bugzilla.gnome.org/show_bug.cgi?id=667817 2012-01-12 00:18:39 +0200 Raimo Järvi * sys/waveform/gstwaveformsink.c: * sys/waveform/gstwaveformsink.h: waveformsink: Fix mingw warnings https://bugzilla.gnome.org/show_bug.cgi?id=667719 2012-01-12 23:55:31 +0000 Tim-Philipp Müller * gst/apetag/gstapedemux.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-read-common.c: GST_TYPE_DATE -> G_TYPE_DATE 2012-01-12 23:48:50 +0000 Tim-Philipp Müller eqMerge remote-tracking branch 'origin/master' into 0.11 Conflicts: ext/jack/gstjackaudiosink.c ext/jack/gstjackaudiosrc.c gst/matroska/matroska-mux.c gst/matroska/matroska-read-common.c gst/rtpmanager/gstrtpssrcdemux.c 2012-01-12 18:23:42 +0000 Vincent Penquerc'h * gst/rtpmanager/gstrtpssrcdemux.c: gstrtpssrcdemux: fix element leak 2012-01-12 14:19:22 +0000 Vincent Penquerc'h * gst/matroska/matroska-read-common.c: matroska: do not leak attachment buffers 2012-01-12 13:17:55 +0100 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: remove obsolete FIXME comments 2012-01-12 10:30:11 +0000 Vincent Penquerc'h * ext/flac/gstflacenc.c: flacenc: do not drop the first data buffer on the floor (and leak it either) 2012-01-12 11:08:38 +0100 Mark Nauwelaerts * gst/flv/gstindex.c: * gst/flv/gstmemindex.c: flvdemux: add prefix to local GstIndex related copies ... to avoid duplicate type names with other such local copies in the wild. 2012-01-12 11:07:33 +0100 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: activate pad before setting caps ... rather than the usual 0.10 other way around. Fixes #667558. 2012-01-11 18:45:33 -0300 Reynaldo H. Verdejo Pinochet * Android.mk: Temporarily disabling multifile for the Android build There is a hard dependency on inotify comming from gio. We are not currently bundling inotify with the Android dist so I'm disabling multifile for now until someone gets around to sort this out. This change fixes building on Android 2010-10-20 02:17:43 -0700 Leo Singer * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioiirfilter.c: * tests/check/elements/audioiirfilter.c: audiofx: Use most common convention for definitions of IIR filter coefficients. Most signal processing texts, including MATLAB, use the following convention for IIR filter coefficients: a_0 y[n] + a_1 y[n-1] + ... + a_M y[n-M] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] Usually, a_0 is set to 1 because the coefficients can always be rescaled, giving y[n] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] - a_1 y[n-1] - ... - a_M y[n-M] The convention that was previously used by audiofxbaseiirfilter and derived class had the a and b coefficients swapped, and did not have the minus signs. This change makes the audiofx plugin use the more common convention described above. 2012-01-11 14:47:36 +0100 Stefan Sauer * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: add a transport mode enum Clients can configure the desired behaviour via "transport" property. The default behaviour is ignoring the transport state. Other modes are master and slave. 2012-01-11 14:10:46 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Fix buffer handling souphttpsrc is now usable again and doesn't crash anymore whenever something is read from a HTTP connection. 2012-01-11 01:45:34 +0000 Tim-Philipp Müller * tests/check/pipelines/wavenc.c: tests: fix wavenc test on big endian wavenc only accepts little-endian PCM, but most of our elements such as audiotestsrc only produce or process audio in native endianness, so we need to plug a converter before wavenc on big endian systems. 2012-01-10 23:02:45 +0100 Stefan Sauer * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: deactivate the request_state code When qjackctl is started, transport is stopped by default. This would be a regression for gstreamer apps that before just started to play right away. 2012-01-10 22:27:11 +0100 Stefan Sauer * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudioclient.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: add transport control handling This feature allows to start and stop playback from other jack applications (e.g. qjackctl). 2012-01-10 18:50:27 +0100 Nicola Murino * gst/matroska/matroska-mux.c: matroskamux: fix codec_priv leaks https://bugzilla.gnome.org/show_bug.cgi?id=667419 2012-01-10 15:17:11 +0100 Sebastian Dröge Merge branch 'master' into 0.11 Conflicts: ext/a52dec/gsta52dec.c ext/a52dec/gsta52dec.h ext/lame/gstlame.c ext/lame/gstlame.h ext/lame/gstlamemp3enc.c ext/mad/gstmad.c ext/mad/gstmad.h gst/mpegaudioparse/gstmpegaudioparse.c gst/mpegstream/gstdvddemux.c gst/realmedia/rdtdepay.c po/es.po po/lv.po po/sr.po 2012-01-10 15:06:39 +0100 Stefan Sauer * ext/jack/gstjackaudioclient.c: jack: use jack type for the callback Jack headers have a typedef for the shutdown callback as well. 2012-01-10 14:32:32 +0100 Sebastian Dröge Merge branch 'master' into 0.11 Conflicts: ext/cairo/gsttextoverlay.c ext/pulse/pulseaudiosink.c gst/audioparsers/gstaacparse.c gst/avi/gstavimux.c gst/flv/gstflvmux.c gst/interleave/interleave.c gst/isomp4/gstqtmux.c gst/matroska/matroska-demux.c gst/matroska/matroska-mux.c gst/matroska/matroska-mux.h gst/matroska/matroska-read-common.c gst/multifile/gstmultifilesink.c gst/multipart/multipartmux.c gst/shapewipe/gstshapewipe.c gst/smpte/gstsmpte.c gst/udp/gstmultiudpsink.c gst/videobox/gstvideobox.c gst/videocrop/gstaspectratiocrop.c gst/videomixer/videomixer.c gst/videomixer/videomixer2.c gst/wavparse/gstwavparse.c po/ja.po po/lv.po po/sr.po tests/check/Makefile.am tests/check/elements/qtmux.c tests/check/elements/rgvolume.c 2012-01-09 22:58:32 +0530 Arun Raghavan * docs/plugins/Makefile.am: docs: Remove old videomixer headers These got removed in the transition to videomixer2. 2012-01-09 17:28:17 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: matroskamux: fix codec string leaks 2012-01-09 14:51:44 +0100 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixerpad.h: videomixer: Remove videomixer and register videomixer2 as videomixer 2012-01-09 11:36:58 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: initialize variable to avoid undefined use 2012-01-06 09:40:22 +0100 Sebastian Dröge * configure.ac: * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flac: Port to the new raw audio caps 2012-01-05 19:25:33 +0000 Vincent Penquerc'h * gst/isomp4/gstqtmux.c: isomp4: fix caps leak 2012-01-05 19:08:03 +0000 Vincent Penquerc'h * gst/isomp4/gstqtmux.c: isomp4: remove dead assignment 2012-01-05 14:18:03 +0100 Wim Taymans * gst/auparse/gstauparse.c: * gst/wavenc/gstwavenc.c: fix pad templates 2012-01-04 15:44:37 +0100 Sebastian Dröge * ext/twolame/gsttwolamemp2enc.c: twolamemp2enc: Update for the new raw audio caps 2012-01-04 15:45:43 +0100 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lamemp3enc: Update for the new raw audio caps 2012-01-04 15:05:41 +0100 Sebastian Dröge * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: Update for the new raw audio caps 2012-01-04 14:54:10 +0100 Sebastian Dröge * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: Add the new layout field to the raw audio caps 2012-01-04 14:52:46 +0100 Sebastian Dröge * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackutil.c: * ext/jack/gstjackutil.h: jackaudiosrc: Port to the new multichannel audio caps 2012-01-04 14:13:54 +0100 Sebastian Dröge * configure.ac: configure: Add FLAC and interleave to the non-ported plugins list Both need to be updated to the audio/x-raw caps and were only half-ported before. 2012-01-04 13:48:36 +0100 Sebastian Dröge * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpchannels.c: * gst/rtp/gstrtpchannels.h: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpvrawpay.c: rtp: Update for the new audio caps 2012-01-04 12:06:12 +0100 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Update for libgstriff API changes Still needs to handle raw audio channel reordering 2012-01-04 12:05:16 +0100 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Update for the new raw audio caps 2012-01-04 12:03:50 +0100 Sebastian Dröge * gst/spectrum/gstspectrum.c: spectrum: Update for the new raw audio caps layout field 2012-01-04 11:57:20 +0100 Sebastian Dröge * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: replaygain: Update for the new audio caps 2012-01-04 11:52:29 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: Update for the new raw audio interleaved caps field Still needs to be fixed to handle the multichannel channel-mask and reordering. 2012-01-04 11:31:07 +0100 Sebastian Dröge * gst/level/gstlevel.c: level: Update for the new raw audio layout field 2012-01-04 11:29:26 +0100 Sebastian Dröge * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/qtdemux.c: isomp4: Port to the new audio caps Still needs to handle the channel positions/masks and channel reordering. 2012-01-04 11:11:06 +0100 Sebastian Dröge * gst/cutter/gstcutter.c: cutter: Update for the new raw audio layout field 2012-01-04 11:09:32 +0100 Sebastian Dröge * gst/goom/gstgoom.c: goom: Port to the new multichannel caps and update for the new raw audio layout field 2012-01-04 11:08:18 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: equalizer: Update for the new raw audio layout field 2012-01-04 11:07:29 +0100 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Update for the libgstriff API changes Still needs to do reordering of channels for raw audio. 2012-01-04 11:06:28 +0100 Sebastian Dröge * gst/auparse/gstauparse.c: auparse: Port to the new multichannel caps and the new raw audio layout field 2012-01-04 11:02:43 +0100 Sebastian Dröge * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: audiofx: Port to the new multichannel caps and the new raw audio layout field 2012-01-04 10:54:46 +0100 Sebastian Dröge * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: oss: Port to the new multichannel caps and the raw audio caps interleaved field 2012-01-04 10:27:09 +0100 Sebastian Dröge * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: pulse: Port to the new multichannel caps 2012-01-04 19:51:46 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 762b692 to 0807187 2012-01-04 17:05:32 +0000 Tim-Philipp Müller * ext/lame/Makefile.am: lame: fix LIBADD order in Makefile.am 2012-01-04 17:59:55 +0000 Tim-Philipp Müller * tests/check/elements/qtmux.c: tests: fix some leaks and remove files when done in qtmux test 2011-12-14 10:14:20 +0100 Peter Seiderer * gst/multifile/gstmultifilesink.c: multifilesink: post better error message when we run out of disk space Map write errno ENOSPC to GST_RESOURCE_ERROR_NO_SPACE_LEFT. 2012-01-04 13:26:45 +0100 Edward Hervey * gst/alpha/gstalphacolor.c: * tests/check/elements/alphacolor.c: alphacolor: More fixes/cleanup 2012-01-04 13:25:40 +0100 Edward Hervey * gst/alpha/gstalpha.c: alpha: Refactor param/process functions When ::set_info() is called, the input/output VideoInfo aren't set yet on the videofilter. 2012-01-04 10:01:48 +0100 Wim Taymans * ext/cairo/gsttextoverlay.c: * ext/dv/gstdvdemux.c: * ext/libpng/gstpngdec.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/wavpack/gstwavpackparse.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/interleave.c: * gst/videomixer/videomixer2.c: GST_FLOW_UNEXPECTED -> GST_FLOW_EOS 2011-12-31 23:33:33 -0500 Matej Knopp * gst/audioparsers/gstdcaparse.c: dcaparse: use right variable Fixes use of unitialized variable. https://bugzilla.gnome.org/show_bug.cgi?id=667085 2012-01-03 15:26:21 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: * ext/soup/gstsouphttpsrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/matroska/ebml-read.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstsplitfilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: * gst/wavparse/gstwavparse.c: GST_FLOW_UNEXPECTED -> GST_FLOW_EOS 2012-01-03 14:42:28 +0100 Wim Taymans * tests/check/pipelines/tagschecking.c: tests: rewrite test a little Rewrite the tag check so that we don't need to deal with tag lists. 2012-01-03 14:16:28 +0100 Wim Taymans * tests/check/Makefile.am: * tests/check/elements/jpegenc.c: * tests/check/elements/multifile.c: * tests/check/elements/qtmux.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtpbin.c: * tests/check/elements/rtpbin_buffer_list.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/shapewipe.c: * tests/check/elements/souphttpsrc.c: * tests/check/elements/udpsink.c: * tests/check/elements/videocrop.c: * tests/check/elements/videofilter.c: * tests/check/elements/y4menc.c: * tests/check/pipelines/flacdec.c: * tests/check/pipelines/tagschecking.c: tests: make more tests compile 2012-01-03 11:56:25 +0100 Wim Taymans * tests/check/Makefile.am: * tests/check/elements/equalizer.c: * tests/check/elements/flacparse.c: * tests/check/elements/flvdemux.c: * tests/check/elements/flvmux.c: * tests/check/elements/icydemux.c: * tests/check/elements/imagefreeze.c: * tests/check/elements/interleave.c: * tests/check/elements/level.c: * tests/check/elements/multifile.c: * tests/check/elements/qtmux.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: test: make more unit tests compile 2012-01-03 10:26:48 +0100 Wim Taymans * tests/check/Makefile.am: * tests/check/elements/audiofirfilter.c: * tests/check/elements/audioiirfilter.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/autodetect.c: * tests/check/elements/avimux.c: * tests/check/elements/avisubtitle.c: * tests/check/elements/capssetter.c: * tests/check/elements/deinterlace.c: * tests/check/elements/deinterleave.c: * tests/check/generic/index.c: * tests/check/generic/states.c: tests: fix some unit tests Remove unit test for GstIndex. Make some other unit tests compile 2012-01-02 14:32:40 +0000 Tim-Philipp Müller * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/rtsp/gstrtspext.c: autodetect, rtsp: gst_registry_get_default() -> gst_registry_get() 2011-12-31 10:00:41 +0100 Stefan Sauer * tests/examples/v4l2/camctrl.c: controller: port to API changes 2011-12-30 17:41:46 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: update for GstIndex removal 2011-12-30 17:23:43 +0000 Tim-Philipp Müller * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: update for GstIndex removal 2011-12-30 17:20:57 +0000 Tim-Philipp Müller * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstindex.c: * gst/flv/gstindex.h: * gst/flv/gstmemindex.c: flvdemux: update for GstIndex removal Add private GstMemIndex for now. 2011-12-30 17:12:03 +0000 Tim-Philipp Müller * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: update for GstIndex removal 2011-12-27 22:59:03 +0000 Tim-Philipp Müller * sys/waveform/gstwaveformsink.c: waveformsink: fix compiler warnings with MingW https://bugzilla.gnome.org/show_bug.cgi?id=666485 2011-12-27 22:54:34 +0000 Tim-Philipp Müller * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: fix printf format in debug statements https://bugzilla.gnome.org/show_bug.cgi?id=666926 2011-12-27 12:06:16 +0000 Tim-Philipp Müller * tests/check/elements/.gitignore: tests: make git ignore new unit test binary 2011-12-27 11:50:03 +0000 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: fix valgrind warning https://bugzilla.gnome.org/show_bug.cgi?id=666644 2011-12-27 11:49:10 +0000 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/udpsrc.c: udpsrc: add unit test that sends 0-size packet https://bugzilla.gnome.org/show_bug.cgi?id=666644 2011-12-21 13:22:03 +0100 John Ogness * gst/udp/gstudpsrc.c: udpsrc: drop dataless UDP packets It is allowed to send/receive UDP packets with no data. When such a packet is available, select() will return with success but ioctl(FIONREAD) will return 0. But a read() must still occur in order to clear off the UDP packet from the queue. This patch will read the dataless packet from the socket. If select() was woken for other reasons (and FIONREAD returns 0), this may result in a UDP packet getting accidentally dropped. But since UDP is not reliable, this is acceptable. NOTE: This patch fixes a nasty bug where sending a dataless UDP packet to a udpsrc instance will cause an infinite loop. https://bugzilla.gnome.org/show_bug.cgi?id=666644 Signed-off-by: John Ogness 2011-12-26 22:22:59 +0000 Yaakov Selkowitz * configure.ac: * sys/Makefile.am: * sys/waveform/Makefile.am: waveform: add autotools bits for waveform plugin https://bugzilla.gnome.org/show_bug.cgi?id=666485 2011-12-21 20:50:21 +0100 Nicola Murino * ext/jpeg/gstjpegdec.c: jpegdec: fix peer_caps leak https://bugzilla.gnome.org/show_bug.cgi?id=666688 2011-12-26 18:24:32 +0100 Mark Nauwelaerts * ext/lame/gstlame.c: * ext/lame/gstlame.h: lame: ensure parsed output ... by doing some basic parsing of encoded lame data. 2011-12-26 16:34:01 +0100 Mark Nauwelaerts * ext/lame/gstlame.h: lame: cleanup unused instance struct fields 2011-12-26 18:23:52 +0100 Mark Nauwelaerts * ext/lame/Makefile.am: * ext/lame/gstlamemp3enc.c: * ext/lame/gstlamemp3enc.h: lamemp3enc: ensure parsed output ... by doing some basic parsing of encoded lame data. Fixes #652150. 2011-12-26 18:15:41 +0100 Mark Nauwelaerts * ext/lame/gstlamemp3enc.c: lamemp3enc: do not leak merged tags 2011-12-25 23:52:46 +0000 Tim-Philipp Müller * configure.ac: configure: remove unnecessary check for gdp library 2011-12-25 22:17:53 +0000 Tim-Philipp Müller * docs/plugins/inspect/plugin-pulseaudio.xml: * ext/pulse/Makefile.am: * ext/pulse/plugin.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulse: remove pulseaudiosink helper bin This is causing us lots of headaches in 0.10 and needs to be done differently and properly in 0.11. playbin or decodebin should reconfigure themselves based on reconfigure events, for example. 2011-12-25 21:45:45 +0000 Tim-Philipp Müller * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: pulse: update for ring buffer audio format type enum rename 2011-12-25 20:34:52 +0100 Stefan Sauer * tests/examples/v4l2/camctrl.c: controller: port to new control source api 2011-12-25 14:23:29 +0000 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: don't try to push already-freed buffers Fixes unit test. 2011-12-24 10:57:42 +0100 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Use scale_ceil() functions from core instead of custom ones 2011-12-21 23:51:03 +0100 Wim Taymans * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavigationtest.h: * gst/effectv/gstaging.c: * gst/effectv/gstaging.h: * gst/effectv/gstdice.c: * gst/effectv/gstdice.h: * gst/effectv/gstedge.c: * gst/effectv/gstedge.h: * gst/effectv/gstop.c: * gst/effectv/gstop.h: * gst/effectv/gstquark.c: * gst/effectv/gstquark.h: * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: * gst/effectv/gstrev.c: * gst/effectv/gstrev.h: * gst/effectv/gstripple.c: * gst/effectv/gstripple.h: * gst/effectv/gstshagadelic.c: * gst/effectv/gstshagadelic.h: * gst/effectv/gststreak.c: * gst/effectv/gststreak.h: * gst/effectv/gstvertigo.c: * gst/effectv/gstvertigo.h: * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: * gst/videofilter/gstgamma.c: * gst/videofilter/gstgamma.h: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: update for videofilter changes. 2011-12-21 17:43:10 +0100 Branko Subasic * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: do not consider duration of non-finalized file ... to avoid it clamping requested seek position. Non-finalized file case, determined by whether _parse_blockgroup_or_simpleblock ever updates the segment duration. Fixes #652195. 2011-12-21 15:06:57 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: improve decision to fall back to scanning when seeking ... which is basically iff not streaming and no entry found in index 2011-12-21 09:09:27 +0100 Oleksij Rempel (Alexey Fisher) * gst/audioparsers/gstaacparse.c: ac3parse: remove unused variable remove unused variable to fix compile error: make -C audioparsers make[3]: Betrete Verzeichnis '/home/lex/tmp/gst-plugins-good/gst/audioparsers' CC libgstaudioparsers_la-gstaacparse.lo gstaacparse.c: In function 'gst_aac_parse_read_loas_audio_specific_config': gstaacparse.c:446:12: error: variable 'sbr' set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors Signed-off-by: Oleksij Rempel (Alexey Fisher) 2011-12-21 11:59:46 +0100 Wim Taymans * ext/pulse/pulsemixer.c: * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * tests/examples/pulse/pulse.c: * tests/examples/v4l2/Makefile.am: * tests/examples/v4l2/probe.c: update for removed property probe 2011-09-09 11:42:09 +0100 Vincent Penquerc'h * gst/audioparsers/gstac3parse.c: ac3parse: let bsid 9 and 10 through Files with 9 and 10 happen, and seem to comply with the <= 8 format, so let them through. The spec says nothing about 9 and 10. https://bugzilla.gnome.org/show_bug.cgi?id=658546 2011-12-19 23:50:19 +0100 Stefan Sauer * tests/examples/v4l2/camctrl.c: controller: port to new interpolation-mode api 2011-12-19 22:53:57 +0100 Stefan Sauer * tests/examples/v4l2/camctrl.c: controller: port to new controller api 2011-12-19 19:03:52 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: update for new interlaced caps 2011-12-16 19:15:38 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: properly determine final duration ... which can be authoratively obtained from our own written timestamps. 2011-12-19 13:56:30 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: only write full metadata at start ... rather than having (potentially) unnecessary duplicates written all over, or even contradictory varying filesize info, or duration info that will not be rewritten upon header rewrite. 2011-12-16 19:15:03 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: use GstCollectPads2 buffer callback and running time clipper ... since the default collection heuristics suffice. 2011-12-16 18:03:01 +0100 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: use GstCollectPads2 buffer callback and running time clipper ... since default collection heuristics suffice. 2011-12-16 17:20:51 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: bring a few debug statements up to specs ... and minor spelling fix. 2011-12-16 16:56:37 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: additional subtitle support 2011-12-15 21:50:42 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: additional buffer handling cleanup 2011-12-15 21:45:17 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: use GstCollectPads2 buffer callback and running time clipper 2011-12-07 13:24:55 +0000 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: parse LOAS variant The LOAS variant seems to have three different subvariants itself, only one of them is implemented as my two samples happen to be using that one. The sample rate is not always reported correctly, as the "main" sample rate is apparently sometimes half what it should be (both of my samples report 24000 Hz there), and there are two other parts of the subvariant with different sampling rates. One of them is parsed, but not the other, as it's located after some other large amount of variable data that needs parsing first, and there seems to be a LOT of it, which is useless for our needs here. This ends up being rather inconsequential, as ffdec_aac_latm, which is the only decoder that can decode such streams, does not need the sample rate on the caps anyway. https://bugzilla.gnome.org/show_bug.cgi?id=665394 2011-12-19 10:48:54 +0100 Wim Taymans * gst/wavparse/gstwavparse.c: wavparse: don't remove srcpad Don't remove the always srcpad in ready and make the element reusable. 2011-12-15 16:40:21 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: use GstCollectPads2 event callback ... in stead of local HACK. 2011-12-15 16:30:17 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: use GstCollectPads2 event callback ... in stead of local HACK. 2011-12-15 16:16:52 +0100 Mark Nauwelaerts * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: use GstCollectPads2 event callback ... in stead of local HACK. 2011-12-15 16:15:22 +0100 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: use GstCollectPads2 event callback ... in stead of local HACK. 2011-12-14 19:13:21 +0000 Vincent Penquerc'h * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: smpte: port to GstCollectPads2 2011-12-14 19:10:53 +0000 Vincent Penquerc'h * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: multipartmux: port to GstCollectPads2 2011-12-14 19:07:23 +0000 Vincent Penquerc'h * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: port to GstCollectPads2 2011-12-14 19:02:23 +0000 Vincent Penquerc'h * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: port to GstCollectPads2 2011-12-14 18:55:36 +0000 Vincent Penquerc'h * gst/interleave/interleave.c: * gst/interleave/interleave.h: interleave: port to GstCollectPads2 2011-12-14 18:52:37 +0000 Vincent Penquerc'h * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flxmux: port to GstCollectPads2 2011-12-14 18:38:09 +0000 Vincent Penquerc'h * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: port to GstCollectPads2 2011-12-14 18:34:25 +0000 Vincent Penquerc'h * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: cairotextoverlay: port to GstCollectPads2 2011-12-13 18:18:45 +0100 Mark Nauwelaerts * gst/matroska/matroska-read-common.c: matroskademux: filter bogus index entries with missing block number ... to avoid contradictory information resulting in seeks sending more downstream than needed for the corresponding segment. 2011-12-13 18:15:18 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: cater for safer arithmetic with global start time 2011-12-13 17:02:01 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: tweak final closing segment sending ... to avoid it interfering with (sparse) stream syncing. 2011-12-12 11:51:06 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: make debug message more useful Add information about the taglist and which pad received the tag event on the debug logging. 2011-12-13 11:46:43 +0000 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: avoid using floating point unnecessarily https://bugzilla.gnome.org/show_bug.cgi?id=665911 2011-12-13 11:42:40 +0000 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: fix format specifier signedness Use unsigned specifiers for all unsigned values. A lot of the values used here are unsigned, and some can take high enough values that their signed counterpart will be negative. https://bugzilla.gnome.org/show_bug.cgi?id=665911 2011-12-12 16:49:19 +0000 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: add a ignore-length property This allows playing broken streams which write an incorrect length in their data chunks (such as, at least, one streaming camera). https://bugzilla.gnome.org/show_bug.cgi?id=665911 2011-12-12 11:54:56 +0100 Sebastian Dröge * gst-libs/gst/glib-compat-private.h: glib-compat: Add license boilerplate for LGPL 2011-12-12 15:15:46 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: mind (un)signed in some timestamp arithmetic ... to avoid ending up with invalid (negative) duration. 2011-02-09 15:31:22 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: increase parse tolerance for fuzzy file cases 2011-12-12 10:38:20 +0000 Tim-Philipp Müller * Makefile.am: build: dist glib-compat-private.h properly Add missing slash. 2011-12-12 10:18:14 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: use atexit, g_atexit has been deprecated in glib master 2011-12-12 02:52:13 +0000 Tim-Philipp Müller * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/wavpack/gstwavpackparse.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtspsrc.c: * gst/videomixer/videomixer2.c: * gst/wavparse/gstwavparse.c: Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly GStaticRecMutex is part of our API/ABI, not much we can do here in 0.10 for most of these. 2011-12-12 02:41:37 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: * tests/icles/equalizer-test.c: * tests/icles/gdkpixbufsink-test.c: * tests/icles/test-oss4.c: * tests/icles/videocrop-test.c: tests: g_thread_init() is deprecated in glib master It's not needed any longer. 2011-12-12 02:38:37 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpclientsink.c: * gst/rtpmanager/gstrtpsession.c: * sys/oss4/oss4-mixer.c: * tests/icles/v4l2src-test.c: Use g_thread_try_new() instead of g_thread_crate() with newer glib versions 2011-12-12 02:31:36 +0000 Tim-Philipp Müller * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: use new glib API for static mutex if available 2011-12-12 02:30:45 +0000 Tim-Philipp Müller * Makefile.am: * ext/jack/gstjackaudioclient.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/soup/gstsouphttpclientsink.c: * gst-libs/gst/glib-compat-private.h: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/equalizer/gstiirequalizer.c: * gst/imagefreeze/gstimagefreeze.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/shapewipe/gstshapewipe.c: * gst/udp/gstmultiudpsink.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: * sys/oss4/oss4-mixer.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2xoverlay.c: * sys/ximage/gstximagesrc.c: Work around deprecated thread API in glib master Add private replacements for deprecated functions such as g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly to avoid the deprecation warnings. We'll change these over to the new API once we depend on glib >= 2.32. 2011-12-12 10:24:45 +0100 Sebastian Dröge * configure.ac: configure: Require GLib >= 2.24 All other modules require this already and nobody is testing with older versions anyway. 2011-12-11 18:40:31 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufsink.c: gdkpixbufsink: fix inverted pixel-aspect-ratio Spotted by Mike Morrison. https://bugzilla.gnome.org/show_bug.cgi?id=665882 2011-12-11 17:55:14 +0000 Tim-Philipp Müller * ext/pulse/pulseaudiosink.c: pulseaudiosink: don't leak pad template 2011-12-10 14:48:57 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpclientsink.c: soup: fix start/stop race in souphttpclientsink Fix crash or hang in generic/states unit test when doing stop() right after start(). Create main loop in the start function already and not just in the thread function, so that stop() always has a valid main loop to quit on. Also, calling g_main_loop_quit() before g_main_loop_run() won't work and result in the stop function waiting for the thread to join forever. Therefore, wait for the thread to be ready and get the main loop running in the start() function, to be sure stop() always works. 2011-12-10 13:35:08 +0000 Tim-Philipp Müller * tests/files/Makefile.am: tests: dist test file used in matroskaparse unit test 2011-12-10 12:32:32 +0000 Tim-Philipp Müller * tests/check/elements/rgvolume.c: tests: fix up rgvolume test for basetransform event caching Some tests assumed that tag events would always pushed through immediately, which isn't the case any longer, so push a newsegment event and an empty buffer first. 2011-12-10 11:12:01 +0100 Wim Taymans * gst/rtpmanager/gstrtpssrcdemux.c: ssrcdemux: fix iterator and caps 2011-12-10 11:11:00 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: forward the caps event 2011-12-10 11:09:43 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: simply forward the caps event forward the caps event we get as input instead of making a new event etc.. 2011-12-09 20:10:19 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: forward caps 2011-12-09 19:46:02 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtp: pass parent to setcaps methods 2011-12-10 02:21:02 +0000 Tim-Philipp Müller * po/LINGUAS: * po/eo.po: * po/ja.po: * po/lv.po: * po/sr.po: po: update translations 2011-12-09 16:04:56 +0000 Tim-Philipp Müller * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: rename "client" properties to "client-name" Better name, but also matches the property on the jack elements (where "client" is used for something else). 2011-12-09 15:50:28 +0000 Tim-Philipp Müller * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: don't leak client name when freeing the element And add gtk-doc chunks for the new property. https://bugzilla.gnome.org/show_bug.cgi?id=665872 2011-12-09 15:45:03 +0000 Nicolas Baron * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: add "client-name" property to jackaudiosink and jackaudiosrc https://bugzilla.gnome.org/show_bug.cgi?id=665872 2011-12-09 12:19:13 +0000 Tim-Philipp Müller * gst/law/Makefile.am: law: fix CFLAGS and LIBS order in Makefile.am 2011-12-09 12:15:30 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 2011-12-09 10:51:14 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtp: fix marshallers Remove custom marshallers for minobject. Init RTCP buffer correctly. Handle results from setcaps Remove asserts. 2011-12-09 10:50:18 +0100 Wim Taymans * gst/law/Makefile.am: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/alaw.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: law: fix negotiation 2011-12-08 11:00:45 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: stream-format=raw goes with aac caps, not mp3 caps 2011-12-08 01:28:26 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 Conflicts: sys/v4l2/gstv4l2object.c 2011-12-02 12:07:24 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2src: do not ignore the highest frame interval https://bugzilla.gnome.org/show_bug.cgi?id=665387 2011-12-02 11:59:03 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2src: do not ignore the largest resolution The 'max' value isn't an STL style "one after the end" bound, but the largest allowed value. https://bugzilla.gnome.org/show_bug.cgi?id=665387 2011-12-06 16:47:25 +0100 Stefan Sauer * gst/multifile/gstmultifilesink.h: docs: add add the two enum values that were just added too 2011-12-06 16:14:54 +0100 Stefan Sauer * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/multifile/gstmultifilesink.h: multifilesink: expose the enum property docs for splitting mode. Fixes #665666. 2011-12-06 14:23:30 +0100 Wim Taymans * gst/rtp/gstrtph263pay.c: h263pay: fix invalid return value 2011-12-06 13:59:52 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: remove unused flush param 2011-12-05 18:40:26 +0100 Edward Hervey * gst/isomp4/gstrtpxqtdepay.c: rtpxqtdepay: Initialize GstRTPBuffer before usage 2011-12-05 18:40:12 +0100 Edward Hervey * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpmanager: Initialize GstRTPBuffer before usage 2011-12-05 18:39:59 +0100 Edward Hervey * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: Initialize GstRTPBuffer before usage 2011-12-05 12:15:21 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2: replace deprecated GST_CLASS_LOCK 2011-11-24 13:58:01 +0100 Sebastian Rasmussen * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Ceil jpeg dimensions, instead of floor A JPEG image inside an RTP stream has a preceeding RFC2435 header that conveys width/height. The dimensions in this header are limited to be multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must already indirectly have image data dimensions that are rounded up in order to contain enough data to render the image. Therefore this fix safely rounds the image dimensions in the RFC2435 header up to the closest multiple of 8. 2011-12-04 12:50:57 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: ensure we only check for sample/block mixup at start Otherwise we might trigger at some point within the file, but the check is only making sense for the second block. 2011-12-03 18:14:59 +0000 Vincent Penquerc'h * gst/matroska/matroska-parse.c: matroskaparse: warn if accumulating headers after they were pushed https://bugzilla.gnome.org/show_bug.cgi?id=665412 2011-10-25 12:54:43 -0700 David Schleef * gst/matroska/matroska-parse.c: matroskaparse: fix parsing Mark more parts as belonging to streamheaders. 2011-12-03 17:30:10 +0000 Vincent Penquerc'h * gst/flv/gstflvdemux.c: flvdemux: fix discontinuity threshold check when timestamps go backwards Since unsigned types are used, a negative value would show as very, very positive. Fixes A/V sync on some... less than well made files where timestamps go backwards. 2011-12-02 22:25:17 +0100 Wim Taymans * ext/soup/gstsouphttpclientsink.c: * gst/debugutils/testplugin.c: * gst/multifile/gstmultifilesink.c: update for basesink event handler changes 2011-12-02 12:01:22 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2src: add a comment about a "hidden" assumption on rank values https://bugzilla.gnome.org/show_bug.cgi?id=665387 2011-12-02 01:58:30 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 Conflicts: docs/plugins/inspect/plugin-esdsink.xml docs/plugins/inspect/plugin-gconfelements.xml ext/pulse/pulseaudiosink.c gst/matroska/matroska-demux.c gst/matroska/matroska-mux.c gst/multifile/gstmultifilesink.c 2011-12-01 18:55:45 +0100 Wim Taymans * gst/isomp4/qtdemux.c: * gst/matroska/matroska-read-common.c: * tests/check/elements/id3demux.c: update for tag API changes 2011-12-01 15:29:15 +0000 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: placate gcc since -Werror is used Initialize values that GCC cannot prove are not used without being initialized, and assert that I did not mess up my proof. 2011-12-01 14:13:05 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: fix up LIBS order som more` 2011-12-01 13:22:42 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroska-mux: fix name of new property and the unit test https://bugzilla.gnome.org/show_bug.cgi?id=654379 2011-09-25 14:57:56 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: add basic buffer list handling We assume for now that all buffers in a buffer list should end up in the same file (so we can group GOPs in buffer lists, for example). Could optimise this a bit to avoid the memcpy. 2011-09-23 18:43:35 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: write stream-headers when switching to the next file in max-size mode 2011-09-23 18:31:01 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: add new 'max-size' mode for switching to the next file 2011-09-23 17:49:05 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: add "max-file-size" property for new next-file mode 2011-12-01 13:38:06 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't forget SSA subtitles in last commit 2011-12-01 13:34:52 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Only check for markup and escape if necessary for plaintext subtitles Otherwise we break USF and ASS/SSA subtitles. 2011-12-01 13:23:33 +0100 Alessandro Decina * gst/multifile/Makefile.am: multifile: fix build in uninstalled setup Add -base libs includes to CFLAGS, fix order of LIBS . 2011-12-01 13:08:01 +0100 Alessandro Decina * tests/check/elements/multifile.c: tests: fix g_mkdtemp presence check in multifile tests g_mkdtemp was added in glib 2.30 even though the doc claims it was added in 2.26. 2011-07-17 23:56:04 +0200 Alessandro Decina * gst/multifile/Makefile.am: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * tests/check/Makefile.am: * tests/check/elements/multifile.c: multifilesink: add flag to cut after a force key unit event 2011-12-01 12:47:26 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Copy all buffer flags when creating a subtitle buffer copy after postprocessing This also copies the caps. Otherwise we could end up pusing the first buffer without any caps, which causes downstream to not get notified about the caps. Fixes bug #664892. 2011-10-11 02:07:13 +0200 Alexey Fisher * gst/matroska/matroska-mux.c: matroskamux: make default framerate optional per stream there is at least two use cases where default frame rate should or may be disabled: - vp8 stream with altref frame enabled. If default frame rate is enabled, some players will missinterprete it (critical!) - for webm container, to reduce micro overhead - for stream with variable frame rate. Signed-off-by: Alexey Fisher 2011-11-30 22:13:11 +0100 Stefan Sauer * gst/effectv/gstripple.c: rippletv: fix CLAMP end-values 2011-11-30 19:25:37 +0000 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update docs 2011-11-30 19:00:42 +0000 Tim-Philipp Müller * gst/multifile/Makefile.am: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/patternspec.c: * gst/multifile/patternspec.h: splitfilesrc: specify filenames via normal wildcards instead of regular expressions Less cracktastic in the end. 2011-10-10 18:28:11 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: check bytes actually read, just in case Handle corner case where we try to read beyond the end of the last file part, in which case we want to return a short read. If we get fewer bytes than expected for any other file part, we should just error out, since something fishy's going on then. 2011-10-06 08:33:19 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: set offsets on buffers Looks like some parsers (in some versions at least) expect the offsets to be set, and behave weird if that's not the case (e.g. off-by-one in h264parse). 2011-07-28 20:19:56 +0100 Tim-Philipp Müller * configure.ac: * gst/multifile/Makefile.am: * gst/multifile/gstmultifile.c: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitfilesrc.h: multifile: add splitfilesrc element Add new splitfilesrc element that presents multiple files (selectable via a location regex) as one single contiguous file. 2011-11-30 07:57:40 +0100 Wim Taymans * ext/pulse/pulsemixerctrl.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: update for moved audio interfaces 2011-11-29 17:34:10 -0300 Thiago Santos * ext/pulse/pulseaudiosink.c: Revert "pulseaudiosink: fix caps leak" This reverts commit d6a9de9e2aedc8b66ab3219902b5a37e8d65ada2. setcaps functions aren't supposed to take ownership of the caps passed 2011-11-29 19:10:58 +0100 Wim Taymans * gst/videofilter/Makefile.am: * gst/videofilter/gstvideobalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2videooverlay.h: * sys/v4l2/gstv4l2vidorient.h: * tests/icles/Makefile.am: * tests/icles/v4l2src-test.c: fix for moved interfaces 2011-11-28 23:20:32 +0000 Tim-Philipp Müller Merge commit '7521b597f4dc49d8d168f368f0e7ebaf98a72156' into 0.11 2011-11-28 21:31:25 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 2011-11-28 21:31:25 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 2011-11-28 21:27:53 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 2011-11-28 21:27:40 +0000 Tim-Philipp Müller Merge commit 'a2337b8af45cb5e8c091ff0e1c3ef4b6cc7b20a3' into 0.11 2011-11-28 18:25:52 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: Update for indexable change 2011-11-28 17:52:06 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtsp/gstrtpdec.c: update for clock provider API change 2011-11-28 16:57:24 +0100 Wim Taymans * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/rtsp/gstrtspsrc.c: fix for element flag updates 2011-11-28 12:58:44 +0000 Vincent Penquerc'h * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairorender.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstswitchsink.c: * ext/gconf/gstswitchsrc.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: * ext/hal/gsthalaudiosrc.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gsttaglibmux.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackparse.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audiopanorama.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/efence.c: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: * tests/check/elements/qtmux.c: various: fix pad template leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 2011-11-28 13:10:01 +0000 Vincent Penquerc'h * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: various: fix pad template ref leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 2011-11-28 13:10:01 +0000 Vincent Penquerc'h * ext/twolame/gsttwolame.c: various: fix pad template ref leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 2011-11-28 13:08:27 +0000 Vincent Penquerc'h * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: various: fix pad template ref leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 2011-11-28 11:47:11 +0100 Chad * gst/debugutils/gsttaginject.c: taginject: set gap-aware The element does not modify the data anyway. 2011-11-27 23:32:18 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update po files 2011-11-27 23:31:43 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 Conflicts: gst/equalizer/gstiirequalizer.c 2011-11-26 21:39:33 +0100 Stefan Sauer * gst/equalizer/gstiirequalizer.c: equalizer: also sync the parameters for the filter bands 2011-11-26 16:06:59 +0000 Tim-Philipp Müller * gst/matroska/matroska-ids.c: matroskademux: initialise seen_markup_tag field on subtitle stream context 2011-11-26 10:01:07 +0100 René Stadler * configure.ac: * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: * gst/matroska/webm-mux.c: * tests/check/elements/matroskamux.c: matroska: port to 0.11 Support for TAG_IMAGE and TAG_ATTACHMENT is commented out; this requires caps on buffers which is gone from 0.11. Segment handling in the demuxer is a bit complex; I added some FIXME comments in places where I'm not yet sure if I ported correctly. 2011-11-26 13:54:22 +0000 Tim-Philipp Müller * configure.ac: * ext/pulse/plugin.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulseaudio: require pulseaudio >= 1.0 2011-11-26 13:34:10 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 Conflicts: ext/pulse/pulseaudiosink.c ext/pulse/pulsesrc.c gst/audioparsers/gstaacparse.c gst/audioparsers/gstamrparse.c gst/audioparsers/gstdcaparse.c gst/audioparsers/gstflacparse.c gst/effectv/gstradioac.c gst/effectv/gstradioac.h gst/effectv/gstripple.c Some possible FIXMEs remaining in the audio parser getcaps functions. 2011-11-25 19:28:55 -0300 Thiago Santos * gst/isomp4/gstqtmuxmap.c: ismlmux: Use iso-fragmented as variant type Using 'iso' conflicts with mp4mux variant type, ismlmux now uses iso-fragmented Fixes #656823 2011-11-24 12:05:33 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: Implement GstStreamVolume interface PulseAudio 1.0 supports per-source-output volumes, and this exposes the functionality via the GstStreamVolume interface. When compiled against pre-1.0 PulseAudio, the interface is not implemented, and the "volume" or "mute" properties are not available. This bit of ugliness will go away when we can depend on PulseAudio 1.0 or greater. https://bugzilla.gnome.org/show_bug.cgi?id=595055 2011-09-10 21:21:38 -0700 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Trivial comment copy-paste-o fix 2011-11-14 12:43:27 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulseaudiosink: Remove redundant code 2011-11-14 12:41:41 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulseaudiosink: Clean up refcounting in event probe Makes sure we don't leak a refcount if the object is disposed before a NEWSEGMENT turns up. 2011-11-24 16:31:38 +0000 Vincent Penquerc'h * gst/flv/gstflvdemux.c: flvdemux: fix seeking Which I accidentally broke when fixing flv videos breaking on spurious timestamp discontinuities in broken files. https://bugzilla.gnome.org/show_bug.cgi?id=631430 2011-11-25 13:13:47 +0100 Stefan Sauer * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: effectv: repair color modes in radioactv by taking rgb,bgr into account 2011-11-25 11:44:49 +0100 Stefan Sauer * gst/effectv/gstradioac.c: radioactv: add one more set of caps It also work in this format. Avoids the need for conversion. 2011-11-25 11:44:18 +0100 Stefan Sauer * gst/effectv/gstradioac.c: * gst/effectv/gstshagadelic.c: effecttv: fix reverse negotiation The plugins were using _fixed_caps_ and thus not adjusting to new upstream sizes. Spotted by Tim Müller. 2011-11-25 11:43:16 +0100 Stefan Sauer * gst/effectv/gstwarp.c: warptv: remove not needed ifdef 2011-11-25 10:15:35 +0100 Stefan Sauer * gst/effectv/gstripple.c: rippletv: clean up the rendering code a bit This is corrrupts the memoy when resizing. Add a FIXME to make it resizeable once that is solved. 2011-11-24 21:41:03 +0100 René Stadler * tests/check/elements/alphacolor.c: * tests/check/elements/audioamplify.c: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioecho.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiopanorama.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/avimux.c: * tests/check/elements/avisubtitle.c: * tests/check/elements/capssetter.c: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: * tests/check/elements/equalizer.c: * tests/check/elements/icydemux.c: * tests/check/elements/jpegenc.c: * tests/check/elements/level.c: * tests/check/elements/parser.c: * tests/check/elements/qtmux.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: * tests/check/elements/rtpjitterbuffer.c: * tests/check/elements/spectrum.c: * tests/check/elements/videofilter.c: * tests/check/elements/y4menc.c: tests: update for gstcheck API change 2011-11-24 20:42:49 +0100 Stefan Sauer * gst/effectv/gstquark.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effecttv: fix reverse negotiation The plugins were using _fixed_caps_ and thus not adjusting to new upstream sizes. Spotted by Tim Müller. 2011-11-24 14:14:53 -0300 Thiago Santos * gst/multifile/gstmultifilesink.c: multifilesink: Fix leak of filename strings Do not forget to free the filename strings when deleting the list of files. 2011-11-24 14:11:33 -0300 Thiago Santos * tests/check/elements/multifile.c: multifile: fix build of tests Tests fail to build because g_mkdtemp is available from glib since 2.26. This patch adds a condition around the redefinition of g_mkdtemp on the tests to only build it if glib is older than 2.26. 2011-09-27 16:49:45 +0100 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: skip id32 tags This allows decoding at least one sample where something has stuffed some ID3 tag before the (supposedly initial) FMT\ . https://bugzilla.gnome.org/show_bug.cgi?id=660249 2011-10-31 17:06:18 +0000 Vincent Penquerc'h * gst/effectv/gstedge.c: edgetv: trivial comment fix for clarity https://bugzilla.gnome.org/show_bug.cgi?id=661841 2011-10-31 17:04:23 +0000 Vincent Penquerc'h * gst/effectv/gstedge.c: edgetv: don't leave bits of the output buffer uninitialized Let's initialize them to zero. It looks alright, but then it also looks alright with v3, or with the corresponding pixels from the source. I don't know what the original intent would be, and the original effectv source also has this bug/feature. https://bugzilla.gnome.org/show_bug.cgi?id=661841 2011-11-24 10:25:02 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparse: Use the sinkpad template caps as fallback, not the srcpad ones 2011-11-24 09:59:40 +0100 Sebastian Dröge * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:57:57 +0100 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:55:47 +0100 Sebastian Dröge * gst/audioparsers/gstdcaparse.c: dcaparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:53:18 +0100 Sebastian Dröge * gst/audioparsers/gstamrparse.c: amrparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:49:27 +0100 Sebastian Dröge * gst/audioparsers/gstamrparse.c: amrparse: Mark some more functions as static 2011-11-24 09:48:33 +0100 Sebastian Dröge * gst/audioparsers/gstac3parse.c: ac3parse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:44:58 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Mark some functions as static and remove unused function declarations 2011-11-24 09:43:14 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 01:48:25 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: update soup test for removed iradio-mode property 2011-11-24 01:45:43 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: get rid of iradio-* properties, post tags instead 2011-11-24 01:40:06 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: always send icecast request header, drop iradio-mode property Server should ignore unknown/unhandled headers.. 2011-11-24 01:19:32 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: make connection-speed property a guint64 2011-11-24 00:52:40 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-rtpmanager.xml: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpmanager.c: * tests/check/elements/rtpbin.c: * tests/examples/rtp/client-PCMA.c: * tests/examples/rtp/client-PCMA.py: * tests/examples/rtp/server-alsasrc-PCMA.c: * tests/examples/rtp/server-alsasrc-PCMA.py: rtpmanager: rename gstrtp* -> rtp* This was done in 0.10 to avoid conflict with the rtp elements in farsight, but the gst-prefixing is no longer needed in 0.11 2011-11-23 23:29:03 +0000 Tim-Philipp Müller * ext/twolame/gsttwolamemp2enc.c: ext: fix more printf format warnings in debug messages 2011-11-23 23:29:03 +0000 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: ext: fix more printf format warnings in debug messages 2011-11-23 10:23:28 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-11-23 09:26:17 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: pulseaudiosink: avoid endless caps loop Check if the caps are the same before adding a new probe. Because of reconfigure events, upstreams sends multiple caps events. 2011-11-23 00:57:39 +0000 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/matroskaparse.c: * tests/files/pinknoise-vorbis.mkv: tests: add basic unit test for matroskaparse 2011-11-23 00:56:26 +0000 Tim-Philipp Müller * gst/matroska/matroska-parse.c: matroskaparse: don't leak stream headers https://bugzilla.gnome.org/show_bug.cgi?id=664548 2011-11-22 01:40:39 +0000 Tim-Philipp Müller * ext/annodex/gstcmmldec.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/jpeg/gstjpegdec.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/gstximagesrc.c: More printf format warning fixes 2011-11-21 20:31:31 +0100 Matej Knopp * configure.ac: * gst/alpha/gstalpha.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavisubtitle.c: * gst/debugutils/breakmydata.c: * gst/debugutils/gstnavigationtest.c: * gst/flv/gstflvdemux.c: * gst/goom/gstgoom.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpsession.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/videofilter/gstvideoflip.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * sys/ximage/gstximagesrc.c: Fix printf format compiler warnings on OS X / 64bit https://bugzilla.gnome.org/show_bug.cgi?id=662615 2011-11-21 13:37:01 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/wavparse/gstwavparse.c: update for activation changes 2011-11-18 17:59:16 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/wavparse/gstwavparse.c: update for new scheduling query 2011-11-18 13:57:20 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: * gst/avi/gstavidemux.c: * gst/debugutils/rndbuffersize.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/wavparse/gstwavparse.c: add parent to activate functions 2011-11-17 17:36:05 +0100 Wim Taymans * gst/isomp4/qtdemux.c: qtdemux: activate pad before setting caps Seting caps on an inactive flushing pad does nothing. 2011-11-17 17:17:11 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/speex/gstspeexenc.c gst/rtpmanager/rtpsession.c 2011-11-17 15:02:55 +0100 Wim Taymans * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/flac/gstflactag.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/pulse/pulseaudiosink.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/gstnavigationtest.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstasteriskh263.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/videocrop/gstaspectratiocrop.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: add parent to pad functions 2011-11-17 08:24:58 +0100 Stefan Sauer * ext/cairo/gsttextoverlay.c: * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-mux.c: * gst/multipart/multipartmux.c: * gst/smpte/gstsmpte.c: * gst/videomixer/videomixer.c: collectpads: port API changes 2011-11-16 19:08:05 +0100 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: ensure to free allocated padded data 2011-11-16 18:57:38 +0100 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: reset tag setter interface when appropriate 2011-11-16 18:57:21 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: reset tag setter interface when appropriate 2011-11-16 17:54:49 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: add parent to internal links 2011-11-16 17:27:13 +0100 Wim Taymans * ext/annodex/gstcmmldec.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/pulse/pulseaudiosink.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/debugutils/gstpushfilesrc.c: * gst/flv/gstflvdemux.c: * gst/goom/gstgoom.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/videocrop/gstaspectratiocrop.c: * gst/wavparse/gstwavparse.c: add parent to query function 2011-11-16 12:40:08 +0100 Wim Taymans * gst/goom/gstgoom.c: goom: update for renamed flags Use the _check_reconfigure method instead of checking flags. Don't need to ref the parent anymore, core does that. 2011-11-15 18:01:16 +0100 Wim Taymans * ext/flac/gstflacenc.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/debugutils/progressreport.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/isomp4/qtdemux.c: * gst/wavparse/gstwavparse.c: _query_peer_*() -> _peer_query_*() 2011-11-15 17:45:31 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: _accept_caps() -> _query_accept_caps() 2011-11-15 17:29:45 +0100 Wim Taymans * ext/jpeg/gstjpegenc.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesrc.c: * gst/goom/gstgoom.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/shapewipe/gstshapewipe.c: * sys/v4l2/gstv4l2src.c: _peer_get_caps() -> _peer_query_caps() 2011-11-15 16:55:27 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/videocrop/gstaspectratiocrop.c: * sys/v4l2/gstv4l2src.c: * tests/icles/gdkpixbufsink-test.c: update for _get_caps() -> _query_caps() 2011-11-15 16:31:45 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/shapewipe/gstshapewipe.c: * gst/videocrop/gstaspectratiocrop.c: change getcaps to query Chain up event function in payloaders. 2011-11-15 13:23:56 +0000 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: fix spurious timestamp discontinuity We need to tell the base class that we're dropping buffers, so it drops the input timestamps corresponding to these. Otherwise, the first actual audio buffers we output will be stamped with those - GST_CLOCK_TIMESTAMP_NONE. That mismatch between input buffer count and output buffer count will stay while playing. With enough headers and long enough buffer durations, the sink will have played enough before receiving the first valid timestamp (usually 0), and will trigger an audible discontinuity. 2011-11-14 15:34:57 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: detect when a file lies about fixed block size If the sample/block number happens to be the same as the block size, we assume variable block size, and thus counters in samples in the headers. This can only get us a false positive for a block size of 1, which is invalid. We can get false negatives more often though (eg, if not starting at the start of the stream), but then that's already GIGO. 2011-09-02 19:20:07 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Add special mode to use FIR as repair as Google does https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-09-01 17:47:38 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: rtpsession: Send FIR requests in response to key unit requests with all-headers=TRUE https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-09-01 16:25:21 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: rtpsession: Put the PLI requests in each RTPSource Also refactor a bit and put all the keyframe request code in one place inside rtpsession.c https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-08-31 14:35:33 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Hack to FIR because Google doesn't set the sender ssrc correctly https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-08-30 19:06:13 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Process received Full Intra Requests Process FIR requests according to RFC 5104 https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-11-07 18:43:26 +0000 Sjoerd Simons * sys/v4l2/gstv4l2object.c: v4l2: Set pixel-aspect-ratio to 1/1 We don't currently support setting the pixel-aspect-ratio from V4L2. So simply set it to be 1/1 in the caps to prevent negotiation failures when fixating to weird values (e.g. when the downstream caps has pixel-aspect-ratio = [ MIN, MAX ] ) https://bugzilla.gnome.org/show_bug.cgi?id=663580 2011-11-14 09:39:15 +0000 Tim-Philipp Müller * tests/check/elements/id3demux.c: tests: make id3demux test compile Still fails though. 2011-11-12 15:42:27 +0200 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: no need to explicitely add controlled properties anymore 2011-11-13 23:42:44 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * gst/debugutils/gstpushfilesrc.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2src.c: Update for GstURIHandler get_protocols() changes 2011-11-13 18:50:51 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: * gst/debugutils/gstpushfilesrc.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2src.c: soup, pushfile, rtsp, udp, v4l2: update for GstURIHandler API changes 2011-11-11 19:24:27 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/pulse/pulseaudiosink.c 2011-11-11 19:21:50 +0100 Wim Taymans * gst/rtp/gstrtpg729pay.c: rtp: fix for rtp header changes 2011-11-11 10:06:25 -0300 Thiago Santos * ext/pulse/pulseaudiosink.c: pulseaudiosink: fix caps leak 2011-11-11 14:55:48 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: do not leak clientname when setting up property 2011-11-11 18:05:35 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulse: Chain up dispose() in pulseaudiosink 2011-11-11 12:32:41 +0100 Wim Taymans * gst/isomp4/gstrtpxqtdepay.h: * gst/rtp/fnv1hash.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpac3depay.h: * gst/rtp/gstrtpac3pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpbvdepay.h: * gst/rtp/gstrtpbvpay.h: * gst/rtp/gstrtpceltdepay.h: * gst/rtp/gstrtpceltpay.h: * gst/rtp/gstrtpdvdepay.h: * gst/rtp/gstrtpdvpay.h: * gst/rtp/gstrtpg722depay.h: * gst/rtp/gstrtpg722pay.h: * gst/rtp/gstrtpg723depay.h: * gst/rtp/gstrtpg723pay.h: * gst/rtp/gstrtpg726depay.h: * gst/rtp/gstrtpg726pay.h: * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtpgstdepay.h: * gst/rtp/gstrtpgstpay.h: * gst/rtp/gstrtph263depay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.h: * gst/rtp/gstrtpilbcdepay.h: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.h: * gst/rtp/gstrtpjpegdepay.h: * gst/rtp/gstrtpjpegpay.h: * gst/rtp/gstrtpmp1sdepay.h: * gst/rtp/gstrtpmp2tdepay.h: * gst/rtp/gstrtpmp2tpay.h: * gst/rtp/gstrtpmp4adepay.h: * gst/rtp/gstrtpmp4apay.h: * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpmparobustdepay.h: * gst/rtp/gstrtpmpvdepay.h: * gst/rtp/gstrtpmpvpay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpqcelpdepay.h: * gst/rtp/gstrtpqdmdepay.h: * gst/rtp/gstrtpsirendepay.h: * gst/rtp/gstrtpsirenpay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: * gst/rtp/gstrtpsv3vdepay.h: * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.h: update for base class rename 2011-11-11 12:25:01 +0100 Wim Taymans * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/gstrtpxqtdepay.h: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3depay.h: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpac3pay.h: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvdepay.h: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpbvpay.h: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltdepay.h: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpceltpay.h: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvdepay.h: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpdvpay.h: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722depay.h: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg722pay.h: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723depay.h: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726depay.h: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg726pay.h: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpg729pay.h: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263depay.h: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcdepay.h: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegdepay.h: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpjpegpay.h: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp1sdepay.h: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tdepay.h: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp2tpay.h: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4adepay.h: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4apay.h: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmparobustdepay.h: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvdepay.h: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpmpvpay.h: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqcelpdepay.h: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpqdmdepay.h: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirendepay.h: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsirenpay.h: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpspeexpay.h: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtpsv3vdepay.h: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: update for base class rename 2011-11-11 12:01:17 +0100 Wim Taymans * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/pulse/pulsesink.c: update for audiobase* rename 2011-11-11 11:53:45 +0100 Wim Taymans * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: audio: update for base class rename 2011-11-11 11:33:44 +0100 Wim Taymans * ext/pulse/pulseutil.h: * gst/equalizer/gstiirequalizer.h: fix for ringbuffer rename 2011-11-11 11:24:00 +0100 Wim Taymans * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackringbuffer.h: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: update for ringbuffer change 2011-11-11 01:27:47 +0000 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: lamemp3enc: cosmetic error message change LET'S TRY TO KEEP CAPITALS TO A MINIMUM. 2011-11-11 00:58:24 +0000 Tim-Philipp Müller * ext/twolame/Makefile.am: * ext/twolame/gsttwolamemp2enc.c: * ext/twolame/gsttwolamemp2enc.h: twolame: rename to twolamemp2enc 2011-11-11 00:51:34 +0000 Tim-Philipp Müller * ext/twolame/gsttwolame.c: twolame: port to 0.11 2011-11-10 23:15:30 +0200 Stefan Sauer * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: port api changes 2011-11-10 23:09:23 +0200 Stefan Sauer * ext/annodex/gstannodex.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audiopanorama.c: * gst/equalizer/gstiirequalizer.c: various: add missing includes 2011-11-10 21:35:24 +0100 René Stadler * ext/pulse/pulsesink.c: pulsesink: fix compilation with pulseaudio 0.9 2011-11-10 18:32:58 +0100 Wim Taymans * ext/flac/gstflactag.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/qtdemux.c: * gst/multipart/multipartdemux.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/wavparse/gstwavparse.c: update for adapter api changes 2011-11-10 17:23:47 +0100 Wim Taymans * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: update for changed base classes 2011-11-10 13:50:34 +0100 Wim Taymans * ext/pulse/pulsesink.c: fix for audio clock change 2011-11-10 11:03:18 +0100 Wim Taymans * ext/aalib/gstaasink.c: * ext/jpeg/gstjpegdec.c: * ext/pulse/pulsesrc.c: * sys/v4l2/gstv4l2src.c: * sys/ximage/gstximagesrc.c: update for removed fixate function 2011-11-09 17:40:10 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-11-09 17:38:03 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: updates for new acceptcaps query 2011-11-08 15:35:26 +0000 Vincent Penquerc'h * gst/avi/gstavidemux.c: avidemux: fix wrong stride when inverting uncompressed video Such frames have a stride multiple of 4, see http://lscube.org/pipermail/ffmpeg-issues/2010-April/010247.html. This showed up on a sample using a odd width of 24 bit video. https://bugzilla.gnome.org/show_bug.cgi?id=652288 2011-11-09 12:25:01 +0100 Wim Taymans * gst/rtp/gstrtph263ppay.c: h263ppay: report to 0.11 2011-11-09 12:18:01 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacdec.c gst/audioparsers/gstflacparse.c gst/isomp4/qtdemux.c 2011-11-09 11:56:07 +0100 Thijs Vermeir * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: fix compiler warning for uninitialized values 2011-11-09 11:53:01 +0100 Wim Taymans * ext/annodex/gstcmmldec.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/wavparse/gstwavparse.c: remove query types 2011-11-09 10:32:06 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: minimal sanity check on creation datetime 2011-11-04 17:54:04 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: dtmfsrc: Reject start/stop requests that come out of order 2011-10-29 18:24:26 +0200 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: Post messages when starting to send/receive DTMF This way, the UI can display the DTMF events as they as being sent. 2011-11-02 12:58:12 -0400 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Return the sink pad template as sink caps, not the src's https://bugzilla.gnome.org/show_bug.cgi?id=577784 2009-03-15 19:26:48 -0400 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Also implement size/framerate restrictions in getcaps https://bugzilla.gnome.org/show_bug.cgi?id=577784 2009-03-04 20:50:19 -0500 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes https://bugzilla.gnome.org/show_bug.cgi?id=577784 2011-11-08 14:31:34 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: also set segment stop at startup rather than only post seek ... so as to ensure consistent playback with or without seek, especially in presence of some bogus edit list entries. 2011-11-08 11:18:06 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: * gst/rtsp/gstrtspsrc.c: update for probe api changes 2011-11-08 08:50:19 +0100 Stefan Sauer * gst/goom/gstgoom.c: goom: code cleanups Move variables to the scope where they are needed. Use our macros and functions more. 2011-11-08 08:49:05 +0100 Stefan Sauer * gst/goom/gstgoom.c: goom: add a sink_query to eat allocation queries We should not forward allocation queries for audio to the video sink. 2011-11-02 17:02:54 +0000 Raul Gutierrez Segales * gst/flv/Makefile.am: gst/flv/: add amfdefs.h to noinst_HEADERS https://bugzilla.gnome.org/show_bug.cgi?id=663334 2011-11-07 17:14:17 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: * gst/rtsp/gstrtspsrc.c: fix for probe updates 2011-10-03 17:50:43 +0100 Vincent Penquerc'h * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: detect large pts gaps and resync Should work on multiple gaps, but tested on only one. https://bugzilla.gnome.org/show_bug.cgi?id=631430 2011-08-22 10:40:45 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: fix off by one between granpos and last_stop 2011-10-07 19:41:35 +0100 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: flacparse: fix last frame timestamp in fixed block size mode The last block may have a different block size, so we should not use it to scale or we'll end up with a wrong timestamp. See comment and quote from the FLAC format documentation in the code. Fixes looped playback of FLAC files (via about-to-finish). https://bugzilla.gnome.org/show_bug.cgi?id=661215 2011-10-27 15:52:47 +0100 Vincent Penquerc'h * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: cairotextoverlay: add a 'silent' property to skip rendering https://bugzilla.gnome.org/show_bug.cgi?id=662856 2011-11-07 12:00:12 +0100 René Stadler * gst/matroska/ebml-write.c: matroskamux: fix regression causing malformed files This was caused by me in 1b213d. It seems I was too focused on 0.11 when I did this and tested the wrong branch. The problem was reported by Alexey Fisher. 2011-11-04 18:41:36 +0100 Stefan Sauer * ext/annodex/gstcmmldec.h: * gst/alpha/Makefile.am: * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: * gst/audiofx/Makefile.am: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofx.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/effectv/Makefile.am: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstvertigo.c: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer.h: * gst/shapewipe/Makefile.am: * gst/shapewipe/gstshapewipe.c: * gst/smpte/Makefile.am: * gst/smpte/gstsmptealpha.c: * gst/videobox/Makefile.am: * gst/videobox/gstvideobox.c: * gst/videofilter/Makefile.am: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/plugin.c: * gst/videomixer/Makefile.am: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * tests/examples/shapewipe/shapewipe-example.c: * tests/examples/v4l2/camctrl.c: controller: port to new controller location and api 2011-11-04 18:52:35 +0100 Stefan Sauer * gst/audiofx/gststereo.c: controller: port to new controller location and api 2011-11-04 17:39:15 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: more template fixes 2011-11-04 16:21:13 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: pulseaudiosink: more 0.11 fixing Make sure the caps event gets to the sink. 2011-11-04 15:35:42 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: pulseaudiosink: port some more Rename decodebin2 -> decodebin some more Cleanup up sinkpad event handling 2011-11-04 13:56:06 +0100 Wim Taymans * ext/pulse/pulseaudiosink.c: pulseaudiosink: port some more to 0.11 We must not forward the caps event. instead we will decide what to do when the pad block is taken. Use decodebin instead of decodebin2 2011-11-04 13:12:37 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/interleave/deinterleave.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: * gst/rtpmanager/gstrtpssrcdemux.c: more template fixes 2011-11-04 11:58:22 +0100 Wim Taymans * gst/avi/gstavimux.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmux.c: * gst/matroska/matroska-mux.c: * gst/matroska/webm-mux.c: * gst/multipart/multipartmux.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/videomixer/videomixer.c: * tests/check/elements/avimux.c: * tests/check/elements/interleave.c: * tests/check/elements/matroskamux.c: * tests/check/elements/qtmux.c: * tests/check/elements/rtpbin.c: make %u in all request pad templates 2011-11-04 11:01:01 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/rtp/gstrtpvrawdepay.c 2011-11-04 10:32:46 +0100 Edward Hervey * configure.ac: * gst/apetag/gstapedemux.c: Port apedemux 2011-11-03 23:28:31 +0000 Tim-Philipp Müller * gst/rtp/gstrtpvrawdepay.c: rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN Fixes compiler warning on mingw32 2011-11-03 16:43:00 +0100 Wim Taymans * common: * configure.ac: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: * gst/udp/Makefile.am: * gst/udp/gstdynudpsink.c: * gst/udp/gstudp.c: * gst/udp/gstudpsrc.c: update for new net library 2011-11-02 12:09:20 +0100 Wim Taymans * ext/annodex/gstcmmldec.c: * ext/flac/gstflactag.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/avi/gstavidemux.c: * gst/debugutils/gsttaginject.c: * gst/flv/gstflvdemux.c: * gst/replaygain/gstrganalysis.c: * gst/wavparse/gstwavparse.c: tags: update for tag API removal 2011-11-02 10:40:12 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-10-31 02:40:08 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstudpsrc.c: update for netbuffer api change 2011-10-31 02:35:51 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstudp.c: * gst/udp/gstudpsrc.c: update for netaddress change 2011-10-31 02:24:04 +0100 Wim Taymans * gst/effectv/gstwarp.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: update for meta api change 2011-10-29 09:29:27 +0200 Wim Taymans * gst/isomp4/gstqtmoovrecover.c: * gst/rtsp/gstrtspsrc.c: update for new task api 2011-10-29 09:09:45 +0200 Wim Taymans * ext/pulse/pulsesink.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtsp/gstrtspsrc.c: * sys/v4l2/gstv4l2object.c: structure: fix for api update 2011-10-29 08:25:27 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: bufferlist: update for new API 2011-11-01 00:40:40 +0000 Tim-Philipp Müller * ext/pulse/pulseaudiosink.c: * gst/rtsp/gstrtspsrc.c: Update for pad API changes GstProbeType, GstProbeReturn and GstActivateMode -> GstPad* 2011-10-31 18:38:55 +0100 René Stadler * gst/audioparsers/gstac3parse.c: ac3parse: fix obvious crash 2011-10-31 16:18:32 +0100 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: avoid shortcut evaluation when adding paired mp4 tag Fixes (part of) #638711. 2011-10-31 15:43:25 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: do not use unoffical V_MJPEG codec id ... but as not spec'ed especially, consider it a VfW compatibility case. Fixes #659837. 2011-10-30 19:30:14 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.h: flacenc: remove dead code from header We require a new-enough libflac that this condition will never apply. 2011-10-30 19:09:03 +0000 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: parse stream headers from caps in set_format function Not that this seems to be actually needed, libflac happily decodes stuff even if we just drop all headers and never feed it to the library. 2011-10-30 18:49:21 +0000 Tim-Philipp Müller * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: don't extract metadata, leave that to the parser or container 2011-10-30 18:45:45 +0000 Tim-Philipp Müller * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: we expect framed input now, remove some more code 2011-10-09 16:18:09 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: naive port to GstAudioDecoder This would probably have been too invasive to do in the 0.10 branch, with all the pull-mode and parser handling code in there. 2011-10-30 12:29:14 +0000 Tim-Philipp Müller * ext/lame/Makefile.am: * ext/lame/README: * ext/lame/gstlame.c: * ext/lame/gstlame.h: * ext/lame/plugin.c: * ext/lame/test-lame.c: * tests/check/pipelines/lame.c: lame: remove lame element, it's been superseded by lamemp3enc 2011-10-30 11:51:58 +0000 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: ext, gst: update for taglist API changes 2011-10-30 11:44:53 +0000 Tim-Philipp Müller * ext/annodex/gstcmmldec.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavisubtitle.c: * gst/debugutils/gsttaginject.c: * gst/flv/gstflvdemux.c: * gst/icydemux/gsticydemux.c: * gst/isomp4/qtdemux.c: * gst/multipart/multipartdemux.c: * gst/replaygain/gstrganalysis.c: * gst/wavparse/gstwavparse.c: ext, gst: update for taglist API changes 2011-10-30 11:41:32 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: fix compilation of audio tests in uninstalled setup 2011-10-28 21:26:33 +0200 René Stadler * gst/audiofx/audiopanorama.c: audiopanorama: simplify get_unit_size 2011-10-28 21:19:42 +0200 René Stadler * tests/check/elements/audioecho.c: tests: audioecho: port to 0.11 2011-10-28 21:18:33 +0200 René Stadler * gst/audiofx/audioecho.c: audioecho: fix internal buffer size calculation 2011-10-28 14:05:48 +0200 René Stadler * tests/check/elements/audiochebband.c: tests: audiochebband: port to 0.11 2011-10-28 16:52:08 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-10-28 15:08:25 +0200 Wim Taymans * ext/pulse/pulseaudiosink.c: pulseaudiosink: fix porting errors The probes were ported wrongly and caused deadlocks. 2011-10-28 09:57:36 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg jpegdec (using libjpeg 6.2/8) can't decode some lossless types of JPEG. https://bugzilla.gnome.org/show_bug.cgi?id=556648 2011-10-28 13:06:20 +0200 René Stadler * tests/check/elements/audiocheblimit.c: tests: audiocheblimit: port to 0.11 2011-10-28 13:02:56 +0200 René Stadler * gst/audiofx/audiofxbaseiirfilter.c: audiofx: fix crash in process() 2011-10-28 11:48:31 +0200 René Stadler * tests/check/elements/audioamplify.c: tests: audioamplify: port to 0.11 2011-10-28 12:51:31 +0200 Wim Taymans * ext/pulse/pulseaudiosink.c: pulse: fix check for empty caps 2011-10-28 12:30:33 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: elaborate some debug statements 2011-10-11 20:56:51 +0400 Stas Sergeev * gst/flv/gstflvdemux.c: flvdemux: be careful with negative cts Fixes #661477. 2011-10-06 13:04:54 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: tune non-update seek handling cases Fixes #661049. 2011-10-28 11:46:40 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/videomixer/gstcollectpads2.c 2011-10-28 11:16:38 +0200 René Stadler * gst/audiofx/audiodynamic.c: audiodynamic: don't set process function too early GstAudioInfo and GstAudioFilter have been changed so that this code doesn't crash anymore when a property is set in NULL state. 2011-10-28 10:42:04 +0200 René Stadler * tests/check/elements/audiodynamic.c: tests: audiodynamic: port to 0.11 2011-10-28 00:24:14 +0200 René Stadler * tests/check/elements/spectrum.c: tests: spectrum: port to 0.11 2011-10-27 23:57:17 +0200 René Stadler * tests/check/elements/audiopanorama.c: tests: audiopanorama: port to 0.11 2011-10-27 23:56:12 +0200 René Stadler * gst/audiofx/audiopanorama.c: audiopanorama: fix get_unit_size 2011-10-28 10:40:36 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer2: Use the clip function instead of the prepare_buffer function 2011-10-28 09:05:27 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpsession.c: * sys/v4l2/gstv4l2object.c: rtpmanager, v4l2: fix compiler warnings after gst_caps_new_simple() change 2011-10-28 09:01:57 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux.c: qtdemux: fix compiler warnings after gst_caps_new_simple() change 2011-10-28 09:36:17 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/gstcollectpads2.c: * gst/videomixer/gstcollectpads2.h: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixer2pad.h: videomixer2: Use collectpads2 from core 2011-10-27 19:39:20 +0200 René Stadler * gst/wavenc/Makefile.am: * gst/wavenc/gstwavenc.c: wavenc: port to 0.11 raw audio caps 2011-10-27 19:06:06 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/flv/gstflvmux.c 2011-10-27 19:00:52 +0200 Wim Taymans * gst/audioparsers/gstaacparse.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/icydemux/gsticydemux.c: * gst/rtp/README: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: make some more things compile again 2011-10-27 16:08:22 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/pulse/pulseaudiosink.c ext/pulse/pulsesink.c 2011-10-27 16:03:17 +0200 Wim Taymans * ext/pulse/pulsesink.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * sys/v4l2/gstv4l2object.c: fix compilation 2011-10-28 00:41:45 +1100 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: deinterlace: Don't pointlessly hold object lock over caps operations Avoids a deadlock when getcaps is recursive due to the getcaps being reflected upstream/downstream. The lock isn't actually protecting anything here. 2011-10-27 00:37:03 +1100 Jan Schmidt * gst/flv/amfdefs.h: * gst/flv/gstflvmux.c: flvmux: add some comments and defines to clarify code. 2011-10-10 15:36:14 +0200 René Stadler * gst/matroska/ebml-write.c: matroska: refactor ebml-write to be more 0.11 friendly Switching to a more 0.11-friendly pattern, where getting the buffer's data pointer and setting the size many times is less natural. This is of course in preparation to the upcoming port of the plugin. 2011-10-11 21:45:46 +0200 René Stadler * gst/matroska/ebml-write.c: matroska: remove stale floatcast include GDOUBLE_TO_BE was moved to core a long time ago. 2011-10-11 22:10:27 +0200 René Stadler * gst/matroska/matroska-mux.c: matroskamux: fix possible crash with malformed dirac codec_data Since size is unsigned, we need to safeguard against wrapping below zero. 2011-10-21 22:33:34 +0200 René Stadler * gst/equalizer/gstiirequalizer.c: equalizer: remove avoidable call to gst_object_set_name 2011-10-21 22:32:38 +0200 René Stadler * gst/deinterlace/gstdeinterlace.c: deinterlace: remove avoidable call to gst_object_set_name 2011-10-21 14:51:23 +0200 Stefan Sauer * ext/pulse/pulsemixerctrl.h: * gst/videofilter/gstvideobalance.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstossmixer.h: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudioelement.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2videooverlay.c: * sys/v4l2/gstv4l2videooverlay.h: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/gstv4l2vidorient.h: interfaces: clean up the use of iface and class/klass 2011-10-21 11:37:05 +0100 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Update spec file so its paralel-installable and only tries to package ported plugins 2011-10-16 20:30:25 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: increase arbitrary resolution limits Apparently libpng can technically do up to 2^31-1 rows and columns. However it imposes an (arbitrary) default limit of 1 million (that could theoretically be lifted by using some additional API). Moved array allocation to the heap now. 2011-10-16 20:25:41 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: don't unconditionally allocate 4096 pointers on the stack Instead allocate as many as needed (on the stack still). 2011-10-16 20:05:28 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: ensure setcaps was called before chain function This is needed to properly error out for e.g. "fakesrc ! pngenc ! fakesink". 2011-10-16 19:44:27 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: validate input buffer size Just for safety; of course such mismatch represents a bug in another element. 2011-10-16 19:41:28 +0200 René Stadler * ext/libpng/Makefile.am: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: pngenc: make setcaps more robust, use gstvideo functions A setcaps function needs to actually verify the caps carefully. In this case, it was possible to e.g. link a video decoder with YUV+RGB template caps to pngenc. That would cause a crash when the decoder pushes a YUV buffer. Same thing when pushing a valid buffer that exceeds the resolution limits. Also, missing framerate caps field would cause a glib critical warning due to invalid GValue. This fails hard now. 2011-10-21 10:01:43 +0200 René Stadler * gst/matroska/matroska-read-common.c: ebml: small correction to previous commit Signal a short read with UNEXPECTED, exactly like the peek_bytes function. 2011-10-19 13:09:51 +0200 Edward Hervey * gst/matroska/matroska-read-common.c: ebml: Fix push-based behaviour The 'peek' method was completely wrong (!?) 2011-10-18 18:31:17 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulse: Get caps correctly on pad block Instead of always going upstream, we should first see if already got caps from a setcaps() call. https://bugzilla.gnome.org/show_bug.cgi?id=661262 2011-10-18 12:25:14 +0100 Tim-Philipp Müller * ext/wavpack/gstwavpackenc.c: wavpackenc: don't unref buffer with gst_object_unref() 2011-10-18 12:05:01 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: only use is_pcm for 1.0 of pulseaudio 2011-10-18 11:58:57 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: only disable trickmodes for !pcm Only disable trickmodes when we are not dealing with raw PCM samples. 2011-10-16 15:32:50 +0200 Wim Taymans * gst/videocrop/gstvideocrop.c: videocrop: fix compilation 2011-10-16 15:26:38 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/rtp/gstrtpvrawdepay.c 2011-10-14 10:56:16 +0530 Arun Raghavan * gst/videomixer/videomixer2.c: videomixer2: Fix a leak Buffers weren't being unref'ed in one case inside, causing memory usage to blow up. 2011-10-14 09:10:01 +0200 Marc Leeman * gst/rtp/gstrtpvrawdepay.c: set colour masks for video/x-raw-rgb in rtpvrawdepay 2011-10-13 01:05:13 -0300 Thiago Santos * configure.ac: configure: re-enable videocrop plugin Already ported to 0.11 2011-10-13 01:05:04 -0300 Thiago Santos * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstaspectratiocrop.h: aspectratiocrop: Port to 0.11 2011-10-13 00:39:28 -0300 Thiago Santos * gst/videocrop/Makefile.am: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: videocrop: Port to 0.11 2011-10-12 17:43:47 -0300 Thiago Santos * tests/check/elements/aspectratiocrop.c: tests: aspectratiocrop: Port to 0.11 2011-10-12 08:24:28 -0300 Thiago Santos * tests/check/elements/alphacolor.c: tests: alphacolor: Port to 0.11 2011-10-13 17:12:23 +0200 Edward Hervey * ext/flac/gstflacenc.c: flacenc: Properly register type It's a subclass of GstAudioEncoder and not of GstElement 2011-10-13 16:59:50 +0530 Arun Raghavan * gst/videomixer/videomixer2.c: videomixer2: Fix incorrect gst_buffer_replace() call This got exposed when gst_buffer_replace() was changed from a macro to a function. 2011-10-13 09:34:04 +0200 Edward Hervey * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Fix wrong usage of gst_iterator_filter It takes a GValue* as the user_data. And don't forget to unref the demuxer before returning. 2011-10-13 09:02:47 +0200 Wim Taymans * ext/jpeg/gstjpegdec.c: fix compile 2011-10-13 08:58:06 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/jpeg/gstjpegdec.c gst/rtp/gstrtpvrawpay.c 2011-10-12 08:09:20 -0300 Thiago Santos * tests/check/elements/cmmlenc.c: tests: cmmlenc: Port to 0.11 2011-10-12 08:02:08 -0300 Thiago Santos * tests/check/elements/cmmldec.c: tests: cmmldec: Port to 0.11 2011-10-12 07:29:30 -0300 Thiago Santos * ext/pulse/pulseaudiosink.c: pulseaudiosink: Use new GstIterator API correctly GstIterator now uses GValue, use it correctly. 2011-10-12 11:26:50 +0200 Edward Hervey * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: Only use 24 LSB for depth=24 RGB caps ... and indent the masks for clarity 2011-10-11 14:58:43 +0200 René Stadler * gst/matroska/matroska-mux.c: matroskamux: fix segment handling, so we actually use running time gst_matroska_mux_best_pad adjusts the buffer timestamp to running time using the segment stored in the pad's collect data. However, the event handler didn't pass the newsegment event on to collectpads' handler, so this segment was never updated at all. Re-fixes bug #432612. 2011-10-10 19:01:23 +0100 Sjoerd Simons * gst/rtp/gstrtpg722pay.c: gstrtpg722pay: Compensate for clockrate vs. samplerate difference The RTP clock-rate used for G722 is 8000, even though the samplerate is 16000. Compensate for this by pretending G722 has 8 bits per sample instead of the 4 bits as if it were a codec that ran at half the speed, but with twice the number of bits. Fixes #661376 2011-09-27 19:25:53 +0100 Sjoerd Simons * ext/jpeg/gstjpegdec.c: jpegdec: Implement upstream negotiation Add upstream negotiation for jpegdec. Fixes #660275 2011-10-10 19:02:58 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: don't leak audio codec_data buffer 2011-10-10 17:41:10 +0200 Edward Hervey alpha: Don't use start() vmethod The only thing we're doing is initializing parameters ... * which won't work because we don't have upstream/downstream caps * which will be initialized when ::set_caps() is called 2011-10-10 14:08:29 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-10-10 13:22:12 +0200 Wim Taymans * configure.ac: * gst/id3demux/gstid3demux.c: id3demux: port to 0.11 2011-10-10 13:20:04 +0200 Stefan Sauer * tests/examples/cairo/Makefile.am: tests: add missing PLUGIN_ASE_LIBS to LDADD 2011-10-10 12:54:22 +0200 Wim Taymans * configure.ac: * gst/icydemux/gsticydemux.c: icydemux: port to 0.11 2011-10-10 12:27:06 +0200 Wim Taymans * configure.ac: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: annodex: port to 0.11 2011-10-10 11:48:20 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/speex/gstspeexenc.c 2011-10-10 00:18:56 -0300 Thiago Santos * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: port pulseutil to 0.11 2011-10-09 21:17:24 -0300 Thiago Santos * ext/pulse/pulseaudiosink.c: pulseaudiosink: port to 0.11 2011-10-09 18:58:29 -0300 Thiago Santos * ext/pulse/pulsesink.c: pulsesink: Fixing getcaps function Update getcaps function to 0.11 API 2011-10-09 21:31:27 +0200 Mark Nauwelaerts * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: speexenc: only push header buffers following initial events 2011-10-09 16:29:05 +0100 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 2011-10-09 16:24:36 +0100 Tim-Philipp Müller * gst/isomp4/qtdemux_dump.c: qtdemux: update for __gst_debug_min name change 2011-10-09 11:18:18 -0300 Thiago Santos * gst/isomp4/atomsrecovery.c: qtmux: Fix memory leak on atoms recovery function Remember to free the ftyp data after writing it to a file. Fixes #660969 2011-10-06 12:26:33 +0200 Wim Taymans * gst/isomp4/gstqtmux.c: qtmux: report new bits 2011-10-06 12:23:39 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/speex/gstspeexdec.c ext/speex/gstspeexenc.c gst/isomp4/atoms.c gst/isomp4/gstqtmux.c 2011-09-21 18:45:42 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: improve segment handling with non-zero starting timestamp ... as well as related items, such as seeking and position reporting. https://bugzilla.gnome.org/show_bug.cgi?id=659808 2011-09-29 18:41:53 +0400 Stas Sergeev * sys/v4l2/gstv4l2object.c: * sys/ximage/gstximagesrc.c: v4l2, ximagesrc: fix some printf format compiler warnings https://bugzilla.gnome.org/show_bug.cgi?id=660150 2011-09-30 12:42:22 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: Refactor bitrate check test Refactor bitrate check test to accomodate multiple tests for bitrate 2011-09-30 13:02:31 -0300 Thiago Santos * gst/isomp4/atoms.c: qtmux: update esds atom under wave atom for aac bitrates AAC in mov format puts an ESDS atom inside of a WAVE atom in STSD atom, we need to update the bitrate on this ESDS. This patch fixes it. 2011-09-30 12:41:52 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/fourcc.h: qtmux: Also update btrt atom When rewriting bitrates, also update the btrt atom under stsd 2011-09-30 10:55:53 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: add tests for bitrate average calculation Adds tests to make sure qtmux/mp4mux sets average bitrate correctly 2011-09-28 11:41:49 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Calculate average bitrate for streams Calculate and use average bitrate for streams when no bitrate tag was received 2011-09-28 10:41:14 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: Avoid a buffer metadata copy if possible If first_ts is 0 there is no need to subtract, so we might skip some copying to make the buffer metadata writable. 2011-09-29 23:21:46 +0100 Tim-Philipp Müller * ext/speex/gstspeexenc.c: speexenc: initialise variable before adding to it 2011-09-29 17:21:22 +0200 Mark Nauwelaerts * ext/speex/gstspeexdec.c: * ext/speex/gstspeexdec.h: speexdec: port to audiodecoder 2011-09-29 16:33:01 +0200 Mark Nauwelaerts * ext/speex/gstspeexenc.h: speexenc: clean up some unused remnants 2011-09-29 17:32:23 +0200 Mark Nauwelaerts * ext/speex/Makefile.am: * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: speexenc: port to audioencoder 2011-09-28 19:10:27 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: get rid of granulepos handling Leave that to the parser or demuxer. There's still some code for operating in DEFAULT (samples) format, but that will be removed later. 2011-09-28 18:32:00 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: get rid of pull-mode support and focus on being a decoder Leave all the other stuff to flacparse. 2011-09-28 17:29:08 +0100 Tim-Philipp Müller * ext/flac/gstflactag.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: flac, jpeg: fix compiler warning 2011-09-28 17:40:01 +0200 Wim Taymans * configure.ac: * ext/flac/gstflacdec.c: * ext/flac/gstflactag.c: flac: port to 0.11 2011-09-28 17:39:12 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacenc.c 2011-09-28 16:18:54 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-09-28 16:09:58 +0200 Mark Nauwelaerts * ext/flac/Makefile.am: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: port to audioencoder 2011-09-27 15:59:24 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-parse.c: matroskademux: ensure minimal alignment for audio/x-raw-* buffers Since matroskademux will attempt to push unaligned buffers, downstream might have trouble with those, especially if downstream uses ORC, such as audioconvert. Ensure we push buffers aligned to the basic type at least for those raw buffers. https://bugzilla.gnome.org/show_bug.cgi?id=659798 2011-09-28 12:44:59 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: common ext/pulse/pulsesink.c ext/soup/gstsouphttpclientsink.c gst/audioparsers/gstaacparse.c gst/audioparsers/gstac3parse.c gst/rtp/gstrtph264depay.c gst/rtpmanager/gstrtpjitterbuffer.c gst/rtpmanager/rtpjitterbuffer.c gst/rtsp/gstrtspsrc.c sys/ximage/gstximagesrc.c 2011-09-28 00:10:09 +0300 Raimo Järvi * gst/goom2k1/goom_core.c: goom2k1: Fix compiler warnings on 64 bit mingw-w64 Fixes bug #660294. 2011-09-27 18:19:50 +0200 Wim Taymans * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: fix raw audio caps too 2011-09-27 18:15:00 +0200 Wim Taymans * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: port to 0.11 2011-09-26 16:29:12 +0200 Sebastian Dröge * ext/twolame/gsttwolame.c: twolame: Simple fix for GstAudioEncoder API change 2011-09-26 16:28:08 +0200 Sebastian Dröge * ext/twolame/gsttwolame.c: twolame: Fix variable 'gstelement_class' set but not used compiler warning 2011-09-26 16:08:20 +0200 Sebastian Dröge * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: Don't get the parent class again, GST_BOILERPLATE does this already 2011-09-26 16:07:54 +0200 Sebastian Dröge * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: Fix variable 'gstelement_class' set but not used compiler warning 2011-09-26 12:07:15 +0200 Mark Nauwelaerts * ext/twolame/gsttwolame.c: twolame: improve output framing and timestamping ... which simply comes down to requesting one frame of input data at a time, since the encoder nicely turns this into 1 encoded frame. 2011-09-26 11:56:23 +0200 Mark Nauwelaerts * ext/twolame/Makefile.am: * ext/twolame/gsttwolame.c: * ext/twolame/gsttwolame.h: twolame: port to audioencoder 2011-09-23 15:32:01 +0200 Mark Nauwelaerts * ext/lame/gstlame.c: lame: use some more boilerplate 2011-09-23 15:26:48 +0200 Mark Nauwelaerts * ext/lame/gstlame.c: * ext/lame/gstlame.h: lame: port to audioencoder 2011-09-23 14:33:55 +0200 Mark Nauwelaerts * ext/lame/gstlamemp3enc.c: lamemp3enc: use some more boilerplate 2011-09-26 14:44:23 +0200 Mark Nauwelaerts * ext/lame/gstlamemp3enc.c: lamemp3enc: really report bitrate rather kbitrate 2011-09-26 14:44:01 +0200 Mark Nauwelaerts * ext/lame/Makefile.am: * ext/lame/gstlamemp3enc.c: * ext/lame/gstlamemp3enc.h: lamemp3enc: port to audioencoder 2011-09-25 15:13:39 +0100 Tim-Philipp Müller * ext/soup/Makefile.am: * ext/soup/gstsoup.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpclientsink.h: soup: rename souphttpsink to souphttpclientsink To avoid confusion, and because we might want a server sink at some point too. https://bugzilla.gnome.org/show_bug.cgi?id=659947 2011-09-23 16:39:46 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsink.c: * ext/soup/gstsouphttpsink.h: souphttpsink: don't create unused second sink pad object The base class will create the sink pad. 2011-09-23 15:36:36 +0200 Julien Isorce * gst/audioparsers/gstac3parse.c: ac3parse: correctly check for ac3/e-ac3 switch https://bugzilla.gnome.org/show_bug.cgi?id=659943 2011-09-21 14:01:20 +0200 Edward Hervey * common: Update common to 0.11 branch 2011-09-20 13:38:53 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: improve downstream flow return feedback to upstream ... although basertpdepay does not really make it easy/possible to do so all the way. 2011-09-20 12:11:47 +0100 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: ximagesrc: add xid and xname properties to allow capturing a particular window A particular window may be selected using the new xid (X-Window XID, eg a pointer) and xname (window title) properties. If both are specified, the XID is used in preference, falling back to xname if not found. Default (if none of xid and xname are specified, or if no such window is found) is to capture the root window. https://bugzilla.gnome.org/show_bug.cgi?id=546932 2011-08-02 17:39:44 +0100 Tim-Philipp Müller * tests/check/elements/qtmux.c: tests: add unit test to make sure encodebin picks mp4mux for variant=iso https://bugzilla.gnome.org/show_bug.cgi?id=651496 2011-09-19 12:15:11 +0200 Ha Nguyen * gst/rtpmanager/gstrtpbin.c: rtpbin: Fix a leaked clock for each buffering message Fixes bug #659237. 2011-09-19 12:11:32 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: parse embedded ID32 tags 2011-09-02 13:41:41 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpsession: avoid source premature timing out Use slightly adjusted sender interval to determine sender timeout rather than our own sender side interval (which may have been forced small). 2011-08-25 12:40:52 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: avoid timing out source too quickly ... following a PAUSE/PLAY cycle, particularly applicable when operating with a short RTCP interval (possibly forced so server-side). 2011-08-24 14:37:52 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer/rtpbin: relax dropping rtcp packets ... to at least having it trigger a/v synchronization, possibly without using provided values which are still not considered sane (as previously dropped). 2011-08-24 14:34:23 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: some more reset when clearing pt map ... which in particular caters for some more reset following a possible rtsp PLAY. 2011-08-21 21:58:38 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: do not set elements to PLAYING when doing seek in PAUSED 2011-09-01 14:47:48 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: only reset skew on gap if input ts available 2011-08-18 14:12:21 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: check some more for possible rtp timestamp discontinuity ... when operating in non slave mode, and reset if detected. This should avoid some (large) bogus outgoing timestamp due to jumps in rtp time, as result of PAUSE/PLAY or seek or ... 2011-08-08 12:48:50 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: switch to rtp time based syncing when guessed appropriate 2011-08-08 12:15:20 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: alternative inter-stream syncing methods ... at least if not syncing to NPT time: * either sync using RTCP SR data (as currently) * only perform the above once using initial RTCP SR packets * discard RTCP and sync by equating provided stream's clock-base rtptime, as provided by jitterbuffer (typically obtained from RTP-Info in RTSP). 2011-08-08 12:11:24 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: also provide clock-base to sync signal 2011-08-08 12:09:41 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: allow configurable rtcp stream syncing interval ... rather than necessarily syncing at each RTCP SR. 2011-08-01 08:35:01 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpsession.c: rtpsession: trigger reconsideration if rtcp interval set 2011-08-01 08:32:24 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: configure rtcp interval if provided ... in PLAY response. 2011-09-16 16:53:22 +0300 Lasse Laukkanen * gst/isomp4/gstqtmux.c: isomp4: Fix allowing zero duration tracks https://bugzilla.gnome.org/show_bug.cgi?id=637486 2011-09-05 10:11:18 +0100 Vincent Penquerc'h * gst/udp/gstudpnetutils.c: udpsrc: error out when no protocol is specified in the uri It is certainly better than to crash. https://bugzilla.gnome.org/show_bug.cgi?id=658178 2011-09-19 09:37:58 +0200 Vincent Penquerc'h * ext/speex/gstspeexenc.c: speexenc: do not use invalid buffer timestamps 2011-03-29 12:09:18 +0530 Arun Raghavan * ext/pulse/Makefile.am: * ext/pulse/plugin.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.h: pulse: New pulseaudiosink element to handle format changes This introduces a new bin which wraps around pulsesink and depending on the formats supported by the sink, plugs in/out a decodebin2 as required. This allows users to switch sinks on the stream and adapts accordingly (for example, you could watch a movie in passthrough mode on your receiver which supports AC3 decode, then plug out and switch to a non-digital profile to continue uninterrupted on analog output). The bin is required because doing the same with playbin2/playsink will require API changes that cannot be made in 0.10. With 0.11/1.0, we should be able to ask for upstream caps renegotiation to deal with all this. https://bugzilla.gnome.org/show_bug.cgi?id=657179 2011-09-16 15:03:23 +0200 Branko Subasic * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-read-common.c: matroskademux: Avoid sending EOS when in paused state Changed the ebml reader's gst_ebml_peek_id_length() function so that it returns the actual reason for why the peek failed, instead of (almost) always returning GST_FLOW_UNEXPECTED. This prevents the pulling task from sending EOS when doing a flushing seek. 2011-09-15 15:53:47 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: fix stuttering A/V Someone got had by implicit promotion to unsigned in ops with a signed and an unsigned value. https://bugzilla.gnome.org/show_bug.cgi?id=659153 2011-09-14 16:37:12 +0100 Vincent Penquerc'h * gst/debugutils/gstnavseek.c: navseek: toggle pause/play on space bar A useful thing to have. https://bugzilla.gnome.org/show_bug.cgi?id=659065 2011-09-14 14:46:00 +0200 David Svensson Fors * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: configurable timestamp gap handling matroskademux performs segment tricks to skip gaps in streams, notably at start for non 0 based files. There may however be cases when full presentation (including intermediate gaps) is desired, so a property allows to configure as of which gap to act (or not at all). API: GstMatroskaDemux::max-gap-time Fixes #659009. 2011-09-12 09:21:47 -0300 Thiago Santos * tests/check/elements/flvmux.c: tests: flvmux: Fix flvmux's tests after fix for request pads handling Now that flvmux doesn't release its request pads on PAUSED->READY the test doesn't need to re-request them for every reuse test start. 2011-09-09 09:12:56 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: Fix ctts generation for streams that don't start at 0 timestamps Subtract the first timestamp of a stream from all input buffers to get 0-based timestamps for creating a sane ctts table. Without this patch the ctts could have larger values than needed, causing the playback to have a delay at startup. As the first timestamp is only found after a few buffers are queued (due to possible reordered buffers), once we find the first timestamp we subtract it from all buffers on the queue, from that point on, all buffers have their timestamps subtract when they are collected. https://bugzilla.gnome.org/show_bug.cgi?id=658659 2011-09-12 07:55:19 +0200 Alessandro Decina * gst/flv/gstflvmux.c: flvmux: don't release request pads going PAUSED->READY Don't release request pads but just reset them. This makes pipelines using flvmux reusable. 2011-09-09 12:35:50 +0100 Vincent Penquerc'h * gst/audioparsers/gstac3parse.c: ac3parse: use bsid 9 and 10 to control sample rate See http://matroska.org/technical/specs/codecid/index.html The spec is silent about this though... https://bugzilla.gnome.org/show_bug.cgi?id=658546 2011-09-07 14:13:03 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: ensure some initial state variable setup ... which might otherwise be skipped if the PLAY command is issued before the OPEN command had a chance to actually be acted upon. Fixes #657376. 2011-09-08 15:02:05 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: tweak gap handling ... so as to avoid buffers before and after gap to have identical running time. 2011-09-08 13:28:24 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: use GST_RESOURCE_ERROR_BUSY if v4l2_ioctl fails with EBUSY https://bugzilla.gnome.org/show_bug.cgi?id=658543 2011-09-07 08:54:17 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: remove one G_UNLIKELY for user property Using G_UNLIKELY on user properties isn't nice, specially when that is the default option. 2011-03-15 11:03:53 +0100 Andoni Morales Alastruey * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: handle GstForceKeyUnit event ... by starting a new cluster after forwarding event. Fixes #644154. 2011-09-07 14:27:36 +0200 Sebastian Dröge * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: cmml: Use complete cmml caps in the unit test 2011-09-07 14:26:01 +0200 Sebastian Dröge * tests/check/elements/qtmux.c: qtmux: Use complete MPEG caps in the unit test 2011-09-07 14:18:58 +0200 Stefan Sauer * docs/plugins/Makefile.am: docs: cleanup makefiles Remove commented out parts that we don't need. Remove "the wingo addition" - no so useful after all. Narrow down file-globs for plugin docs. 2011-08-29 14:12:22 +0200 Konstantin Miller * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't handle HTTP response 407 as error if proxy authentication data is available Fixes bug #657422. 2011-09-07 12:11:39 +0200 Sebastian Dröge * gst/audioparsers/gstac3parse.c: ac3parse: Add Converter to the classification because it can convert between different alignments This allows decodebin2 to let it negotiate properly. 2011-09-07 12:10:48 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparsers: Improve src template caps Remove the parsed/framed fields and add all fields to the template caps that always exist. 2011-09-06 15:59:49 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: parse codec_data to determine number of samples per frame Fixes #656734. 2011-09-06 21:24:46 +0200 Stefan Sauer * common: Automatic update of common submodule From a39eb83 to 11f0cd5 2011-09-06 16:57:12 +0200 Wim Taymans * configure.ac: configure: try to disable deinterlace.. 2011-09-06 15:40:32 +0200 Stefan Sauer * common: Automatic update of common submodule From 605cd9a to a39eb83 2011-09-06 16:37:03 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: common 2011-09-06 16:06:25 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/audioparsers/gstamrparse.c gst/isomp4/qtdemux.c 2011-09-06 15:40:32 +0200 Stefan Sauer * common: Automatic update of common submodule From 605cd9a to a39eb83 2011-09-06 15:05:37 +0200 Mark Nauwelaerts * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: make default duration check less sensitive Frame duration might vary for 1 usecond, in this case matroskamux decides to create BLOCKGROUP instead of SIMPLEBLOCK. Convert duration to timecodescale which is (typically) less precise, and then also allow the difference of 1/-1 to arrange for less sensitive check. Based on patch by Alexey Fisher Fixes #653080. 2011-09-06 13:18:40 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: improve bogus interleaved index compensating Patch by Fixes #654585. 2011-09-06 13:16:27 +0200 Wim Taymans * ext/jack/gstjack.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiopanorama.h: * gst/auparse/gstauparse.c: * gst/avi/gstavimux.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/law/alaw.c: * gst/law/mulaw-decode.c: * gst/law/mulaw.c: * gst/spectrum/gstspectrum.c: * gst/wavparse/gstwavparse.c: -good: port to new audio caps 2011-09-06 10:33:21 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Allow positive, non-1.0 segment rates Only negative rates are not supported. Fixes bug #658305. 2011-09-05 15:50:56 +0200 Mark Nauwelaerts * tests/check/elements/parser.c: tests: parsers: provide more real data when testing draining of garbage 2011-09-05 15:50:04 +0200 Mark Nauwelaerts * gst/audioparsers/gstamrparse.c: amrparse: fix and streamline valid frame checking ... to handle various combinations of sync or not, and sufficient data or not as might be expected. Fixes #650714. 2011-09-05 14:49:32 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: fragmented support; avoid adjustment for keyframe seek ... since all index data may not yet be available at that time. 2011-09-05 14:48:02 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: fragmented support; mark all audio track samples as keyframe 2011-09-05 14:46:29 +0200 Brian Li * gst/isomp4/qtdemux.c: qtdemux: fragmented support; properly init return variable value Fixes #655918. 2011-09-05 13:31:20 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: add gtk-doc for new short-header property 2011-09-05 13:18:39 +0200 Marc Leeman * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: allow sending short RTSP requests to a server Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by GStreamer, but do accept the short header as sent by Live555. This patch makes the extending the request optional by adding a property (short-header). Fixes #655805. API: GstRTSPSrc:short-header 2009-03-04 14:51:09 -0500 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Set H263-2000 if thats what the other side wants The static caps states this element supports H263-2000, but setcaps never sets it, so it was lie. See https://bugzilla.gnome.org/show_bug.cgi?id=577784 2011-08-30 19:02:51 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Initialise the last_keyframe_request variable 2011-08-31 16:04:24 +0200 Peter Korsgaard * gst/udp/gstmultiudpsink.c: multiudpsink: make add/remove/clear/get-stats action signals http://bugzilla.gnome.org/show_bug.cgi?id=657830 Signed-off-by: Peter Korsgaard 2011-08-31 18:45:15 +0200 Wim Taymans * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: mp2t: fix encoding name according to RFC3551 2011-08-30 13:33:49 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: push mode; perform some extra checks prior to upstream seeking 2011-08-30 13:28:21 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: push mode; fix buffered streaming That is, in case where no seek is peformed to moov, but preceding limited mdat is buffered. 2011-08-30 14:06:12 +0200 Wim Taymans * configure.ac: * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: shapewipe: port to 0.11 2011-08-30 12:49:08 +0200 Wim Taymans * configure.ac: law is ported now 2011-08-30 12:25:35 +0200 Wim Taymans * gst/law/alaw.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/law/mulaw.c: law: port to 0.11 2011-08-29 19:11:25 +0200 Wim Taymans * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: alaw: port to 0.11 2011-08-29 19:10:35 +0200 Wim Taymans * gst/goom/gstgoom.c: goom: fix comment 2011-08-29 18:02:15 +0200 Wim Taymans * configure.ac: * ext/soup/gstsouphttpsink.c: * ext/soup/gstsouphttpsrc.c: soup: port soup elements to 0.11 2011-08-29 15:13:56 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: avoid overflow wraparound in timestamp when adding durations Do some type juggling to avoid overflow, while still allowing for 'negative' durations (which would need a wraparound effect). 2011-08-29 13:43:59 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: sys/v4l2/v4l2src_calls.c 2011-08-26 14:20:49 +0200 Wim Taymans * gst/effectv/gstwarp.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: allocation: fix for vmethod changes 2011-08-25 23:37:47 +0100 Vincent Penquerc'h * sys/v4l2/v4l2src_calls.c: v4l2src: make this work more than once in a row We used to skip frame rate setup if the camera was already setup with the requested frame rate. This breaks some cameras though, causing them to not output data (several models of Thinkpad cameras have this problem at least). So, don't skip. https://bugzilla.gnome.org/show_bug.cgi?id=638300 2011-08-25 16:41:23 +0200 Wim Taymans * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/y4m/gsty4mencode.c: * sys/v4l2/gstv4l2bufferpool.c: port to new video flags 2011-08-24 18:40:07 +0200 Wim Taymans * ext/pulse/pulseutil.c: pulse: add some more channels 2011-07-12 21:48:37 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: Add more debug 2011-07-12 19:09:02 -0400 Olivier Crête * gst/dtmf/gstdtmfcommon.h: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: Max event type is 15 2011-04-14 15:46:08 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: dtmfsrc: Align DTMF sound buffers with last-stop from event Also make sure the timestamps never go backwards 2011-07-11 21:31:07 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Correctly recognize the end of a buffer 2011-07-11 20:47:23 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Make sure rtpdtmfsrc timestamps don't overlap 2011-07-11 20:46:20 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Put the inter digit interval at the end, not at the start The reason is to let rtpdtmfmux drop buffers during the inter digit interval, this way, there will be more silence around the DTMF tones so IVFs will have a better chance recognizing them. 2011-04-14 17:08:57 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: rtpdtmfsrc: Start at the last_stop from the start event if there was one The goal is to try to not have a GAP between the audio and the DTMF 2011-04-14 16:49:39 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: rtpdtmfsrc: Respect ptime from the caps Respect the ptime from the caps for the DTMF packets 2011-07-11 21:30:28 -0400 Olivier Crête * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Just error out if there is no clock 2011-08-24 14:16:44 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-08-23 12:12:15 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: only require two frames in a row when we do not have sync This avoids a single bit error dropping two frames unnecessarily. The two consecutive frames check is still required when we don't have sync. https://bugzilla.gnome.org/show_bug.cgi?id=657080 2011-08-23 21:41:15 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Trivial indentation fix 2011-08-23 19:09:31 +0200 Wim Taymans * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/rtp/gstrtpvrawpay.c: video: port to new colorimetry info 2011-07-21 17:23:28 -0400 Monty Montgomery * ext/flac/gstflacdec.c: flacdec: Correct sample number rounding resulting in timestamp jitter flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer. Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample. This corrects the time->sample convesion 2011-08-22 13:10:07 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-08-22 12:24:15 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: * gst/rtp/gstrtpj2kdepay.c: fourcc: remove fourcc from caps 2011-08-20 14:48:20 -0700 David Schleef * gst/debugutils/breakmydata.c: breakmydata: element is not passthrough 2011-07-13 11:20:34 -0700 David Schleef * gst/multifile/gstmultifilesrc.c: multifilesrc: quiet debugging 2011-07-10 21:40:20 -0700 David Schleef * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: change field handling through methods This likely breaks stuff. The good: all of the methods now create field images aligned with input frames, without timestamp mangling. The bad: this touches a lot of code, much of which is hairy and in need of cleanup. However, at this point we can reasonably create a PSNR-based test. 2011-08-21 14:41:14 +0200 Alessandro Decina * gst/multifile/gstmultifilesink.c: multifilesink: reset ->streamheaders to NULL on _stop Fixes invalid memory access reusing multifilesink 2011-08-20 10:46:18 +0200 Wim Taymans * gst/cutter/gstcutter.c: * gst/cutter/gstcutter.h: cutter: bring cutter somewhat into this millennium 2011-08-19 16:27:20 +0200 Wim Taymans * gst/replaygain/gstrganalysis.c: rg: fix caps 2011-08-19 16:13:23 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: port after merge 2011-08-19 16:12:01 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-08-19 16:09:48 +0200 Wim Taymans * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/auparse/Makefile.am: * gst/equalizer/gstiirequalizer.c: * gst/goom/gstgoom.c: * gst/level/Makefile.am: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/spectrum/gstspectrum.c: port to more audio api changes 2011-08-19 14:01:45 +0200 Wim Taymans * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/auparse/gstauparse.c: * gst/auparse/gstauparse.h: * gst/cutter/gstcutter.c: * gst/equalizer/gstiirequalizer.c: * gst/level/gstlevel.c: * gst/level/gstlevel.h: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/spectrum/gstspectrum.c: * sys/oss/gstosshelper.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * tests/check/elements/audioinvert.c: * tests/check/elements/level.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/rtpjitterbuffer.c: * tests/examples/level/level-example.c: * tests/examples/spectrum/spectrum-example.c: port more elements to new audio caps and API 2011-08-19 11:49:44 +0200 Wim Taymans * gst/audiofx/audioamplify.c: * gst/audiofx/audioamplify.h: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofirfilter.h: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioiirfilter.h: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiokaraoke.h: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: port to new audio API and caps 2011-08-18 13:37:39 +0200 David Henningsson * ext/pulse/pulsesink.c: pulsesink: Allow writes in bigger chunks There's no use in splitting the incoming data down to the segsize limit - by writing as much as possible in one chunk, we increase performance and avoid PulseAudio unnecessary rewinds. Signed-off-by: David Henningsson 2011-08-18 19:37:39 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-08-18 19:21:07 +0200 Wim Taymans * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: port to new audio caps. 2011-08-08 22:14:28 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: ensure no-more-pads is always emitted In particular, do so even if failing to read while prerolling, such as when reading from a partial file (eg, while it is being downloaded). This fixes a wedge in playbin2. https://bugzilla.gnome.org/show_bug.cgi?id=651965 2011-08-17 17:57:11 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2: improve fixate function Use new core function to fixate a field. Chain up to parent fixate function. 2011-08-17 15:52:18 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacdec.c 2011-08-17 15:39:27 +0200 Wim Taymans * configure.ac: * ext/jpeg/Makefile.am: * ext/jpeg/gstjpeg.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpeg: port to 0.11 Also disable smoke for now. 2011-08-16 17:27:13 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: avoid timestamp/offset tracking going out of sync The libFLAC API is callback based, and we must only call it to output data when we know we have enough input data. For this reason, a single processing step is done when receiving a buffer. However, if there were metadata buffers still pending, a step intended for the first audio frame might end up writing that leftover metadata. Since a single step is done per buffer, this will cause every buffer to be written one step late. This would add some latency (a bufferfull's worth), possibly lose a buffer when seeking or the like, and also cause timestamp and offset to be applied to the wrong buffer, as updates to the "current" segment last_stop (from incoming buffer timestamp) will be applied to an output buffer originating from the previous incoming buffer. This fixes the issue by ensuring that, upon receiving the first audio frame, processing is done till all metadata is processed, so the next "single step" done will be for the audio frame. After this, we should keep to 1 input buffer -> 1 output buffer and so avoid getting out of sync. https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-17 11:17:38 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-08-16 15:32:07 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: bail on reserved value Now that we look at the right bits, we can test against the reserved value as we do for other fields. https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 15:27:43 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: fix bit twiddling Right shifting a 8 bit value by 8 bits is twice too much to get the high 4 bits. https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 15:22:46 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: warn if we see a variable block size where unsupported https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 18:25:29 +0100 Vincent Penquerc'h * gst/spectrum/gstspectrum.c: spectrum: avoid crashing by resetting the correct number of channels https://bugzilla.gnome.org/show_bug.cgi?id=656606 2011-08-16 18:35:53 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: sys/v4l2/v4l2src_calls.c 2011-08-16 13:16:22 +0100 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: flacparse: fix off by one in frame size check Yes, I was tracking another bug and the small test file I generated to test with improbably just happened to trigger this, with a second and last frame of 1615 bytes. https://bugzilla.gnome.org/show_bug.cgi?id=656649 2011-08-15 12:19:14 +0200 Wim Taymans * tests/check/elements/parser.c: tests: update for _negotiated_caps() change 2011-08-14 20:46:01 +0100 Tim-Philipp Müller * gst/id3demux/id3v2.3.0.html: * gst/id3demux/id3v2.4.0-frames.txt: * gst/id3demux/id3v2.4.0-structure.txt: id3demux: remove specs from git as well now that parsing code is in -base 2011-07-14 15:42:36 +0200 Mark Nauwelaerts * configure.ac: * gst/id3demux/Makefile.am: * gst/id3demux/gstid3demux.c: * gst/id3demux/id3tags.c: * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: id3demux: use -base provided id3 tag parsing https://bugzilla.gnome.org/show_bug.cgi?id=654388 2011-08-13 16:51:22 +0100 Tim-Philipp Müller * ext/jack/gstjackaudiosrc.c: jackaudiosrc: fix error message code And also post 'not found' error if jackd is not even installed. 2011-08-12 16:32:58 +0200 Stefan Kost * gst/isomp4/qtdemux.c: qtdemux: initialize bitrate variable and reset for each loop Don't check eventually unset variable and don't accidentially use values from last cycle. 2011-08-10 11:28:26 +0200 Edward Hervey * ext/aalib/gstaasink.c: aasink: Remove unused variables 2011-08-09 11:28:17 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Properly error out if SDP contains no streams Also fixes unitialized variable error on macosx. 2011-08-09 09:05:31 +0100 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: ximagesrc: clear flags on buffer reuse This will ensure a logically new buffer does not keep flags from a previous use of that buffer (eg, DISCONT would be set on the first buffer, and mistakenly kept when reused). https://bugzilla.gnome.org/show_bug.cgi?id=653709 2011-08-08 10:54:26 +0100 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2: take care not to change the current format where appropriate Some drivers are buggy are will change the current format when processing VIDIOC_TRY_FMT. Save and restore the current format to ensure the format is kept unchanged. https://bugzilla.gnome.org/show_bug.cgi?id=649067 2011-08-08 15:27:11 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2011-08-08 15:26:00 +0100 Tim-Philipp Müller * ext/aalib/Makefile.am: aalib: make sure -DGST_USE_UNSTABLE_API is defined So we don't get warnings. 2011-08-08 15:25:31 +0100 Tim-Philipp Müller * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2videooverlay.c: * sys/v4l2/gstv4l2videooverlay.h: v4l2: update for GstXOverlay => GstVideoOverlay rename 2011-08-07 12:23:26 +0200 Sjoerd Simons * sys/v4l2/v4l2src_calls.c: v4l2src: Use fraction compare util function. Use the fraction compare utility to compare function, not the handcrafted one. The handcrafted one is buggy as it doesn't take into account rounding error. For example comparing a framerate of 20/1 on a camera configured as 30/1 fps would yield true: 1 == (1 * 20)/30 and not re-configure the camera. Fixes #656104 2011-08-07 11:14:50 +0200 Wim Taymans * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: avoid race in starting Sine the base class now does the negotiation from the streaming thread we have to be careful and check if the stream is ready before changing its corked state. 2011-08-05 12:27:18 +0200 Edward Hervey * tests/check/Makefile.am: check: Use GST_CFLAGS when building tests Ensures we have the proper define for using unstable API 2011-08-05 08:59:59 +0200 Wim Taymans * configure.ac: * gst/isomp4/gstqtmux.c: * gst/isomp4/qtdemux.c: isomp4: fixup after small api changes Port to recently changed api so that it compiles again. 2011-08-05 11:32:45 +0200 Edward Hervey * gst/y4m/Makefile.am: y4menc: Now depends on libgstvideo 2011-08-04 18:41:29 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulse: more cleanups 2011-08-04 18:15:55 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: small cleanups 2011-08-04 16:35:46 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2src: call set_caps method of baseclass Call the baseclass set_caps function to make it send the caps event and properly trigger the negotiation functions. 2011-08-04 16:25:04 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: small cleanups 2011-08-04 15:25:20 +0200 Wim Taymans * configure.ac: * gst/goom/gstgoom.c: goom: port to new caps 2011-08-04 13:52:18 +0200 Edward Hervey * sys/v4l2/gstv4l2sink.c: v4l2sink: Size variable should be a guint and not a gsize 2011-08-04 12:50:01 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.c: v4l2: activate the pool in fallback When nobody is using our pool, activate it ourselves. Avoid leaking the buffer array. Set default pool configuration with caps. Don't keep current_caps, core does that for us now. 2011-08-03 22:57:48 +0200 Wim Taymans * docs/plugins/Makefile.am: * tests/icles/videocrop-test.c: fix compilation hal elements were removed, remove them from docs too change example for pad-block API (actually remove the pad block, an application should not be bothered with working around bugs in elements) 2011-08-03 18:37:27 +0200 Wim Taymans * ext/pulse/pulsesink.c: * gst/audioparsers/gstac3parse.c: * gst/rtp/gstrtph264depay.c: port to new API 2011-08-03 18:25:30 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/pulse/pulsesink.c ext/pulse/pulsesrc.c gst/audioparsers/gstac3parse.c gst/rtp/gstrtph264depay.c gst/rtp/gstrtph264pay.c gst/rtpmanager/gstrtpssrcdemux.c 2011-08-03 22:50:05 +1000 Jan Schmidt * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: * gst/matroska/matroska.c: matroska: Register new debug category Register the matroskareadcommon debug category when the plugin is loaded to avoid assertion output when debug is turned on. 2011-08-03 13:38:01 +0200 Edward Hervey * tests/icles/gdkpixbufsink-test.c: test/ickles: Port gdkpixbufsink test 2011-08-03 13:33:59 +0200 Edward Hervey * tests/check/Makefile.am: * tests/check/elements/autodetect.c: Revert "tests/check/Makefile.am: Disable autodetect test temporarily, so that the build bots update -bad and the ranks of unr..." This reverts commit 475aed8af6d2a57c1d21490c824e754a6b2367a9. It won't consider elements from anywhere else anymore 2011-08-03 13:10:46 +0200 Edward Hervey * tests/check/Makefile.am: * tests/check/elements/parser.c: check: Update parser mini-lib to 0.11 API 2011-08-03 13:09:07 +0200 Edward Hervey * po/POTFILES.in: po: update for modified source file location 2011-08-03 13:08:43 +0200 Edward Hervey * configure.ac: configure.ac: cairo_gobject isn't ported either 2011-08-03 10:59:56 +0200 Sebastian Dröge * configure.ac: * ext/Makefile.am: * ext/hal/Makefile.am: * ext/hal/gsthalaudiosink.c: * ext/hal/gsthalaudiosink.h: * ext/hal/gsthalaudiosrc.c: * ext/hal/gsthalaudiosrc.h: * ext/hal/gsthalelements.c: * ext/hal/gsthalelements.h: * ext/hal/hal.c: * ext/hal/hal.h: hal: Remove hal plugin hal is not developed anymore and nobody is using the plugin nowadays. 2011-07-29 13:03:55 +0200 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: soften assertion check on stream size https://bugzilla.gnome.org/show_bug.cgi?id=655570 2011-08-03 10:09:42 +0200 Robert Krakora * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Add support for H.264 payload in MJPEG container See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf Fixes bug #655530. 2011-08-02 22:05:08 -0400 Tristan Matthews * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: jackaudiosink: Don't call g_alloca() in process_cb g_alloca() is not RT-safe, so instead we should allocate the memory needed in advance. Fixes #655866 2011-08-03 08:58:00 +0200 Sebastian Dröge * configure.ac: configure: Add hal to the list of non-ported plugins 2011-08-03 08:53:24 +0200 Sebastian Dröge * configure.ac: configure: Add monoscope to the list of non-ported plugins 2011-08-03 08:51:19 +0200 Sebastian Dröge * gst/effectv/gstquark.c: * gst/effectv/gstwarp.c: effectv: Fix unused but set variable compiler warnings 2011-08-02 23:42:58 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * sys/v4l2/gstv4l2object.c: docs: fix two more Since: tags 2011-07-31 04:19:00 +0300 Mart Raudsepp * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix Since tags for fieldanalysis related new properties commit c1b100cf9c is after 0.10.29 and 0.10.30 was a branched release. So fix Since tags from 0.10.29 to 0.10.31 for the new properties. 2011-08-02 11:51:45 +0200 Wim Taymans * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: fix porting error 2011-08-02 11:29:40 +0200 Edward Hervey * configure.ac: configure.ac: Define list of non-ported plugins 2011-08-02 11:29:25 +0200 Edward Hervey * common: Update common submodule 2011-08-02 11:17:38 +0200 Edward Hervey * configure.ac: configure.ac: Sort AG_GST_CHECK_PLUGIN alphabetically 2011-07-29 17:27:07 +0200 Wim Taymans * gst/effectv/gstwarp.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: -good: fix for bufferpool API change 2011-07-29 17:21:03 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2src.c: v4l: change for new API 2011-07-29 13:05:42 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: fix variable-set-but-not-used compiler warning with older pulse versions 2011-07-29 12:07:24 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpsession.c: rtpsession: properly init rtcp_min_interval 2011-03-09 11:04:36 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.c: pulsesink: Add support for compressed formats This adds support for various compressed formats (AC3, E-AC3, DTS and MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF, HDMI and Bluetooth). The acceptcaps() function allows bins to probe for what formats the sink being connected to support. This only works after the element is set to at least READY. If the underlying sink changes and the format we are streaming is not available, we emit a message that will allow upstream elements/bins to block and renegotiate a new format. 2011-03-01 15:34:46 +0530 Arun Raghavan * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulsesink: Use the extended stream API if available This uses the new extended API for creating streams. This will allow us to support compressed formats natively in pulsesink as well. 2011-07-29 00:07:52 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: Add a source-output-index property This exposes the source output index of the record stream that we open so that clients can use this with the introspection if they want (to move the stream, for example). 2011-07-28 14:44:57 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: keep a ref on the src pad while using it Prevent a possible race if clear_ssrc() is called between getting the pad and doing the push. Based on patch by https://bugzilla.gnome.org/show_bug.cgi?id=650916 2011-05-24 11:29:57 +0300 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit We need to keep the lock held because we don't want a push before the "new-ssrc-pad" handler has completed. But we may want to push an event from inside that handler, hence the recursive mutex. https://bugzilla.gnome.org/show_bug.cgi?id=650916 2011-05-24 11:17:25 +0300 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Use PADs lock https://bugzilla.gnome.org/show_bug.cgi?id=650916 2011-07-28 11:09:08 +0100 Tim-Philipp Müller * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: update for position/query/convert API changes 2011-07-28 10:54:38 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/progressreport.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/isomp4/qtdemux.c: * gst/wavparse/gstwavparse.c: gst: udpate for position/duration/convert query API changes 2011-07-28 00:37:13 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: fix compiler warning gstavidemux.c: In function 'gst_avi_demux_parse_stream': gstavidemux.c:1261:24: error: 'data' may be used uninitialized in this function [-Werror=uninitialized] gstavidemux.c:1204:11: note: 'data' was declared here 2011-07-27 18:15:20 +0100 Sjoerd Simons * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: Cope with FU-A E bit not being set Some h264 payloaders are unfortunately buggy and don't correctly set the E bit in FU-A NAL when they have ended. Work around this by assuming such a fragmentation unit has ended when there was no packet loss and a new NAL is started 2011-04-12 17:01:47 +0530 Arun Raghavan * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: Support switching alignment on-the-fly This allows switching of alignment for E-AC3 streams at run-time. This is requested by downstream elements via a custom event. https://bugzilla.gnome.org/show_bug.cgi?id=650313 2011-07-27 16:46:46 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: v4l2: remove unused variables Use the more specialized type for the bufferpool. Use the size from the driver as the size of the image to read. Don't configure the pool when created. This will be done in the setup_allocation method later or by upstream for sinks. Remove unused properties and variables. Bufferpool sizes are now configured in the bufferpool by the elements in the pipeline. We might want to influence the pool size later somehow. 2011-07-27 13:46:09 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.h: v4l2bufferpool: remove unused variable 2011-07-27 13:43:08 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2src: add metadata 2011-07-27 13:41:28 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: bufferpool: check for metadata Only add video metadata when it was configured in the pool. Fail if there was no video metadata configured and the strides are not the default ones. 2011-07-27 12:42:21 +0100 Wim Taymans * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: warp: add stride support 2011-07-27 12:41:33 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: add colorspace to debug 2011-07-26 17:45:01 +0200 Wim Taymans * gst/rtp/gstrtph264pay.c: rtp: fix compilation 2011-07-26 16:15:05 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: v4l2: rename a variable Rename the size variable to sizeimage and fill it with the size that has been given to use by the v4l2 driver instead of making something up.. 2011-07-26 13:18:55 +0200 Wim Taymans * sys/v4l2/gstv4l2sink.c: v4l2: use new setup_allocation vmethod 2011-07-26 10:56:07 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: implement more bits of RW I/O mode Implement the relaese of RW buffers in the pool. Warn for unsupported write() mode for sinks. 2011-07-26 10:54:23 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: improve IO mode error handling Error out when an unsupported IO mode was selected 2011-04-09 12:26:56 +0530 Arun Raghavan * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: * tests/check/elements/ac3parse.c: ac3parse: Add support for IEC 61937 alignment When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading requires each buffer to contain 6 blocks from each substream. This adds code to collect all the frames needed to meet this requirement before pushing out a buffer. https://bugzilla.gnome.org/show_bug.cgi?id=650313 2011-06-08 15:57:37 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Always send application requested feedback in immediate mode Send as many application requested feedback messages in immediate mode, even if they have already been sent. https://bugzilla.gnome.org/show_bug.cgi?id=654583 2011-06-08 14:48:01 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Don't let the computed RTP bandwidth fall too low If it falls too low, the computed RTCP bandwidth will be near zero and the RTCP thread will be stopped. https://bugzilla.gnome.org/show_bug.cgi?id=654583 2011-04-25 16:13:38 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Wait longer to timeout SSRC collision Using the current RTCP interval to timeout SSRC collision can lead to collisions being timed out immediately if a BYE packet is sent because it is sent immediately, so the interval is 0. This is not what we want. So just set a static 10 times the default RTCP interval, it should be enough https://bugzilla.gnome.org/show_bug.cgi?id=648642 2011-07-25 15:51:22 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: remove unused method 2011-07-25 15:38:38 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: fix flushing start and stop Move the flushing calls to the right place in the bufferpool. Fix the min and max buffer sizes. 2011-07-25 14:47:05 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: dequeue buffers when all are queued Prefer to always use the default bufferpool queue for the _acquire function because it properly supports unblocking when setting inactive etc. As a result, we need to dequeue buffers and put them back in the bufferpool queue when we have queued all buffers in the sink. Rename some variables to more meaningfull names to avoid a problem with freeing the wrong amount of buffers. 2011-07-19 13:38:01 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: set SOURCE flag at init time Fixes #654816. 2011-07-25 10:10:58 +0100 Wim Taymans * gst/effectv/gstvertigo.c: vertigotv: add stride support 2011-07-19 18:25:29 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: only to STREAMOFF when streaming Only call STREAMOFF when we previously called STREAMON 2011-07-22 21:26:32 +0200 Wim Taymans * gst/replaygain/gstrganalysis.c: replay: fix for event handler 2011-07-22 21:19:45 +0200 Wim Taymans * gst/audiofx/audiofxbasefirfilter.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/progressreport.c: fixes for event handler changes 2011-07-18 16:46:27 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: rtph264depay: Complete merged AU on marker bit The marker bit on a RTP packet means the AU has been completed, so push it out immediately to reduce the latency. https://bugzilla.gnome.org/show_bug.cgi?id=654850 2011-07-18 20:27:38 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit An access unit could contain multiple NAL units, in that case, only the last RTP packet of the last NALU should have its marker bit set. https://bugzilla.gnome.org/show_bug.cgi?id=654850 2011-07-20 08:52:58 +0200 Alessandro Decina * gst/multipart/multipartmux.c: multipart: fix compiler warning 2011-07-19 18:20:43 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: v4l2: handle unsupported formats 2011-07-19 16:59:55 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: v4l2: Fix sink bufferpool handling Remove old method, use neww _process method for the sink. Inform the parent bufferpool class about the settings too. This is needed to let it know about the max-buffers. Allocate the negotiated max-buffers and initially mmap min-buffers. The idea is that the bufferpool will allocate more when needed. Improve debugging. Only poll in capture mode, it does not seem to work in playback mode on this beagleboard. 2011-07-19 12:05:51 +0200 Mark Nauwelaerts * gst/auparse/gstauparse.c: auparse: avoid hanging on invalid short input ... as in such case there is no srcpad yet on which to forward EOS. 2011-07-18 15:13:33 -0300 Thiago Santos * ext/pulse/pulsesrc.c: pulsesrc: Fix default value leaking Remember to free the default value of client name, avoiding a leak 2011-07-18 18:54:49 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: More work on bufferpools Add different transport methods to the bufferpool (MMAP and READ/WRITE) Do more parsing of the bufferpool config. Start and stop streaming based on the bufferpool state. Make separate methods for getting a buffer from the pool and filling it with data. This allows us to fill buffers from other pools too. Either use copy or read to fill up the target buffers. Add property to force a transfer mode in v4l2src. Increase default number of buffers to 4. Negotiate bufferpool and its properties in v4l2src. 2011-07-18 14:24:48 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: reset upon FLUSH_STOP ... which is particularly needed when merging NAL units, where not resetting would lead to output of an older (pre-flush) AU (with unintended timestamp). 2011-07-18 14:30:51 +0200 Mark Nauwelaerts * gst/multifile/gstmultifilesink.c: multifilesink: do not use g_slist_free_full ... as that is only in GLib 2.28, which is not yet required at this time. 2011-07-18 10:52:23 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: add IO method enum 2011-07-18 10:51:21 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: bufferpool: improve _new function 2011-07-18 09:38:26 +0200 Alessandro Decina * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * tests/check/elements/multifile.c: multifilesink: add max-files property Add max-files property to limit the number of files saved on disk. API: multifilesink::max-files 2011-07-17 23:36:55 +0200 Alessandro Decina * gst/multifile/gstmultifilesink.c: multifilesink: refactor file opening and closing code 2011-07-16 19:38:51 +0200 Alexey Fisher * gst/matroska/matroska-demux.c: matroskademux: fix pixel-aspect-ratio if header has only one display variable Current matroska demux calculates the pixel aspect ratio only if both DisplayHeight and DisplayWidth are set, but it is legal to use only one variable if the other is equal to PixelWidth or PixelHeight, at least the mkclean utility is doing that. So this makse mkcleaned files play correctly. https://bugzilla.gnome.org/show_bug.cgi?id=654744 2011-07-16 23:47:50 +0100 Antoine Jacoutot * gst/goom/plugin_info.c: goom: fix build on PPC on openbsd A missing sys/param.h include results in: /usr/include/sys/proc.h:64: error: 'MAXLOGNAME' undeclared here (not in a function) /usr/include/sys/proc.h:285: error: 'MAXCOMLEN' undeclared here (not in a function) when compiling goom on openbsd/ppc. We can just remove the two sys/ includes here, they are not needed for anything. https://bugzilla.gnome.org/show_bug.cgi?id=654749 2011-07-15 17:06:39 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-07-15 16:55:50 +0200 Wim Taymans * sys/v4l2/gstv4l2src.c: v4l2: implement setup_allocation Implement the setup_allocation vmethod, we'll hopefully do something clever in there later. 2011-07-15 16:26:06 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: improve bufferpool config setting Pass the caps and the default video size to the bufferpool config. Don't activate the bufferpool, this will be done by the object that decides to use the bufferpool. Improve debugging and error reporting. 2011-07-15 13:52:38 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: handle dequeueing correcly First clean up the buffers in the queue, then the remaining ones in the device. 2011-07-15 13:29:42 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: unref copied buffer After we copy the incomming buffer to one of our bufferpool buffers, unref the target buffer after rendering so that it is put back in the pool. 2011-07-15 13:07:11 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: dequeue buffers for the sink When we have all buffers queued for playback and we need a new empty buffer, dequeue one and return it. Set the right size for sink buffers. Improve counting of queued buffers. 2011-07-15 12:35:14 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: use the parent queue for the sink We want to maintain a queue of free buffers for the sink, use the parent methods to do that. 2011-07-15 12:00:54 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: fix error messages 2011-07-15 11:30:25 +0200 Wim Taymans * sys/v4l2/gstv4l2sink.c: v4l2: add ALLOCATION query to the sink 2011-07-15 11:27:18 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: convert to GstBufferPool Extend from GstBufferPool. Handle the lifetime of the pool buffers correctly with the start/stop vmethods. Map acquire and release directly to QBUF and DQBUF. We still expose an explicit qbuf for the v4l2sink for now. 2011-07-15 11:18:03 +0200 Wim Taymans * sys/v4l2/v4l2_calls.c: v4l2: remove experimental markers 2011-07-14 20:10:02 -0400 Olivier Crête * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic Partially reverts 397dc60b 2011-07-14 16:21:36 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: merge code 2011-07-14 16:12:15 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: v4l2: Move output details to device object Move the details of how a buffer is rendered to the device object. 2011-03-04 15:41:22 -0500 Olivier Crête * gst/rtp/Makefile.am: * gst/rtp/gstrtph264pay.c: rtph264pay: Implement getcaps Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level) 2011-07-13 18:32:00 +0200 Wim Taymans * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: move capture code to device object Move the details of how to capture to the device object. Remove the v4l2src_calls.[ch] files because they are empty now. Provide two simple methods to get and return a buffer to the device. Also do a slow copy when the buffer is not from our pool. 2011-07-13 16:58:08 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: add some more debug 2011-07-13 16:56:21 +0100 Wim Taymans * sys/v4l2/gstv4l2sink.c: v4l2: stop streaming in READY and NULL 2011-07-13 16:40:39 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: start streaming for the output as well 2011-07-13 16:33:58 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: Let the device object manage the pool Rename start and stop methods to open and close because that is what they do. After setting the format on the device object, setup the bufferpools. Move this code from the v4l2src_calls.c file, it is shared between source and sink. Make new device start and stop method that merges various bits of common code spread over several files. 2011-07-13 13:52:30 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: don't store stride in the videoinfo We want to keep the default strides in the videoinfo. Keep the stride of the video frames separate so that we can use both to copy a video frame and do correct stride conversion. 2011-07-13 13:38:15 +0200 Wim Taymans * sys/v4l2/gstv4l2sink.c: v4l2: Use video frame copy for raw video Use the video frame copy API for raw video frames so that we copy with the right strides. 2011-07-13 13:37:58 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: add video metadata to raw video buffers 2011-07-13 13:15:05 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: v4l2: small cleanups 2011-07-13 13:00:42 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: v4l2: improve caps parsing Use GstVideoInfo to store the parsed caps. Remove outsize from the caps parsing code, it's wrong because it does not use the stride given by the driver. 2011-07-13 11:40:11 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: use errno 2011-07-13 11:36:54 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: handle EINVAL without posting a warning EINVAL means that a call is not supported, we only want to post a WARNING when something is really wrong. 2011-07-13 11:29:26 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: only set framerate for capture for now 2011-07-13 11:19:28 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: Move configuration of framerate to _set_format Move the configuration of the framerate to where we set the other format parameters. Remove hack to check if the device is active. Store streamparm in the device info. Use some macros to access the current device configuration. Remove some duplicate fields in src and sink and use the device configuration instead. 2011-07-12 19:13:45 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: fix return value... 2011-07-12 19:03:32 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: simplify setting the capture format Pass the caps to the set_format function and make _set_format parse the caps. Also keep the parsed values in the v4l2object so that we can refer to them when we want. 2011-07-12 18:41:47 +0100 Wim Taymans * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: remove more unused parameters 2011-07-12 18:29:35 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l: handle object out of the normal flow 2011-07-12 18:13:42 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2src_calls.c: v4l2: Let the bufferpool own the V4l2Object Keep track of the currently configured format and setting in the v4l2object. Pass the v4l2object to the bufferpool constructor so that the bufferpool can know everything about the currently configured settings. This also allows us to remove some awkward code. 2011-07-12 17:06:41 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2src_calls.c: v4l: remove caps argument, it's not needed Remove the caps parameter, we don't need it anymore because we don't set caps on buffers anymore. 2011-07-12 16:46:21 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2src_calls.c: v4l: pass the bytesperline around When setting a format, return the bytesperline to the caller so that it can be used to allocate buffers. 2011-07-12 16:43:04 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: pool: make buffer writable We need writable buffers when we need to do a slow memcpy. 2011-07-12 15:04:38 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: fix seeking regression ... introduced when shuffling around code for the async implementation by setting state of source (and udp sources) in _play before downstream flushing is undone. 2011-07-11 15:23:41 +0300 René Stadler * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: fix buffer duration on blocks-per-frame change The gst_base_parse_set_frame_rate call was predicated on a change to sample rate, duration or profile. However, the block count per frame can also change between packets, which would result in incorrect buffer durations. 2011-07-11 13:51:52 +0200 Wim Taymans * sys/v4l2/gstv4l2sink.c: v4l2sink: handle pools Create a new pool in setcaps and stop/destroy the old one. Remove buffer_alloc functions. Check that we have v4l2 metadata in show_frame and fall back to memcpy into a buffer from our pool if we don't receive one of our own buffers. 2011-07-11 12:04:57 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: v4l2: various cleanups Various cleanups, avoids useless casts, move error handling outside of the main code flow. Negotiate to a resonable resolution instead of the max resolution. 2011-07-10 21:50:19 +0200 Mark Nauwelaerts * gst/rtp/Makefile.am: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: rtp: port remaining to 0.11 2011-07-10 14:56:00 +0200 Wim Taymans * sys/ximage/gstximagesrc.c: * sys/ximage/ximageutil.c: ximage: port to 0.11 2011-07-10 13:44:49 +0200 Wim Taymans * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: y4m: port some more Use video helpers. 2011-07-10 13:28:27 +0200 Wim Taymans * gst/y4m/gsty4mencode.c: y4m: port to 0.11 2011-07-10 12:46:03 +0200 Wim Taymans * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: multipart: port to 0.11 2011-07-10 11:42:37 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-07-10 11:40:40 +0200 Wim Taymans * gst/debugutils/Makefile.am: * gst/debugutils/breakmydata.c: * gst/debugutils/efence.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstdebug.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavigationtest.h: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/tests.c: debug: port to 0.11, disable others Diasable the efence and capsdebug elements, port them later. 2011-07-09 19:23:41 -0700 David Schleef * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: multifilesrc: Improve looping Add start-index and stop-index properties. 2011-06-16 13:57:03 +0100 Jonny Lamb * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: multifile: add loop property to multifilesrc Fixes: #652727 Signed-off-by: Jonny Lamb Signed-off-by: David Schleef 2009-11-20 10:07:43 +0100 Philip Jägenstedt * sys/directsound/gstdirectsoundsink.c: directsoundsink: 16-bit audio is signed, 8-bit is unsigned. Pretending to handle 8-bit signed causes distorted audio when actually given such audio, which you will get if passing 8-bit unsigned through audioconvert ! audioresample, as audioresample only handles 8-bit signed. Fixes #605834. Signed-off-by: David Schleef 2011-07-08 16:37:11 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: v4l2: fix gray format, use filter in getcaps 2011-07-08 16:10:47 +0200 Wim Taymans * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: v4l2: port and enable v4l2sink 2011-07-08 14:34:40 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: v4l2src: port to new video formats 2011-07-08 12:51:14 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-07-08 12:49:12 +0200 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: v4l2: port to 0.11 2011-07-07 18:27:36 +0200 Alexey Fisher * gst/matroska/matroska-demux.c: matroskademux: handle blocks with duration=0 Some video frames, for example alt-ref frame in VP8, will be never displayed. This is why it has duration=0. This patch allow to use this duration. Bug: 654175 Signed-off-by: Alexey Fisher 2011-07-06 17:18:05 -0700 David Schleef * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Add direct dirac mapping 2011-07-07 17:59:04 +0200 Wim Taymans * gst/effectv/gstripple.c: * gst/effectv/gstripple.h: effectv: port last effectv element to 0.11 2011-07-07 17:49:34 +0200 Wim Taymans * gst/effectv/gstradioac.c: * gst/effectv/gststreak.c: * gst/effectv/gststreak.h: effectv: port streaktv to 0.11 2011-07-07 17:40:22 +0200 Wim Taymans * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: effectv: port radioactv to 0.11 2011-07-07 17:29:58 +0200 Wim Taymans * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effectv: fix docs 2011-07-07 17:29:44 +0200 Wim Taymans * gst/effectv/gstop.c: * gst/effectv/gstop.h: effectv: port op to 0.11 2011-07-07 17:18:21 +0200 Wim Taymans * gst/effectv/gstquark.c: * gst/effectv/gstquark.h: * gst/effectv/gstrev.c: effectv: port quark tv 2011-07-07 16:57:39 +0200 Wim Taymans * gst/effectv/gstrev.c: * gst/effectv/gstrev.h: effectv: port revtv to 0.11 2011-07-07 16:46:51 +0200 Wim Taymans * gst/effectv/gstvertigo.c: * gst/effectv/gstvertigo.h: effectv: port vertigotv to 0.11 2011-07-07 16:38:10 +0200 Wim Taymans * gst/effectv/gstaging.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstshagadelic.h: effectv: port shagadelictv to 0.11 2011-07-07 11:22:26 +0200 Mark Nauwelaerts * gst/auparse/gstauparse.c: auparse: use ALWAYS src pad rather than SOMETIMES 2011-07-07 11:14:16 +0200 Mark Nauwelaerts * gst/auparse/gstauparse.c: auparse: port to 0.11 2011-07-06 19:03:52 +0200 Wim Taymans * gst/shapewipe/gstshapewipe.c: shapewipe: beginnings of porting 2011-07-06 18:50:26 +0200 Wim Taymans * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: warptv: port to 0.11 2011-07-06 18:50:15 +0200 Wim Taymans * gst/effectv/gstdice.c: dice: keep track of info 2011-07-06 18:32:45 +0200 Wim Taymans * gst/effectv/gstdice.c: * gst/effectv/gstdice.h: effectv: port dice 2011-07-06 18:09:49 +0200 Wim Taymans * gst/effectv/gstaging.c: * gst/effectv/gstaging.h: effectv: port agingtv 2011-07-06 17:50:54 +0200 Wim Taymans * ext/aalib/Makefile.am: * ext/aalib/gstaasink.c: * ext/aalib/gstaasink.h: aasink: port to new video API 2011-07-06 17:40:20 +0200 Wim Taymans * ext/libcaca/Makefile.am: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacasink.h: cacasink: port to 0.11 2011-07-06 16:50:32 +0200 Wim Taymans * ext/jpeg/gstjpegenc.c: jpeg: beginnings of porting to 0.11 2011-07-06 16:31:18 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: use ALWAYS source pad rather than SOMETIMES 2011-07-06 16:10:34 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: port to 0.11 2011-07-06 16:10:23 +0200 Mark Nauwelaerts * gst/wavenc/gstwavenc.c: wavenc: port to 0.11 2011-07-06 12:22:43 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: adjust to unsigned segment fields 2011-07-06 15:57:23 +0200 Wim Taymans * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: port speex elements 2011-07-06 12:05:12 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-07-06 10:11:52 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpmanager: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces * misc caps and segment handling changes FIXME: also relies on being able to pass caps along with a buffer, which has no evident equivalent yet, so that either needs one, or still needs quite some code path modification to drag along caps. 2011-06-29 20:59:26 +0300 René Stadler * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: prevent race condition causing ref leak Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the deferred call to be run before returning. This causes a race when READY->NULL is executed shortly after, which stops the mainloop. This leaks the element reference which is passed as userdata for the callback (introduced in commit 7cf996, bug #614765). The correct fix is to wait in READY->NULL for all outstanding calls to be fired (since libpulse doesn't provide a DestroyNotify for the userdata). We get rid of the reference passing from 7cf996 altogether, since finalization from the callback would anyways lead to a deadlock. Re-fixes bug #614765. 2011-07-04 08:58:14 +0300 René Stadler * ext/pulse/pulsesink.c: pulsesink: small cleanup of copy-paste code 2011-06-29 19:50:42 +0300 René Stadler * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: remove unused member variable and misleading log message Wim changed it in commit 8bfd80 so that pa_defer_ran is not read anywhere. The log message used to annotate a mainloop_wait call which is gone. 2011-07-05 15:37:52 +0200 Wim Taymans * gst/videofilter/gstvideoflip.c: videoflip: fix caps 2011-07-05 11:40:56 +0200 Wim Taymans * gst/effectv/gstedge.c: * gst/effectv/gstedge.h: effectv: port edgetv 2011-07-05 10:12:25 +0100 Tim-Philipp Müller * configure.ac: Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings 2011-07-04 12:58:38 -0700 David Schleef * gst/goom/gstgoom.c: goom: Don't answer lantency queries before negotiation 2011-07-04 18:15:42 +0200 Wim Taymans * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: udp: port to new API 2011-07-04 18:12:56 +0200 Wim Taymans * ext/pulse/pulsemixer.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: remove implementsinterface 2011-07-04 18:10:55 +0200 Wim Taymans * gst/alpha/gstalpha.c: alpha: fix caps 2011-07-04 18:06:48 +0200 Wim Taymans * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: alpha: port to new video API 2011-07-04 17:00:34 +0200 Wim Taymans * gst/alpha/gstalpha.c: alpha: more porting 2011-07-04 16:09:33 +0200 Wim Taymans * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: port to new video api 2011-06-28 14:03:43 +0200 Wim Taymans * gst/videofilter/gstgamma.c: * gst/videofilter/gstgamma.h: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: video: port to new video apis 2011-07-04 14:30:09 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: avoid crashing on invalid input without components 2011-07-04 11:09:19 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvmux.c: flv: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer * misc segment and caps changes 2011-07-04 11:48:13 +0200 Mark Nauwelaerts Merge branch 'master' into 0.11 Conflicts: ext/pulse/pulsesink.c 2011-07-04 11:25:28 +0200 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: pass along segment info to collectpads ... so it can track this and be subsequently used to determine running time etc. 2011-07-04 11:24:23 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: indicate raw format in aac caps 2011-07-04 11:07:13 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: mind requested name for request pad 2011-07-04 11:06:54 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: free scheduling query 2011-07-03 19:51:32 -0700 David Schleef * ext/pulse/plugin.c: pulse: Increase ranks to PRIMARY + 10 So that pulsesrc/pulsesink get chosen over other possible PRIMARY src/sinks by autoaudiosink. Presumably, if pulse is available, it is always preferred over another src/sink. Fixes: #647540. 2011-06-30 18:47:48 -0700 David Schleef * gst/multipart/multipartmux.c: multipartmux: Add \r\n to tail of pushed buffers Clients such as Firefox require the \r\n after the payload. 2011-06-16 14:52:51 +0200 Branko Subasic * gst/matroska/ebml-read.c: * gst/matroska/matroska-demux.c: matroskademux: avoid looping when searching for clusters Fixes some bugs that results in the demuxer looping when seaching for clusters in non-finalized files. https://bugzilla.gnome.org/show_bug.cgi?id=652195 2011-06-30 12:30:22 +0200 Mark Nauwelaerts * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: multifile: port to 0.10 * use G_DEFINE_TYPE * adjust to new GstBuffer * misc caps handling 2011-06-30 11:35:21 +0200 Mark Nauwelaerts * gst/cutter/gstcutter.c: cutter: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer * minor misc 2011-06-30 11:17:19 +0200 Mark Nauwelaerts * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: replaygain: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer 2011-06-30 10:53:09 +0200 Mark Nauwelaerts * gst/spectrum/gstspectrum.c: spectrum: remove deprecated property 2011-06-30 10:51:55 +0200 Mark Nauwelaerts * gst/spectrum/gstspectrum.c: spectrum: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer 2011-06-30 10:38:49 +0200 Mark Nauwelaerts * gst/level/gstlevel.c: level: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer 2011-06-30 10:30:16 +0200 Mark Nauwelaerts * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: equalizer: port to 0.11 2011-06-10 18:54:48 +0530 Debarshi Ray * gst/matroska/matroska-parse.c: matroskaparse: fix reference counting of parse->streamheader https://bugzilla.gnome.org/show_bug.cgi?id=652286 Signed-off-by: David Schleef 2011-06-29 14:39:52 -0700 David Schleef * ext/jpeg/gstjpegenc.c: jpegenc: Don't round up size of encoded buffers For some reason, in code dating to 2001, encoded jpeg buffers were rounded up to multiples of 4 bytes. With the added bonus that the extra bytes are unwritten, causing valgrind issues. Oops. I can't think of any reason why JPEG buffers need to be multiples of 4 bytes, so I removed the padding. There might be some code somewhere that depends on this behavior, so if this needs to be reverted, please fix the valgrind issues. 2011-06-29 12:46:20 +0200 Mark Nauwelaerts * gst/isomp4/Makefile.am: * gst/isomp4/atoms.c: * gst/isomp4/atomsrecovery.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: isomp4: port to 0.11 2011-06-28 12:55:45 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: tweak some ported segment handling ... to avoid losing duration during push mode seeking, and to properly accumulate running time when segment seeking. 2011-06-29 12:05:04 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: free date tag 2011-06-28 12:26:37 +0200 Jonas Larsson * gst/audioparsers/gstaacparse.c: aacparse: not so greedy minimum frame size Fixes #653559. 2011-06-25 11:39:23 -0700 David Schleef * configure.ac: configure: remove non-pkg-config check for shout Fixes: 653327 2011-06-20 18:49:57 +0200 Andoni Morales Alastruey * ext/raw1394/gst1394clock.c: dv1394src: make the internal clock thread safe Fixes: #653091. 2011-06-24 11:54:29 +0200 Miguel Angel Cabrera Moya * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: return correct type when assertion fails 2011-06-23 11:28:27 -0700 David Schleef * common: Automatic update of common submodule From 69b981f to 605cd9a 2011-06-22 16:41:13 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: fix for uri changes 2011-02-02 16:18:54 +0530 Arun Raghavan * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: Drop support for PA versions before 0.9.16 This drops support fof PulseAudio versions prior to 0.9.16, which was released about 1.5 years ago. Testing with very old versions is not feasible and we don't want to maintain 2 independent code-paths. 2011-06-21 18:24:41 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: configure.ac docs/plugins/inspect/plugin-esdsink.xml docs/plugins/inspect/plugin-gconfelements.xml 2011-06-21 18:19:02 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: fix for header cleanups 2011-06-21 15:15:06 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: fix output buffer timestamps in case of multiple frames 2011-06-20 16:47:36 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: The signal has 5 arguments, not 4 2011-06-20 12:13:11 +0200 Wim Taymans * gst/avi/gstavimux.c: avimux: use string for video format now 2011-06-20 12:04:48 +0200 Wim Taymans * gst/avi/Makefile.am: avi: link against gstvideo now 2011-06-20 12:03:24 +0200 Wim Taymans * gst/avi/gstavimux.c: avi: port to new caps 2011-06-18 13:43:02 +0100 Tim-Philipp Müller Bump git version after unplanned 0.10.30 release Merge branch '0.10.30' Conflicts: configure.ac docs/plugins/inspect/plugin-1394.xml docs/plugins/inspect/plugin-aasink.xml docs/plugins/inspect/plugin-alaw.xml docs/plugins/inspect/plugin-alpha.xml docs/plugins/inspect/plugin-alphacolor.xml docs/plugins/inspect/plugin-annodex.xml docs/plugins/inspect/plugin-apetag.xml docs/plugins/inspect/plugin-audiofx.xml docs/plugins/inspect/plugin-audioparsers.xml docs/plugins/inspect/plugin-auparse.xml docs/plugins/inspect/plugin-autodetect.xml docs/plugins/inspect/plugin-avi.xml docs/plugins/inspect/plugin-cacasink.xml docs/plugins/inspect/plugin-cairo.xml docs/plugins/inspect/plugin-cutter.xml docs/plugins/inspect/plugin-debug.xml docs/plugins/inspect/plugin-deinterlace.xml docs/plugins/inspect/plugin-dv.xml docs/plugins/inspect/plugin-efence.xml docs/plugins/inspect/plugin-effectv.xml docs/plugins/inspect/plugin-equalizer.xml docs/plugins/inspect/plugin-esdsink.xml docs/plugins/inspect/plugin-flac.xml docs/plugins/inspect/plugin-flv.xml docs/plugins/inspect/plugin-flxdec.xml docs/plugins/inspect/plugin-gconfelements.xml docs/plugins/inspect/plugin-gdkpixbuf.xml docs/plugins/inspect/plugin-goom.xml docs/plugins/inspect/plugin-goom2k1.xml docs/plugins/inspect/plugin-gstrtpmanager.xml docs/plugins/inspect/plugin-halelements.xml docs/plugins/inspect/plugin-icydemux.xml docs/plugins/inspect/plugin-id3demux.xml docs/plugins/inspect/plugin-imagefreeze.xml docs/plugins/inspect/plugin-interleave.xml docs/plugins/inspect/plugin-isomp4.xml docs/plugins/inspect/plugin-jack.xml docs/plugins/inspect/plugin-jpeg.xml docs/plugins/inspect/plugin-level.xml docs/plugins/inspect/plugin-matroska.xml docs/plugins/inspect/plugin-mulaw.xml docs/plugins/inspect/plugin-multifile.xml docs/plugins/inspect/plugin-multipart.xml docs/plugins/inspect/plugin-navigationtest.xml docs/plugins/inspect/plugin-oss4.xml docs/plugins/inspect/plugin-ossaudio.xml docs/plugins/inspect/plugin-png.xml docs/plugins/inspect/plugin-pulseaudio.xml docs/plugins/inspect/plugin-replaygain.xml docs/plugins/inspect/plugin-rtp.xml docs/plugins/inspect/plugin-rtsp.xml docs/plugins/inspect/plugin-shapewipe.xml docs/plugins/inspect/plugin-shout2send.xml docs/plugins/inspect/plugin-smpte.xml docs/plugins/inspect/plugin-soup.xml docs/plugins/inspect/plugin-spectrum.xml docs/plugins/inspect/plugin-speex.xml docs/plugins/inspect/plugin-taglib.xml docs/plugins/inspect/plugin-udp.xml docs/plugins/inspect/plugin-video4linux2.xml docs/plugins/inspect/plugin-videobox.xml docs/plugins/inspect/plugin-videocrop.xml docs/plugins/inspect/plugin-videofilter.xml docs/plugins/inspect/plugin-videomixer.xml docs/plugins/inspect/plugin-wavenc.xml docs/plugins/inspect/plugin-wavpack.xml docs/plugins/inspect/plugin-wavparse.xml docs/plugins/inspect/plugin-ximagesrc.xml docs/plugins/inspect/plugin-y4menc.xml win32/common/config.h 2011-06-17 10:37:33 +0100 Tim-Philipp Müller * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: sunaudio: fix typo in comment 2011-06-17 18:12:50 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-06-17 18:11:38 +0200 Wim Taymans * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: autodetect: fix caps 2011-06-16 15:38:10 +0200 Luis de Bethencourt * gst/goom/gstgoom.c: goom: fix unused-but-set-compiler warnings Remove unnecessary res variables, core checks existance and type of these fields for us already via the template caps, and we know that these fields exist because we've fixated them before in _negotiate(). 2011-06-17 03:07:09 +0300 Stefan Kost * gst/audiofx/audioecho.c: audioecho: fix param flags If the parameter cannot be changed in paused&playing, it is not controlable. Set the appropriate mutability flag instead. === release 0.10.30 === 2011-06-15 23:57:34 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.30 This is an ad-hoc release that is almost identical to 0.10.29: * work around GLib atomic ops API change * better handling of malformed buffers in RTP depayloders * some minor compilation fixes 2011-06-08 18:33:10 +0300 Raimo Järvi * gst/udp/gstudpnetutils.h: udp: Fix compiler warning on mingw-w64 Fixes: #652144. gstudpnetutils.h:32:0: error: "WINVER" redefined /usr/i686-w64-mingw32/sys-root/mingw/include/_mingw.h:231:0: note: this is the location of the previous definition 2011-06-04 13:49:52 -0700 David Schleef * gst/interleave/interleave.c: interleave: Work around changes in g_atomic API See #651514 for details. 2011-05-18 12:36:40 +0200 Jose Antonio Santos Cadenas * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpspeexdepay.c: rtp: Fix segmentation fault processing payload buffers This commit checks if the value returned by gst_rtp_buffer_get_payload_buffer and gst_rtp_buffer_get_payload_subbuffer is NULL before using it. 2011-05-16 09:04:31 +0200 Pino Toscano * ext/pulse/pulseutil.c: pulse: Define PATH_MAX if it isn't defined GNU Hurd for example doesn't define it. 2011-04-29 08:55:19 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Allow setcaps to be called after a format was negotiated if it's compatible Otherwise wavenc will fail if upstream decides to set equivalent caps or caps with additional information later. Thanks to Alexander Schremmer for finding this bug. 2011-06-15 15:06:23 +0100 Tim-Philipp Müller * REQUIREMENTS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-esdsink.xml: * ext/Makefile.am: * ext/esd/Makefile.am: * ext/esd/esdmon.c: * ext/esd/esdmon.h: * ext/esd/esdsink.c: * ext/esd/esdsink.h: * ext/esd/gstesd.c: * gst-plugins-good.spec.in: * m4/Makefile.am: * m4/as-arts.m4: * m4/esd.m4: * po/POTFILES.in: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Remove esound/esdsink plugin 2011-06-15 14:37:29 +0100 Tim-Philipp Müller * Makefile.am: * REQUIREMENTS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-gconfelements.xml: * ext/Makefile.am: * ext/gconf/Makefile.am: * ext/gconf/gstgconf.c: * ext/gconf/gstgconf.h: * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfaudiosrc.c: * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfelements.c: * ext/gconf/gstgconfelements.h: * ext/gconf/gstgconfvideosink.c: * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.c: * ext/gconf/gstgconfvideosrc.h: * ext/gconf/gstswitchsink.c: * ext/gconf/gstswitchsink.h: * ext/gconf/gstswitchsrc.c: * ext/gconf/gstswitchsrc.h: * gconf/.gitignore: * gconf/Makefile.am: * gconf/gstreamer.schemas.in: * gst-plugins-good.spec.in: * m4/Makefile.am: * m4/gconf-2.m4: * po/POTFILES.in: * tests/check/Makefile.am: Remove gconf elements and plugin GConf was deprecated in favour of GSettings etc. 2011-06-15 15:17:19 +0200 Wim Taymans * gst/audioparsers/gstflacparse.c: flacparse: fix unitialized access 2011-06-09 21:06:28 +0300 Stefan Kost * gst/matroska/matroska-read-common.c: matroska: add missing stdio include for sscanf 2011-06-13 19:08:38 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-06-13 17:51:40 +0200 Wim Taymans * gst/audiofx/audiopanorama.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: -good: port some more plugins 2011-06-13 17:14:51 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: fix for flush_stop API change 2011-06-13 17:14:00 +0200 Wim Taymans * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: rtp: port some more (de)payloader 2011-06-13 17:05:19 +0200 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparsers: not so greedy minimum frame size ... which will be determined by parsing anyway, and avoids introducing redundant additional latency. 2011-06-13 16:33:57 +0200 Wim Taymans * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsrc.c: -good: update for buffer API change 2011-06-13 16:33:46 +0200 Wim Taymans * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: rtp: port to 0.11 2011-06-13 13:25:49 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: rtp: fix for API changes in the base classes 2011-06-13 13:07:50 +0200 Wim Taymans * gst/avi/gstavimux.c: avimux: use caps event for negotiation 2011-06-13 13:07:27 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: fix for flush stop event changes 2011-06-08 18:33:10 +0300 Raimo Järvi * gst/udp/gstudpnetutils.h: udp: Fix compiler warning on mingw-w64 Fixes: #652144. gstudpnetutils.h:32:0: error: "WINVER" redefined /usr/i686-w64-mingw32/sys-root/mingw/include/_mingw.h:231:0: note: this is the location of the previous definition 2011-06-11 18:58:07 +0200 Wim Taymans * gst/goom/gstgoom.c: goom: fix for bufferpool update 2011-06-10 18:05:01 +0200 Wim Taymans * gst/goom/gstgoom.c: goom: update for alignment change 2011-06-09 17:56:18 +0200 Wim Taymans * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: port some more 2011-06-09 17:52:34 +0200 Wim Taymans * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtsp: port to 0.11 2011-06-09 17:50:08 +0200 Wim Taymans * gst/udp/gstudpsrc.c: udp: port to 0.11 2011-06-09 11:37:47 +0200 Wim Taymans * ext/aalib/gstaasink.c: aasink: register template and klass correctly 2011-06-09 10:50:44 +0200 Wim Taymans * gst/goom/gstgoom.c: * gst/goom/gstgoom.h: goom: port goom 2011-06-08 18:06:56 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-06-08 18:05:20 +0200 Wim Taymans * ext/aalib/gstaasink.c: assink: port aasink to 0.11 2011-06-07 12:06:08 +0200 Edward Hervey * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: debugutils: Switch from GST_BOILERPLATE to G_DEFINE_TYPE 2011-06-07 11:25:18 +0200 Edward Hervey * gst/videofilter/gstvideoflip.c: videofilter: Use new GstBaseTransform::transform_caps API 2011-06-07 11:23:55 +0200 Edward Hervey * gst/auparse/gstauparse.c: auparse: Don't use GST_BOILERPLATE 2011-06-07 11:22:35 +0200 Edward Hervey * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Buffers no longer have caps 2011-06-07 11:20:00 +0200 Edward Hervey * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: alpha: Use new transform_caps vmethod (with filter) 2011-06-06 20:43:31 +0200 Edward Hervey * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparsers: fix some more parsers 2011-06-06 18:21:04 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_chapters https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-06 14:47:27 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_attachments https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-06 12:43:14 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_attached_file https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-05 22:45:55 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_info https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-05 10:15:23 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_metadata https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-05 09:54:42 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_metadata_id_tag https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-05 02:24:41 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_metadata_id_simple_tag https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-06-06 12:42:53 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: reset state tracking variable when appropriate ... so we don't end up interrupting an operation that should not be interrupted based on the indication of a previous interruptable operation. 2011-06-04 13:49:52 -0700 David Schleef * gst/interleave/interleave.c: interleave: Work around changes in g_atomic API See #651514 for details. 2011-06-04 13:43:00 -0700 David Schleef * ext/soup/gstsouphttpsink.c: * ext/soup/gstsouphttpsink.h: souphttpsink: code cleanup 2011-06-05 02:00:08 +0530 Debarshi Ray * gst/matroska/matroska-parse.c: matroskaparse: Use ARTIST tag instead of AUTHOR for GST_TAG_ARTIST AUTHOR only existed in an old version of the spec and ARTIST is the new replacement for this. We are still reading both to still be compatible with old files. Fixes bug #644875. 2011-06-02 18:51:29 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: sys/ximage/ximageutil.c 2011-06-02 18:47:36 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: avi: port AVI elements to new API 2011-06-02 13:38:30 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: First query the peer duration in the requested format before converting to BYTES Fixes usage of dvdemux after another demuxer, e.g. mxfdemux. Fixes bug #650503. 2011-06-02 10:41:52 +0200 Sebastian Dröge * ext/soup/gstsouphttpsink.c: souphttpsink: Fix refcounting of the "session" property Properties should never take ownership of the values passed to them. 2011-06-01 17:04:27 -0700 David Schleef * gst/matroska/matroska-mux.c: matroskamux: For streaming files, push tags first 2011-05-24 14:52:01 -0700 David Schleef * ext/soup/Makefile.am: * ext/soup/gstsoup.c: * ext/soup/gstsouphttpsink.c: * ext/soup/gstsouphttpsink.h: * ext/soup/gstsouphttpsrc.c: soup: Add souphttpsink 2011-06-01 10:19:31 +0200 Thijs Vermeir * gst/udp/gstudpsrc.c: udpsrc: allow skip-first-bytes of full buffer size 2011-05-30 18:31:50 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_parse_header https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-30 12:09:31 +0200 Antonio Frediani * gst/isomp4/gstqtmux.c: qtmux: Use GST_TAG_IMAGE for coverart too Fixes bug #638107. 2011-05-30 10:40:08 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_get_seek_track - gst_matroska_{demux,parse}_reset_streams https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-28 22:04:34 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska{demux,parse}_found_global_tag https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-28 10:59:09 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_index_seek_find - gst_matroska{demux,parse}_do_index_seek https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-27 23:15:23 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_tracknumber_unique https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-27 20:28:19 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_decode_data https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-27 19:30:48 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_get_length https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-27 09:17:46 +0200 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Revert 1a90a6c4 and drop Dirac support again It does not work at all (A/V sync issues), is not very useful, other containers work much better with Dirac and Dirac in AVI is not supported by other software. Fixes bug #541215. 2011-05-26 23:35:52 +0530 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_encoding_cmp - gst_matroska_{demux,parse}_read_track_encodings https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_peek_id_length_pull - gst_matroska_{demux,parse}_peek_id_length_push https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_peek_adapter https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-26 12:48:36 +0200 Sebastian Dröge * sys/ximage/ximageutil.c: xvimagesink: Fallback to non-XShm mode if allocating the XShm image failed Fixes bug #630456. 2011-05-26 12:22:52 +0200 Marc Leeman * gst/rtp/gstrtpmp4vpay.c: rtpmp4vpay: Deprecated send-config property and replace by config-interval Fixes bug #622412. 2010-06-23 11:12:00 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: UTF-8 subtitles may have markup Fixes #616936. 2011-01-23 15:56:49 +0000 Vincent Penquerc'h * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: cairotextoverlay: forward new segment events from the sink to the source Not doing so will cause buffers to be received by downstream without a time base set. We use the same method avimux uses to get access to the event when collectpads got the sink event function. https://bugzilla.gnome.org/show_bug.cgi?id=640323 2011-01-24 11:11:48 +0000 Vincent Penquerc'h * ext/cairo/gsttextoverlay.c: textoverlay: forward source events to sinks Events are passed to the video sink, and to the text sink if it is linked. This will allow seeking, for instance. https://bugzilla.gnome.org/show_bug.cgi?id=586450 2011-05-25 21:12:12 +0200 David Hoyt * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: Add property to assume a single stream and emit no-more-pads Fixes bug #616686. 2011-05-25 14:50:26 +0200 Miguel Angel Cabrera Moya * gst/rtsp/gstrtspsrc.c: rtspsrc: uniform unknown message handling Do the same processing in all the cases when an unknown message is received. That is, give a warning. https://bugzilla.gnome.org/show_bug.cgi?id=651059 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_peek_pull https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following function to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_peek_bytes https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_{demux,parse}_encoding_order_unique - gst_matroska_{demux,parse}_read_track_encoding https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-24 18:27:10 +0200 Wim Taymans * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: autodetect: port to new API 2011-05-24 17:34:19 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/avi/gstavidemux.c gst/rtp/gstrtpac3depay.c gst/rtp/gstrtpg726depay.c gst/rtp/gstrtpmpvdepay.c gst/videofilter/gstgamma.c 2011-05-24 13:12:19 +0200 Mark Nauwelaerts * gst/rtp/gstrtppcmudepay.c: pcmudepay: allow variable sample rate 2011-05-24 13:11:54 +0200 Mark Nauwelaerts * gst/rtp/gstrtppcmadepay.c: pcmadepay: allow variable sample rate 2010-04-04 06:43:41 -0500 Rob Clark * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/v4l2_calls.c: v4l2: add norm property Based on a patch by Guennadi Liakhovetski. v2: updates because I forgot to add GstTuner interface to v4l2sink v3: update to add all possible values to norm enum 2011-05-23 20:46:04 +0300 Debarshi Ray * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: fixed copyright headers https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Move the following functions to matroska-read-common.[ch] from matroska-demux.c and matroska-parse.c: - gst_matroska_decode_content_encodings - gst_matroska_decompress_data https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 18:48:57 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.h: matroska: move GstMatroska{Demux,Parse}::state to GstMatroskaReadCommon https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-24 09:48:56 +0200 Jonas Larsson * gst/isomp4/qtdemux.c: qtdemux: Fix buffer leak with corrupted files Fixes bug #650912. 2011-05-23 02:46:38 -0700 Miguel Angel Cabrera Moya * gst/deinterlace/gstdeinterlace.c: deinterlace: fix parameter type in trace https://bugzilla.gnome.org/show_bug.cgi?id=650937 2011-05-23 18:06:44 +0300 Debarshi Ray * gst/matroska/Makefile.am: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: matroska: refactor code common to matroskademux and matroskaparse Replace the following functions with their gst_matroska_read_common_* counterparts: - gst_matroska_{demux,parse}_parse_index - gst_matroska_{demux,parse}_parse_skip - gst_matroska_{demux,parse}_stream_from_num Introduce GstMatroskaReadCommon to contain those members of GstMatroskaDemux and GstMatroskaParse that were used by the above functions. https://bugzilla.gnome.org/show_bug.cgi?id=650877 2011-05-23 13:50:46 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: tell baseparse the duration in samples for better accuracy Tell GstBaseParse the duration in samples instead of time, so that a duration query in DEFAULT format will return the correct number of samples without rounding errors. Baseparse will convert this into time itself when needed. https://bugzilla.gnome.org/show_bug.cgi?id=650785 2011-05-23 13:25:44 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: also try upstream first for duration query in DEFAULT format https://bugzilla.gnome.org/show_bug.cgi?id=650785 2011-05-23 13:23:21 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: make conversion from TIME to DEFAULT format (samples) work Fix copy'n'paste error in the previous commit. 2011-05-23 11:36:36 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Implement conversions between TIME and DEFAULT format Fixes bug #650785. 2011-05-22 18:50:51 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: don't error out on invalid minimum_blocksize value in streaminfo header We don't use it, so may just as well accept an invalid value of 0 here, which is likely inconsequential anyway. https://bugzilla.gnome.org/show_bug.cgi?id=650691 2011-05-20 10:34:47 +0300 Stefan Kost * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpqcelpdepay.c: rtp: fix static array overruns in a nicer way Use G_N_ELEMENTS instead of hard-coding the array size. 2011-05-20 00:53:44 +0300 Stefan Kost * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpqcelpdepay.c: rtp: fix static array overruns Yes array[10] has elements from 0...9. 2011-05-19 23:31:19 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: docs: update plugin introspection data Now more files are merged and produced in a canonical fashion, which hopefully creates less or no delta in the future. 2011-05-19 22:57:15 +0300 Stefan Kost * common: Automatic update of common submodule From 9e5bbd5 to 69b981f 2011-05-19 18:21:33 +0300 Stefan Kost * gst/isomp4/qtdemux.c: qtdemux: add missing break 2010-11-08 14:06:15 +0100 Robert Swain * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Add support for deinterlacing using buffer caps/flags When not using the fieldanalysis element immediately upstream of deinterlace, behaviour should remain unchanged. fieldanalysis will set the caps and flags on the buffers such that they can be interpreted and acted upon to produce progressive output. There are two main modes of operation: - Passive pattern locking Passive pattern locking is a non-blocking, low-latency mode of operation that is suitable for close-to-live usage. Initially a telecine stream will be output as variable framerate with naïve timestamp adjustment. With each incoming buffer, an attempt is made to lock onto a pattern. When a lock is obtained, the src pad and output buffer caps will reflect the pattern and timestamps will be accurately interpolated between pattern repeats. This means that initially and at pattern transitions there will be short periods of inaccurate timestamping. - Active pattern locking Active pattern locking is a blocking, high-latency mode of operation that is targeted at use-cases where timestamp accuracy is paramount. Buffers will be queued until enough are present to make a lock. When locked, timestamps will be accurately interpolated between pattern repeats. Orphan fields can be dropped or deinterlaced. If no lock can be obtained, a single field might be pushed through to be deinterlaced. Locking can also be disabled or 'auto' chooses between passive and active locking modes depending on whether upstream is live. 2011-05-10 16:25:40 -0700 David Schleef * configure.ac: configure: Remove config script check for caca 2011-05-18 12:36:40 +0200 Jose Antonio Santos Cadenas * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpspeexdepay.c: rtp: Fix segmentation fault processing payload buffers This commit checks if the value returned by gst_rtp_buffer_get_payload_buffer and gst_rtp_buffer_get_payload_subbuffer is NULL before using it. 2011-05-18 14:49:17 +0200 Sebastian Dröge * ext/lame/Makefile.am: * ext/lame/gstlamemp3enc.c: lamemp3enc: Post CODEC and BITRATE tags Also filter any CODEC/AUDIO_CODEC tags from incoming tag events. Fixes bug #391543. 2011-05-18 16:10:07 +0300 Stefan Kost * common: Automatic update of common submodule From fd35073 to 9e5bbd5 2011-05-18 12:52:31 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: ensure 0-padding when correcting dubious list size 2011-05-18 12:24:25 +0300 Stefan Kost * common: Automatic update of common submodule From 46dfcea to fd35073 2011-05-18 10:22:27 +0300 Stefan Kost * gst/rtsp/gstrtspsrc.c: rtspsrc: use EINVAL for missing url parameter Fixes gcc warning about using uninitialized variable 'res'. 2011-04-28 15:37:40 +0300 Stefan Kost * gst/debugutils/rndbuffersize.c: * gst/videofilter/gstgamma.c: various: fix author tag in element details 2011-04-20 15:25:58 -0400 Chris E Jones * gst/auparse/gstauparse.c: auparse: implement seeking Implement seeking and seeking query. Fixes #644512 2011-05-17 16:13:59 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-04-06 16:05:55 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: also allow PAUSE to be interrupted ... as it is on the way out to NULL. See #632504. 2011-04-06 15:51:49 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: ensure proper closing and cleanup ... since the TEARDOWN sequence might not have had a chance to even start, but at least connections should be closed (synchronously) and state cleaned up. See #632504. 2011-04-06 15:49:01 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: fix and improve async handling Simplify the command handling; passing a command to thread means we really want it to get the message, which means to always flush provided the command can handle being interrupted. Command thread indicates whether command allows interruption and ensure non-flushing connection as it subsequently needs it. In particular, this also makes the TEARDOWN sequence interruptable and also prevents races where _loop_ could miss a command and would continue receiving (or at least trying to). See #632504. 2011-04-06 14:53:27 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: tweak post-seek loop handling 2011-01-10 12:46:37 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: open on play and pause when not done yet With the async state changes, it is possible that we need to open the stream before play and pause. Also make sure we remember a previous open failure so that we don't keep trying again. 2011-01-10 11:45:03 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: improve async handling Simplify the command handling, only continue looping when we have not received another command or when the previous loop was successfull. Avoid looping on a disconnected socket. 2011-01-07 18:02:49 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: rework reconnect code Use the same async code path to implement reconnects. Make sure we only post progress messages when doing async things. 2011-01-07 17:19:59 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: small cleanups Make sure we cancel the previous task when queuing a new one. Move the messages to a central place so we can more easily post them. 2011-01-07 15:15:49 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't post errors when interrupting 2011-01-07 13:43:06 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: implement more async handling Remove some old locks. Make sure we never go into the loop function when flushing. 2011-01-07 11:40:32 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: first attempt at async implementation 2011-01-07 11:40:11 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.h: rtspsrc: small header cleanups 2011-05-17 10:47:32 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpssrcdemux.c: ssrcdemux: Fix uninitialized variable compiler warning for (pre-) releases too 2011-04-28 15:57:04 +0200 Edward Hervey * sys/v4l2/gstv4l2object.c: v4l2objects: Only allow mpeg-ts on source objects Ugly fix for #648312 2011-05-17 09:24:08 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Fix uninitialized variable compiler warning 2011-05-06 19:09:17 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: ssrcdemux: Implement iterate internal links for sink pads https://bugzilla.gnome.org/show_bug.cgi?id=649617 2011-05-06 18:41:01 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: iterate pad function is only valid for src pads The iterate function is only used for src pads, so mark it as such and remove dead code. https://bugzilla.gnome.org/show_bug.cgi?id=649617 2011-05-06 18:12:53 -0400 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Release lock before emitting signal If the lock is not released before emitting a signal, it may cause a deadlock if any other function in the element is called. Also removed an unused timestamp parameter https://bugzilla.gnome.org/show_bug.cgi?id=649617 2011-05-15 23:25:15 +0300 Debarshi Ray * gst/matroska/matroska-parse.c: matroskaparse: calculate segment duration after parsing all the IDs Since the segment duration is given in terms of the GST_MATROSKA_ID_TIMECODESCALE we should only convert it into nanoseconds when we are sure that any scale specified in the file has been read. https://bugzilla.gnome.org/show_bug.cgi?id=650258 2011-05-16 17:52:11 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: configure.ac 2011-05-16 17:50:15 +0200 Wim Taymans * ext/pulse/pulsesrc.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: -good: fix for new API 2011-05-04 11:55:21 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: additional lock safety Fixes #619590. 2011-04-26 16:06:56 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: also check for bitrate info in caps 2010-05-25 01:04:43 +0530 Arun Raghavan * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: guess bitrate if only one stream's bitrate is unknown If the bitrates for all but one audio/video streams are known, and the total stream size and duration can be determined, this calculates the unkown bitrate as (stream size / duration) - (sum of known bitrates). While this is not guaranteed to be very accurate, it should be good enough for most purposes. For example, this is useful for H.263 + AAC streams where no 'btrt' atom is available for the video portion. https://bugzilla.gnome.org/show_bug.cgi?id=619548 2010-05-31 23:59:59 +0530 Arun Raghavan * gst/isomp4/qtdemux.c: qtdemux: Export max bitrate for AMR-NB/-WB streams This parses the 'damr' atom if present, and exports the maximum bitrate of the stream using the mode set field to determine the highest bitrate frame type that might be present. https://bugzilla.gnome.org/show_bug.cgi?id=620186 2011-05-16 09:04:31 +0200 Pino Toscano * ext/pulse/pulseutil.c: pulse: Define PATH_MAX if it isn't defined GNU Hurd for example doesn't define it. 2011-05-15 23:25:15 +0300 Debarshi Ray * gst/matroska/matroska-demux.c: matroskademux: calculate segment duration after parsing all the IDs Since the segment duration is given in terms of the GST_MATROSKA_ID_TIMECODESCALE we should only convert it into nanoseconds when we are sure that any scale specified in the file has been read. https://bugzilla.gnome.org/show_bug.cgi?id=650258 2011-05-09 19:00:45 +0200 Andoni Morales Alastruey * gst/flv/gstflvmux.c: flvmux: Add support for mpegversion 2, which is also AAC 2011-05-11 10:25:15 +0200 Sebastian Dröge * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: Send EOS when seeking after the end of file instead of failing Fixes bug #649780. 2011-04-29 08:59:20 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Set fixedcaps getcaps function on the sinkpad wavenc does not allow to change the caps during playback and always returning the template caps is just wrong. 2011-04-29 08:55:19 +0200 Sebastian Dröge * gst/wavenc/gstwavenc.c: wavenc: Allow setcaps to be called after a format was negotiated if it's compatible Otherwise wavenc will fail if upstream decides to set equivalent caps or caps with additional information later. Thanks to Alexander Schremmer for finding this bug. 2011-05-14 10:02:22 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development === release 0.10.29 === 2011-05-10 10:04:28 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: Release 0.10.29 Highlights: - amrparse, aacparse, ac3parse, flacparse, mpegaudioparse, dcaparse audio parsers (moved from -bad) - muxers now mux based on running time - ISO MP4 muxers: mp4mux/3gppmux/qtmux/mj2mux (moved from -bad) - new matroskaparse element - new v4l2radio element - rtpsession: support RTCP Early Feedback (the AVPF profile) - orc 0.4.14 or newer recommended - many other fixes and improvements 2011-05-05 13:24:23 +0200 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Fix signed floating point values writing You would end up on some architectures with 0 being written out instead of the proper value. https://bugzilla.gnome.org/show_bug.cgi?id=649449 2011-05-04 12:04:15 +0200 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: avoid building index when streamable ... as it will not be written anyway. Fixes #648937 (?). 2011-05-02 12:09:02 +0100 Tim-Philipp Müller * Makefile.am: build: add old qtdemux/quicktime directories to CRUFT_DIRS and CRUFT_FILES 2011-05-01 00:04:03 -0400 Tom Janiszewski * gst/flv/gstflvmux.c: flvmux: don't overwrite metadata tag with duration in streaming mode A duration tag gets inserted only for streamable=false, so only update/write the duration later if we actually inserted that tag, otherwise we write garbage into other tags. https://bugzilla.gnome.org/show_bug.cgi?id=649060 2011-04-30 18:16:36 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * po/fr.po: * win32/common/config.h: 0.10.28.4 pre-release 2011-04-30 17:46:36 +0100 Tim-Philipp Müller * Android.mk: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect/plugin-isomp4.xml: * gst-plugins-good.spec.in: * gst/isomp4/LEGAL: * gst/isomp4/Makefile.am: * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/atomsrecovery.c: * gst/isomp4/atomsrecovery.h: * gst/isomp4/descriptors.c: * gst/isomp4/descriptors.h: * gst/isomp4/fourcc.h: * gst/isomp4/ftypcc.h: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/gstqtmoovrecover.h: * gst/isomp4/gstqtmux-doc.c: * gst/isomp4/gstqtmux-doc.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: * gst/isomp4/gstqtmuxmap.c: * gst/isomp4/gstqtmuxmap.h: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/gstrtpxqtdepay.h: * gst/isomp4/isomp4-plugin.c: * gst/isomp4/properties.c: * gst/isomp4/properties.h: * gst/isomp4/qtatomparser.h: * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux.vcproj: * gst/isomp4/qtdemux_dump.c: * gst/isomp4/qtdemux_dump.h: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_lang.c: * gst/isomp4/qtdemux_lang.h: * gst/isomp4/qtdemux_types.c: * gst/isomp4/qtdemux_types.h: * gst/isomp4/qtpalette.h: * po/POTFILES.in: quicktime: rename plugin to isomp4 https://bugzilla.gnome.org/show_bug.cgi?id=648004 2011-04-29 17:55:28 +0200 Wim Taymans * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: audioparsers: fix some parsers 2011-04-29 17:54:42 +0200 Wim Taymans * configure.ac: fix error caused by merging 2011-04-29 15:49:41 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: configure.ac gst/rtp/gstrtpgstpay.c 2011-04-29 15:46:21 +0200 Wim Taymans * gst/audiofx/audiofxbasefirfilter.c: audiofx: fix pad_alloc 2011-04-27 12:45:51 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * po/bg.po: * po/ja.po: * po/nl.po: * po/ru.po: * win32/common/config.h: 0.10.28.3 pre-release 2011-04-26 15:58:12 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: fix buffer leak 2011-04-26 15:58:12 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: fix buffer leak 2011-04-26 15:42:47 +0200 Wim Taymans * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: port jack elements 2011-04-25 10:04:52 +0200 Philip Jägenstedt * ext/jpeg/gstjpegdec.c: jpegdec: documentation typo "jpegddec" https://bugzilla.gnome.org/show_bug.cgi?id=648589 2011-04-25 18:14:45 +0200 Wim Taymans * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: rtp: port some more elements 2011-04-25 17:27:40 +0200 Wim Taymans * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: rtp: port more to 0.11 2011-04-25 13:16:58 +0200 Wim Taymans * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: rtp: port some more (de)payloaders 2011-04-25 12:49:36 +0200 Wim Taymans * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: port some more elements to 0.11 2011-04-25 11:38:28 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-04-24 16:45:07 -0700 David Schleef * gst/avi/gstavimux.c: * gst/matroska/matroska-mux.c: avimux,matroskamux: Add stream-format to h264 caps Fixes #606662. 2011-02-20 12:13:49 -0800 David Schleef * ext/libpng/gstpngdec.c: pngdec: Remove temporary code Now that we depend on (what will be) -base-0.10.33. 2011-04-24 14:03:56 +0100 Tim-Philipp Müller * configure.ac: configure: don't pass -Waddress to ObjC compiler on OSX when compiling osxvideosink Temporary workaround until we fix this properly and check for the ObjC warning/error flags instead of just passing CFLAGS to the ObjC compiler. https://bugzilla.gnome.org/show_bug.cgi?id=643939 2011-04-24 13:29:32 +0100 Tim-Philipp Müller * docs/plugins/inspect/plugin-quicktime.xml: * gst-plugins-good.spec.in: * gst/quicktime/Makefile.am: quicktime: rename plugin filename from *qtdemux* to *quicktime* https://bugzilla.gnome.org/show_bug.cgi?id=648004 2011-04-24 14:03:41 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From c3cafe1 to 46dfcea 2011-04-21 23:30:26 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/quicktime/Makefile.am: * gst/quicktime/gstqtmoovrecover.c: * gst/quicktime/gstqtmux-doc.c: * gst/quicktime/gstqtmux-doc.h: docs: add various qtmux variants to documentation 2011-04-21 22:51:52 +0100 Tim-Philipp Müller * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: quicktime: register 3gppmux element in addition to the misnamed gppmux 2011-04-18 18:08:30 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Remove incomplete support for RTCP FIR Remove bits that were meant to suppport RTCP FIR https://bugzilla.gnome.org/show_bug.cgi?id=648160 2011-04-19 18:55:31 +0200 Wim Taymans * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: flac: port to 0.11 2011-04-19 17:35:47 +0200 Wim Taymans * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: use G_DEFINE_TYPE some more 2011-04-19 17:20:19 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: avi: use G_DEFINE_TYPE 2011-04-19 17:07:18 +0200 Wim Taymans * ext/pulse/pulsemixer.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: use G_DEFINE_TYPE 2011-04-19 16:25:28 +0200 Wim Taymans Merge branch 'master' into 0.11 2011-04-19 14:33:25 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/generic/.gitignore: * tests/check/generic/index.c: tests: add generic set_index test 2011-04-19 14:33:42 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: fix deadlock on setting index on flvdemux 2011-04-19 14:16:11 +0100 Tim-Philipp Müller * tests/check/elements/flacparse.c: tests: add index-setting test for baseparse/flacparse https://bugzilla.gnome.org/show_bug.cgi?id=646811 2011-04-18 11:29:15 +0200 Sebastian Dröge * tests/check/pipelines/wavpack.c: wavpack: Remove bus GSource to prevent a valgrind warning 2011-04-18 11:14:32 +0200 Sebastian Dröge * tests/check/pipelines/wavenc.c: wavenc: Remove bus GSource to prevent a valgrind warning 2011-04-18 11:11:53 +0200 Sebastian Dröge * tests/check/pipelines/tagschecking.c: tagschecking: Remove bus GSource to prevent a valgrind warning 2011-04-18 11:10:01 +0200 Sebastian Dröge * tests/check/elements/imagefreeze.c: imagefreeze: Remove bus GSource to prevent a valgrind warning 2011-04-18 10:54:43 +0200 Wim Taymans * gst/audiofx/audiopanorama.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: port more plugins to 0.11 2011-04-18 10:23:45 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: android/apetag.mk android/avi.mk android/flv.mk android/icydemux.mk android/id3demux.mk android/qtdemux.mk android/rtp.mk android/rtpmanager.mk android/rtsp.mk android/soup.mk android/udp.mk android/wavenc.mk android/wavparse.mk configure.ac 2011-04-17 01:29:01 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: fix 'variable may be used uninitialized' warnings caused by -DG_DISABLE_ASSERT 2011-04-16 18:50:11 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: * win32/common/gstrtpbin-marshal.c: * win32/common/gstrtpbin-marshal.h: 0.10.28.2 pre-release 2011-04-16 18:49:27 +0100 Tim-Philipp Müller * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: gst: update disted orc backup code 2011-04-16 18:29:45 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update for pre-release 2011-04-16 18:27:54 +0100 Tim-Philipp Müller * po/bg.po: * po/cs.po: * po/de.po: * po/es.po: * po/id.po: * po/sl.po: po: update translations 2011-04-16 18:17:01 +0100 Tim-Philipp Müller * gst/quicktime/gstqtmux.c: qtmux: refuse incomplete legacy h264 caps Refuse h264 caps without stream-format and codec_data fields for now, to avoid creating broken files. This might cause some pipelines that worked previously to fail. However, the move from -bad to -good is our only chance to fix this up, so make it strict for now. We can always change it back to be less strict in future. https://bugzilla.gnome.org/show_bug.cgi?id=647919 2011-04-16 18:16:11 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2sink.c: v4l2sink: fix another unused-but-set-variable warning 2011-04-16 18:10:24 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/speex/gstspeexenc.c: * gst/rtp/gstrtpgsmpay.c: pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling Don't use g_assert() for error handling, even if they're highly unlikely. Either we *know* that something can't happen, in which case we should just not handle it, or we think something can happen, but it is very very unlikely that it will ever happen, in which case we should handle it like any other error instead of asserting. g_assert() is best left for conditions we have control of, like checking internal consistency of our code, not checking return values of external code. Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT: gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer': gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used gstspeexenc.c: In function 'gst_speex_enc_encode': gstspeexenc.c:904:19: warning: variable 'written' set but not used pulsesink.c: In function 'gst_pulsesink_change_state': pulsesink.c:2725:9: warning: variable 'res' set but not used pulsesrc.c: In function 'gst_pulsesrc_change_state': pulsesrc.c:1253:7: warning: variable 'e' set but not used 2011-04-16 18:07:35 +0100 Tim-Philipp Müller * tests/examples/rtp/server-alsasrc-PCMA.c: examples: fix some warnings in rtp example Caused by -DG_DISABLE_ASSERT 2011-04-16 17:57:32 +0100 Tim-Philipp Müller * tests/examples/level/level-example.c: examples: don't put code with side-effects into g_assert() Otherwise things won't work too well when compiling with -DG_DISABLE_ASSERT (as we do for pre-releases and releases). 2011-04-16 16:51:32 +0100 Tim-Philipp Müller * gst/deinterlace/tvtime/greedyh.c: * gst/matroska/matroska-mux.c: deinterlace, matroska: fix two variable-may-be-used-uninitialized compiler warnings We use -DG_DISABLE_ASSERT for the pre-releases, which makes these warnings pop up in cases that were previously covered by g_assert_not_reached() and the like: tvtime/greedyh.c:801:14: warning: 'scanline' may be used uninitialized in this function matroska-mux.c:501:19: warning: 'context' may be used uninitialized in this function 2011-04-16 14:45:25 +0200 Sebastian Dröge * gst/apetag/gstapedemux.c: apedemux: Port to 0.11 2011-04-16 13:33:45 +0100 Tim-Philipp Müller * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: fix unused-but-set-variable warnings with gcc-4.6 2011-04-16 13:23:50 +0100 Tim-Philipp Müller * tests/examples/cairo/cairo_overlay.c: examples: fix 'control reaches end of non-void function' warning in cairo example 2011-04-15 15:47:24 +0200 Robert Swain * sys/v4l2/gstv4l2src.c: v4l2src: Address unused but set variable The v4l2object formats list was being obtained into a local variable and then still used from the context. Make use of the local variable. 2011-04-15 15:17:34 +0200 Robert Swain * sys/oss4/oss4-mixer-slider.c: * sys/oss4/oss4-mixer-switch.c: * sys/oss4/oss4-property-probe.c: * sys/oss4/oss4-source.c: oss4: Address unused but set variables GCC 4.6.x complains about such variable usage. Unused but set variables were removed except that gst_oss4_mixer_slider_set_mute () now returns the value from the call to gst_oss4_mixer_set_control_val (). 2011-04-15 15:14:13 +0200 Robert Swain * ext/jpeg/gstjpegenc.c: * ext/pulse/pulsesink.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: jpegenc: pulsesink: raw1394: Address unused but set variables GCC 4.6.x spits warnings about such usage of variables. The variables in raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate. The others were removed. 2011-04-15 15:12:44 +0200 Robert Swain * gst/shapewipe/gstshapewipe.c: * gst/y4m/gsty4mencode.c: y4mencode: shapewipe: Address unused but set variables GCC 4.6.x complains about such usage. 2011-04-15 15:11:35 +0200 Robert Swain * tests/check/elements/deinterlace.c: * tests/check/elements/rtp-payloading.c: * tests/check/pipelines/flacdec.c: * tests/examples/level/level-example.c: * tests/icles/videocrop-test.c: * tests/icles/ximagesrc-test.c: tests: Address unused but set variables GCC 4.6.x spits warnings about such usage of variables. 2011-04-15 15:36:41 +0200 Robert Swain * gst/videomixer/blendorc.orc: videomixer: Fix argb/rgba overlay orc code Remove some redundant operations (convubw) and use the correct variable, t2, in the orc_overlay_bgra function. 2011-04-15 15:33:35 +0200 Robert Swain * gst/videomixer/blend.c: * gst/videomixer/gstcollectpads2.c: * gst/videomixer/videomixer2.c: videomixer: address unused but set variables GCC 4.6.x spits warnings about variables that are set but unused. Such variables have been removed in blend, collectpads2 and videomixer2. 2011-04-15 14:57:20 +0200 Robert Swain * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: rtp, rtpmanager: Address unused but set variables GCC 4.6.x spits warnings about variables that are unused but set. Such variables have been removed where trivial but with comments left behind for informational purposes in some cases. gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4 to always return GST_FLOW_OK instead of the return value of rtp_session_process_rtcp (), so we'll keep it that way. 2011-04-15 11:29:30 +0200 Robert Swain * gst/quicktime/descriptors.c: * gst/quicktime/gstrtpxqtdepay.c: * gst/quicktime/qtdemux.c: quicktime: Remove unused but set variables GCC 4.6.x spits warnings about such variable usage. Note that some calculations are left as comments for informative purposes. 2011-04-15 11:23:38 +0200 Robert Swain * gst/matroska/matroska-demux.c: * gst/matroska/matroska-parse.c: matroska: Remove unused but set variables GCC 4.6.x spits warnings about such variable usage. 2011-04-15 11:19:26 +0200 Robert Swain * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Remove unused but set duration variable GCC 4.6.x spits warnings about such variable usage. 2011-04-15 11:18:19 +0200 Robert Swain * gst/flv/gstflvdemux.c: flxdemux: Remove unused but set keyframe variables The FIXMEs about the keyframe flag never being used are left for later fixing, at which point the keyframe variables could be added back. 2011-04-15 11:16:42 +0200 Robert Swain * gst/effectv/gstedge.c: edgetv: Remove unused but set height variable GCC 4.6.x spits warnings about such variables. 2011-04-15 18:51:20 +0100 Tim-Philipp Müller * gst/audioparsers/gstflacparse.c: flacparse: update for gst_base_parse_frame_init() API change 2011-02-01 15:57:01 -0500 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Use existing functions to parse RTCP FB packets Use existing functions to get the FCI from FB packets. https://bugzilla.gnome.org/show_bug.cgi?id=622553 2011-02-01 16:23:52 -0500 Olivier Crête * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/rtpsession.c: rtpsession: marshal GstBuffer as a MiniObject instead of a pointer https://bugzilla.gnome.org/show_bug.cgi?id=622553 2011-04-14 23:24:56 -0700 David Schleef * gst/matroska/matroska-demux.c: matroskademux: Better calculation of framerate https://bugzilla.gnome.org/show_bug.cgi?id=647833 2011-04-13 12:37:09 +0100 Tim-Philipp Müller * gst/quicktime/gstqtmux.c: qtmux: default to dts-method=reorder and presentation-time=true https://bugzilla.gnome.org/show_bug.cgi?id=636699 2011-04-15 12:47:52 +0200 Mark Nauwelaerts * tests/check/elements/qtmux.c: tests: qtmux: test various dts-methods 2011-04-15 12:34:05 +0200 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: fix corner case buffer handling for reorder method 2011-04-14 13:47:05 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Don't leak the SEEKING query 2011-04-14 13:43:06 +0200 Sebastian Dröge * gst/quicktime/gstqtmoovrecover.c: * gst/quicktime/gstqtmoovrecover.h: qtmoovrecover: Don't leak the static recursive mutex 2011-04-14 13:37:52 +0200 Sebastian Dröge * sys/v4l2/gstv4l2radio.c: v4l2radio: Free videodev string before replacing it 2011-04-14 13:24:21 +0200 Sebastian Dröge * gst/matroska/matroska-parse.c: matroskaparse: Allow webm and matroska caps and don't leak caps 2011-04-14 07:35:29 +0100 Christian Fredrik Kalager Schaller * gst-plugins-good.spec.in: Add parser plugin 2011-04-13 21:58:36 -0400 Olivier Crête * gst/dtmf/Makefile.am: * gst/dtmf/gstdtmfcommon.h: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: dtmf: Move duplicate #defines into a common include Centralize duplicated constants so they have the same value. Also standardise minimum tone duration to 250ms and minimum inter-tone interval to 100ms. 2011-03-24 14:34:24 -0700 David Schleef * sys/directsound/gstdirectsoundsink.c: directsoundsink: Add conditionals on WAVE_FORMAT_DOLBY_AC3_SPDIF 2011-04-11 20:09:14 +0100 Tim-Philipp Müller * gst/debugutils/gstcapsdebug.c: capsdebug: fix unused-but-set-variable warnings with gcc 4.6 2011-04-11 20:05:54 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: fix unused-but-set-variable warning with gcc 4.6 Most likely a leftover from when the index parsing code was rewritten. 2011-04-11 19:54:00 +0100 Tim-Philipp Müller * gst/audioparsers/gstac3parse.c: ac3parse: fix unused-but-set-variable warning with gcc 4.6 2011-04-11 19:50:07 +0100 Tim-Philipp Müller * gst/videofilter/gstvideobalance.c: videobalance: fix handling of YUV images with 'odd' widths Fixes unused-but-set-variable warnings with gcc 4.6. 2011-04-11 19:49:22 +0100 Tim-Philipp Müller * gst/videofilter/gstvideoflip.c: videoflip: fix unused-but-set-variable warnings with gcc 4.6 2011-04-13 18:11:34 +0200 Sebastian Dröge * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: audiowsinc{band,limit}: Fix check for divison by zero 2011-04-13 18:01:01 +0200 Sebastian Dröge * gst/audiofx/audiowsincband.c: audiowsincband: Fix range of kernel elements (lim -> lim-1) 2011-04-13 18:00:44 +0200 Sebastian Dröge * gst/audiofx/audiowsinclimit.c: audiowsinclimit: Add some more braces to make the code more readable 2011-04-11 18:40:30 -0500 Jordi Burguet-Castell * gst/audiofx/audiowsinclimit.c: audiowsinclimit: Fix range of kernel elements (lim -> lim-1) in high/low-pass filters 2011-04-13 17:49:22 +0200 Sebastian Dröge * gst/audiofx/audiowsincband.c: audiowsincband: Add new windowing functions: gaussian, cos and hann 2011-04-11 18:41:43 -0500 Jordi Burguet-Castell * gst/audiofx/audiowsinclimit.c: audiowsinclimimt: Add new windows to high/low-pass filters: gaussian, cosine, hann 2011-04-13 16:47:05 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: set stream-format=byte-stream on h264 caps if there's no codec data https://bugzilla.gnome.org/show_bug.cgi?id=606662 2011-04-13 16:37:07 +0100 Thiago Santos * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: restrict h264 some more to only accept AU-aligned AVC https://bugzilla.gnome.org/show_bug.cgi?id=606662 2011-04-13 17:11:26 +0200 Sebastian Dröge * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: The VBRI header is always at offset 0x20, independent of MPEG version Also clean up advancing of the data pointer a bit. Fixes bug #647659. 2011-04-13 15:18:11 +0100 Tim-Philipp Müller * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: * tests/check/Makefile.am: * tests/check/elements/qtmux.c: qtmux: add variant-less video/quicktime to source pad template caps This is needed for automatic transcoding using encodebin. Our typefinder does not always add a variant to the found caps, and encodebin needs an *exact* match to the caps on the source pad template, so we need to add the variant-less video/quicktime caps to the template as well for encodebin to be able to find it. Add unit test for this as well. https://bugzilla.gnome.org/show_bug.cgi?id=642879 2011-04-13 16:17:41 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Properly interprete the result of strcmp() 2011-04-13 16:09:04 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Don't store image tags inside the vorbiscomments and the flac metadata Instead only store them inside the flac metadata. There's no point in storing them twice and the flac metadata is still the official way to store image tags inside flac. 2011-04-13 12:38:15 +0100 Tim-Philipp Müller * tests/check/elements/.gitignore: * tests/check/pipelines/.gitignore: tests: ignore new qtmux-related test binaries 2011-04-13 11:25:11 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-quicktime.xml: * gst/quicktime/Makefile.am: * gst/quicktime/gstqtmuxplugin.c: * gst/quicktime/quicktime.c: * tests/check/Makefile.am: quicktime: move qtmux plugin from -bad to -good https://bugzilla.gnome.org/show_bug.cgi?id=636699 2011-04-12 16:42:17 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: Remove leftover MAEMO_BROKEN defines Remove defines to work around bugs in old Maemo releases 2011-04-04 12:21:23 +0200 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: more helpful debug error message when no needed duration on input buffers Fixes #646256. 2011-03-21 10:56:51 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: qtmux: Adding GstTagXmpWriter interface Adds GstTagXmpWriter interface support to qtmux 2011-03-22 20:53:08 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: use running time for synchronization See also #432612. 2011-03-10 16:03:58 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: provide for PTS metadata when so configured ... and not only when sort-of feeling like it. In any case, if it turns out all really is in order, and presumably DTS == PTS, then no ctts will be produced anyway. 2011-03-10 16:02:42 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: also track original PTS buffer timestamp in reorder dts-method 2011-02-21 12:14:59 +0100 Edward Hervey * gst/quicktime/gstqtmux.c: Revert "Check that collectpads exists before removing pad" This reverts commit 6d8740476ccd3a3498dc4f18c19733643825c7b8. Depends on a core commit that was reverted 2011-02-20 23:57:19 -0800 David Schleef * gst/quicktime/gstqtmux.c: Check that collectpads exists before removing pad The core now calls release pad from finalize, at which point the collectpads might have already been freed. 2011-01-13 11:28:32 -0300 Thiago Santos * tests/check/elements/qtmux.c: test: qtmux: Tests qtmux reuse Forces the use of qtmux after it has been put to PLAYING and back to NULL once https://bugzilla.gnome.org/show_bug.cgi?id=639338 2011-01-13 15:27:36 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: set src pads when starting file ... rather than at _init time, so they are also available following a pad (de)activation cycle. https://bugzilla.gnome.org/show_bug.cgi?id=639338 2011-01-03 17:24:23 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: adjust nasty case timestamp tracking That is, all sorts of problems arise with re-ordered input timestamps that tend to defy automagic handling for every case, so allow for a few variations that can be tried depending on circumstances. Also try to document accordingly. Also fixes #638288. 2010-12-30 21:48:41 +0200 Felipe Contreras * gst/quicktime/gstqtmux.c: qtmux: get rid of timestamp overprotectiveness Signed-off-by: Felipe Contreras 2011-01-03 16:56:57 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/atomsrecovery.c: * gst/quicktime/gstqtmux.c: qtmux: simplify and fix pts_offset storing In particular, only write a ctts atom if and only if ever a non-zero offset. 2011-01-03 10:43:15 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: add some more documentation 2010-12-03 15:23:00 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: remove large-file property Rather, auto-determine if 64-bits fields are needed for a valid result, and stick to plain 32-bits if not needed. API: GstQTMux:large-file (removed) 2010-12-19 12:53:34 +0100 Sebastian Dröge * gst/quicktime/gstqtmux.c: qtmux: Free AtomInfo structs 2010-12-19 12:50:30 +0100 Sebastian Dröge * gst/quicktime/gstqtmux.c: qtmux: Free tag string after use 2010-12-19 12:12:25 +0100 Sebastian Dröge * tests/check/pipelines/tagschecking.c: tagschecking: Fix some more memory leaks 2010-12-17 19:41:25 +0200 Lasse Laukkanen * gst/quicktime/gstqtmux.c: qtmux: allow zero duration tracks 2010-12-03 18:09:41 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: add documentation 2010-12-01 10:45:49 +0100 David Hoyt * gst/quicktime/gstqtmux.c: qtmux: handle msvc ftruncate incompatibility Fixes #636185. 2010-11-27 16:07:19 -0600 Alejandro Gonzalez * gst/quicktime/gstqtmux.c: qtmux: gst_qtmux_check_difference verify before subtract Avoid negative overflow by checking the order of operands on subtraction of unsigned integers. https://bugzilla.gnome.org/show_bug.cgi?id=635878 2010-11-19 17:55:36 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: remove remnant of obsolete property 2010-11-19 15:18:58 +0100 Mark Nauwelaerts * tests/check/elements/qtmux.c: tests: qtmux: also unit test fragmented file cases 2010-07-30 12:48:29 +0200 Marc-André Lureau * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: allow specifying trak timescale This is mainly because Smoothstreaming client are broken and don't take the TimeScale property into account. 2010-11-19 17:41:41 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: qtmux: include sdtp atoms for ismv fragmented files Based on patch by Marc-André Lureau 2010-11-19 19:17:45 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: enable default fragmented file for ismlmux 2010-09-02 13:58:05 +0200 Marc-André Lureau * gst/quicktime/atoms.h: * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: qtmux: add ismlmux, for fragmented isml major brand 2010-11-19 14:44:45 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: finalize sinkpads list 2010-07-22 19:40:07 +0200 Marc-André Lureau * gst/quicktime/gstqtmux.c: qtmux: add moov in streamheader 2010-08-06 13:26:27 +0200 Marc-André Lureau * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: add streamable property to avoid building fragmented mfra index 2010-11-18 16:48:06 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: add mfra to fragmented file Based on patch by Marc-André Lureau 2010-11-15 15:17:59 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: optionally create fragmented file In this mode, an initial empty moov (containing only stream metadata) is written, followed by fragments containing actual data (along with required metadata). New fragments are started either at keyframe (if such are sparse) or when property configured duration exceeded. Based on patch by Marc-André Lureau Fixes #632911. 2010-11-15 15:12:45 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: qtmux: use helper to set atom flags from given uint 2010-11-09 16:49:07 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: refactor configuring and sending of moov Based on patch by Marc-André Lureau 2010-11-09 15:54:44 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: refactor extra top-level atom handling Also check a bit more for possible errors, and free proper items in such case. 2010-11-09 15:01:15 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: refactor slightly using buffer helper 2010-11-05 13:48:57 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: fix misinforming comment 2010-11-05 12:08:15 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: qtmux: delegate mvex handling to atoms ... which keeps qtmux simpler. 2009-09-28 16:11:35 +0200 Marc-André Lureau * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: qtmux: add mvex/trex in header if fragmented One "trex" is added per "trak". We don't support default values, but the "trex" box is mandatory. 2009-09-28 13:01:30 +0200 Marc-André Lureau * gst/quicktime/fourcc.h: qtmux: add a couple of fourcc for fragmented mp4 2010-11-05 11:08:01 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: avoid removing temp file when error occurred 2009-09-30 17:16:30 +0200 Marc-André Lureau * gst/quicktime/gstqtmux.c: qtmux: truncate buffer file after each send 2009-09-28 16:53:51 +0200 Marc-André Lureau * gst/quicktime/gstqtmux.c: qtmux: remove temp file when reset/finalize 2010-10-19 13:43:14 +0300 Stefan Kost * gst/quicktime/gstqtmoovrecover.c: various (gst): add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. 2010-10-13 17:47:29 +0200 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: prevent infinite loop when adjusting framerate Fixes #632070. 2010-10-03 23:45:46 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: Add G_PARAM_STATIC_STRINGS Add G_PARAM_STATIC_STRINGS to qtmux properties 2010-09-15 17:54:49 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: Follow xmp serialization guidelines closer qt and isom variants have different ways of serializing xmp, follow these guidelines. Those can be found in Adobe's xmp docs. 2010-08-16 12:36:24 +0200 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: autodetect out-of-order input timestamps and determine DTS accordingly Favour using input buffer timestamps for DTS, but fallback to using buffer duration (accumulation) if input ts detected out-of-order. Fixes #624212. 2010-07-28 16:15:53 +0200 Marc-André Lureau * gst/quicktime/gstqtmux.c: qtmux: use caps bitrate at last chance If we didn't get the stream's bitrate from one of the atoms, try getting it from the caps as a last resort. https://bugzilla.gnome.org/show_bug.cgi?id=625496 2010-07-28 16:12:11 +0200 Marc-André Lureau * gst/quicktime/atoms.c: qtmux: btrt - max bitrate before average According to iso base media file format, the max bitrate is before the avg https://bugzilla.gnome.org/show_bug.cgi?id=625496 2010-07-06 14:48:08 +0530 Arun Raghavan * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: qtmux: Write 'btrt' atom for H.264 media if possible This writes out the optional 'btrt' atom (MPEG4BitrateBox) for H.264 media if either or both of average and maximum bitrate are available for the stream. https://bugzilla.gnome.org/show_bug.cgi?id=623678 2010-07-05 14:09:50 +0530 Arun Raghavan * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: Write avg/max bitrate to ESDS if available This collects the 'bitrate' and 'maximum-bitrate' tags on the corresponding pad and uses these to populate these fields in the ESDS where applicable. https://bugzilla.gnome.org/show_bug.cgi?id=623678 2010-07-02 12:45:20 +0200 Edward Hervey * gst/quicktime/gstqtmux.c: qtmux: Don't use bogus codec/format tags https://bugzilla.gnome.org/show_bug.cgi?id=623365 2010-06-25 20:19:20 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: Write uint tags that don't have a complement Write uint tags that have complements (e.g. track-number/ track-count) even when we only have one of them available and set the other one to 0. Fixes #622484 2010-06-21 19:39:54 +0200 Edward Hervey * gst/quicktime/gstqtmux.c: qtmux: Remove the pad from our internal list before calling collectpads Previously we would end up with the collectpaddata structure already freed. This would result in a bogus iteration of mux->sinkpads (all the GstQTPad being freed) and it wouldn't be removed from that list. Finally, due to it not being removed from that list, we would end up calling a bogus gst_qt_mux_pad_reset on those structures => SEGFAULT 2010-05-12 18:50:34 -0700 David Schleef * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: Add VP8 2010-05-11 13:15:37 +0100 Tim-Philipp Müller * tests/check/pipelines/tagschecking.c: tests: don't fail tagschecking test if qtdemux is not available or too old 2010-03-27 09:46:30 +0000 Tim-Philipp Müller * gst/quicktime/gstqtmuxplugin.c: qtmux: use GStreamer package name and origin in the plugin info 2010-03-23 17:34:30 -0300 Thiago Santos * tests/check/pipelines/tagschecking.c: tests: tagschecking: New tags tests Adds new tags checking tests. 2010-03-25 00:20:54 +0000 Tim-Philipp Müller * gst/quicktime/gstqtmux.c: qtmux: init debug category before using it 2010-03-22 16:56:03 +0100 Benjamin Otte * gst/quicktime/atoms.c: Add -Wold-style-definition and fix the warnings 2010-03-22 13:16:33 +0100 Benjamin Otte * gst/quicktime/atoms.c: * gst/quicktime/gstqtmuxmap.h: * tests/check/elements/qtmux.c: Add -Wwrite-strings and fix its warnings 2010-03-21 21:39:18 +0100 Benjamin Otte * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/atomsrecovery.c: * gst/quicktime/descriptors.c: * tests/check/elements/qtmux.c: * tests/check/pipelines/tagschecking.c: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/quicktime/gstqtmoovrecover.c: * gst/quicktime/gstqtmux.c: gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-12 11:28:51 -0300 Thiago Santos * tests/check/pipelines/tagschecking.c: tests: tagschecking: Improvements and new geo-location tests Makes some improvements to tagschecking.c, making it use fakesrc instead of videotestsrc and allowing to set input caps so that more muxers can be used. Previously we could only use those that accepted raw video caps. Also adds some tests for geo-location tags 2010-03-12 10:53:36 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: Use xmp on mp4mux and gppmux too Do not restrict xmp to qtmux, but use it too on mp4mux and gppmux 2010-03-05 13:33:37 -0300 Thiago Santos * tests/check/pipelines/tagschecking.c: check: tagschecking: tests for tags serialization in muxers Adds a check unit test that aims to test tags serialization and deserialization consistency (in muxers). It provides a basic function that allows one to easily specify tags, a muxer and a demuxer and a test will be done to check if the tags have been consistently muxed and demuxed 2010-02-22 16:45:34 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: qtmux: add xmp support Adds xmp metatags adding to qtmux. Fixes #609539 2010-03-11 17:17:15 +0000 Tim-Philipp Müller * gst/quicktime/gstqtmoovrecover.c: qtmux: fix GST_ELEMENT_ERROR usage We need to pass (NULL) rather than NULL for empty arguments. 2010-03-10 10:23:23 -0600 Rob Clark * gst/quicktime/gstqtmoovrecover.c: qtmux: fix compile error gst/quicktime/gstqtmoovrecover.c:268: warning: format not a string literal and no format arguments https://bugzilla.gnome.org/show_bug.cgi?id=612454 2010-02-22 19:38:15 -0300 Thiago Santos * gst/quicktime/gstqtmuxmap.c: qtmux: Rename 'avc-sample' to 'avc' in caps Fixes #606662 2010-02-26 11:50:25 -0800 Michael Smith * gst/quicktime/gstqtmux.c: qtmux: Take lock around use of (non-threadsafe) tagsetter interface. 2010-02-22 16:51:00 -0300 Thiago Santos * gst/quicktime/atoms.c: qtmux: write all udta children atoms UDTA might have META and other children atoms together, write them all. 2010-02-22 10:48:11 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: Use internal sink pads list Due to GstCollectPads sink pads list being not reliably iteratable (when not inside the collected function) this patch adds a sink pads list to qtmux to be used when iterating sink pads on reset function. Fixes #609055 2010-02-16 17:13:09 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: qtmux: prevent leaking hdlr name 2010-02-16 16:24:12 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: support for ALAC Fixes #580731. 2010-02-16 14:19:04 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: qtmux: refactor building stsd entry 'wave' extension 2010-02-08 11:51:52 -0300 Thiago Santos * gst/quicktime/atomsrecovery.c: qtmux: atomsrecovery: Fix compilation problem Fixes a compilation error due to unused function result. 2009-12-12 16:07:15 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/atomsrecovery.c: * gst/quicktime/atomsrecovery.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmoovrecover.c: * gst/quicktime/gstqtmoovrecover.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxplugin.c: qtmux: Adds moov recovery feature Adds a new property to qtmux that sets a path to a file to write and update data about the moov atom (that is not writen till the end of the file). If the pipeline/app crashes during execution it might be possible to recover the movie using the qtmoovrecover element. qtmoovrecover is an element that is also a pipeline. It is not meant to be used with other elements (it has no pads). It is merely a tool/utilitary to recover unfinished qtmux files. Fixes #601576 2010-01-27 19:06:53 -0800 Michael Smith * gst/quicktime/atoms.c: qtmux: for fixed-sample size streams (PCM audio, etc) don't allocate an enormous buffer that we then won't use at all. 2010-01-27 15:37:37 -0800 Michael Smith * gst/quicktime/gstqtmux.c: qtmux: handle muxing adpcm correctly. 2010-01-22 13:36:04 -0800 Michael Smith * gst/quicktime/atoms.c: qtmux: Set the mdia hdlr name field to what quicktime uses. Fix writing it since it's not null-terminated. Improves compatibility with some hardware players. 2010-01-22 13:30:07 -0800 Michael Smith * gst/quicktime/gstqtmux.c: qtmux: endianness in gstreamer is an int, not boolean. 2010-01-26 17:54:28 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: qtmux: streamline moov data memory storage In particular, use arrays rather than (double) linked lists. 2010-01-26 13:44:04 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: g_free is NULL safe 2010-01-20 13:30:48 +0100 Benjamin Otte * gst/quicktime/descriptors.c: * gst/quicktime/descriptors.h: * gst/quicktime/properties.c: [cleanup] Various style and cleanups Various fixes for gtk-doc warnings and making functions without arguments take void as parameter. 2010-01-14 08:09:03 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/gstqtmux.c: qtmux: Actually use new caps info on renegotiation Following the previous qtmux commit, this patch tries to use the new info added to the caps to fill the 'trak' atom's fields and children atoms. This way qtmux will use the late added 'codec_data' when h264parse adds it in the following pipeline: videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \ h264parse output-format=0 ! qtmux ! \ filesink location=test.mov 2010-01-13 23:33:51 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/gstqtmux.c: qtmux: Do caps renegotiation when it only adds fields Qtmux can accept caps renegotiation if the new caps is a superset of the old one, meaning upstream added new info to the caps. This patch still doesn't make qtmux update any atoms info from the new info, but at least it doesn't reject the new caps anymore. A pipeline that reproduces this use case is: videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \ h264parse output-format=0 ! qtmux ! \ filesink location=test.mov 2010-01-13 19:30:45 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: provide request pads under wider conditions Fixes #606859. 2010-01-13 10:35:00 -0300 Thiago Santos * gst/quicktime/gstqtmuxmap.c: qtmux: Only accept avc-sample h264 qtmux and mp4mux should only accept h264 in avc-sample format 2010-01-11 13:13:41 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: Rename aac's stream-format 'none' to 'raw' Renames aac's stream-format from previous commits from none to raw 2010-01-11 10:34:32 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: Only accept stream-format='none' aac Only accept raw aac streams (stream-format=none) to avoid generating invalid files. Fixes #604925 2009-12-28 11:34:35 +0200 Stefan Kost * gst/quicktime/gstqtmux.h: qtmux: also add .h file changes to unbreak the build 2009-12-27 23:51:50 +0200 Stefan Kost * gst/quicktime/gstqtmux.c: qtmux: use correct names from template for request pads The pads where names pad0, pad1, ... 2009-12-27 23:32:58 +0200 Stefan Kost * gst/quicktime/gstqtmux.c: qtmux: move errors _new_pad to the end 2009-12-21 13:58:30 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: Accept non-paired uint tags Adds support for unpaired unsigned interger tags 2009-12-21 12:05:37 -0300 Thiago Santos * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: qtmux: Adds new tags Maps more tags that are already posted by qtdemux Fixes #599759 2009-12-10 22:20:45 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxmap.c: qtmux: support more of j2k Reads the new caps added to qtdemux by commit c917d65e6df0b5d585f905c7ad78a8a0a44b2cb0 and adds its corresponding atoms. Also adds support for image/x-jpc as it is the same as image/x-jp2, except that the buffers need to be boxed inside a jp2c isom box before muxing. To solve this the QTPads now have a function that (if not NULL) is called when a buffer is collected. This function returns a replacement to the current collected buffer. Fixes #598916 2009-12-10 16:53:19 -0300 Thiago Santos * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: Maps 'classification' tag for 3gpp files Adds the mapping of 'classification' tags to writing of 'clsf' atoms for gppmux. Based on a patch by: Lasse Laukkanen 2009-12-08 17:59:04 -0800 Michael Smith * gst/quicktime/atoms.c: * gst/quicktime/gstqtmux.c: qtmux: remove c++ comments and add some more comments. 2009-12-08 17:55:56 -0800 Michael Smith * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: add ima adpcm support 2009-11-25 21:41:27 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: replace _scale with _scale_round Use the rounding version for improved sync between streams. Small variations in the duration when muxing might lead to cumullative wrong timestamping when demuxing. Fixes #602936 2009-11-24 16:16:56 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: use timestamps for muxing Try to use timestamps even when the stream has out of order timestamps, only fall back to durations when we detect an out of order buffer. Improves sync between streams. 2009-11-19 18:28:52 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: fix missing debug argument Adds a missing debug argument 2009-11-19 11:36:14 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: fix misinforming debug statement 2009-11-19 11:14:57 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: ensure writable buffer metadata before setting caps 2009-10-29 08:36:02 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: support for SVQ3 Adds support for muxing SVQ3 content. Usually this format has decoder info that must be passed in the 'seqh' field in the caps. It is also good to add the gama atom to make quicktime not crash. Fixes #587922 2009-11-17 09:26:05 -0300 Thiago Sousa Santos * gst/quicktime/gstqtmux.c: qtmux: do not leak a string Frees a string after use. Also does some code organization 2009-11-16 14:57:53 -0300 Thiago Sousa Santos * gst/quicktime/atoms.c: qtmux: do not add size to the pointer variable Do not wrongly add the result of the function to the pointer to the buffer size. Instead, check the result to see if the serialization was ok. Based on a patch by: "Carsten Kroll " Fixes #602106 2009-11-06 10:34:39 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: handle 'late' streams When muxing streams, some can start later than others. qtmux now handle this by adding an empty edts entry with the duration of the 'lateness' to the stream's trak. It tolerates a stream to be up to 0.1s late. Fixes #586848 2009-11-05 21:35:56 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: qtmux: adds the EDTS and ELTS atoms to atoms.c These atoms will be useful for signaling streams that start later in the file. As well for adding edit lists if needed sometime later. 2009-11-06 00:46:12 -0300 Thiago Santos * gst/quicktime/atoms.c: * gst/quicktime/gstqtmux.c: qtmux: Adding some ifs for protection Adding somes ifs to protect against warning conditions that might happen when upstream element is not sane Fixes #600895 2009-10-16 10:47:32 -0300 Thiago Santos * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: gppmux: Add support for 3gr6 Keep track of the chunk durations to be able to add 3gr6 brand if it is a faststart file and the longest chunk is smaller than a sec. Implemented according to 3gpp TS 26.244 v6.4.0 (2005-09) Fixes #584361 2009-10-15 21:11:16 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: Only push ftyp later (in faststart mode) In faststart mode, there is no need to send the ftyp right at the beginning of the stream. Waiting and sending it only later (when the moov atom is ready to be sent) provides us with more information about the stream and we can better select the compatible brands. 2009-10-15 17:51:39 -0300 Thiago Santos * gst/quicktime/gstqtmux.c: qtmux: Improve error message Improve error message when we can't get or estimate the timestamp/duration of a buffer 2009-09-29 15:47:13 +0200 Marc-André Lureau * gst/quicktime/atoms.c: qtmux: fix flags_as_uint to flags[] 2009-08-04 12:58:35 +0200 Jan Urbanski * gst/quicktime/gstqtmux.c: qtmux: Don't require endianness field for 8 bit raw audio Fixes bug #590360. 2009-06-25 08:38:21 +0200 Edward Hervey * gst/quicktime/atoms.c: qtmux: Remove unused variable. 2009-06-25 08:38:10 +0200 Edward Hervey * gst/quicktime/gstqtmux.c: qtmux: Fix debug statement. 2009-06-11 15:54:42 +0200 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: qtmux: only use (64-bit) extended (mdat) atom size if needed. Fixes #585319. 2009-06-10 14:46:14 +0200 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: set default movie timescale to microsecond units 2009-06-10 13:24:20 +0200 Mark Nauwelaerts * gst/quicktime/atoms.c: qtmux: compress/optimize stsc writing 2009-06-10 12:42:44 +0200 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: qtmux: add 3GP style tagging (and refactor appropriately) 2009-06-01 23:00:44 +0200 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: qtmux (and variants): handle pixel-aspect-ratio. Fixes #584358. 2009-06-01 22:42:08 +0200 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmuxmap.c: gppmux: enhance ftyp brand heuristic. Fixes #584360. 2009-05-28 13:56:10 +0200 Mark Nauwelaerts * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: qtmux: use different stsd atom type for H263 for ISO and QT variants Fixes #584114. 2009-05-15 01:54:44 -0300 Thiago Santos * gst/quicktime/atoms.c: [qtmux] Fixes segfault when adding a blob as first tag. Moves tags data initialization to the function that actually appends the tags to the list. Fixes #582702 Also fixes some style caught by the pre-commit hook. 2009-05-10 21:21:36 +0200 Mark Nauwelaerts * gst/quicktime/gstqtmuxmap.c: gppmux: Add MPEG-4 part 2 to supported formats. Fixes #581593. 2009-05-07 17:53:42 +0100 Christian Schaller * gst/quicktime/gstqtmux.c: Add ranks to various muxers and encoders in -bad 2009-04-30 14:43:36 -0300 Thiago Santos * gst/quicktime/gstqtmuxmap.c: qtmux: changes caps of src pads to video/quicktime, variant=something Take a look at bug #580005 for further info. 2009-04-24 18:53:36 -0300 Thiago Santos * gst/quicktime/gstqtmuxmap.c: mp4mux: Changes src caps to application/x-iso-mp4 Fixes #580005 2009-03-25 21:24:44 +0100 Mark Nauwelaerts * gst/quicktime/gstqtmux.c: qtmux: fix reusing element State change to READY and then back to PAUSED should still provide the proper structures as are otherwise freshly available following a request_new_pad. Pointed out by Thiago Santos. 2009-03-23 11:17:39 +0100 Wim Taymans * gst/quicktime/gstqtmux.c: qtmux: fix includes for lseek -- 2009-03-20 14:20:16 +0100 LRN * gst/quicktime/gstqtmux.c: win32: fix seeking in large files Use _lseeki64() on Windows to seek in large files. Fixes #576021. 2009-03-02 10:57:35 +0100 Edward Hervey * gst/quicktime/gstqtmux.c: qtmux: Be a bit more verbose in our debug message when failing to renegotiate 2009-01-28 13:25:14 +0100 Mark Nauwelaerts * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmuxmap.c: Additional media type support in qtmux (and friends). Support AMR and H263 for both qtmux and gppmux, and add extensions in sample table description. 2009-01-09 21:59:48 +0000 David Schleef gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part to caps so schroenc/schroparse can use it. Fixes #5... Original commit message from CVS: * gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part to caps so schroenc/schroparse can use it. Fixes #566958 2008-12-19 18:53:47 +0000 Mark Nauwelaerts gst/quicktime/gstqtmux.c: Do not tempt or suggest to violate gst_collect_pads API specification. Original commit message from CVS: * gst/quicktime/gstqtmux.c: (gst_qt_mux_change_state): Do not tempt or suggest to violate gst_collect_pads API specification. 2008-12-19 18:33:47 +0000 Mark Nauwelaerts gst/quicktime/: Dual license qtmux LGPL/MIT. Fixes #564232. Original commit message from CVS: * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/descriptors.c: * gst/quicktime/descriptors.h: * gst/quicktime/fourcc.h: * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: * gst/quicktime/properties.c: * gst/quicktime/properties.h: Dual license qtmux LGPL/MIT. Fixes #564232. 2008-12-16 16:26:52 +0000 Stefan Kost Totally remove the internal taglists and fully use tagsetter. Fixes various tag muxing issues. Original commit message from CVS: * ext/celt/gstceltenc.c: * ext/celt/gstceltenc.h: * ext/metadata/gstmetadatamux.c: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: Totally remove the internal taglists and fully use tagsetter. Fixes various tag muxing issues. 2008-12-01 16:37:45 +0000 Mark Nauwelaerts gst/quicktime/atoms.c: Fix mj2 sample description metadata construction. Original commit message from CVS: * gst/quicktime/atoms.c: (build_jp2h_extension): Fix mj2 sample description metadata construction. 2008-11-18 01:09:09 +0000 David Schleef gst/quicktime/gstqtmux.c: Quiet a debugging message that I recently added. Original commit message from CVS: * gst/quicktime/gstqtmux.c: Quiet a debugging message that I recently added. 2008-11-15 02:56:31 +0000 David Schleef gst/quicktime/gstqtmux.*: Use dts from GST_BUFFER_OFFSET_END() for video/x-qt-part. Original commit message from CVS: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: Use dts from GST_BUFFER_OFFSET_END() for video/x-qt-part. 2008-11-14 21:24:51 +0000 Mark Nauwelaerts gst/quicktime/: Revert previous commit. Original commit message from CVS: * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/descriptors.c: * gst/quicktime/descriptors.h: * gst/quicktime/fourcc.h: * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: * gst/quicktime/properties.c: * gst/quicktime/properties.h: Revert previous commit. 2008-11-14 20:38:18 +0000 Mark Nauwelaerts gst/quicktime/: Dual license LGPL/MIT, as apparently supposed to. Original commit message from CVS: * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/descriptors.c: * gst/quicktime/descriptors.h: * gst/quicktime/fourcc.h: * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: * gst/quicktime/properties.c: * gst/quicktime/properties.h: Dual license LGPL/MIT, as apparently supposed to. 2008-11-14 20:17:10 +0000 Mark Nauwelaerts gst/quicktime/: Cut detour in sample description extension construction. Original commit message from CVS: * gst/quicktime/atoms.c: (build_esds_extension), (build_mov_aac_extension), (build_jp2h_extension), (build_codec_data_extension): * gst/quicktime/atoms.h: * gst/quicktime/fourcc.h: * gst/quicktime/gstqtmux.c: (gst_qt_mux_audio_sink_set_caps), (gst_qt_mux_video_sink_set_caps): * gst/quicktime/gstqtmuxmap.c: (gst_qt_mux_map_format_to_header): Cut detour in sample description extension construction. Also actually implement ISO JPEG2000 mj2 format. 2008-11-11 19:31:35 +0000 Mark Nauwelaerts tests/check/: Add unit test for qtmux. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/qtmux.c: (setup_src_pad), (teardown_src_pad), (setup_qtmux), (cleanup_qtmux), (check_qtmux_pad), (GST_START_TEST), (qtmux_suite), (main): Add unit test for qtmux. 2008-11-11 19:24:12 +0000 Mark Nauwelaerts gst/quicktime/gstqtmux.c: Add some more safety/sanity checks in tag manipulation. Original commit message from CVS: * gst/quicktime/gstqtmux.c: (gst_qt_mux_add_metadata_tags): Add some more safety/sanity checks in tag manipulation. 2008-11-08 02:00:58 +0000 Thiago Sousa Santos Copy qtmux from revision 148 of the gst-qtmux repository. Original commit message from CVS: patch by: Thiago Sousa Santos * configure.ac: * gst/quicktime/Makefile.am: * gst/quicktime/atoms.c: * gst/quicktime/atoms.h: * gst/quicktime/descriptors.c: * gst/quicktime/descriptors.h: * gst/quicktime/fourcc.h: * gst/quicktime/ftypcc.h: * gst/quicktime/gstqtmux.c: * gst/quicktime/gstqtmux.h: * gst/quicktime/gstqtmuxmap.c: * gst/quicktime/gstqtmuxmap.h: * gst/quicktime/properties.c: * gst/quicktime/properties.h: Copy qtmux from revision 148 of the gst-qtmux repository. Fixes #550280. 2011-04-12 18:25:34 +0100 Tim-Philipp Müller * Android.mk: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/inspect/plugin-quicktime.xml: * gst/quicktime/LEGAL: * gst/quicktime/Makefile.am: * gst/quicktime/gstrtpxqtdepay.c: * gst/quicktime/gstrtpxqtdepay.h: * gst/quicktime/qtatomparser.h: * gst/quicktime/qtdemux.c: * gst/quicktime/qtdemux.h: * gst/quicktime/qtdemux.vcproj: * gst/quicktime/qtdemux_dump.c: * gst/quicktime/qtdemux_dump.h: * gst/quicktime/qtdemux_fourcc.h: * gst/quicktime/qtdemux_lang.c: * gst/quicktime/qtdemux_lang.h: * gst/quicktime/qtdemux_types.c: * gst/quicktime/qtdemux_types.h: * gst/quicktime/qtpalette.h: * gst/quicktime/quicktime.c: * po/POTFILES.in: qtdemux: rename directory to quicktime to match plugin name In preparation for qtmux moving to -good. 2011-04-12 11:49:54 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: simplify framerate fraction calculation 2011-01-24 15:45:28 -0600 Leonardo Sandoval * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: add width, height and framerate to caps when present on onMetaData Fixes #640483. 2010-08-24 13:57:55 +0200 Pascal Buhler * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Unknown SSRC is not fatal https://bugzilla.gnome.org/show_bug.cgi?id=646966 2010-08-24 13:54:58 +0200 Pascal Buhler * gst/rtpmanager/rtpsession.c: rtpsession: Number of active sources should be updated whenever the status of the source changes to active Forward-ported by Olivier Crête https://bugzilla.gnome.org/show_bug.cgi?id=646965 2010-06-23 11:29:58 +0200 Havard Graff * gst/rtpmanager/rtpsession.c: rtpmanager: ignore a BYE if it is sent with our internal SSRC https://bugzilla.gnome.org/show_bug.cgi?id=646964 2010-01-29 09:49:48 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Adds more h264 fields to its caps Adds alignment=au and stream-format=avc to h264 caps Fixes #606662 2011-04-11 12:44:19 +0300 Stefan Kost * configure.ac: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: also handle deprecations for jack 1.9.7 Jack 1.9.7 was released 20.Mar.2011, need to handle the deprecated api for this version too. 2011-04-11 00:36:35 -0400 Thibault Saunier * gst/dtmf/Makefile.am: android: make it ready for androgenizer Remove the android/ top dir Fixe the Makefile.am to be androgenized To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files. Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git 2011-04-10 18:56:52 -0400 Thibault Saunier * Android.mk: * android/NOTICE: * android/apetag.mk: * android/avi.mk: * android/flv.mk: * android/gst/rtpmanager/gstrtpbin-marshal.c: * android/gst/rtpmanager/gstrtpbin-marshal.h: * android/gst/udp/gstudp-enumtypes.c: * android/gst/udp/gstudp-enumtypes.h: * android/gst/udp/gstudp-marshal.c: * android/gst/udp/gstudp-marshal.h: * android/icydemux.mk: * android/id3demux.mk: * android/qtdemux.mk: * android/rtp.mk: * android/rtpmanager.mk: * android/rtsp.mk: * android/soup.mk: * android/udp.mk: * android/wavenc.mk: * android/wavparse.mk: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debugutils/Makefile.am: * gst/deinterlace/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flv/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/imagefreeze/Makefile.am: * gst/interleave/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/qtdemux/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/rtsp/Makefile.am: * gst/shapewipe/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * gst/y4m/Makefile.am: android: Make it ready for androgenizer Remove the android/ top dir Fixe the Makefile.am to be androgenized To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files. Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git 2011-04-05 21:14:43 +0200 Haakon Sporsheim * gst/rtp/gstrtpgstpay.c: rtpgstpay: declare frag_offset to hold 32bits. As specified in documenation above and below. https://bugzilla.gnome.org/show_bug.cgi?id=646954 2011-04-09 12:41:48 +0200 Havard Graff * gst/rtpmanager/gstrtpsession.c: rtpsession: fix wrongly applied patch Obviously recv_rtp_sink does not have much to do with send_rtcp_src... See commit 046ff170. https://bugzilla.gnome.org/show_bug.cgi?id=647263 2011-04-08 15:59:58 +0100 Tim-Philipp Müller * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparsers: update for set_frame_props -> set_frame_rate API change 2011-04-08 00:03:21 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: tests: hook up audioparser unit tests 2011-04-07 18:30:49 +0200 Mark Nauwelaerts * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: relax sync match a bit when draining ... to at least allow initial caps change (but no further caps jitter). Fixes unit test again after previous change. 2011-04-07 15:21:10 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videofilter.xml: docs: update for changes in git 2011-04-07 15:20:19 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-audioparsers.xml: docs: add audioparsers to docs 2011-04-07 15:07:15 +0100 Tim-Philipp Müller * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstamrparse.h: * gst/audioparsers/plugin.c: aacparse, amrparse: gst_fooparse_xyz -> gst_foo_parse_xyz to match GstFooParse See moving-plugins checklist. 2011-04-07 14:43:42 +0100 Tim-Philipp Müller * configure.ac: * gst/audioparsers/Makefile.am: * gst/audioparsers/plugin.c: audioparsers: hook up to build 2011-04-07 13:26:41 +0100 Tim-Philipp Müller * gst/audioparsers/Makefile.am: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstamrparse.h: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: audioparsers: port to new GstBaseParse in core 2011-04-04 20:55:39 +0200 Mark Nauwelaerts * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: require tighter sync match when draining 2011-04-01 14:47:43 +0200 Sebastian Dröge * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: mpegaudioparse: Parse encoder delay and encoder padding from the LAME header if present 2011-03-09 23:06:14 +0530 Arun Raghavan * gst/audioparsers/plugin.c: dcaparse: Bump rank to primary+1 Seems to work fine with a reasonably wide range of media, so bumping rank. 2011-03-23 22:02:37 +0530 Arun Raghavan * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: dcaparse: Expose frame size in caps This exports the size of the frame (number of bytes from one sync point to the next) as the "frame_size" field in caps. 2011-03-09 23:03:10 +0530 Arun Raghavan * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: dcaparse: Expose block size in caps This sets the "block_size" field on caps as the number of samples encoded in one frame. 2011-03-16 15:53:13 +0000 Tim-Philipp Müller * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: add FIXME for making the base class use xing seek tables better 2011-03-14 18:25:25 +0100 Sebastian Dröge * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: dcaparse: Add depth and endianness to the caps Some decoders can only handle specific endianness or a fixed depth and this allows better negotiation. Fixes bug #644208. 2011-02-26 13:53:44 -0800 David Schleef * gst/audioparsers/gstaacparse.c: Revert "aacparse: allow parsed frames on sink pad" This reverts commit e49b89d5c5a1244fa0dcb8bb4996e38fb9bff9e5. 2011-02-23 17:25:03 -0800 David Schleef * gst/audioparsers/gstaacparse.c: aacparse: allow parsed frames on sink pad 2010-10-13 16:12:02 -0700 David Schleef * tests/check/elements/parser.c: tests: fix baseparse test 2010-10-13 15:39:55 -0700 David Schleef * gst/audioparsers/Makefile.am: * gst/audioparsers/gstaacparse.h: * gst/audioparsers/gstac3parse.h: * gst/audioparsers/gstamrparse.h: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: * gst/audioparsers/gstdcaparse.h: * gst/audioparsers/gstflacparse.h: * gst/audioparsers/gstmpegaudioparse.h: baseparse: Create baseparse library 2011-02-07 14:46:57 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: tune QUERY_SEEKING response Even if we currently do not have a duration yet, assume seekable if it looks like we'll likely be able to determine it later on (which coincides with needed information to perform seeking). Fixes #641047. 2011-02-08 23:39:24 +0530 Arun Raghavan * gst/audioparsers/gstbaseparse.c: baseparse: Update min/max bitrate before first posting them This avoids posting an initial min-bitrate of G_UINTMAX and max-bitrate of 0. https://bugzilla.gnome.org/show_bug.cgi?id=641857 2011-02-08 23:50:13 +0530 Arun Raghavan * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: mpegaudioparse: Post CBR bitrate as nominal bitrate Even if VBR headers are missing, we can't guarantee that a stream is in fact a CBR stream, so it's safer to let baseparse calculate the average bitrate rather than assume a CBR stream. However, in order to make /some/ metadata available before the requisite number of frames have been parsed, this posts the bitrate from the non-VBR headers as the nominal bitrate. https://bugzilla.gnome.org/show_bug.cgi?id=641858 2010-09-06 14:10:11 +0200 Mark Nauwelaerts * gst/audioparsers/gstamrparse.c: amrparse: a valid amr-wb frame should not have reserved frame type index See #639715. 2011-01-27 16:52:34 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: improve handling of dependent substream frames In particular, timestamps of these should track main-stream timestamps. 2011-01-21 14:53:39 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: tune default duration estimate update interval Rather than a fixed default frame count, estimate frame count to aim for an interval duration depending on fps if available, otherwise use old fixed default. 2011-01-14 15:16:04 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: reverse playback; mind keyframes for fragment boundary 2011-01-13 15:26:21 +0100 Mark Nauwelaerts * gst/audioparsers/gstamrparse.c: amrparse: properly check for sufficient available data prior to access 2011-01-12 14:40:37 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: ensure non-empty candidate frames 2011-01-11 15:24:23 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: clarify some debug statements 2011-01-11 15:24:02 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: properly track upstream timestamps ... rather than with a delay. 2011-01-11 15:23:29 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: need proper frame duration to obtain sensible frame bitrate 2011-01-11 15:22:51 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: proper initial values for index tracking variables 2011-01-11 12:05:13 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: arrange for consistent event handling 2011-01-10 16:59:59 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.h: baseparse: header style cleaning 2011-01-10 17:07:38 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: provide some more initial frame metadata in parse_frame ... and document accordingly. 2011-01-10 16:56:36 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: * gst/audioparsers/gstflacparse.c: baseparse: refactor passthrough into format flags Also add a format flag to signal baseparse that subclass/format can provide (parsed) timestamp rather than an estimated one. In particular, such "strong" timestamp then allows to e.g. determine duration. 2011-01-10 15:34:48 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: baseparse: introduce a baseparse frame to serve as context ... and adjust subclass parsers accordingly 2011-01-07 16:39:51 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: restrict duration scanning to pull mode and avoid extra set_caps call 2011-01-07 15:58:49 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: update some documentation Also add some more debug. 2011-01-06 11:41:44 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: allow increasing min_size for current frame parsing only Also check that subclass actually either directs to skip bytes or increases expected frame size to avoid going nowhere in bogus indefinite looping. 2011-01-14 15:26:37 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baesparse: fix refactor regression in loop based parsing 2011-01-06 11:16:56 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: pass all available data to subclass rather than minimum Also reduce some adapter calls and add a few debug statements. 2010-12-10 15:59:49 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: fix reverse playback handling 2010-12-10 14:56:13 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: minor typo and debug statement cleanup 2010-12-10 14:40:05 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: reduce locking ... which is either already mute and/or implicitly handled by STREAM_LOCK. 2011-01-14 14:08:38 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: avoid loop in frame locating interpolation 2011-01-19 18:26:30 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: mind gst_buffer_unref not liking NULL Fixes #639950. 2011-01-14 16:30:11 -0300 Thiago Santos * gst/audioparsers/gstbaseparse.c: audioparsers: baseparse: Be careful to not lose the event ref Don't unref the event if it hasn't been handled, because the caller assumes it is still valid and might reuse it. I ran into this problem when transcoding an AVI (with mp3 inside) to gpp. https://bugzilla.gnome.org/show_bug.cgi?id=639555 2011-01-13 17:10:13 +0000 Tim-Philipp Müller * gst/audioparsers/gstdcaparse.c: dcaparse: fix sync word for 14-bit little endian coding Fix copy'n'paste bug that made us look for the raw little endian sync word twice instead of looking for the 14-bit LE sync word as well. Fixes parsing of such streams (see #636234 for sample file). 2011-01-13 16:27:04 +0000 Tim-Philipp Müller * gst/audioparsers/gstbaseparse.c: docs: minor baseparse docs/comment fixes Remove copy'n'paste leftovers. 2011-01-06 12:49:43 +0100 Edward Hervey * gst/audioparsers/gstflacparse.c: flacparse: Fix unitialized variable on macosx 2010-12-13 15:17:29 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: relax bsid checking ... to the widest possible spec interpretation. Fixes #637062. 2010-12-03 18:11:56 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: audioparsers: update some documentation 2010-12-03 18:11:38 +0100 Mark Nauwelaerts * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: add to documentation 2010-12-03 18:11:09 +0100 Mark Nauwelaerts * gst/audioparsers/gstdcaparse.c: dcaparse: add to documentation 2010-11-08 19:58:31 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: increase keyframe awareness ... which is not particular relevant for audio parsing, but more so in video cases. In particular, auto-determine if dealing with video (caps). 2010-12-01 15:28:53 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: use proper EAC-3 caps 2010-11-30 15:41:02 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: avoid unexpected stray metadata 2010-11-30 15:40:28 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: use proper _NONE output value when applicable 2010-11-25 18:56:42 +0100 Edward Hervey * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstbaseparse.c: audioparsers: Remove dead assignments 2010-11-25 17:14:23 +0100 Andoni Morales Alastruey * gst/audioparsers/gstbaseparse.c: audioparse: fix possible division-by-zero https://bugzilla.gnome.org/show_bug.cgi?id=635786 2010-11-17 16:23:42 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: use correct offset when adding index entry ... bearing in mind that BUFFER_OFFSET is media specific and may not reflect the basic offset after having been parsed. 2010-11-17 14:30:09 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: enhancements for timestamp marked framed formats That is, as such formats allow subclass to extract position from frame, it is possible to extract duration (if not otherwise provided) from (near) last frame, and a seek can fairly accurately target the required position. Fixes #631389. 2010-11-16 17:06:14 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: refactor frame scanning peformed by _loop 2010-11-16 18:04:00 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: slightly optimize sending of pending newsegment events 2010-11-16 17:04:35 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: minor fixes and enhancements Arrange for upstream as well as downstream flushing when seeking. Also determine upstream size as well as seekability. Adjust some comments to reality and employ debug statement in proper order. 2010-11-17 15:33:36 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: aacparse: minor cleanups 2010-11-17 15:24:37 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: aacparse: fix regression in ADIF src caps setting 2010-11-16 12:11:53 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: parse seektable Fixes #631389 (partially). 2010-11-16 12:08:54 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: minor refactor and enable default baseparse segment clipping 2010-11-09 19:38:25 +0100 Mark Nauwelaerts * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: fix silly leak in _reset 2010-10-29 14:08:58 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: use only upstream duration if it provides one 2010-10-25 14:15:50 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: reflow update_bitrate code ... which makes local variables represent real state better, and avoids triggering unneeded updates/actions. 2010-10-25 14:13:51 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: add some debug statements 2010-10-19 23:25:54 +0100 Tim-Philipp Müller * gst/audioparsers/gstdcaparse.c: dcaparse: init variable to make osx build bot happy gstdcaparse.c: In function 'gst_dca_parse_check_valid_frame': gstdcaparse.c:246: warning: 'best_sync' may be used uninitialized in this function 2010-10-19 00:15:20 +0100 Tim-Philipp Müller * gst/audioparsers/Makefile.am: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstdcaparse.h: * gst/audioparsers/plugin.c: audioparsers: add very basic dts/dca parser Still some issues, e.g. with seekable queries in totem, but also processing already-chunked input (created with matroskademux ! gdppay). 2010-10-14 16:48:21 +0200 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: properly parse e-ac3 frame header Also add a few debug statements. 2010-10-13 11:00:01 +0200 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: tweak setting buffer metadata; avoid timestamp jitter Fixes #631993. 2010-10-12 18:07:49 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: streamline src caps setting In particular, also set src caps whenever changes in stream warrant doing so. 2010-10-12 10:28:33 +0200 Sebastian Dröge * tests/check/elements/flacparse.c: flacparse: Adjust unit tests to new flacparse behaviour Garbage after frames is now included in the frames because flacparse has no easy way to detect the real end of a frame. Decoders are expected to everything after the frame because only decoding the bitstream will reveal the real end of the frame. Fixes bug #631814. 2010-10-12 10:27:53 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Don't drop the last frame if it is followed by garbage See bug #631814. 2010-10-11 17:49:46 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: perform bitrate handling and posting after newsegment sending 2010-10-11 17:36:19 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: immediately post subclass provided bitrate 2010-10-11 17:06:48 +0200 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: fix parsing with unknown framesizes Fixes #631814 (mostly). 2010-10-07 23:37:36 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Simplify frame header parsing by using lookup tables Based on a patch by Felipe Contreras. See bug #631200. 2010-10-07 23:28:08 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: Don't parse the complete FLAC frames but only look for valid frame headers Thanks to Felipe Contreras for the suggestion. This is partially based on his patches and makes flacparse more than 3.5 times faster. Looking for valid frame headers is unlikely to give false positives because every frame header is at least 9 bytes long, contains a 14 bit sync code and a 8 bit checksum over the first 8 bytes. Fixes bug #631200. 2010-10-06 18:32:51 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Really post tags only after the initial newsegment event The first newsegment event will be send by the first call to gst_base_parse_push_buffer() if necessary, posting the tags before that is not a good idea. Instead do it from the GstBaseParse::pre_push_buffer vfunc. 2010-10-05 11:17:52 +0100 Tim-Philipp Müller * gst/audioparsers/gstbaseparse.c: Revert "baseparse: add skip property" This reverts commit b5a3d60363d837a10f0533c141ec93d10b742312. Reverting this for now, since no one really seems to remember why this property exists or what it could possibly be good for. It seems to have been in the original mp3parse since the beginning of time and was back- ported from there. 2010-10-04 10:41:52 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Fix uninitialized variable compiler warnings These warnings are wrong, the variables are only used if they were initialized by the bit reader. 2010-09-14 02:48:58 +0300 Felipe Contreras * gst/audioparsers/gstflacparse.c: flacparse: fix picture parsing Signed-off-by: Felipe Contreras 2010-10-03 23:54:49 +0200 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Push tags before the header buffers are pushed 2010-08-02 20:50:21 +0300 Felipe Contreras * gst/audioparsers/gstflacparse.c: flacparse: trivial caps fix Signed-off-by: Felipe Contreras 2010-10-03 23:50:29 +0200 Sebastian Dröge * gst/audioparsers/gstbaseparse.c: audioparser: Let the format string agree with the parameters to fix compiler warning 2010-10-03 15:41:20 +0200 Sebastian Dröge * gst/audioparsers/gstac3parse.c: ac3parse: Use unchecked versions of the bitreader get functions We didn't check the return values anyway... 2010-09-22 15:44:43 +0530 Arun Raghavan * gst/audioparsers/gstbaseparse.c: baseparse: Fix debug output We lose the reference to the buffer after gst_pad_push(), so the debug print should happen before. https://bugzilla.gnome.org/show_bug.cgi?id=622276 2010-10-01 12:34:55 +0200 Mark Nauwelaerts * tests/check/elements/flacparse.c: * tests/check/elements/parser.c: * tests/check/elements/parser.h: audioparsers: add flacparse unit test ... and tweak parser test helper in the process. 2010-09-29 16:12:42 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: support reverse playback ... in pull mode or upstream driven. 2010-09-27 12:16:43 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: remove done TODOs and update documentation 2010-09-25 14:40:54 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: use determined seekability in answering SEEKING query 2010-09-25 14:32:06 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: add skip property 2010-09-25 13:59:39 +0200 Mark Nauwelaerts * tests/check/elements/ac3parse.c: * tests/check/elements/mpegaudioparse.c: audioparsers: add ac3parse and mpegaudioparse unit test 2010-09-25 13:59:18 +0200 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/gstmpegaudioparse.c: * gst/audioparsers/gstmpegaudioparse.h: * gst/audioparsers/plugin.c: mpegaudioparse: initial version ... adequately equivalent to mp3parse, so lets boldly set it to higher rank. 2010-09-25 14:01:07 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: aacparse: set minimum frame size at _start ... rather than one time at _init. 2010-09-25 13:50:51 +0200 Mark Nauwelaerts * tests/check/elements/aacparse.c: * tests/check/elements/amrparse.c: * tests/check/elements/parser.c: * tests/check/elements/parser.h: audioparsers: refactor existing unit tests using common helper 2010-09-22 15:07:09 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: use _set_frame_props to configure frame lead_in and lead_out ... provided a corresponding decoder with sufficient leading and following frames to carry out full decoding for a particular segment. 2010-09-22 14:13:17 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: * gst/audioparsers/gstflacparse.c: baseparse: use _set_duration to configure duration update interval ... as it logically belongs there as one or the other; either subclass can provide a duration, or an estimate must be made (reguarly updated). 2010-09-22 13:55:20 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: localize use of provided fps information 2010-09-22 12:13:12 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: seek table and accurate seek support 2010-09-21 13:57:10 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: proper and more extended segment and seek handling That is, loop pause handling, segment seek support, newsegment for gaps, etc 2010-09-21 10:57:04 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: add index support 2010-09-21 09:59:56 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: refactor state reset 2010-09-20 16:39:37 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: prevent indefinite resyncing 2010-09-20 13:57:55 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: specific EOS handling if no output so far 2010-09-20 13:31:57 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: adjust _set_frame_prop documentation and set default as claimed 2010-09-20 13:30:54 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: fix bitrate copy-and-paste and update heuristic 2010-09-17 18:33:29 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: post duration message if average bitrates is updated 2010-09-17 18:24:22 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: remove is_seekable vmethod and use a set_seek instead Seekability, like duration, etc is unlikely to change (frequently), and the default assumption covers most cases, so let subclass set when needed. At the same time, allow subclass to indicate if it has seek-metadata (table) available, and possibly have it provide an average bitrate. 2010-09-17 17:35:40 +0200 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: remove redundant default is_seekable 2010-09-17 17:21:46 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: add another hook for subclass prior to pushing buffer ... and allow subclass to perform custom segment clipping, or to emit tags or messages at this time. 2010-09-17 17:19:37 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: 0 converts to 0 by default 2010-09-16 18:56:46 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: basepase: refactor conversion using helper function and export default convert 2010-09-16 18:35:47 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: streamline query handling 2010-09-16 11:51:20 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: cleanup struct and remove unused member 2010-08-16 11:04:37 +0200 Mark Nauwelaerts * gst/audioparsers/plugin.c: audioparsers: increase ranks to enable auto-plugging Because we can, and should, have some shakedown testing before having these make it into -good later on ... 2010-09-22 16:07:24 +0530 Arun Raghavan * gst/audioparsers/gstbaseparse.c: baseparse: Allow chaining of subclass event handlers This allows the child class to chain its event handler with GstBaseParse, so that subclasses don't have to duplicate all the default event handling logic. https://bugzilla.gnome.org/show_bug.cgi?id=622276 2010-08-27 18:35:10 +0200 Sebastian Dröge * gst/audioparsers/gstbaseparse.c: baseparse: Don't use GST_FLOW_IS_FATAL() Also don't post an error message for UNEXPECTED and do it for NOT_LINKED. 2010-09-06 14:12:00 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: non-TIME seek event is simply not handled 2010-06-15 15:34:05 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: fix seek event ref handling 2010-06-15 15:33:37 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: prevent arithmetic overflows in pull mode buffer cache handling 2010-06-15 15:32:34 +0200 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: fix seek handling Allow a few more seek event type combinations, and really use the result of gst_segment_set_seek to perform the seek. Also add some debug. 2010-04-12 18:07:29 +0200 Edward Hervey * tests/check/elements/aacparse.c: * tests/check/elements/amrparse.c: check: Don't re-declare 'GList *buffers' in the tests It's an external which lives in gstcheck.c. Redeclaring it makes some compilers/architectures think the 'buffers' in the individual tests are a different symbol... and therefore we end up comparing holodecks with oranges. 2010-03-26 18:56:49 +0000 Arun Raghavan * gst/audioparsers/gstbaseparse.c: baseparse: Don't emit bitrate tags too early We wait to parse a minimum number of frames (10, arbitrarily) before emiting bitrate tags so that our early estimates are not wildly inaccurate for streams that start with a silence. If the stream ends before that, we just emit the tags anyway. While it _would_ be nicer to be specify the threshold to start pushing the tags in terms of duration, this would introduce more complexity than this merits. https://bugzilla.gnome.org/show_bug.cgi?id=614991 2010-03-26 18:58:35 +0100 Sebastian Dröge * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: Optionally check the overall frame checksums too before accepting a frame as valid This is optional because it's a quite expensive operation and it's very unlikely that a non-frame is detected as frame after the header CRC check and checking all bits for valid values. The overall frame checksums are mainly useful to detect inconsistencies in the encoded payload. 2010-03-26 18:42:28 +0100 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Check the CRC-8 of the headers before accepting a frame as valid This makes false-positives during seeking much less likely and detection of them much faster. 2010-03-26 18:20:24 +0100 Sebastian Dröge * gst/audioparsers/gstbaseparse.c: baseparse: Set the last stop to the buffer starttime if the duration is invalid ...instead of not setting it at all. 2010-03-26 18:19:00 +0100 Joshua M. Doe * gst/audioparsers/gstbaseparse.c: baseparse: Send NEWSEGMENT event with correct start and position Instead of taking the last stop (which could be buffer endtime instead of starttime) always take the buffer starttime. Fixes bug #614016. 2010-03-26 16:49:01 +0000 Arun Raghavan * gst/audioparsers/gstflacparse.c: flacparse: Fix buffer refcount issue When called from the GST_FLAC_PARSE_STATE_HEADERS case, gst_flac_parse_hand_headers() does a gst_buffer_set_caps() on a buffer with refcount > 1. This change handles this case by making the buffer metadata_Writable. https://bugzilla.gnome.org/show_bug.cgi?id=614037 2010-03-25 17:09:17 +0000 Tim-Philipp Müller * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: audioparsers: remove unused GstBaseParseClassPrivate structure 2010-03-25 12:55:02 +0000 Arun Raghavan * gst/audioparsers/gstflacparse.c: flacparse: Make bitrate estimation more accurate This implements the get_frame_overhead() vfunc so that baseparse can make more accurate bitrate estimates. 2010-03-25 11:48:46 +0000 Arun Raghavan * gst/audioparsers/gstaacparse.c: aacparse: Fix bitrate calculation This patch adds the get_frame_overhead() vfunc so that baseparse can accurately calculate the min/avg/max bitrates for aacparse. Note: The bitrate was being incorrectly calculated for ADTS streams (it's not in the header as the code suggests). 2010-03-25 11:22:58 +0000 Arun Raghavan * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: audioparsers: Add bitrate calculation to baseparse This makes baseparse keep a running average of the stream bitrate, as well as the minimum and maximum bitrates. Subclasses can override a vfunc to make sure that per-frame overhead from the container is not accounted for in the bitrate calculation. We take care not to override the bitrate, minimum-bitrate, and maximum-bitrate tags if they have been posted upstream. We also rate-limit the emission of bitrate so that it is only triggered by a change of >10 kbps. 2010-03-22 16:56:03 +0100 Benjamin Otte * tests/check/elements/amrparse.c: Add -Wold-style-definition and fix the warnings 2010-03-21 21:39:18 +0100 Benjamin Otte * tests/check/elements/aacparse.c: * tests/check/elements/amrparse.c: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstamrparse.c: gst_element_class_set_details => gst_element_class_set_details_simple 2010-01-14 11:50:33 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: audioparsers: rename baseparse GType name to avoid possible conflicts 2010-01-12 18:55:53 +0100 Edward Hervey * gst/audioparsers/gstflacparse.c: flacparse: Initialize variables. Fixes build on $#@*( macosx 2010-01-11 22:41:57 +0300 ������ ��������� * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstamrparse.c: win32: Include config.h before anything else. Fix mpegdemux LIBADD Because config.h defines __MSVCRT_VERSION__, which should be defined before inclusion of any system header. Also fixes mpegdemux Makefile.am LIBADD typo. Fixes #606665 2010-01-11 13:20:26 -0300 Thiago Santos * gst/audioparsers/gstaacparse.c: aacparse: Also add stream-format to template caps Do not forget to add stream-format to template caps off aacparse 2010-01-11 13:13:41 -0300 Thiago Santos * gst/audioparsers/gstaacparse.c: * tests/check/elements/aacparse.c: Rename aac's stream-format 'none' to 'raw' Renames aac's stream-format from previous commits from none to raw 2010-01-11 12:10:02 -0300 Thiago Santos * tests/check/elements/aacparse.c: aacparse: update tests to stream-format changes Updates aacparse unit tests to check for stream-format correctness as well. 2010-01-11 10:51:18 -0300 Thiago Santos * gst/audioparsers/gstaacparse.c: aacparse: Add stream-format to output caps Adds stream-format field to output caps 2010-01-05 15:05:05 +0100 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstbaseparse.c: audioparsers: documentation fixes 2010-01-05 15:04:38 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: add documentation 2010-01-05 14:48:49 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: add documentation 2009-12-21 18:29:43 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: perform additional frame checks when resyncing 2010-01-05 16:35:52 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: fix (multiple channel) frame parsing 2010-01-05 16:35:44 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: declare unparsed input and parsed output 2009-12-21 18:19:23 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: fix scanning for next syncword 2009-12-21 18:18:39 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: adjust seek handling and newsegment sending Perform sanity check on type of seek, and only perform one that is appropriately supported. Adjust downstream newsegment event to first buffer timestamp that is sent downstream. 2009-12-21 11:59:45 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: minor refactor cleanup Also add some debug logging. 2009-12-18 21:05:11 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: locate next sync code more efficiently 2009-12-18 21:04:12 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: baseparse takes care of handling leftover pieces 2009-12-18 21:02:40 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: implement leftover draining in pull mode 2009-12-17 12:45:36 +0100 Mark Nauwelaerts * gst/audioparsers/gstflacparse.c: flacparse: set _OFFSET and _OFFSET_END on outgoing buffers 2009-12-17 12:44:20 +0100 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: * gst/audioparsers/plugin.c: audioparsers: move 'flacparse' into it 2009-12-16 18:38:33 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: provide default conversion using bps if no fps available Also store estimated duration as such, rather than pretending otherwise (e.g. set by subclass). 2009-12-18 13:30:29 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: check for remaining data when draining in push mode 2009-12-18 13:30:07 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: baseparse: fix pull mode cache size comparison 2009-12-18 13:01:17 +0100 Edward Hervey * gst/audioparsers/gstac3parse.c: ac3parse: Fix unitialized variable. 2009-12-17 14:46:01 +0000 Christian Schaller * gst/audioparsers/Makefile.am: Update spec file and fix ac3parser header listing in Makefile.am 2009-12-11 10:25:16 -0800 Michael Smith * gst/audioparsers/gstbaseparse.c: audioparse: fix a format string as reported on irc. 2009-11-23 16:34:50 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: ensure sufficient data available for parsing 2009-10-29 15:19:04 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: extract and use some more details for Enhanced Ac-3 streams 2009-10-29 15:18:37 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: baseparse: custom bufferflag indicates not to count frame in stats 2009-10-28 14:08:43 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: perform additional frame checks when resyncing 2009-10-28 14:07:17 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: inform base parser of frame duration 2009-10-27 16:16:50 +0100 Mark Nauwelaerts * gst/audioparsers/gstac3parse.c: ac3parse: improve src caps settings 2009-11-27 17:59:03 +0100 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: * gst/audioparsers/plugin.c: ac3parse: initial version MARGINAL rank for now; might take some time for some (useful) framed=true/false to appear here and there. 2009-11-26 18:34:45 +0100 Mark Nauwelaerts * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstamrparse.h: amrparse: use (default) time handling of baseparser class 2009-11-26 18:15:21 +0100 Mark Nauwelaerts * gst/audioparsers/Makefile.am: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstamrparse.h: * gst/audioparsers/plugin.c: audioparsers: move 'amrparse' into it 2009-11-27 17:27:32 +0100 Mark Nauwelaerts * gst/audioparsers/gstbaseparse.c: audioparsers: reference GstBaseParse now lives here 2009-11-28 18:13:31 +0100 Mark Nauwelaerts * gst/aacparse/Makefile.am: * gst/audioparsers/Makefile.am: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: * gst/audioparsers/gstbaseparse.c: * gst/audioparsers/gstbaseparse.h: * gst/audioparsers/plugin.c: audioparsers: rename 'aacparse' plugin to generic 'audioparsers' plugin 2009-11-26 17:04:43 +0100 Mark Nauwelaerts * gst/aacparse/Makefile.am: * gst/aacparse/gstaacparse.c: * gst/aacparse/plugin.c: aacparse: separate plugin registration and rename plugin 2009-11-26 17:04:36 +0100 Mark Nauwelaerts * gst/aacparse/gstaacparse.c: aacparse: ensure sufficient data available before accessing 2009-11-05 14:31:40 +0100 Mark Nauwelaerts * gst/aacparse/gstaacparse.c: * gst/aacparse/gstaacparse.h: aacparse: use (default) time handling of baseparser class 2009-10-29 15:19:35 +0100 Mark Nauwelaerts * gst/aacparse/gstaacparse.c: aacparse: fixup comments to C-style 2009-10-29 16:05:00 +0100 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: reset passthrough mode to default (disabled) on activation 2009-10-29 15:16:59 +0100 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: ensure buffer metadata is writable 2009-10-28 14:06:13 +0100 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: * gst/aacparse/gstbaseparse.h: baseparse: fix/enhance DISCONT marking In particular, consider DISCONT == !sync, and allow subclass to query sync state, as it may want to perform additional checks depending on whether sync was achieved earlier on. Also arrange for subclass to query whether leftover data is being drained. 2009-11-23 15:48:25 +0100 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: * gst/aacparse/gstbaseparse.h: baseparse: add timestamp handling, and default conversion In particular, (optionally) provide baseparse with a notion of frames per second (and therefore also frame duration) and have it track frame and byte counts. This way, subclass can provide baseparse with fps and have it provide default buffer time metadata and conversions, though subclass can still install callbacks to handle such itself. 2009-10-28 12:02:03 +0100 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: documentation fixes 2009-10-28 12:00:08 +0100 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: use_fixed_caps for src pad After all, stream is as-is, and there is little molding to downstream's taste that can be done. If subclass can and wants to do so, it can still override as such. 2009-11-20 17:32:13 +0100 Julien Moutte * gst/aacparse/gstbaseparse.c: aacparse: Fix compilation warnings 2009-10-11 11:22:11 +0200 Josep Torra * gst/aacparse/gstaacparse.c: * gst/aacparse/gstbaseparse.c: aacparse: fix warnings in macosx snow leopard 2009-09-25 17:02:53 +0200 Mark Nauwelaerts * gst/aacparse/gstaacparse.c: * gst/aacparse/gstbaseparse.c: * gst/aacparse/gstbaseparse.h: aacparse: forego (bogus) parsing of already parsed (raw) input 2009-08-07 13:07:17 +0200 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: prevent infinite loop when draining 2009-08-07 13:06:28 +0200 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: fix minor memory leak 2009-07-14 14:08:04 +0200 Sebastian Dröge * gst/aacparse/gstbaseparse.c: * gst/aacparse/gstbaseparse.h: aacparse: Add function for the baseparse subclass to push buffers downstream Also handle the case gracefully where the subclass decides to drop the first buffers and has no caps set yet. It's still required to have valid caps set when the first buffer should be passed downstream. 2009-07-14 14:07:44 +0200 Sebastian Dröge * gst/aacparse/gstbaseparse.c: baseparse: Fix seek event leaking 2009-06-18 12:13:28 +0200 Mark Nauwelaerts * gst/aacparse/gstaacparse.c: aacparse: ADIF: do not send bogus timestamps, leave to downstream (decoder) 2009-06-01 15:53:27 +0100 Tim-Philipp Müller * gst/aacparse/gstaacparse.c: aacparse: fix sample rate extraction from codec data In one case we extracted the sample rate index from the codec data and saved it as sample rate rather than getting the real sample rate from the table. Fix that, and also make sure we don't access non-existant table entries by adding a small helper function that guards against out-of-bounds access in case of invalid input data. 2009-06-01 14:02:33 +0100 Tim-Philipp Müller * gst/aacparse/gstaacparse.c: aacparse, amrparse: remove bogus gst_pad_fixate_caps() calls 2009-06-01 13:56:18 +0100 Tim-Philipp Müller * gst/aacparse/gstbaseparse.c: baseparse: propagate return value of GstBaseParse::set_sink_caps() gst_base_parse_sink_setcaps() presumably should fail if the subclass returns FALSE from its ::set_sink_caps() function. 2009-06-01 13:47:01 +0100 Tim-Philipp Müller * gst/aacparse/gstbaseparse.c: baseparse: don't try to GST_LOG an already-freed caps string The proper way to log caps is via GST_PTR_FORMAT anyway. 2009-06-01 13:05:35 +0100 Tim-Philipp Müller * gst/aacparse/gstaacparse.c: * tests/check/elements/aacparse.c: aacparse: set channels and rate on output caps, and keep codec_data Create output caps from input caps, so we maintain any fields we might get on the input caps, such as codec_data or rate and channels. Set channels and rate on the output caps if we don't have input caps or they don't contain such fields. We do this partly because we can, but also because some muxers need this information. Tagreadbin will also be happy about this. 2009-05-26 19:43:53 +0200 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: fix debug category 2009-04-27 22:39:15 +0200 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: fix (regression in) newsegment handling (aacparse, amrparse, flacparse). Fixes #580133. 2009-04-07 04:53:02 +0300 René Stadler * gst/aacparse/gstbaseparse.c: baseparse: Fix slightly broken buffer-in-segment check (aacparse, amrparse, flacparse) 2009-04-05 03:50:19 +0300 René Stadler * gst/aacparse/gstbaseparse.c: baseparse: Fix push mode seeking (aacparse, amrparse) Sending the flush-start event forward before taking the stream lock actually works, in contrast to deadlocking in downstream preroll_wait (hunk 1). After that we get the chain function being stuck in a busy loop. This is fixed by updating the minimum frame size inside the synchronization loop because the subclass asks for more data in this way (hunk 2). Finally, this leads to a very probable crash because the subclass can find a valid frame with a size greater than the currently available data in the adapter. This makes the subsequent gst_adapter_take_buffer call return NULL, which is not expected (hunk 3). 2009-03-31 16:07:46 +0200 Mark Nauwelaerts * gst/aacparse/gstbaseparse.c: baseparse: Delay newsegment as long as possible. If newsegment is sent (too) early, caps may not yet be fixed/set, and downstream may not have been linked. 2009-03-19 01:17:25 +0200 René Stadler * gst/aacparse/gstaacparse.c: aacparse: Fix busyloop when seeking. Fixes #575388 The problem is that after a discont, set_min_frame_size(1024) is called when detect_stream returns FALSE. However, detect_stream calls check_adts_frame which sets the frame size on its own to something larger than 1024. This is the same situation as in the beginning, so the base class ends up calling check_valid_frame in an endless loop. 2009-03-19 00:32:40 +0200 René Stadler * gst/aacparse/gstaacparse.c: aacparse: Refactor check_valid_frame to expose broken code Just moving code around and removing an unhelpful/misleading comment. 2009-02-27 11:24:37 +0200 Stefan Kost * gst/aacparse/gstbaseparse.c: baseparse: revert last change and properly fix Baseparse internaly breaks the semantics of a _chain function by calling it with buffer==NULL. The reson I belived it was okay to remove it was that there is also an unchecked access to buffer later in _chain. Actually that code is wrong, as it most probably wants to set discont on the outgoing buffer. 2009-02-26 11:02:06 +0200 Stefan Kost * gst/aacparse/gstbaseparse.c: baseparse: remove checks for buffer==NULL Accordifn to docs for GstPadChainFunction buffer cannot be NULL. If we would leave the check, we would also need more such check below. 2009-02-11 00:15:43 +0200 René Stadler * gst/aacparse/gstaacparse.c: aacparse: Fix license specified in plugin details. 2009-01-30 18:18:10 +0000 Jan Schmidt * gst/aacparse/gstbaseparse.c: Fix the return value of the default parse_frame function. Fix the return value of the default parse_frame function in both copies of GstBaseParse 2009-01-23 16:00:10 +0200 Stefan Kost * gst/aacparse/gstaacparse.c: Log aac details found in codec_data. 2008-11-13 17:24:58 +0000 Wim Taymans gst/aacparse/gstaacparse.c: Don't autoplug aacparse until it works. Original commit message from CVS: * gst/aacparse/gstaacparse.c: (plugin_init): Don't autoplug aacparse until it works. 2008-11-13 15:20:15 +0000 Stefan Kost tests/check/: Add unit tests for new parsers. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/aacparse.c: * tests/check/elements/amrparse.c: Add unit tests for new parsers. 2008-11-13 14:21:39 +0000 Stefan Kost gst/: Fix baseparse type name. Original commit message from CVS: * gst/aacparse/gstbaseparse.c: * gst/amrparse/gstbaseparse.c: Fix baseparse type name. 2008-11-13 12:59:34 +0000 Stefan Kost Add two new baseparse based parsers (aac and amr) from Bug #518857. Original commit message from CVS: * configure.ac: * gst/aacparse/Makefile.am: * gst/aacparse/gstaacparse.c: * gst/aacparse/gstaacparse.h: * gst/aacparse/gstbaseparse.c: * gst/aacparse/gstbaseparse.h: * gst/amrparse/Makefile.am: * gst/amrparse/gstamrparse.c: * gst/amrparse/gstamrparse.h: * gst/amrparse/gstbaseparse.c: * gst/amrparse/gstbaseparse.h: Add two new baseparse based parsers (aac and amr) from Bug #518857. 2011-03-20 01:08:38 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Make src_query MT-safe It is possible that the element might be going down while the event arrives 2011-04-08 15:22:47 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Unref event if the parent element disappeared 2011-04-08 15:22:19 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Unref event if the parent element disappeared 2011-03-21 16:04:34 +0100 Havard Graff * ext/jpeg/gstjpegdec.c: jpegdec: Make upstream events MT-safe 2011-03-21 16:04:34 +0100 Havard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Make upstream events MT-safe 2011-04-08 15:20:51 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: rtp: Unref events if the parent element disappeared 2011-01-06 18:24:36 +0100 Ole André Vadla Ravnås * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: rtpmanager: fix pad callbacks so they handle when parent goes away 1) We need to lock and get a strong ref to the parent, if still there. 2) If it has gone away, we need to handle that gracefully. This is necessary in order to safely modify a running pipeline. Has been observed when a streaming thread is doing a buffer_alloc() while an application thread sends an event on a pad further downstream, and from within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing while the streaming thread has its buffer_alloc() in progress. 2010-11-26 15:20:04 +0100 Havard Graff * gst/rtpmanager/gstrtpsession.c: rtpsession: make iterate_internal_links MT-safe 2011-04-08 14:35:04 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: Revert "Pulsesink: Allow chunks up to bufsize instead of segsize" This reverts commit 1e2c1467ae042a3c6bb1a6bc0c07aeff13ec5edb. The commit causes pulsesink to ignore the latency-time baseaudiosink property. 2011-04-08 11:13:07 +0200 Alexey Fisher * gst/rtp/gstrtpspeexpay.c: rtpspeexpay: Do not transmitt samples with GAP flag If we get GAP samples, there is no need to transmitt it. In some situations, microphone is muted, we can drop net traffick usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s 2011-04-08 11:11:58 +0200 Alexey Fisher * ext/speex/gstspeexenc.c: speexenc: Use speex intern silence detection Speex has build in silence detection. If speex_encode_int returns 0, than there is silence and sample do not need to be transmitted. This work only if vbr=1 and dtx=1 optionas are enabled. So if we get 0, we add GAP flag to the sample. 2011-04-07 19:04:33 +0200 Wim Taymans * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: rtp: port some pay/depayloaders 2011-04-05 19:15:11 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: udpsink: handle scather gather from buffers Iterate the memory blocks on the buffer and send them using sendmsg. 2011-04-05 17:26:44 +0200 Wim Taymans * gst/rtsp/gstrtpdec.c: rtpdec: reset structure before use 2011-04-05 17:20:08 +0200 Wim Taymans Merge branch 'master' into 0.11 Conflicts: gst/rtsp/gstrtspsrc.c 2011-04-05 17:12:28 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle * control correctly Parse session control attributes when no media control attribute is present. Threat * control attributes as an empty string, just like the spec says. Fixes #646800 2011-04-05 17:06:41 +0200 Wim Taymans * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: rtsp/udp: port to 0.11 2011-04-05 14:28:54 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Add support for A-Law and µ-Law Fixes bug #646567. 2011-04-05 09:44:01 +0200 Jon Nordby * configure.ac: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: Fix build with jack 0.120.1 9544622674c0d0a3147a9b51145159b02eec68e9 checked for 0.120.2 and later, but the deprecation was introduced in 0.120.1 2011-04-05 11:13:36 +0200 Wim Taymans * gst/avi/gstavisubtitle.c: avi: more porting to 0.11 2011-04-05 12:05:19 +0300 Stefan Kost * sys/v4l2/gstv4l2radio.h: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2xoverlay.c: docs: fix docuemntation warnings (and reindent) 2011-04-04 19:17:43 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: avi: port to 0.11 API 2011-04-04 17:34:17 +0200 Alessandro Decina * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: videomixer: update orc dist files 2011-04-04 15:57:10 +0300 Stefan Kost * common: Automatic update of common submodule From 1ccbe09 to c3cafe1 2011-03-01 14:08:12 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Always call pa_stream_new_with_proplist() pa_stream_new_with_proplist() can take a NULL proplist, so we don't need to concern ourselves with whether it's NULL or not. 2011-04-04 11:33:10 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: perform post-flush state tricks downstream to upstream ... so downstream is set when upstream resumes data flow. 2011-04-04 11:27:29 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: distribute new base_time to manager children following flush seek ... by forcing a state changed to PLAYING, which should otherwise be a no-op as elements should already be in that state. In particular, jitterbuffer needs new base_time as soon as possible to perform proper timing (e.g. eos timeout handling) and can't wait for the new base_time that will be distributed when the whole pipeline returns to PLAYING. See bug #646397. 2011-04-04 11:35:59 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: Revert "jitterbuffer: reset element base_time upon flush" This reverts commit f84b8a69cba9c538f5546869cb4ef454ad5efb9d. Fixes bug #646397. 2011-04-04 10:31:44 +0100 Zaheer Abbas Merali * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: flv: Specify the only possible stream-format for h264 in the pad templates. 2011-04-04 10:07:42 +0200 Sebastian Dröge * gst/qtdemux/qtdemux.c: qtdemux: Check for invalid (empty) classification info entity strings Otherwise the classification string can be empty and gst_tag_list_add() will complain or have a \0 in the first four bytes, which is wrong too. 2011-04-04 10:01:26 +0200 Sebastian Dröge * gst/qtdemux/qtdemux.c: qtdemux: Year 0 is not a valid year for GDate and the proleptic gregorian calendar 2011-04-01 13:18:55 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Add support for writing METADATA_BLOCK_PICTURE blocks for GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE 2011-04-01 11:33:54 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: videomixer[2]: Use orc_memset() instead of memset() 2011-01-19 18:06:45 -0700 Lane Brooks * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Add transparent background option for alpha channel formats 2011-01-19 12:07:17 -0700 Lane Brooks * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/blendorc.orc: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer2: Add transparent background option for alpha channel formats This option allows the videomixer2 element to output a valid alpha channel when the inputs contain a valid alpha channel. This allows mixing to occur in multiple stages serially. The following pipeline shows an example of such a pipeline: gst-launch videotestsrc background-color=0x000000 pattern=ball ! video/x-raw-yuv,format=\(fourcc\)AYUV ! videomixer2 background=transparent name=mix1 ! videomixer2 name=mix2 ! ffmpegcolorspace ! autovideosink videotestsrc ! video/x-raw-yuv,format=\(fourcc\)AYUV ! mix2. The first videotestsrc in this pipeline creates a moving ball on a transparent background. It is then passed to the first videomixer2. Previously, this videomixer2 would have forced the alpha channel to 1.0 and given a background of checker, black, or white to the stream. With this patch, however, you can now specify the background as transparent, and the alpha channel of the input will be preserved. This allows for further mixing downstream, as is shown in the above pipeline where the a second videomixer2 is used to mix in a background of an smpte videotestsrc. So the result is a ball hovering over the smpte test source. This could, of course, have been accomplished with a single mixer element, but staged mixing is useful when it is not convenient to mix all video at once (e.g. a pipeline where a foreground and background bin exist and are mixed at the final output, but the foreground bin needs an internal mixer to create transitions between clips). Fixes bug #639994. 2011-03-31 13:25:00 +0200 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: also uncork during EOS waiting (and after EOS is rendered) Pulsesink was recently changed to defer uncorking until there is data to write. This condition will however never occur when EOS in being rendered (since that marks the end of data). Changing to PAUSED state while EOS is being waited on results in a hang: pausing corks the stream, which will never be undone since there is no more data when going back to PLAYING. If pulsesink is the clock provider, deadlock ensues since time doesn't continue in corked state and the clock id for EOS wait never fires. Fixes #645961. 2011-03-29 16:33:43 +0200 Sebastian Dröge * tests/check/elements/rtpbin.c: rtpbin: Don't try to request the same request pad twice 2011-03-28 23:46:47 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: fix issues with large metadata blocks when streaming unframed flac Parse metadata blocks when handling unparsed flac in push mode. This works around a bunch of issues with the flac decoder when handling metadata blocks that are larger than the max. flac framesize, which coverart blocks often are. We need to have all the data for these blocks available when we pass data to libflac. http://gstreamer-devel.966125.n4.nabble.com/Flac-files-that-will-playback-but-not-stream-td3338198.html#a3395276 https://bugzilla.gnome.org/show_bug.cgi?id=566769 2011-03-28 21:05:31 +0200 Wim Taymans * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: plugins: port to new memory API 2011-03-28 20:50:59 +0200 Wim Taymans Merge branch 'master' into 0.11-fdo 2011-03-27 21:39:50 +0200 Jan Urbański * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Do not build an index if upstream is not seekable An index is not useful if upstream cannot handle seeks and building it for infinite files, for instance FLV streams, results in a memory leak. 2011-03-27 01:19:58 +0300 Alexey Chernov <4ernov@gmail.com> * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-video4linux2.xml: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2radio.c: * sys/v4l2/gstv4l2radio.h: v4l2: new v4l2radio element to control analog radio devices https://bugzilla.gnome.org/show_bug.cgi?id=640118 2011-03-25 22:22:43 +0100 Sebastian Dröge * common: Automatic update of common submodule From 193b717 to 1ccbe09 2011-03-25 14:56:06 +0200 Stefan Kost * common: Automatic update of common submodule From b77e2bf to 193b717 2011-03-25 12:53:43 +0200 Stefan Kost * ext/cairo/Makefile.am: cairo: fix the name of the *-marshall.list file to unbreak make distcheck 2011-03-25 09:31:03 +0100 Sebastian Dröge * common: Automatic update of common submodule From d8814b6 to b77e2bf 2011-03-25 09:06:16 +0100 Sebastian Dröge * common: Automatic update of common submodule From 6aaa286 to d8814b6 2011-03-25 00:10:56 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: refactor processing loop for block based operation Previously the chain function was working sample frame based. In each cycle it was checking if it is time to run a fft or if it is time to send a message. Now we changed the data transform functions to work on a block of data and calculate the max length until either {end-of-data, do-fft, do-msg}. This allows us also to avoid the duplicated code for the single and multi-channel case (as the transformers have the same signature now). 2011-03-24 23:47:33 +0200 Stefan Kost * configure.ac: jack: unbreak the build for jack2 users Jack2 (versions 1.X.X) does only have that API in svn. Limmit the use of the new API for jack1 versions. 2011-03-24 18:49:19 +0200 Stefan Kost * common: Automatic update of common submodule From 6aec6b9 to 6aaa286 2011-03-24 14:14:09 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: fix the error accumulation and frames_todo handling Even though we wrap around the accumulated second, we still need to add the error in the same cycle. Increase the todo in the same conditional as afterwards the accumulated error will be below one second. 2011-03-24 13:53:12 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: fix broken code resulting for a wrong splitup of changes 2011-03-22 16:29:53 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: simplify the have_interval calculation Move some of the conditions to the places where the dependent variables change. 2011-03-22 16:26:45 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: use local var for input_data function Avoid dereferencing the input_data from the instance from within an inner loop. 2011-03-23 16:34:16 +0100 Sebastian Dröge * ext/speex/gstspeexdec.c: * ext/speex/gstspeexdec.h: speexdec: Get and use streamheader from the caps if possible This allows playback of streams where the streamheader buffers were dropped from the stream for some reason. 2011-03-22 19:36:31 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: use running time for synchronization Fixes #432612. 2011-03-22 19:36:21 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: use running time for synchronization Fixes #432612. 2011-03-22 19:35:58 +0100 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: use running time for synchronization See bug #432612. 2011-03-22 12:53:22 +0100 Luis de Bethencourt * configure.ac: configure.ac: redundant uses of AC_MSG_RESULT() cleaned the redundant uses of AC_MSG_RESULT() in configure.ac 2011-03-18 19:34:57 +0100 Luis de Bethencourt * autogen.sh: autogen: wingo signed comment 2011-03-16 10:43:47 +0100 Robert Swain * ext/jack/gstjackaudiosink.c: jackaudiosink: Fix typo from 9544622674c0d0a3147a9b51145159b02eec68e9 2011-03-16 09:38:43 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: Mark tag mapping tables as static const 2011-03-16 09:37:58 +0100 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Use ARTIST instead of AUTHOR for GST_TAG_ARTIST 2011-03-16 09:35:50 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Use ARTIST Matroska tag instead of AUTHOR for GST_TAG_ARTIST AUTHOR only existed in an old version of the spec and ARTIST is the new replacement for this. We are still reading both to still be compatible with old files. Fixes bug #644875. 2011-03-15 20:19:48 +0000 Tim-Philipp Müller * tests/check/elements/videofilter.c: tests: enable more formats in videofilter unit test, check more resolutions 2011-03-14 19:14:07 -0400 Youness Alaoui * gst/videofilter/gstvideoflip.c: videoflip: Fix buffer overflow bug for odd resolutions and Y422 colorspaces https://bugzilla.gnome.org/show_bug.cgi?id=644773 2011-03-15 19:36:01 +0200 Vincent Penquerc'h * ext/speex/gstspeexdec.c: speexdec: silence warning message when appropriate If we did not know how many frames to expect, then we get an unexpected end of stream when trying to decode more frames that are there, if there are leftover bits to pad to the next byte 2011-03-14 19:14:07 -0400 Youness Alaoui * gst/videofilter/gstvideoflip.c: videoflip: Add support for YUY2, UVYV and YVYU colorspaces https://bugzilla.gnome.org/show_bug.cgi?id=644773 2011-03-15 09:43:35 +0000 Tim-Philipp Müller * tests/check/elements/videofilter.c: tests: in videofilter unit test also check with 'odd' widths and heights And only use one test suite. 2011-03-14 19:28:07 +0100 Sebastian Dröge * ext/speex/gstspeexdec.c: speexdec: Always process the number of frames per packet as specified in the header Looking at the remaining bits in the bitstream after decoding a single frame can't be used as loop condition. The remaining bits might not give a complete frame and the speex decoder will then output nothing but access uninitialized memory, which leads to valgrind warnings. Fixes bug #644669. 2011-03-14 15:46:50 +0100 Andoni Morales Alastruey * gst/matroska/matroska-mux.c: matroskamux: return TRUE from sink pad event function for tag events, which are handled https://bugzilla.gnome.org/show_bug.cgi?id=644730 2011-03-12 00:44:31 +0530 Philip Jägenstedt * ext/pulse/pulsesink.c: pulsesink: Better fix for deadlock on failed connect This reverts the previous fix that would cause a double-unlock when the stream connect failed. https://bugzilla.gnome.org/show_bug.cgi?id=644510 2011-03-11 23:06:31 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Fix deadlock if connecting to PA fails Commit dd4ec22e introduced a deadlock in the failure path while trying to connect to PulseAudio. This makes sure we drop the lock on the resource mutex to avoid this. https://bugzilla.gnome.org/show_bug.cgi?id=644510 2011-03-11 16:59:10 +0200 Stefan Kost * tests/check/Makefile.am: tests: order state-test blacklist and add jack elements Jack audio src/sink elements recently got moved from bad and should be excluded from the test (like the other device specific source and sinks). Fixes #644288 2011-03-11 13:47:26 +0100 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Chain up to the parent class' ::send_event for non-seek events 2011-03-11 13:46:05 +0100 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Fix refcount issues with the seek event Fixes bug #642963. 2011-03-11 09:54:02 +0000 Tim-Philipp Müller * ext/pulse/pulsesink.c: docs: fix pulsesink gtk-doc markup 2011-03-11 10:29:08 +0100 Philippe Normand * configure.ac: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: fix build against jack 0.120.2 jack_port_get_total_latency() has been deprecated in favor of jack_port_get_latency_range(). https://bugzilla.gnome.org/show_bug.cgi?id=644477 2011-03-10 14:29:25 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: more comments and tune and logging 2011-03-10 14:15:42 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: avoid unneccesary extra fft runs Before it was possible that we run an extra fft when the time for sending a new message is due. Only do this if we have not run the fft for the interval at all. 2011-03-10 14:12:01 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: only scale the vectors that we are processing Phase is not produced by default, so lets not scale it unconditionally to save a few cycles. 2011-03-10 14:10:25 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: put number of channels to instance variable When freeing data the format might have changed. Thus we need to remember for which format we allocated memory. 2011-03-10 10:27:14 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: update doc review stamp 2011-03-10 10:22:29 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: use function pointers for data readers Don't check the format for each sample frame to read. We can make that decission in _setup already. This is still not ideal as we call the function per frame. Ideally we determine how many samples we can copy and have a loop in the input reader. As an alternative we might also consider to use the fft variants for the various formats and not convert to float for all cases - we would still need to mix or deinterleave though. 2011-03-09 17:07:47 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: improve recovery from failed seek In case server-side fails to perform seek, i.e. PLAY at non-zero requested position, recovery so far would arrange for streaming to continue, albeit having lost position tracking in the process. So, query position prior to seek and use upon failed seek. 2011-03-09 16:51:00 +0100 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: handle position query 2011-03-09 16:57:28 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: multi-channel support Add a boolean multi-channel property with a default of FALSE. When set to TRUE the element won't mix all input channels to mono, but instead run a FFT on each channel. In that case the result message would contain a 2 dimensional array of channel x data for magnitude and phase. API: GstSpectrum:multi-channel https://bugzilla.gnome.org/show_bug.cgi?id=593482 2011-03-09 16:55:56 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: more xrefs in the docs 2011-03-09 12:41:15 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: factor out the code that accumulated samples into the ring-buffer Use a separate function to read a sample frame into a ringbuffer slot. In the future we can use format-specific function pointer to avoid the reoccuring format checks. 2011-03-09 12:38:52 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: pull format to temp var to improve readability of lines using it 2011-03-09 12:20:11 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: code cleanup for copying data to ring-buffer Rename fp to is_float and restructure if-else part for handling the different formats. 2011-03-09 11:40:48 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: add a GstSpecrtumChannel context structure We now keep the fft data that is related to one channel in a separate structure to prepare for multichannel support. We also refactor the code to operate more often on the channel context. 2011-03-09 11:18:19 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: call the instance var spectrum instead of filter 2011-03-09 11:14:37 +0200 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: don't value we already took from the gvalue 2011-03-08 17:26:17 +0000 Wim Taymans Merge branch 'master' into 0.11 Conflicts: configure.ac 2011-03-08 17:02:30 +0000 Wim Taymans * gst/debugutils/efence.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/ximage/ximageutil.c: meta: update for new API 2011-03-08 16:28:27 +0000 Tim-Philipp Müller Merge ad-hoc release branch '0.10.28' === release 0.10.28 === 2011-03-08 15:47:52 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.28 Ad-hoc release to fix build issue with newer kernels. 2011-03-03 00:16:47 +0000 Tim-Philipp Müller * sys/v4l2/v4l2_calls.h: v4l2: remove unnecessary linux/videodev.h include Causes compilation issues with newer kernel headers where the old v4l interface has been removed. https://bugzilla.gnome.org/show_bug.cgi?id=643716 2011-03-08 10:14:20 +0000 Wim Taymans Merge branch 'master' into 0.11 Conflicts: tests/examples/cairo/Makefile.am 2011-03-07 16:56:43 +0100 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: also estimate eos if very near eos 2011-03-07 16:56:18 +0100 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: avoid trying to buffer more than is available. That is, in case of short (or near eos of) stream, deadlock (until timeout) would occur trying to buffer more than is yet forthcoming. 2011-03-07 11:01:06 +0100 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reset element base_time upon flush ... to arrange for properly scheduled timeout (following seek). 2011-03-07 10:54:22 +0100 Sebastian Dröge * tests/examples/cairo/cairo_overlay.c: cairooverlay: Add a bus handler to the example to handle EOS/ERROR/WARNING Also clean up the pipeline properly. 2011-03-07 10:47:23 +0100 Sebastian Dröge * tests/examples/Makefile.am: examples: Always dist the cairo example 2011-03-07 10:46:12 +0100 Sebastian Dröge * tests/examples/cairo/Makefile.am: cairooverlay: Use LDADD instead of LDFLAGS for libs and add $(GST_LIBS) 2011-03-05 23:22:58 +0000 Jon Nordby * tests/examples/Makefile.am: * tests/examples/cairo/Makefile.am: * tests/examples/cairo/cairo_overlay.c: cairooverlay: Remove unnecessary gtk/gtk-x11 use in example. This removes code, and allows the example to be used on any platform. Fixes bug #643981. 2011-03-04 18:37:38 -0800 David Schleef * sys/v4l2/gstv4l2object.c: v4l2: Use #ifdefs for V4L2_PIX_FMT_PJPG It's only recently added to kernel headers. 2011-02-23 16:50:43 +0100 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: tune output max buffer size to material ... to avoid ending up with tons of short time buffers for e.g. high sample rate audio. 2011-03-04 17:04:37 +0100 Wim Taymans * tests/examples/cairo/Makefile.am: examples: don't use hardcodec 0.10 2011-03-04 16:30:36 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-03-04 15:50:01 +0200 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: add a doc example for setting stream-properties 2011-03-04 15:42:19 +0200 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: fix the xml in the docs 2011-03-03 00:16:47 +0000 Tim-Philipp Müller * sys/v4l2/v4l2_calls.h: v4l2: remove unnecessary linux/videodev.h include Causes compilation issues with newer kernel headers where the old v4l interface has been removed. https://bugzilla.gnome.org/show_bug.cgi?id=643716 2011-03-02 23:21:15 +0100 Sebastian Dröge * configure.ac: * tests/examples/Makefile.am: * tests/examples/cairo/Makefile.am: * tests/examples/cairo/cairo_overlay.c: cairooverlay: The example always requires gtk-x11 Check for gtk-x11 and only build the example if it's available. 2011-03-02 23:14:36 +0100 Sebastian Dröge * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: cairooverlay: Some minor cleanup 2011-03-02 23:09:21 +0100 Sebastian Dröge * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-deinterlace.xml: docs: Update inspected plugin data 2011-01-28 02:14:04 +0200 Jon Nordby * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/cairo/.gitignore: * ext/cairo/Makefile.am: * ext/cairo/gstcairo-marshal.list: * ext/cairo/gstcairo.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairooverlay.h: * tests/examples/Makefile.am: * tests/examples/cairo/.gitignore: * tests/examples/cairo/Makefile.am: * tests/examples/cairo/cairo_overlay.c: cairooverlay: Add generic Cairo overlay video element. Allows applications to connect to the "draw" signal of the element and do their custom drawing there. Includes an example application demonstrating usage. Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=595520 2011-03-02 13:00:31 +0200 Stefan Kost * gst/monoscope/monoscope.c: monoscope: don't leak the monoscope_state data The monoscope_close() implementation was empty. 2011-03-02 12:59:35 +0200 Stefan Kost * gst/monoscope/monoscope.c: monoscope: we have 64 colors, don't access colors[64] Fixes remaining invalid read. 2011-03-02 10:25:29 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: arrange for non-fatal error when parsing non-vital parts 2011-03-02 10:56:33 +0200 Stefan Kost * gst/monoscope/convolve.c: monoscope: stack needs to be size+1 as we put a end-marker into it Valgrind is still complaining about one bad read, but this takes care of the crash mentioned in the comment and in bug #564122. 2011-03-01 22:40:19 +0200 Stefan Kost * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: example: fix the variable name for the ip-address Fix the name in the launch pipeline and use a value of "localhost" by default. 2011-02-28 19:16:00 +0100 Mark Nauwelaerts * configure.ac: configure.ac: cygwin/mingw; enable plugin linking to static lib Useful for DirectX plugin(s). Fixes #642507. 2011-02-28 19:13:41 +0100 Mark Nauwelaerts * configure.ac: configure.ac: export plugin description more platform independent Fixes #642504. 2011-02-28 18:32:54 +0100 Mark Nauwelaerts * common: Automatic update of common submodule From 1de7f6a to 6aec6b9 2011-02-28 13:29:47 +0100 Wim Taymans Merge branch 'master' into 0.11 2011-02-28 13:28:29 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: use NetAddress metadata 2011-02-28 13:14:37 +0100 Wim Taymans * gst/udp/gstdynudpsink.c: * gst/udp/gstudp.c: * gst/udp/gstudpsrc.c: udp: implement NetAddress with metadata 2011-02-28 10:16:52 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: v4l2: register metadata 2011-02-27 19:43:13 +0100 Wim Taymans * gst/debugutils/efence.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/v4l2src_calls.c: * sys/ximage/gstximagesrc.c: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: meta: fix for new API 2011-02-25 16:29:38 +0100 Wim Taymans * gst/debugutils/efence.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/v4l2src_calls.c: * sys/ximage/gstximagesrc.c: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: metadata: use metadata for private buffer data Use buffer metadata to store element private data. 2011-02-24 13:51:32 +0100 Wim Taymans * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/v4l2src_calls.c: * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: * sys/ximage/ximageutil.c: * sys/ximage/ximageutil.h: miniobject: port to 0.11 Use buffer private data instead of subclassing. 2011-02-24 13:50:48 +0100 Wim Taymans * tests/examples/pulse/Makefile.am: * tests/examples/v4l2/Makefile.am: * tests/icles/Makefile.am: build: don't hardcode version number 2011-02-24 13:03:44 +0100 Wim Taymans * ext/taglib/gstid3v2mux.cc: id3: use boxed type instead of miniobject 2011-02-24 13:00:48 +0100 Wim Taymans * gst/debugutils/efence.c: * gst/replaygain/Makefile.am: * gst/rtpmanager/rtpsession.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstudp.c: * gst/udp/gstudpsrc.c: miniobject: use buffer private field for extra data Use the owner private field to store extra buffer data instead of using subclassing. 2011-02-24 12:23:44 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: jpegdec: add duration when extimating QoS time When we need to decide on the next QoS time, take into account the duration of the buffers. 2011-02-28 11:58:05 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: configure.ac 2011-02-23 17:41:22 +0100 Philip Jägenstedt * ext/pulse/pulsesink.c: pulsesink: release pa_shared_resource_mutex before pa_threaded_mainloop_wait Not doing so can result in a deadlock when two threads enter gst_pulseringbuffer_open_device at the same time, as pa_threaded_mainloop_wait releases the mainloop lock while waiting, allowing another thread to take it, resulting in a deadlock as two threads waits for the lock the other is holding. https://bugzilla.gnome.org/show_bug.cgi?id=643087 2011-02-23 17:18:19 +0100 Philip Jägenstedt * ext/pulse/pulsesink.c: pulsesink: s/ressource/resource/ https://bugzilla.gnome.org/show_bug.cgi?id=643087 2011-02-25 20:12:35 -0800 David Schleef * gst/qtdemux/qtdemux.c: qtdemux: remove accidental debug message in previous commit 2011-02-25 19:35:51 -0800 David Schleef * gst/qtdemux/qtdemux.c: qtdemux: Add support for 2Vuy and r210 2011-02-24 14:08:25 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Add support for NV21 colorspace 2011-02-24 14:00:37 +0100 Carsten Kroll * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Add support for NV12 colorspace Fixes bug #642961. 2011-02-24 13:56:04 +0100 Carsten Kroll * ext/dv/gstdvdemux.c: dvdemux: First try if upstream handles TIME seeks before handling them here Fixes bug #642963. 2010-11-08 14:25:59 +0100 Robert Swain * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Simplify setcaps The current code never uses upstream negotiation so the code can be significantly simplified. 2011-01-24 12:48:18 +0100 Robert Swain * gst/deinterlace/tvtime/greedy.c: deinterlace: Port greedyl to GstDeinterlaceSimpleMethod The main goal of this change is to reuse the complex but now neatly written scanline pointer calculation code from the simple methods. 2011-02-22 15:20:11 +0200 Stefan Kost * gst/id3demux/gstid3demux.c: Revert "id3demux: ensure a taglist before adding the container tag" This reverts commit a86bab66893bb1a3323a756410573c117b8219ef. The issue is fixed with commit ff5e5a8f0daa1fdf89792d0726ea063bbd99db18 instead. 2011-02-22 15:19:00 +0200 Stefan Kost * gst/id3demux/id3tags.c: id3demux: return ID3TAGS_BROKEN_TAG for unsupported versions This prevents us for trying to work with a NULL taglist. 2011-02-22 14:15:27 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Fix unitialized variable. 2011-02-22 14:01:27 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: ensure sane parameters when parsing superindex 2011-02-22 14:00:11 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: check for NULL audio stream format header when parsing stream 2011-02-22 14:52:18 +0200 Stefan Kost * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: rtp-examples: move capsfilter behind converters We need to have the capsfilter behin the converters to make the converters convert from the formats v4l2src can do to what we request with the capsfilter. 2011-02-22 14:50:59 +0200 Stefan Kost * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: rtp-examples: fix ascii-art Some boxes where misaligned due to long "audiotetssrc" name. Trim trailing whitespace. 2011-02-22 13:29:26 +0100 Blaise Gassend * gst/rtpmanager/gstrtpbin.c: rtpbin: handle NULL demux elements When using gstrtpbin with ignore-pt=true, the free_stream function tries to call gst_element_set_locked_state and gst_element_set_state on a stream->demux which is NULL. fixes #642412 2011-01-24 12:18:39 +0100 Robert Swain * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: small clean-ups Improve debug output by printing the buffer pointer when popping a buffer and simplify code to use scanlines.bottom_field as appropriate. https://bugzilla.gnome.org/show_bug.cgi?id=642691 2011-01-24 12:18:39 +0100 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: fix assigned method_id when using fallback https://bugzilla.gnome.org/show_bug.cgi?id=642691 2011-02-21 17:17:32 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: fix setting the SDES property Only the sdes veriable is protected with the object lock. Use the right object when setting the sdes property. 2011-02-21 12:09:07 +0100 Edward Hervey * ext/cairo/gsttextoverlay.c: * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/interleave/interleave.c: * gst/matroska/matroska-mux.c: * gst/videomixer/videomixer.c: Revert "Check that collectpads exists before removing pad" This reverts commit 8e6b876e76c94410db160afe5eb30f21452e419f. Depends on a core commit that was reverted 2011-02-21 00:55:49 +0000 Tim-Philipp Müller * gst/icydemux/gsticydemux.c: icydemux: fix tag list handling issues that might have caused crashes Fix slightly confused tag handling in some places: make it clear when we're taking ownership of a tag list and when not. For example, gst_icydemux_tag_found() was taking ownership when the source pad existed, but otherwise not (leak). Also, gst_event_parse_tag() does not return a newly-allocated taglist, but a tag list that belongs to the tag event, so don't give ownership of it away. While we're at it, some minor clean-ups: don't re-invent g_strndup() and simplify gst_icydemux_parse_and_send_tags() a bit, and don't leak the tag list in case no valid tags where found. https://bugzilla.gnome.org/show_bug.cgi?id=641330 2011-02-20 23:39:41 -0800 David Schleef * ext/cairo/gsttextoverlay.c: * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/interleave/interleave.c: * gst/matroska/matroska-mux.c: * gst/videomixer/videomixer.c: Check that collectpads exists before removing pad The core now calls release pad from finalize, at which point the collectpads might have already been freed. 2011-02-19 15:48:22 -0800 David Schleef * ext/libpng/gstpngdec.c: pngdec: Handle 16-bit-per-channel images 2011-02-18 10:12:47 +0200 Stefan Kost * gst/avi/gstavidemux.c: avidemux: stream->current_total is accumulated byte size and not time Use timestamp for the stream index as well. 2011-02-15 19:33:45 -0800 David Schleef * gst/udp/gstmultiudpsink.c: udpsink: warn when packet is too large 2011-02-17 17:59:25 -0800 David Schleef * gst/matroska/Makefile.am: * gst/matroska/matroska-parse.c: * gst/matroska/matroska-parse.h: * gst/matroska/matroska.c: matroskaparse: New element Copied from demux. Duplicates much code, also some dead code remaining. 2011-02-17 17:57:55 -0800 David Schleef * gst/matroska/matroska-demux.c: matroskademux: Earlier debug category initialization 2011-01-22 00:13:16 -0800 David Schleef * gst/flv/gstflvmux.c: flvmux: don't set duration for live stream 2011-01-06 15:44:24 -0800 David Schleef * gst/debugutils/Makefile.am: * gst/debugutils/negotiation.c: debugutils: remove bitrotten negotiation element Wasn't enabled, didn't work, and planned features have been superceded by capsfilter and capsdebug. 2010-09-17 12:10:38 -0700 David Schleef * gst/rtp/gstrtpvrawpay.c: * gst/rtp/gstrtpvrawpay.h: rtpvrawpay: Implement interlacing 2011-02-17 17:57:42 +0200 Stefan Kost * gst/avi/gstavidemux.c: avidemux: also add the frame-type for the stream index 2011-02-17 17:56:29 +0200 Stefan Kost * gst/avi/gstavidemux.c: avidemux: get the index writer id when the pad has a parent Otherwise the index writer has a weired name, as the pad has no parent yet. 2011-02-17 14:00:48 +0200 Stefan Kost * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: avidemux, flvdemux: formatting cleanup Trim trailing whitespaces and fix the formatting of double negation. 2011-02-17 13:57:37 +0200 Stefan Kost * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: avidemux, flvdemux: mark delta-units in the index We need to use the 'delta' flag for delta units and not the 'none' flag. 2011-02-17 11:58:42 +0000 Tim-Philipp Müller * tests/icles/.gitignore: .gitignore: ignore moved equalizer test binary 2011-02-17 12:46:14 +0200 Stefan Kost * gst/qtdemux/qtdemux.c: qtdemux: mark delta-unit in the index We need to use the delta flag fro delta units and not none. Print more details to the debug log. 2011-02-17 12:44:01 +0200 Stefan Kost * gst/qtdemux/qtdemux.c: qtdemux: formatting cleanup Trim trailing whitespaces and fix the formatting of double negation. 2011-02-16 17:09:20 +0200 Stefan Kost * gst/matroska/matroska-mux.c: matroskamux: rework _request_new_pad to handle explict req-pad-names Don't ignore explicit pad-names. 2011-02-16 17:06:51 +0200 Stefan Kost * gst/avi/gstavimux.c: avimux: rework _request_new_pad to handle explict req-pad-names Don't ignore explicit pad-names. Rearrange the code and the error handling a bit. Add a FIXME-0.11 for the bad pad-names. 2011-02-16 15:28:53 +0100 Sebastian Dröge * tests/icles/Makefile.am: icles: Add equalizer-test to the build system 2011-02-16 15:23:50 +0100 Sebastian Dröge * tests/icles/equalizer-test.c: [MOVED FROM BAD 5/5] equalizer-test: Initialize debug category after gst_init() to fix segfault 2007-11-07 15:36:59 +0000 Sebastian Dröge [MOVED FROM BAD 4/5] tests/icles/equalizer-test.c: Fix gain ranges for the latest equalizer changes. Original commit message from CVS: * tests/icles/equalizer-test.c: (do_slider_fiddling): Fix gain ranges for the latest equalizer changes. 2007-05-21 14:01:16 +0000 Stefan Kost [MOVED FROM BAD 3/5] ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa... Original commit message from CVS: * ChangeLog: ChangeLog surgery. * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBand, object, _GstIirEqualizerBandClass, parent_class, gst_iir_equalizer_band_set_property, gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type, gst_iir_equalizer_child_proxy_get_child_by_index, gst_iir_equalizer_child_proxy_get_children_count, gst_iir_equalizer_child_proxy_interface_init, setup_filter, gst_iir_equalizer_compute_frequencies, plugin_init): * tests/icles/equalizer-test.c: Add fixme and comment for example. 2007-03-14 16:33:03 +0000 Stefan Kost [MOVED FROM BAD 2/5] tests/icles/equalizer-test.c: Port the example to new equalizer api. Original commit message from CVS: * tests/icles/equalizer-test.c: (equalizer_set_band_value), (equalizer_set_all_band_values), (equalizer_set_band_value_and_wait), (equalizer_set_all_band_values_and_wait), (do_slider_fiddling), (main): Port the example to new equalizer api. 2007-02-03 23:35:26 +0000 Tim-Philipp Müller [MOVED FROM BAD 1/5] Fix up to use the newly ported (actually working) GstAudioFilter. Original commit message from CVS: * configure.ac: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_init), (setup_filter), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup), (plugin_init): * gst/equalizer/gstiirequalizer.h: Fix up to use the newly ported (actually working) GstAudioFilter. Bump core/base requirements to CVS for this. * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/equalizer-test.c: (check_bus), (equalizer_set_band_value), (equalizer_set_all_band_values), (equalizer_set_band_value_and_wait), (equalizer_set_all_band_values_and_wait), (do_slider_fiddling), (main): Add brain-dead interactive test for equalizer. 2011-02-15 15:59:32 -0300 Thiago Santos * sys/v4l2/gstv4l2object.c: v4l2: Add PJPG mapping Adds mapping of progressive jpeg format 2011-02-15 16:30:20 +0100 Andy Wingo plug qtdemux refcount leaks * gst/qtdemux/qtdemux.c (gst_qtdemux_src_convert): Unref the qtdemux; we weren't doing so before. (gst_qtdemux_handle_src_event, gst_qtdemux_chain): Fix some error cases which would leak a ref to the qtdemux. 2011-02-14 20:20:08 +0100 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Add URI query handler Fixes bug #642337. 2011-02-14 17:49:54 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: avoid sorting NULL array of cluster positions 2011-02-14 16:46:46 +0100 Wim Taymans * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: theorapay: handle 0 sized packets Handle 0 sized packets (repeat frame) in the payloader and depayloader. Fixes #641827 2011-02-14 15:21:29 +0200 Tuukka Pasanen * gst/debugutils/gsttaginject.c: taginject: resend tags when they are changed Allow setting new tags on the property while running and send them. Fixes #640249 2011-02-14 12:53:27 +0200 Stefan Kost * common: Automatic update of common submodule From f94d739 to 1de7f6a 2011-02-07 23:32:53 +0100 Miguel Angel Cabrera Moya * gst/rtsp/gstrtspsrc.c: rtspsrc: fix minor leaks when handling server requests. https://bugzilla.gnome.org/show_bug.cgi?id=640163 2011-02-14 00:49:00 +0000 Heath Nielson * gst/qtdemux/qtdemux.c: qtdemux: extract MusicBrainz tags Extract MusicBrainz tags added by MusicBrainz's Picard tagger application. These tags (esp. the album id) are helpful for rhythmbox et.al. to automatically downloads cover art. https://bugzilla.gnome.org/show_bug.cgi?id=642205 2011-02-14 00:38:45 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: refactor iTunes tag parsing a bit 2011-02-10 23:52:51 +0000 Tim-Philipp Müller * gst-plugins-good.doap: doap: update mailing list location 2011-02-10 18:11:46 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: propagate error during expose_streams ... as it may occur during initial parsing of fragmented file. 2011-02-10 18:00:11 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: avoid skipping exposing a stream following a removed stream 2011-02-10 11:56:33 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: store cluster positions provided by SeekHead ... and use those, if available, to locate a cluster rather than scanning. 2011-02-09 16:22:47 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: properly resume cluster scanning ... rather than getting offset tracking messed up, and then likely failing a subsequent assert. 2011-02-08 10:07:43 +0200 Stefan Kost * gst/id3demux/gstid3demux.c: id3demux: ensure a taglist before adding the container tag In the case of id3v1 also don't return NULL on empty tags, but also create a new taglist and add the container tag for consistency. 2011-02-07 17:08:47 +0200 Stefan Kost * gst/rtsp/gstrtspsrc.c: rtspsrc: strip trailing spaces 2011-02-07 17:07:42 +0200 Stefan Kost * gst/rtsp/gstrtspsrc.c: rtpsrc: set multiple properties in one go There is no need for separate g_object_set() calls here. 2011-02-03 16:10:49 -0300 Thiago Santos * gst/deinterlace/gstdeinterlace.c: * tests/check/elements/deinterlace.c: deinterlace: Handle image caps without asserting Images might have framerate=0/1 in the caps, which caused an assertion on deinterlace. I don't know of interlaced image formats but deinterlace might be hardcoded on some generic pipelines and it shouldn't assert. The fix was to set field_duration to 0 if the input has a framerate with a 0 numerator. This patch also adds checks for this situation on the unit tests. https://bugzilla.gnome.org/show_bug.cgi?id=641400 2011-02-04 12:33:09 +0200 Stefan Kost * gst/udp/gstudpsrc.c: docs: fix parameter name in udpsrc docs It is "buffer-size" and not "buffer". Also trim trailing whitespace. 2011-02-03 23:42:59 +0100 Mark Nauwelaerts * sys/v4l2/gstv4l2object.c: v4l2: fix interlaced set_format configuration Commit 6c8268dbfd5c88fac28c882ef2e4598a6522e2d6 broke recording from interlaced v4l2 source (e.g. typical tv capture card) since V4L2_FIELD_SEQ_TB (with fields stored separately) does not map to currently defined interlaced format (fields stored interleaved). Besides this mismatch, hardware might quite likely not support or appreciate this field value, since querying supported formats mapped _INTERLACED field formats to interlaced=true caps (so the latter should not be mapped to field value that is not known to be supported). 2011-02-03 18:25:00 +0000 Tim-Philipp Müller * tests/check/pipelines/lame.c: tests: add unit test for lamemp3enc negotiation issue https://bugzilla.gnome.org/show_bug.cgi?id=641151 2011-02-03 18:18:35 +0000 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: lamemp3enc: implement sinkpad get_caps() function to proxy rate and channels restrictions from downstream The element downstream of mp3enc might only accept certain sample rates or channels, make sure we relay any restrictions that do exist to upstream when it does a get_caps() on the sink pad. That way upstream elements like audioresample or audioconvert can pick a sample rate / channel configuration that will be accepted, instead of just negotiating to the highest, which might then be rejected. https://bugzilla.gnome.org/show_bug.cgi?id=641151 2011-02-02 18:27:52 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: source: fix type of ntpnstime 2011-02-02 18:21:26 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: rtpbin: Get and use the NTP time when receiving RTCP When we receive an RTCP packet, get the current NTP time in nanseconds so that we can correctly calculate the round-trip time. 2011-02-01 19:40:58 +0100 Mark Nauwelaerts * sys/directsound/gstdirectsoundsink.c: directsound: arrange for definition of _swab on Cygwin gstdirectsoundsink.c: In function 'gst_directsound_sink_write': gstdirectsoundsink.c:557: error: implicit declaration of function '_swab' gstdirectsoundsink.c:557: error: nested extern declaration of '_swab' 2010-10-06 21:17:28 -0400 Olivier Crête * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheoradepay.h: rtptheoradepay: Request new keyframe on lost packets Theora can only use the last frame (or the keyframe) as a reference, so in practice. If we receive a buffer that references an unknown codebook, request new headers. It probably means that headers were lost. 2010-08-27 14:11:53 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Add action signal to request early RTCP 2010-08-27 16:11:06 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Add callback to get the current time 2010-10-19 22:21:54 +0200 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Don't relay more than one PLI request per RTT Drop PLI requests if one was relay in the last RTT, the other side may just not have received the keyframe yet. 2010-06-23 16:43:24 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI 2010-11-24 15:27:46 -0500 Sjoerd Simons * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Fallback for FIR to PLI if PLI isn't available 2010-06-22 19:56:50 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Implement sending PLI packets in response to GstForceKeyUnit 2010-06-22 13:33:32 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsource: Retain RTCP Feedback packets for a specified amount of time 2010-09-07 13:35:16 +0300 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Make rtcp buffer metadata writable after processing it Functions that process the rtcp buffer could decide to keep a ref on the buffer for further processing. So make the metadata writable only after they are done. 2010-06-17 17:34:19 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Emit signal on incoming RTCP FB packet 2011-02-01 18:17:13 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: fix compilation 2010-06-15 18:39:47 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Add method to request early RTCP packet Implement the early mode defined in RFC 4585. In this mode, RTCP feedback packets are sent early to notifier. 2010-06-01 19:28:01 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpsession: Add property for minimum interval between Regular RTCP messages This can be changed according to RFC 4585 2010-06-14 18:40:33 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Emit signal when sending a compound RTCP packet This allows users to add extra RTCP packets to the compound RTCP packet. 2010-06-19 19:11:06 -0400 Olivier Crête * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: Tag upstream custom events with payload type 2010-06-18 19:12:40 -0400 Olivier Crete * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Tag upstream custom events with SSRC 2010-10-01 17:19:16 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Emit "on-ssrc-validated" when validating by RTCP Emit "on-ssrc-validated" if the SSRC is validated by receiving a RTCP SDES packet. 2011-02-01 16:38:20 +0100 Wim Taymans * gst/rtp/gstrtpj2kpay.c: j2kpay: skip EPH packets Include EPH markers into the previous chunk of packets. 2011-01-31 17:56:18 -0500 Olivier Crête * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: rtppcmapay: Rename the class to have the right name It was name pmca instead of pcma and made debug logs hard to search. 2011-01-31 05:58:36 +0100 David Henningsson * ext/pulse/pulsesink.c: Pulsesink: Allow chunks up to bufsize instead of segsize By allowing larger chunks to be sent, PulseAudio will have a lower CPU usage. This is especially important on low-end machines, where PulseAudio can crash if packets are coming in at a higher rate than PulseAudio can process them. Signed-off-by: David Henningsson 2011-01-31 13:44:45 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: simplify template caps We can merge all the YUV variants into one single structure. 2011-01-27 15:35:06 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: win32: fix DEFAULT_AUDIOSINK, should be direct*sound*sink https://bugzilla.gnome.org/show_bug.cgi?id=640705 2011-01-27 16:02:46 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: initialize local variable to please mingw32 compiler 2011-01-26 22:21:31 +0100 Mark Nauwelaerts * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpnetutils.h: * gst/udp/gstudpsrc.c: udp: use socklen_t where appropriate rather than custom type In particular, fixes Cygwin build where socklen_t is defined as int in line with native win32 api definition. 2011-01-27 12:16:46 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: mind rounding issues when converting from global time to mov time In particular, this avoids missing the intended keyframe when first converting from the frame's mov time to global segment time, and then back from global time to mov time when activating the segment. 2011-01-26 08:48:43 +0000 Ognyan Tonchev * gst/matroska/ebml-write.c: * tests/check/elements/matroskamux.c: matroskamux: don't leak ebml writer caps when re-using matroskamux https://bugzilla.gnome.org/show_bug.cgi?id=640542 2011-01-25 21:56:19 +0200 Stefan Kost * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: don't divide by 0 2011-01-18 14:48:04 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: pull mode should always report seekable ... as it no longer requires an index, but can seek by scanning as well. 2011-01-10 12:34:22 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: support some more mpeg-4 fourcc variants 2011-01-10 12:34:03 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: simplify retrieving stsd child entry atom 2011-01-24 18:27:52 +0100 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Don't consider 0 fcc_handler as uncompressed. Just avoids a warning 2011-01-20 12:14:08 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: take configured start time into account when creating the newsegment event, take the configured start time into account. 2011-01-24 15:11:02 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: fix printf format warning on mingw32 Make win32 build bot happy again, and nicefy output while we're at it. qtdemux.c: In function 'qtdemux_parse_trun': qtdemux.c:2162:3: error: format '%lu' expects type 'long unsigned int', but argument 9 has type 'guint32' 2011-01-24 13:39:58 +0000 Tim-Philipp Müller * tests/examples/rtp/client-H263p-AMR.sh: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-PCMA.sh: examples: autoaudisink -> autoaudiosink in RTP examples 2011-01-24 00:32:41 +0000 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development === release 0.10.27 === 2011-01-21 12:54:16 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.27 2011-01-20 14:10:55 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264depay.c: h264depay: don't leak codec data buffer in byte-stream=true mode https://bugzilla.gnome.org/show_bug.cgi?id=640063 2011-01-20 13:41:33 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: don't leak url string https://bugzilla.gnome.org/show_bug.cgi?id=640064 2011-01-20 11:45:47 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Gracefully handle mov files misusing the WAVE atoms Check that the WAVEHEADER node is present instead of blindly using it. If not present we won't be able to provide a more refined caps, but at least we won't crash. https://bugzilla.gnome.org/show_bug.cgi?id=640028 2011-01-20 00:07:33 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2sink.c: v4l2sink: fix accidental breakage of navigation interface support 2011-01-18 12:58:29 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.26.4 pre-release 2011-01-12 14:03:57 -0800 David Schleef * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: rewrite how neighboring scan lines are calculated Old code was difficult to understand exactly how the neighboring scan lines are calculated, and it appeared that some were off by +2 or -2, depending on the field flag. Fixes #639321. 2011-01-18 09:33:06 +0000 Tim-Philipp Müller * gst/avi/gstavisubtitle.c: avisubtitle: set caps on srcpad to fix issue with discoverer Set caps from the start so discoverer doesn't blow up on seeing no negotiated caps between elements on preroll, which might happen if no subtitle buffers have been pushed yet at the time. See file from bug #603308. 2011-01-17 20:09:16 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Uncork stream while flushing the ringbuffer After starting the ringbuffer, we wait for enough data to arrive before uncorking the stream. This will cause the pipeline to stall if we get an EOS (or otherwise need to flush the stream) before sufficient data becomes available. This patch makes sure that the stream is uncorked while flushing to avoid this problem. Fixes issue with a webkit unit test testing reverse playback of an MP4 H.264/AAC file. https://bugzilla.gnome.org/show_bug.cgi?id=639740 2011-01-14 14:51:51 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: avoid creating caps from string when possible Fixes #639516. 2011-01-14 14:48:49 +0100 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: set src pad caps when starting file Fixes #639516. 2011-01-12 20:38:59 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: v4l2: define V4L2_FIELD_INTERLACED_{TB,BT} if not available in header Older kernels don't have these, and there's no easy way to check for the existance of enums that doesn't involve a configure check, so just define these if the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define is not there, which was added in the same commit as the TB/BT enum. Fixes compilation on CentOS 5. https://bugzilla.gnome.org/show_bug.cgi?id=639339 2011-01-11 23:18:59 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.26.3 pre-release 2011-01-11 22:42:42 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update docs 2011-01-11 23:39:12 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Make corking during pause synchronous This makes the call to pa_stream_cork() during ringbuffer pause() synchronous, which makes sure that the clock does not advance after we take a snapshot for start_time. https://bugzilla.gnome.org/show_bug.cgi?id=639240 2011-01-11 19:33:16 +0000 Tim-Philipp Müller * po/da.po: * po/gl.po: * po/pl.po: * po/pt_BR.po: * po/sl.po: * po/sv.po: * po/tr.po: po: update translations 2011-01-11 15:50:28 +0200 Stefan Kost * common: Automatic update of common submodule From e572c87 to f94d739 2011-01-10 16:36:19 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From ccbaa85 to e572c87 2011-01-10 14:53:39 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 46445ad to ccbaa85 2011-01-07 13:24:02 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.26.2 pre-release 2011-01-07 13:06:38 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2011-01-07 02:32:20 +0000 Tim-Philipp Müller * gst/alpha/gstalpha.c: alpha: fix compiler warnings caused by -DG_DISABLE_ASSERT 2011-01-07 02:06:51 +0000 Tim-Philipp Müller * gst/matroska/ebml-read.c: matroska: don't put essential function calls into g_assert() g_assert() will expand to NOOPs if -DG_DISABLE_ASSERT is passed. 2011-01-07 01:35:45 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2sink.c: v4l2sink: don't put functional code like ioctl calls into g_return_if_fail() These macros will expand to NOOPs given the right defines. Also, g_return_if_fail() and friends are meant to be used to catch programming errors (like invalid input to functions), not runtime error handling. 2011-01-07 01:11:02 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: never disable g_assert() and cast checks for the unit tests The unit tests are riddled with g_assert() and friends, make sure we don't disable assert and cast checks for the unit tests even if this has been specified for the rest of the code base, e.g. via --disable-glib-asserts. 2011-01-06 12:29:21 +0100 Edward Hervey * gst/rtp/gstrtpmp4adepay.c: rtp: Fix unitialized variables on macosx 2011-01-06 12:28:58 +0100 Edward Hervey * gst/qtdemux/qtdemux_dump.c: qtdemux: Fix unitialized variables on macosx 2011-01-05 17:49:16 -0800 David Schleef * gst/debugutils/gstcapsdebug.c: capsdebug: Add capdebug debug category 2010-12-11 12:42:10 -0800 David Schleef * gst/deinterlace/gstdeinterlace.c: deinterlace: Change the default to linear The previous default, greedyh, takes 4 times as long as MPEG-2 video decoding, and is unlikely fast enough on any current CPU to play 1080i video in real-time. greedyl isn't much faster. linear was chosen over vfir, since the quality advantage of vfir is minimal compared to the occasional visual artifacts and slower processing. 2011-01-05 18:32:58 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't confuse return values Return a return value of the right type. 2011-01-05 16:24:13 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: qtdemux: Fix unitialized variables on macosx 2011-01-05 15:03:32 +0100 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: fix length check Add some more debugging. Add the length check so we don't cause unneeded warnings. 2011-01-05 12:04:03 +0100 Wim Taymans * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multiudpsink: add buffer-size property Add buffer-size property to configure the kernel send buffer. 2011-01-03 20:16:22 +0200 Stefan Kost * gst/rtsp/gstrtspsrc.c: rtspsrc: remove unused variables when debug-logging disabled 2011-01-03 20:06:35 +0200 Stefan Kost * gst/matroska/matroska-demux.c: matroska-demux: remove unused variables when debug-logging disabled 2011-01-03 18:05:15 +0100 Wim Taymans * ext/libcaca/gstcacasink.c: cacasink: fix masks and strides Use the right endianness to read the masks. Use the right strides for the bitmap. Fixes #638569 2011-01-03 01:18:06 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2src.c: v4l2src: undo presumably accidental enablement of the GstXOverlay interface Looks like this got enabled by accident when adding it to v4l2sink, so undo this for now. Not sure it makes much sense in a GStreamer context with current hardware. 2011-01-03 15:40:11 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: increase udp buffer size Set a bigger UDP buffer size by default to reduce packet loss with high bitrate streams. 2011-01-02 19:19:27 -0800 David Schleef * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: send stream headers in key-frame mode 2011-01-02 19:43:02 +0000 Tim-Philipp Müller * ext/jack/Makefile.am: * ext/jack/README: * ext/jack/gstjack.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: fix up element details and some other minor clean-ups 2011-01-02 19:23:51 +0000 Erich Schubert * gst/id3demux/id3v2frames.c: id3demux: fix parsing of ID3v2.4 genre frames with multiple genres We'd only extract the first genre (multiple times) instead of all genres. https://bugzilla.gnome.org/show_bug.cgi?id=638535 2011-01-02 17:40:41 +0000 Tim-Philipp Müller * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: template caps had lists with one value, just use value directly 2011-01-02 17:07:19 +0000 Tim-Philipp Müller * ext/jack/gstjack.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: make get_type functions thread-safe Because we can (shouldn't be needed with other workarounds still there). 2011-01-02 15:27:19 +0000 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-rtp.xml: docs: update plugin docs 2011-01-02 15:25:41 +0000 Tim-Philipp Müller * .gitignore: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-jack.xml: * ext/Makefile.am: * gst-plugins-good.spec.in: * tests/examples/Makefile.am: * tests/examples/jack/Makefile.am: jack: new jackaudiosrc and jackaudiosink elements, moved from gst-plugins-bad https://bugzilla.gnome.org/show_bug.cgi?id=621929 2010-10-19 16:23:23 +0300 Stefan Kost * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: various (ext): add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. 2010-09-09 14:49:06 -0400 Tristan Matthews * ext/jack/Makefile.am: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: added translatable text for server not found error 2010-09-06 17:17:54 -0400 Tristan Matthews * tests/examples/jack/Makefile.am: * tests/examples/jack/jack_client.c: examples: add test to demonstrate jack_client_t usage 2010-09-06 16:11:31 -0400 Tristan Matthews * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudioclient.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: added client property 2010-06-17 16:26:07 -0400 Tristan Matthews * ext/jack/gstjackbin.c: jack: removed unused file gstjackbin.c This is a 0.8 leftover. 2010-05-13 12:55:29 +0200 Wim Taymans * ext/jack/gstjackaudiosrc.c: jacksrc: make sure we always read nframes Error out when we are asked to read a different size that what was configured as the jack period size because that would mean something else is wrong. Fixes #618409 2010-05-11 17:56:31 -0400 Tristan Matthews * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: improve process_cb 2010-04-27 10:48:32 -0400 Tristan Matthews * ext/jack/Makefile.am: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackutil.c: * ext/jack/gstjackutil.h: jack: implement multichannel support correctly for jackaudiosrc Fixes parts of bug #616541. 2010-04-27 11:21:16 +0300 Stefan Kost * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackringbuffer.h: jack: remove empty dispose and finalize methods 2010-04-27 10:59:00 +0300 Stefan Kost * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: don't leak caps Add dispose methods to clear caps. 2010-04-27 10:34:24 +0300 Stefan Kost * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: don't use GST_DEBUG_FUNCPTR for gobject vmethods 2010-03-24 15:59:53 +0200 Stefan Kost * ext/jack/gstjackaudiosrc.c: jack: fix element name in section doc blob 2010-03-22 16:56:03 +0100 Benjamin Otte * ext/jack/gstjackaudiosrc.c: Add -Wold-style-definition and fix the warnings 2010-03-21 21:39:18 +0100 Benjamin Otte * ext/jack/gstjack.h: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-03-18 17:30:26 +0100 Benjamin Otte * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: gst_element_class_set_details => gst_element_class_set_details_simple 2009-10-12 09:06:37 +0300 Stefan Kost * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: ensure segtotal is at least 2 Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is atleast 2. Do same change on jacksrc. We could also check the latency and buffer time configured by the client and adjust buffer-time so that we get to the same number of segments. 2009-10-12 00:51:27 +0300 Stefan Kost * ext/jack/gstjackaudiosink.c: jack: don't crash in ringbuffer with SIGFPE on small buffer-times Jack overrides user-specified latency-time with the one it gets from jack itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0 2009-05-11 16:12:54 +0300 Stefan Kost * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudiosink.c: jack: when stopping playback, do one more cycle to flush the port. Fixes #582167 The gst_jack_audio_client_set_active() flags the port as deactivating and uses a GCond to wait until the jack_process_cb() has run once more and cleared the flag. This way the client zero's the buffer. This happens if one manyally go to PAUSED and then to READY, while leting the mainloop run inbetween. 2009-03-16 11:21:02 +0100 Wim Taymans * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: Add new connection mode Add a new connection mode to jacksrc and jacksink. In this new auto-force connection mode jack will create as many ports as requested/needed in the pipeline and will then connect as many physical ports as possible, possibly leaving some ports unconnected. Also get rid of some leftover g_print. Fixes #575284. 2008-11-23 17:50:08 +0000 Stefan Kost ext/jack/: Query port latencies for sink/src delays. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: Query port latencies for sink/src delays. * ext/jack/gstjackbin.c: No printf please. 2008-11-04 12:42:30 +0000 Stefan Kost Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-09-17 13:59:21 +0000 Jan Schmidt Fix compiler warnings on OS/X Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (jack_process_cb): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Fix compiler warnings on OS/X 2008-08-07 13:15:21 +0000 Stefan Kost ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. Original commit message from CVS: * ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. 2008-08-07 09:09:44 +0000 Stefan Kost docs/plugins/: docs/plugins/inspect/plugin-jack.xml Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/inspect/plugin-jack.xml Add new element to docs. * ext/jack/gstjack.h Add missing file. * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: Rename jackaudiosrc to jack_audio_src. 2008-08-07 08:47:40 +0000 Tristan Matthews ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse. Original commit message from CVS: patch by: Tristan Matthews * ext/jack/Makefile.am: * ext/jack/gstjack.c: * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: * ext/jack/gstjackringbuffer.h: Add a jackaudiosrc. Refactor sink slightly for better code reuse. Fixes #545197. 2008-06-13 11:59:23 +0000 Stefan Kost docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-app.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdaudio.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstinterlace.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-mythtv.xml * docs/plugins/inspect/plugin-nas.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-oss4.xml * docs/plugins/inspect/plugin-rawparse.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rfbsrc.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-selector.xml: * docs/plugins/inspect/plugin-sndfile.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-subenc.xml * docs/plugins/inspect/plugin-timidity.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-vcdsrc.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/dc1394/gstdc1394.c: * ext/directfb/dfbvideosink.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/mplex/gstmplex.cc: * ext/musicbrainz/gsttrm.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst-libs/gst/app/gstappsink.c: * gst/deinterlace/gstdeinterlace.c: * gst/dvdspu/gstdvdspu.c: * gst/festival/gstfestival.c: * gst/freeze/gstfreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/modplug/gstmodplug.cc: * gst/nuvdemux/gstnuvdemux.c: Add missing elements to docs. Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types. 2008-06-12 14:49:18 +0000 Stefan Kost Do not use short_description in section docs for elements. We extract them from element details and there will be war... Original commit message from CVS: * ext/dc1394/gstdc1394.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/metadata/gstmetadatademux.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * gst-libs/gst/app/gstappsink.c: * gst/bayer/gstbayer2rgb.c: * gst/deinterlace/gstdeinterlace.c: * gst/rawparse/gstaudioparse.c: * gst/rawparse/gstvideoparse.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/selector/gstinputselector.c: * gst/selector/gstoutputselector.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Do not use short_description in section docs for elements. We extract them from element details and there will be warnings if they differ. Also fixing up the ChangeLog order. 2008-05-26 17:52:21 +0000 Wim Taymans ext/jack/gstjackaudiosink.c: Include the element name in the port name to avoid duplicate port names. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_allocate_channels): Include the element name in the port name to avoid duplicate port names. 2008-04-06 20:18:16 +0000 Tim-Philipp Müller ext/jack/gstjackaudiosink.c: Work around missing bits of thread-safety on older GLibs some more to avoid assertions w... Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init): Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multiple playbin objects concurrently (see #512382). 2008-03-13 14:25:20 +0000 Sebastian Dröge Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. Original commit message from CVS: * ext/alsaspdif/alsaspdifsink.c: * ext/gsm/gstgsm.c: * ext/jack/gstjack.c: * ext/libmms/gstmms.c: * ext/neon/gstneonhttpsrc.c: * ext/shout/gstshout.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst/nuvdemux/gstnuvdemux.c: * gst/tta/gsttta.c: Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. 2007-07-18 07:42:47 +0000 Stefan Kost ext/jack/gstjackaudiosink.c: Add stdlib include here too. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_acquire): Add stdlib include here too. 2007-04-04 07:36:28 +0000 Stefan Kost ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_acquire): Try t better name clients. properly handle return codes when re- establishing links. 2007-03-18 17:57:48 +0000 Paul Davis ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks. Original commit message from CVS: Based on patch by: Paul Davis * ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection): Don't need to take the connection lock, it will not be used and could cause deadlocks. 2007-03-08 15:24:52 +0000 Paul Davis ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele... Original commit message from CVS: Includes patch by: Paul Davis * ext/jack/Makefile.am: * ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init), (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (connection_find), (gst_jack_audio_make_connection), (gst_jack_audio_get_connection), (gst_jack_audio_unref_connection), (gst_jack_audio_connection_add_client), (gst_jack_audio_connection_remove_client), (gst_jack_audio_client_new), (gst_jack_audio_client_free), (gst_jack_audio_client_get_client), (gst_jack_audio_client_set_active): * ext/jack/gstjackaudioclient.h: Make an object to manage client connections to the jack server which we will use in the future to run selected jack elements with the same jack connection. Make some stuff a bit more threadsafe. Activate the jack client ASAP. * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_allocate_channels), (gst_jack_audio_sink_free_channels), (jack_process_cb), (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_close_device), (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release), (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init), (gst_jack_audio_sink_getcaps): * ext/jack/gstjackaudiosink.h: Use new client object to manage connections. Don't remove and recreate all ports, try to reuse them. 2007-01-12 10:25:40 +0000 Wim Taymans ext/jack/gstjackaudiosink.*: Improve docs. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (gst_jack_ring_buffer_acquire): * ext/jack/gstjackaudiosink.h: Improve docs. 2006-12-06 16:57:17 +0000 Jan Schmidt ext/jack/.cvsignore: Ignore old files as requested by the build slave. Original commit message from CVS: * ext/jack/.cvsignore: Ignore old files as requested by the build slave. 2006-11-30 11:59:04 +0000 Wim Taymans ext/Makefile.am: Fix build. Original commit message from CVS: * ext/Makefile.am: Fix build. * ext/jack/gstjackaudiosink.c: (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (gst_jack_ring_buffer_acquire): Small cleanups. 2006-11-30 11:49:36 +0000 Wim Taymans Added fully functional jackaudiosink. Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/jack/Makefile.am: * ext/jack/gstjack.c: (plugin_init): * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type), (gst_jack_ring_buffer_class_init), (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose), (gst_jack_ring_buffer_finalize), (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_close_device), (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release), (gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause), (gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay), (gst_jack_connect_get_type), (gst_jack_audio_sink_base_init), (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init), (gst_jack_audio_sink_set_property), (gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps), (gst_jack_audio_sink_create_ringbuffer): * ext/jack/gstjackaudiosink.h: Added fully functional jackaudiosink. 2006-04-08 21:48:01 +0000 Stefan Kost Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): * ext/arts/gst_arts.c: (gst_arts_class_init): * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): * ext/audiofile/gstafsink.c: (gst_afsink_class_init): * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): * ext/audioresample/gstaudioresample.c: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_class_init): * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): * ext/jack/gstjack.c: (gst_jack_class_init): * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): * ext/nas/nassink.c: (gst_nassink_class_init): * ext/shout/gstshout.c: (gst_icecastsend_class_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): * ext/sndfile/gstsf.c: (gst_sf_class_init): * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), (gst_swfdec_class_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): * gst/chart/gstchart.c: (gst_chart_class_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): * gst/festival/gstfestival.c: (gst_festival_class_init): * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstiir.c: (gst_iir_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_class_init): * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_class_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init): * gst/overlay/gstoverlay.c: (gst_overlay_class_init): * gst/passthrough/gstpassthrough.c: (passthrough_class_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): * gst/smooth/gstsmooth.c: (gst_smooth_class_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): * gst/stereo/gststereo.c: (gst_stereo_class_init): * gst/switch/gstswitch.c: (gst_switch_class_init): * gst/tta/gstttadec.c: (gst_tta_dec_class_init): * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_class_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): * sys/v4l2/gstv4l2colorbalance.c: (gst_v4l2_color_balance_channel_class_init): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), (gst_v4l2_tuner_norm_class_init): * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 2006-04-01 10:09:11 +0000 Thomas Vander Stichele * ext/jack/gstjack.c: rework build; add translations for v4l2 Original commit message from CVS: rework build; add translations for v4l2 2005-10-12 14:29:55 +0000 Stefan Kost renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio), (gst_artsdsink_close_audio), (gst_artsdsink_change_state): * ext/artsd/gstartsdsink.h: * ext/audiofile/gstafparse.c: (gst_afparse_open_file), (gst_afparse_close_file): * ext/audiofile/gstafparse.h: * ext/audiofile/gstafsink.c: (gst_afsink_open_file), (gst_afsink_close_file), (gst_afsink_chain), (gst_afsink_change_state): * ext/audiofile/gstafsink.h: * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file), (gst_afsrc_close_file), (gst_afsrc_change_state): * ext/audiofile/gstafsrc.h: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init): * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init): * ext/dts/gstdtsdec.c: (gst_dtsdec_init): * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: (gst_jack_bin_init), (gst_jack_bin_change_state): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init): * ext/nas/nassink.c: (gst_nassink_open_audio), (gst_nassink_close_audio), (gst_nassink_change_state): * ext/nas/nassink.h: * ext/polyp/polypsink.c: (gst_polypsink_init): * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state): * ext/sdl/sdlvideosink.h: * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): * ext/sndfile/gstsf.c: (gst_sf_set_property), (gst_sf_change_state), (gst_sf_release_request_pad), (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop): * ext/sndfile/gstsf.h: * ext/swfdec/gstswfdec.c: (gst_swfdec_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init): * gst/apetag/apedemux.c: (gst_ape_demux_init): * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init): * gst/festival/gstfestival.c: (gst_festival_change_state): * gst/festival/gstfestival.h: * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init): * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init), (gst_multifilesink_set_location), (gst_multifilesink_open_file), (gst_multifilesink_close_file), (gst_multifilesink_next_file), (gst_multifilesink_pad_query), (gst_multifilesink_handle_event), (gst_multifilesink_chain), (gst_multifilesink_change_state): * gst/multifilesink/gstmultifilesink.h: * gst/videodrop/gstvideodrop.c: (gst_videodrop_init): * sys/cdrom/gstcdplayer.c: (cdplayer_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init), (dxr3audiosink_open), (dxr3audiosink_close), (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3), (dxr3audiosink_change_state): * sys/dxr3/dxr3audiosink.h: * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open), (dxr3spusink_close), (dxr3spusink_chain), (dxr3spusink_change_state): * sys/dxr3/dxr3spusink.h: * sys/dxr3/dxr3videosink.c: (dxr3videosink_init), (dxr3videosink_open), (dxr3videosink_close), (dxr3videosink_write_data), (dxr3videosink_change_state): * sys/dxr3/dxr3videosink.h: * sys/glsink/glimagesink.c: (gst_glimagesink_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state), (gst_qcamsrc_open), (gst_qcamsrc_close): * sys/qcam/gstqcamsrc.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get), (gst_vcdsrc_open_file), (gst_vcdsrc_close_file), (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate): * sys/vcd/vcdsrc.h: renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition 2005-09-05 17:20:29 +0000 Jan Schmidt * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: Fix up all the state change functions. Original commit message from CVS: Fix up all the state change functions. 2004-08-03 14:28:12 +0000 Benjamin Otte fixes for G_DISABLE_ASSERT and friends Original commit message from CVS: * examples/dynparams/filter.c: (ui_control_create): * examples/gstplay/player.c: (print_tag): * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad): * ext/gdk_pixbuf/gstgdkanimation.c: (gst_gdk_animation_iter_may_advance): * ext/jack/gstjack.c: (gst_jack_request_new_pad): * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list), (tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event): * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value): * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain): * gst-libs/gst/media-info/media-info-test.c: (print_tag): * gst/sine/demo-dparams.c: (main): * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments): * testsuite/alsa/formats.c: (create_pipeline): * testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get): fixes for G_DISABLE_ASSERT and friends * gst/typefind/gsttypefindfunctions.c: (aac_type_find), (mp3_type_frame_length_from_header), (mp3_type_find), (plugin_init): require mp3 typefinding to have at least MIN_HEADERS valid headers add typefinding for AAC adts files 2004-05-21 23:28:57 +0000 Stéphane Loeuillet * ext/jack/gstjack.c: * ext/jack/gstjack.h: second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ... Original commit message from CVS: second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in gst-plugins/ext/ this time) 2004-03-15 19:32:27 +0000 Thomas Vander Stichele * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: don't mix tabs and spaces Original commit message from CVS: don't mix tabs and spaces 2004-03-15 16:32:54 +0000 Johan Dahlin *.h: Revert indenting Original commit message from CVS: * *.h: Revert indenting 2004-03-14 22:34:33 +0000 Thomas Vander Stichele * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: gst-indent Original commit message from CVS: gst-indent 2004-01-12 03:40:18 +0000 David Schleef * ext/jack/gstjack.c: Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). Original commit message from CVS: Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). 2003-12-22 01:47:09 +0000 David Schleef * ext/jack/gstjack.c: Merge CAPS branch Original commit message from CVS: Merge CAPS branch 2003-12-13 16:59:51 +0000 Benjamin Otte * ext/jack/gstjackbin.c: removed GST_*_CAST. Disabling of type checking is done in glib. Original commit message from CVS: removed GST_*_CAST. Disabling of type checking is done in glib. 2003-12-04 10:37:38 +0000 Andy Wingo * ext/jack/gstjack.c: remove copyright field from plugins Original commit message from CVS: remove copyright field from plugins 2003-11-07 12:47:02 +0000 Ronald S. Bultje * ext/jack/gstjackbin.c: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... Original commit message from CVS: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files 2003-11-01 23:43:13 +0000 Iain Holmes * ext/jack/gstjack.c: Jack fixed too Original commit message from CVS: Jack fixed too 2003-10-29 03:15:55 +0000 David Schleef * ext/jack/gstjack.h: change gst/bytestream.h to gst/bytestream/bytestream.h Original commit message from CVS: change gst/bytestream.h to gst/bytestream/bytestream.h 2003-10-28 20:52:41 +0000 Benjamin Otte * ext/jack/gstjack.h: merge TYPEFIND branch. Major changes: Original commit message from CVS: merge TYPEFIND branch. Major changes: - totally reworked type(find) system - all typefind functions are in gst/typefind now - more typefind functions then before - some plugins might fail to compile now because I don't have them installed and they a) require bytestream or b) haven't had their typefind fixed. Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies 2003-10-08 16:08:19 +0000 Andy Wingo * ext/jack/gstjack.c: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. Original commit message from CVS: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. 2003-10-01 13:14:50 +0000 Ronald S. Bultje * ext/jack/gstjack.h: New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste... Original commit message from CVS: New typefind system: * bytestream is now part of the core * all plugins have been modified to use this new typefind system * asf typefinding added * mpeg video stream typefiding removed because it's broken * duplicate typefind entries removed * extra id3 typefinding added, because we've seen 4 types of files (riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs to work. Instead, I've added an id3 element and let it redo typefiding after the id3 header. this needs a hack because spider only typefinds once. We can remove this hack once spider supports multiple typefinds. * with all this, mp3 typefinding is semi-rewritten * id3 typefinding in flac/vorbis is removed, it's no longer needed * fixed spider and gst-typefind to use this, too. * Other general cleanups 2003-09-30 12:56:27 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: conform to the buffer-frames props entry -- much nicer now... Original commit message from CVS: conform to the buffer-frames props entry -- much nicer now... 2003-08-10 00:01:58 +0000 David Schleef * ext/jack/Makefile.am: Remove redundant plugindir definition Original commit message from CVS: Remove redundant plugindir definition 2003-07-19 23:25:25 +0000 Leif Johnson * ext/jack/gstjack.c: * ext/jack/gstjack.h: + changes for new float caps without slope/intercept + some category changes for plugins Original commit message from CVS: + changes for new float caps without slope/intercept + some category changes for plugins 2003-07-06 20:49:52 +0000 Ronald S. Bultje * ext/jack/gstjack.c: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri... Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs 2003-07-01 02:27:06 +0000 David Schleef * ext/jack/gstjack.c: fix type punning Original commit message from CVS: fix type punning 2003-06-29 19:46:13 +0000 Benjamin Otte * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: compatibility fix for new GST_DEBUG stuff. Original commit message from CVS: compatibility fix for new GST_DEBUG stuff. Includes fixes for missing includes for config.h and unistd.h I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately. 2003-06-13 21:21:17 +0000 Wim Taymans * ext/jack/gstjack.c: Removed ugly caps fixed flag hack, will be done automatically in core soon Original commit message from CVS: Removed ugly caps fixed flag hack, will be done automatically in core soon 2003-03-04 15:34:20 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: update for the latest jack cvs and non-cothreaded gst scheduler Original commit message from CVS: update for the latest jack cvs and non-cothreaded gst scheduler 2003-02-05 20:38:41 +0000 Jan Schmidt * ext/jack/gstjack.c: Changed caps->fixed to use FLAG_SET Original commit message from CVS: Changed caps->fixed to use FLAG_SET 2003-01-10 13:38:32 +0000 Thomas Vander Stichele * ext/jack/gstjack.c: PadConnect -> PadLink Original commit message from CVS: PadConnect -> PadLink 2003-01-10 10:22:25 +0000 Thomas Vander Stichele * ext/jack/gstjack.c: another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t... Original commit message from CVS: another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so that I don't spend double the time resolving conflicts 2002-12-08 14:50:10 +0000 Thomas Vander Stichele * ext/jack/Makefile.am: parallel install fixes Original commit message from CVS: parallel install fixes 2002-09-29 18:12:18 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: licenses again Original commit message from CVS: licenses again 2002-09-18 19:02:52 +0000 Christian Schaller * ext/jack/gstjack.c: plugins part of license field patch Original commit message from CVS: plugins part of license field patch 2002-09-10 09:31:40 +0000 Ronald S. Bultje * ext/jack/gstjack.c: This updates all plugins to the new API for gst_pad_try_set_caps Original commit message from CVS: This updates all plugins to the new API for gst_pad_try_set_caps 2002-09-09 23:27:38 +0000 Thomas Vander Stichele * ext/jack/gstjack.c: removing warnings as approved by wim Original commit message from CVS: removing warnings as approved by wim 2002-08-23 04:04:11 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: fix jack input port connection Original commit message from CVS: fix jack input port connection 2002-07-09 17:39:17 +0000 Andy Wingo * ext/jack/gstjack.c: compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4 Original commit message from CVS: compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4 2002-07-02 23:35:07 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: make jack work in all its full duplex glory Original commit message from CVS: make jack work in all its full duplex glory 2002-06-12 03:32:02 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjackbin.c: working jack elements (fixed a problem in upstream jack) random other fixen... Original commit message from CVS: * working jack elements (fixed a problem in upstream jack) * random other fixen... 2002-05-15 19:08:49 +0000 Steve Baker * ext/jack/gstjack.c: use new bytestream api Original commit message from CVS: use new bytestream api 2002-05-13 18:08:33 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: update to new jack api Original commit message from CVS: update to new jack api 2002-05-05 19:39:17 +0000 Andy Wingo * ext/jack/gstjack.c: add some includes Original commit message from CVS: add some includes 2002-05-05 01:08:05 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: better initialization. it doesn't work over here, though. Original commit message from CVS: better initialization. it doesn't work over here, though. 2002-05-04 21:38:56 +0000 Andy Wingo * ext/jack/gstjackbin.c: a commit so that jack will build without errors on Uraeus's system ;) Original commit message from CVS: a commit so that jack will build without errors on Uraeus's system ;) 2002-05-04 20:53:35 +0000 Andy Wingo * ext/jack/gstjack.c: set caps once we know the sample rate of the system Original commit message from CVS: set caps once we know the sample rate of the system 2002-05-04 18:57:44 +0000 Andy Wingo * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: some jack fixes, alsa touchups, and add rtp by default to the build if there are any problems building rtp, we're mov... Original commit message from CVS: some jack fixes, alsa touchups, and add rtp by default to the build if there are any problems building rtp, we're moving it back to experimental ;) 2002-04-20 21:42:51 +0000 Andy Wingo * ext/jack/gstjack.c: a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br... Original commit message from CVS: * a hack to work around intltool's brokenness * a current check for mpeg2dec * details->klass reorganizations * an element browser that uses details->klass * separated cdxa parse out from the avi directory 2002-04-16 17:14:05 +0000 Andy Wingo * ext/jack/Makefile.am: * ext/jack/gstjack.c: * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: Finally we're on to a proper jack setup, with a specialized bin and elements that can only go in a jack bin. I had to... Original commit message from CVS: Finally we're on to a proper jack setup, with a specialized bin and elements that can only go in a jack bin. I had to fix the parser first to do this, but to run it, the syntax is like so: gst-launch jackbin.( filesrc ! mad ! jacksink ) But of course it's not fully functional yet. Sigh. 2002-04-11 20:42:26 +0000 Andy Wingo * ext/jack/gstjack.c: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. Original commit message from CVS: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. also, some -Werror fixes. 2002-03-30 21:07:51 +0000 Andy Wingo * ext/jack/gstjack.c: alphabetization fixen a jack caps fix Original commit message from CVS: * alphabetization fixen * a jack caps fix 2002-03-30 19:31:13 +0000 Andy Wingo * ext/jack/gstjack.c: add notify back to filesrc, it's needed for MVC applications remove notify printouts from gst-launch cleanup in gst-p... Original commit message from CVS: * add notify back to filesrc, it's needed for MVC applications * remove notify printouts from gst-launch * cleanup in gst-plugins configure.ac * some jack updates * remove SELF_ITERATING flag in favor of SEF_SCHEDULABLE (not a clear name, but it's what we have for the moment) * improve parsing of request pad names, no more sscanf * fixes to the fastscheduler Makefile.am 2002-03-20 21:45:04 +0000 Andy Wingo * ext/jack/gstjack.c: s/Gnome-Streamer/GStreamer/ Original commit message from CVS: s/Gnome-Streamer/GStreamer/ 2002-03-19 04:10:06 +0000 Andy Wingo * ext/jack/Makefile.am: * ext/jack/gstjack.c: removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with... Original commit message from CVS: * removal of //-style comments * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct, and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory. 2002-03-19 01:39:43 +0000 Andy Wingo * ext/jack/Makefile.am: s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi... Original commit message from CVS: s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagically, and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag 2002-03-18 04:41:35 +0000 Andy Wingo * ext/jack/Makefile.am: * ext/jack/README: * ext/jack/gstjack.c: * ext/jack/gstjack.h: s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way added jack ... Original commit message from CVS: * s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way * added jack element, doesn't quite work right yet but i didn't want to lose the work -- it does build, register, and attempt to run though * imposed some restrictions on the naming of request pads to better allow for reverse parsing * added '%s' to reverse parsing * added new bin flag to indicate that it is self-iterating, and some lame code in gst-launch to test it out * fixen on launch-gui * added pkg-config stuff for the editor's libs 2011-01-02 11:37:14 +0000 Tim-Philipp Müller * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/v4l2_calls.c: v4l2: mark v4l2sink as experimental and build only if --enable-experimental is passed It's not really of 'good' quality yet, but there's a lot of code shared with v4l2src, so not so easy to move it elswhere. https://bugzilla.gnome.org/show_bug.cgi?id=612244 2011-01-02 01:24:21 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/v4l2_calls.c: Revert "v4l2: add norm property" This reverts commit 9e1d419d07337e6db2cc3936472be205ce927e54. Reverting this since it adds unreviewed and bad API to v4l2src (property of type enum, with seemingly random and unsorted values). 2011-01-01 23:26:33 +0000 Tim-Philipp Müller * tools/.gitignore: * tools/Makefile.am: * tools/README.filterstamp: * tools/filterstamp.sh: * tools/gst-launch-ext-m.m: * tools/gst-launch-ext.1.in: * tools/gst-visualise-m.m: * tools/gst-visualise.1.in: tools: remove unused left-over directory These are all in -base/tools. 2010-12-31 13:57:05 +0100 Wim Taymans * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4adepay.h: mp4adepay: improve timestamps on outgoing packets Improve parsing of the samplerate. Parse the framelen so that we can calculate timestamps. When interpollate the incomming timestamp on outgoing buffers when there are multiple subframes. fixes #625825 2010-12-31 00:12:53 -0800 David Schleef * gst/dtmf/tone_detect.c: dtmf: Fix build failure caused by previous commit 2010-12-30 18:20:47 -0800 David Schleef * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/tone_detect.c: * gst/dtmf/tone_detect.h: dtmf: build fixes for MSVC Use gint16 and G_PI. 2010-12-30 18:19:47 -0800 David Schleef * gst/dtmf/tone_detect.c: dtmf: reindent 2010-12-31 02:16:54 +0000 Tim-Philipp Müller * ext/cairo/gsttimeoverlay.c: * gst/videofilter/gstvideobalance.c: cairo, videofilter: use gst/math-compat.h header for rint 2010-12-30 14:30:27 -0800 David Schleef * gst/videofilter/gstvideobalance.c: videobalance: Check for HAVE_RINT instead Also change M_PI to G_PI for giggles. 2010-12-30 14:21:37 -0800 David Schleef * ext/cairo/gstcairorender.c: cairo: Don't use #ifdefs inside macros 2010-12-30 14:20:52 -0800 David Schleef * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/effectv/gstop.c: * gst/equalizer/gstiirequalizer.c: * gst/goom/convolve_fx.c: * gst/goom/ifs.c: * gst/goom/lines.c: * gst/goom/tentacle3d.c: * tests/examples/audiofx/firfilter-example.c: * tests/examples/audiofx/iirfilter-example.c: Change M_PI to G_PI 2010-12-30 12:07:52 -0800 David Schleef * gst/videofilter/gstvideobalance.c: videobalance: use G_OS_WIN32 for windows check 2010-12-30 16:24:16 +0100 Wim Taymans * gst/rtp/gstrtpmp4adepay.c: mp4adepay: fix timestamps on buffers 2010-12-30 16:22:48 +0100 Wim Taymans * gst/rtp/gstrtpmpvpay.c: mpvpay: fix flushing and discont Fix flushing and disconts. Clean up in state changes. 2010-12-29 23:38:18 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: increase allowed max. block size for push mode from 10M to 15M It was an arbitrary limit from the start, meant as a basic sanity check, so may just as well increase it a little. Would be good to provide progress reporting while completing the block in any case.. https://bugzilla.gnome.org/show_bug.cgi?id=637060 2010-12-29 23:09:04 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: assume matroska if no doctype is specified https://bugzilla.gnome.org/show_bug.cgi?id=638019 2010-12-04 13:43:11 -0600 Rob Clark * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: add interlaced support 2010-10-02 14:45:14 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/gstv4l2xoverlay.h: v4l2sink: add navigation support 2010-04-04 06:43:41 -0500 Rob Clark * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/v4l2_calls.c: v4l2: add norm property Based on a patch by Guennadi Liakhovetski. 2010-07-13 10:03:51 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: v4l2: cleanup get/set input/output output devices should use get/set output, and in either case we should not print a warning message if the ioctl fails but the device does not claim to support the tuner interface 2010-06-10 11:15:46 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/gstv4l2xoverlay.h: v4l2xoverlay: add support to create window If xoverlay is available, v4l2sink should create a window for the overlay to display in. The window automatically tries to make itself as large as possible. This works well on a small screen, but perhaps should first attempt to use the size of the video that is played (no scaling). 2010-04-04 06:41:28 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: v4l2sink: special handling for cases gst_buffer_make_metadata_writable() Special case check for sub-buffers: In certain cases, places like GstBaseTransform, which might check that the buffer is writable before copying metadata, timestamp, and such, will find that the buffer has more than one reference to it. In these cases, they will create a sub-buffer with an offset=0 and length equal to the original buffer size. This could happen in two scenarios: (1) a tee in the pipeline, and (2) because the refcnt is incremented in gst_mini_object_free() before the finalize function is called, and decremented after it returns.. but returning this buffer to the buffer pool in the finalize function, could wake up a thread blocked in _buffer_alloc() which could run and get a buffer w/ refcnt==2 before the thread originally unref'ing the buffer returns from finalize function and decrements the refcnt back to 1! This is related to issue #545501 2010-04-04 06:39:52 -0500 Rob Clark * sys/v4l2/gstv4l2bufferpool.c: v4l2: fix race condition The size of the buffer would be zero'd out in gst_v4l2_buffer_finalize() after the buffer is qbuf'd or pushed onto the queue of available buffers.. leaving a race condition where the thread waiting for the buffer could awake and set back a valid size before the finalizing thread zeros out the length. This would result that the newly allocated buffer has length of zero. 2010-04-04 06:39:08 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: v4l2sink: add properties to control crop 2010-04-04 06:37:16 -0500 Rob Clark * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2xoverlay.c: v4l2: re-enable x-overlay support 2010-12-25 11:52:36 -0600 Rob Clark * sys/v4l2/gstv4l2sink.c: v4l2sink: fix for PAUSED->READY->PAUSED state transitions When v4l2sink goes to PAUSED->READY it only stops streaming, so the state should be set to STATE_PENDING_STREAMON in case the element transitions back to PLAYING. 2010-04-04 06:28:51 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: v4l2sink: add "min-queued-bufs" property 2010-04-04 06:26:50 -0500 Rob Clark * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2src_calls.c: v4l2sink: Add support for blocking dequeue. We'd prefer to throttle the decoder if we run out of buffers, to keep a bound on memory usage. Also, for OMAP4 it is a requirement of the decoder to not alternate between memory alloced by the display driver and malloc'd userspace memory. 2010-04-04 06:24:41 -0500 Rob Clark * sys/v4l2/gstv4l2bufferpool.c: v4l2: clear flags before reusing buffer from buffer pool note: this really only affects v4l2sink since gst_v4l2_buffer_pool_get() is only called once per buffer in the v4l2src case (in gst_v4l2src_buffer_pool_activate()) 2010-04-04 06:23:31 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: v4l2sink: don't render preroll buffers Most v4l2 drivers will get upset when you queue the same buffer twice in a row without first dequeueing it. Rendering of pre-roll buffers can be re-introduced later, but will require tracking the state of the buffer, and avoiding to re-QBUF if the buffer has already been passed to the driver. 2010-04-04 06:22:43 -0500 Rob Clark * sys/v4l2/gstv4l2sink.c: v4l2sink: Improve behavior for shared buffers. When the decoder is using pad_alloc(), v4l2sink would behave badly if the number of buffers ('queue-size' property) was not high enough to account for all the buffers needed by the decoder, and other elements (such as queues) between the decoder and v4l2sink. This patch slightly increases the default number of buffers, and changes v4l2sink to drop frames rather than return an error in case the number of buffers is not high enough. 2010-11-15 15:58:28 +0100 Andy Wingo * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: add "client" property * ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init) (gst_pulsesrc_set_property, gst_pulsesrc_get_property) (gst_pulsesrc_open): Add a "client" property, as in pulsesink. Fixes #634914 2010-12-29 15:54:46 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: serialise/deserialise floats without changing locale Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise floating point numbers, instead of ugly hacks that switch locale before and after calling libc functions (which is not a good idea in a multi-threaded application). 2010-12-29 14:40:05 +0000 Tim-Philipp Müller * gst/rtp/gstrtpjpegdepay.c: rtpjpegdepay: fix framerate parsing for locales that use a comma as floating point atof() converts strings according to the current locale, but the framerate string will likely always use a dot as floating point separator, so use g_ascii_strtod() instead (but also canonicalise the string before, so we can handle both formats as input). 2010-12-27 13:11:59 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: use the right variable Use the right variable for specifying that we sent a receiver report. 2010-12-23 16:42:29 -0600 Rob Clark * sys/v4l2/gstv4l2bufferpool.c: v4l2: fix typo 2010-12-23 16:03:00 -0600 Rob Clark * gst/matroska/matroska-demux.c: matroska-demux: add stream-format and alignment properties for h264 2010-12-22 11:41:59 +0100 Wim Taymans * gst/rtp/gstrtpgstpay.c: gstpay: fix klass, add RTP as a use case 2010-12-12 15:10:47 +0100 Wim Taymans * gst/rtp/gstrtpgstdepay.c: gstdepay: cleanup the cache 2010-12-12 05:10:01 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: gstpay/depay: add generic gstreamer payloader Add the beginnings of a generic GStreamer buffers payloader. 2010-12-23 17:06:58 +0100 Wim Taymans * gst/rtp/gstrtpmp4gpay.c: mp4gpay: reset state on flush-stop 2010-12-23 16:26:07 +0100 Wim Taymans * gst/rtp/gstrtpmp4gdepay.c: mp4gdepay: flush state on flush-stop 2010-12-23 16:25:15 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: on-npt-stop is a manager signal 2010-12-23 15:24:29 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: improve RTP session handling Store the RTP session in the stream so that we can more efficiently perform actions on the stream based on RTP signals. 2010-12-23 13:55:31 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: include last send RB block Only report RB values for non-internal sources. Report not only the RB blocks we last received from but also the last RB block we sent to a source. 2010-12-23 13:52:57 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.h: rtpsession: remember last sent RB values. 2010-12-23 13:00:49 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: include all stats and document Include all possible stats of a source in the stats structure because we might be interested in what happened in the past. Document the stats property and the fields. 2010-12-23 12:59:59 +0100 Wim Taymans * tests/examples/rtp/client-PCMA.c: examples: add example RTP stats Add some more RTP examples for how to retrieve RTP stats in a receiver. 2010-12-23 12:58:05 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: also emit RTCP activity on SR Also emit RTCP activity signals when we receive an SR packet without RB blocks, such as from a sender that is not receiving anything. 2010-12-23 11:10:55 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: docs: add some more gstrtpbin docs 2010-12-22 21:27:11 +0100 Edward Hervey * sys/ximage/gstximagesrc.c: ximagesrc: remote is a boolean (and not uint) property 2010-12-22 19:58:21 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't use gst_pad_alloc_buffer() Using this in a demuxer will cause deadlocks if there's a pad with a pending pad-block downstream, no matter if there is a queue between the pad or not. Queues pass bufferalloc downstream from the same thread and only act as a thread boundary for events and buffers. 2010-12-22 14:14:08 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: fix subtitle pad template, we only handle kate for now 2010-12-16 11:44:44 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: docs: update rtspsrc docs, rtpbin is not in -bad any more 2010-12-22 11:42:31 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: unlock before emitting signals 2010-12-21 22:34:49 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpac3pay.h: rtpac3pay: add AC3 payloader 2010-12-21 22:17:19 +0100 Wim Taymans * gst/rtp/gstrtpac3depay.c: ac3depay: fix debug category description 2010-12-21 22:16:42 +0100 Wim Taymans * gst/rtp/gstrtpmpapay.c: mpapay: add debug category 2010-12-20 14:49:02 -0300 Thiago Santos * tests/check/Makefile.am: * tests/check/elements/jpegenc.c: jpegenc: Adds another test case Adds a test for jpegenc to check that is possible to negotiate and push buffers with different resolution one after another. https://bugzilla.gnome.org/show_bug.cgi?id=637686 2010-12-21 13:37:40 -0300 Thiago Santos * ext/jpeg/gstjpegenc.c: jpegenc: sink pad's getcaps shouldn't use the src pad getcaps Instead of using get_allowed_caps on the srcpad, the sinkpad getcaps should use the getcaps of the srcpad's peer. This way the srcpad can keep using fixed_caps and sinkpad getcaps exposes all caps that can be negotiated https://bugzilla.gnome.org/show_bug.cgi?id=637686 2010-12-21 16:58:47 +0100 Wim Taymans * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: add RTP hint to the klass 2010-12-21 16:49:28 +0100 Wim Taymans * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: fix rank of payloaders and depayloaders Set the payloaders and depayloaders to a reasonable rank. 2010-12-21 15:24:18 +0100 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: reset depayloader state Reset the depayloader state on flush-stop. 2010-12-21 15:07:14 +0100 Wim Taymans * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: mp4pay: use vmethod for intercepting events 2010-12-21 13:55:40 +0100 Wim Taymans * gst/rtp/gstrtptheorapay.c: theorapay: clear packet on flush-stop 2010-12-21 13:49:41 +0100 Wim Taymans * gst/rtp/gstrtpvorbispay.c: vorbispay: clear packet on flush-stop 2010-12-21 12:31:44 +0100 Wim Taymans * gst/rtp/gstrtpmp4gdepay.c: mp4gdepay: reset depayloader state 2010-12-21 12:29:58 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: h264pay: flush adapter on flush-stop 2010-12-20 18:49:49 +0100 Wim Taymans * gst/rtp/gstrtpmpapay.c: mpapay: flush last packets on EOS 2010-12-20 17:47:05 +0100 Edward Hervey * common: Automatic update of common submodule From 169462a to 46445ad 2010-12-20 16:51:47 +0100 Wim Taymans * gst/rtp/gstrtpmpapay.c: mpapay: reset payloader on state change 2010-12-20 16:05:36 +0100 Wim Taymans * gst/rtp/gstrtpmpapay.c: mpapay: reset payloader on flush Reset the payloader on a flush event. Handle DISCONT better. 2010-12-20 15:54:45 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: get better buffering level When the jitterbuffer contains -1 timestamps, make sure we still calculate the buffer fill level by skipping the -1 buffers. Try to be more resilient to weird input timestamps. 2010-12-20 11:10:22 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: provide a clock. since we are using the clock for sync, we need to also provide a clock for good measure. The reason is that even if downstream elements provide a clock, we don't want to have that clock selected because it might not be running yet. 2010-12-20 10:49:56 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: copy buffering stats when we create an aggregate buffering message, copy the buffering stats form the last message. At least we get correct buffering mode then. 2010-12-19 11:02:41 +0100 Sebastian Dröge * tests/check/pipelines/wavenc.c: wavenc: Fix memory leaks in the unit test 2010-12-19 10:58:16 +0100 Sebastian Dröge * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: radioactv: Prevent use of uninitialized values Fixes bug #618652. 2010-12-19 10:22:29 +0100 Sebastian Dröge * gst/debugutils/gstcapsdebug.c: capsdebug: Don't leak pad templates created from static pad templates 2010-11-29 12:36:06 +0000 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: ximagesrc: change from XGetImage to XGetSubImage dependant on a property ximagesrc: change from XGetImage to XGetSubImage dependant on a property to avoid unnecessary performance hits by default. 2010-11-28 16:04:35 +0000 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: ximagesrc: use XGetSubImage instead of XGetImage, works with remote X ximagesrc: use XGetSubImage instead of XGetImage, works with remote X (on my setup anyway...) 2010-11-27 17:15:32 +0000 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: ximagesrc: fix various width/height calculations being off by one, ximagesrc: fix various width/height calculations being off by one, and make it so a single pixel width/height can be captured (except the top left one, as 0,0,0,0 is reserved for full screen as per the property comments). 2010-12-17 19:19:35 -0600 Rob Clark * sys/v4l2/gstv4l2object.c: fix compile errors on macosx with i686-apple-darwin10-gcc-4.2.1: gstv4l2object.c: In function 'gst_v4l2_object_get_nearest_size': gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 12 has type 'gint *' gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 13 has type 'gint *' 2010-12-17 15:38:15 +0100 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: determine output h264 layout using caps negotiation ... thereby (partially) deprecating properties currently controlling whether or not byte-stream output or NAL/AU alignment (though properties still determine fallback if nothing specified in caps). Fixes #606662. 2010-12-16 18:55:43 +0100 Wim Taymans * gst/rtp/gstrtpj2kpay.c: j2kpay: handle EOC correctly Don't include the next 2 bytes when we are at the end of the data and there are no more bytes left. 2010-12-16 15:15:49 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: flush remaining buffered samples on EOS ... which can make a difference between all or nothing when dealing with short streams and relatively large ringbuffer segment. 2010-12-16 10:04:19 +0100 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Change classification to Filter/Effect/Video/Deinterlace 2010-12-15 18:21:34 +0100 Edward Hervey * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: Initialize all fields Makes sad compliers happy 2010-12-15 16:22:54 +0100 Wim Taymans * gst/rtp/gstrtpj2kpay.c: j2kpay: cleanup header construction Use a simpler way of constructing the header that doesn't depend on the endianness. 2010-12-15 13:30:50 +0000 Tim-Philipp Müller * configure.ac: configure: depend on -base from git for new rtp base depayloader features This is ok in this case, since the plan is to release core/base again along with good/ugly/bad in the next cycle. 2010-12-15 14:55:58 +0200 Stefan Kost * common: Automatic update of common submodule From 20742ae to 169462a 2010-12-15 13:12:09 +0100 Wim Taymans * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: j2kdepay: add support for buffer lists 2010-12-14 18:12:43 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: session: fix average RTCP packet size some more. Fix stupid error in averaging macro. Include udp headers in packet length estimation. 2010-12-14 17:15:23 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.c: rtpbin: correctly calculate RTCP packet size 2010-12-14 15:27:52 +0100 Wim Taymans * gst/rtp/gstrtpj2kpay.c: j2kpay: stop scanning when we reached the end Stop scanning for markers when we reached the end of the data. 2010-12-13 16:23:24 +0200 Stefan Kost * common: Automatic update of common submodule From 011bcc8 to 20742ae 2010-12-13 12:56:12 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: avoid leaking sink events Avoid leaking the newsegment event when it has the wrong format. 2010-12-12 14:53:17 +0100 Wim Taymans * gst/rtp/gstrtpmp4vpay.c: mp4vpay: we can also accept xvid caps 2010-12-12 01:39:06 +1100 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: deinterlace: Avoid infinite loop draining frames When the pipeline is flushed just as we're draining history, don't loop infinitely, just discard the history and abort. 2010-12-11 17:39:20 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: add "max-errors" property to ignore decoding errors Add property to ignore decoding errors. Default is to ignore a few decoding errors if the input is packetized, but error out immediately if the input is not packetized. Ignoring errors for packetized input most likely doesn't work properly yet, so don't do that for now. https://bugzilla.gnome.org/show_bug.cgi?id=623063 2010-05-28 15:27:14 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegenc.c: jpegenc: free/malloc instead of realloc, avoids memcpy 2010-12-11 17:49:03 +0100 Sebastian Dröge * gst/qtdemux/qtdemux.c: qtdemux: Check if there's actually a seek table before parsing it 2010-12-11 17:46:17 +0100 Kishore Arepalli * gst/qtdemux/qtdemux.c: qtdemux: Implement CONVERT and FORMATS query Fixes bug #636784. 2010-07-01 00:22:07 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: put unrecognised RIFF format IDs into the unknown caps Extra info can't hurt. Field names aren't necessarily consistent with what's used elsewhere though (e.g. avidemux), but then neither are the caps. https://bugzilla.gnome.org/show_bug.cgi?id=623178 2010-10-29 22:50:14 +0100 Jan Schmidt * ext/pulse/pulsemixerctrl.c: * ext/pulse/pulsemixerctrl.h: pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS Add the mixer flag and send notifications when either the volume or muted status changes. https://bugzilla.gnome.org/show_bug.cgi?id=618389 2010-02-08 21:41:29 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: mark DISCONT when resuming PLAY In particular, when streaming interleaved, this arranges for setting a new timestamp on outgoing buffer so downstream can appropriate reset to a change in (rtp)time. 2010-12-02 16:08:34 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response 2010-10-25 11:51:06 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: add and use auto buffering mode ... which selects BUFFER for a non-live stream, and otherwise SLAVE. Fixes #633088. 2010-12-06 12:16:12 +0100 Wim Taymans * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: j2kdepay: make the depayloader more resilient Use 3 adapters, one to accumulate paketization units, another on to accumulate tiles and a last one to accumulate the final frame. Don't just blindly flush the adapter on DISCONT but only discard the current packetization unit. When we dropped jpeg2000 packets between SOP markers, adjust the SOT header with the new lenght. 2010-12-09 13:49:04 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix flow return aggregation 2010-12-08 11:35:33 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix handling near end-of-file corner cases Also, relax some error handling to not bail out completely when something feels amiss, but consider this EOF and continue with was obtained so far. 2010-12-07 17:19:00 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; fix offset handling and relax error raising In particular, accept unknown stream in track fragment, and only error out if that raises problems later on with respect to offset tracking. Fixes #620283. 2010-12-07 15:39:32 +0100 Wim Taymans * tests/check/pipelines/lame.c: check: don't use deprecated method 2010-12-07 13:11:48 +0100 Mark Nauwelaerts * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: flvdemux: use aac codec-data to adjust samplerate if needed Based on patch by Fabien Lebaillif-Delamare Fixes #636621. 2010-12-07 11:43:13 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: don't uncork in _start Don't uncork in the _start method just yet but wait until we have written some samples to pulseaudio. This avoid underruns on pulseaudio and less crackling noises when starting. 2010-12-07 11:47:41 +0100 Wim Taymans Merge branch 'master' into 0.11 2010-12-07 11:43:13 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: don't uncork in _start Don't uncork in the _start method just yet but wait until we have written some samples to pulseaudio. This avoid underruns on pulseaudio and less crackling noises when starting. 2010-12-07 11:42:15 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use _object_ref_sink() when we can 2010-12-07 11:40:58 +0100 Wim Taymans * sys/v4l2/gstv4l2object.c: v4l2: don't abuse the class lock Use a new static lock to protect the probed device list instead of the object class lock. 2010-12-06 19:59:49 +0100 Alessandro Decina * gst/qtdemux/qtdemux.c: qtdemux: fix compiler warnings on OSX. 2010-12-06 18:17:24 +0100 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: add debug to notify when skipping to jpeg header 2010-12-06 18:16:19 +0100 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: discard incomplete image ... as determined when finding SOI next image before an EOI. Based on patch by David Hoyt Fixes #635734. 2010-12-06 17:45:38 +0100 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: avoid infinite loop when resyncing Fixes #635734 (partly). 2010-12-06 17:28:32 +0100 Wim Taymans Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2010-12-06 17:27:51 +0100 Wim Taymans * android/apetag.mk: * android/avi.mk: * android/flv.mk: * android/icydemux.mk: * android/id3demux.mk: * android/qtdemux.mk: * android/rtp.mk: * android/rtpmanager.mk: * android/rtsp.mk: * android/soup.mk: * android/udp.mk: * android/wavenc.mk: * android/wavparse.mk: * configure.ac: more 0.10 -> 0.11 changes 2010-12-06 15:21:53 +0100 David Hoyt * gst/imagefreeze/gstimagefreeze.c: imagefreeze: pass along eos if received before buffer arrives Fixes #636172. 2010-10-20 11:05:49 +0200 Andoni Morales Alastruey * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: matroskamux: try to write timestamps in all the outgoing buffers Fixes #632654. 2010-12-06 12:21:00 +0100 Wim Taymans * configure.ac: configure: start 0.11 branch 2010-12-06 12:17:21 +0100 Mark Nauwelaerts * gst/debugutils/progressreport.c: * gst/debugutils/progressreport.h: progressreport: optionally determine progress using buffer metadata Based on patch by Leo Singer Fixes #629418. 2010-12-05 14:39:19 +0100 Edward Hervey * tests/check/elements/interleave.c: check: Fixup the shutting down order First bring down everything to NULL before attempting to unlink or unref anything. Avoids the tests just hanging there for ever waiting to acquire a lock that doesn't exist anymore. 2010-11-04 19:31:45 +0100 Janne Grunau * sys/v4l2/gstv4l2bufferpool.c: v4l2src: set top field first for interlaced buffers if v4l2 exports it https://bugzilla.gnome.org/show_bug.cgi?id=634393 2010-11-04 18:36:09 +0100 Janne Grunau * sys/v4l2/gstv4l2object.c: v4l2src: check field information and set interlaced caps accordingly Reject the format if the field type is not supported. https://bugzilla.gnome.org/show_bug.cgi?id=634391 2010-12-03 17:42:14 +0100 Benjamin Gaignard * Android.mk: * android/NOTICE: * android/apetag.mk: * android/avi.mk: * android/flv.mk: * android/gst/rtpmanager/gstrtpbin-marshal.c: * android/gst/rtpmanager/gstrtpbin-marshal.h: * android/gst/udp/gstudp-enumtypes.c: * android/gst/udp/gstudp-enumtypes.h: * android/gst/udp/gstudp-marshal.c: * android/gst/udp/gstudp-marshal.h: * android/icydemux.mk: * android/id3demux.mk: * android/qtdemux.mk: * android/rtp.mk: * android/rtpmanager.mk: * android/rtsp.mk: * android/soup.mk: * android/udp.mk: * android/wavenc.mk: * android/wavparse.mk: Add build system for Android 2010-03-26 13:51:58 +0100 Guillaume Emont * gst/debugutils/gstnavseek.c: navseek: add basic support to change playback rate The following keys will now be interpreted by navseek: 'f' means fast forward: the stream gets played at rate 2.0 'r' means rewind: the stream gets played at rate -2.0 'n' means normal: the stream gets played at rate 1.0 Fixes #631516. 2010-12-01 13:12:04 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: add support for e(a)c-3 audio 2010-11-19 12:44:35 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: avoid sending EOS event twice 2010-11-19 12:44:18 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: remove dead code trying to update stream duration On the one hand, it insufficiently checks whether it only updates a dummy segment. On the other hand, only doing this at the time the last sampled is prepared (and sent downstream) is too little too late. 2010-11-09 10:58:57 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; handle ismv sample flags 2010-11-08 11:41:21 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; handle ismv stbl atoms ... or lack of some thereof, such as mandatory stsz. Shuffle some code in _stbl_init to detect this early enough. 2010-11-08 11:39:37 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; compensate for ismv offset handling ... or lack thereof, which according to specs would put media data in unlikely position. 2010-11-04 14:07:56 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: fragmented support for push mode 2010-11-04 10:17:37 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: fragmented support; proper and incremental moof parsing That is, parse each moof in one pass (considering all contained streams' metadata), and do so incrementally as needed for playback rather than an initial complete scan of all moof (though all moov sample metadata is fully parsed at startup). 2010-11-04 10:06:30 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: refactor stream freeing 2010-11-04 10:05:15 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: delegate linear search for sample to binary search when possible Also arrange for parsing a sample prior to taking a reference to it, which requires less memory layout assumptions for correctness. 2010-11-01 15:52:29 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; handle moov samples and proper stream duration 2010-11-01 13:40:05 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; consider mvex and handle flags and offset fields 2010-10-28 16:49:41 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fragmented support; forego check for short streams ... as some bogus files may indicate streams of 0 duration in moov, while indicating the complete movie duration in mvhd (the latter should be in mehd). 2010-10-28 16:46:48 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_types.h: qtdemux: fragmented support; code cleanups and optimizations in atom parsing Avoid extra allocation in _parse_trun, add more checks for parsing errors, add or adjust some debug statement, fix comments, sprinkle some branch prediction. 2010-09-13 23:19:44 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: parse_moof should return TRUE on success 2010-09-10 22:41:03 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Fix iteration bug Avoid infinite loop when iterating traf 2010-09-10 21:32:26 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Refactor trun parsing The allocation of the samples can be placed out of the loop. Makes the code clearer. Also avoid relying on traf information as it is placed on the end of the file and might not be acessible on push mode. 2010-09-10 00:29:26 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Remove parsing of unused atom sdtp atom is parsed but not used, so we don't have to parse it. 2010-11-09 11:45:00 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: tweak wam support ... with some comment and portability macros. 2009-09-23 18:47:42 +0200 Marc-André Lureau * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: qtdemux: support wma & vc-1 https://bugzilla.gnome.org/show_bug.cgi?id=596321 2010-03-11 09:56:04 +0100 Andoni Morales Alastruey * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: parse fmp4 samples information The fragmented mp4 format stores the tracks and samples information in the 'moof' boxes, which are appended before each fragment (fragment->'moof'+'mdat'). The 'mfra' box stores the offset of each 'moof' box and their presentation time. The location of this box can be retrieved from the 'mfro' box, which is located at the end of the file. The 'mfra' box is parsed to get the offset of each 'moof' box and their presentation time. Each 'moof' box can contain information for one or more tracks inside 'tfhd' boxes. For each track in a 'moof', we have a 'trun' box, which contains information of each sample (offset and duration) used to build the samples table. Based on patch by Marc-André Lureau https://bugzilla.gnome.org/show_bug.cgi?id=596321 2010-03-11 15:34:49 +0100 Marc-André Lureau * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux_dump.c: * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: * gst/qtdemux/qtdemux_types.h: qtdemux: add fragmented mp4 fourccs Adds fourcc's for tfra, tfhd, trun, sdtp, trex, mehd and their dumps https://bugzilla.gnome.org/show_bug.cgi?id=596321 2010-03-11 10:24:56 +0100 Marc-André Lureau * gst/qtdemux/qtdemux.c: qtdemux: parse the track id from the track header Signed-off-by: Andoni Morales Alastruey https://bugzilla.gnome.org/show_bug.cgi?id=596321 2010-03-11 14:10:12 +0100 Marc-André Lureau * gst/qtdemux/qtdemux.c: qtdemux: allow pulling atoms with unknown size Signed-off-by: Andoni Morales Alastruey https://bugzilla.gnome.org/show_bug.cgi?id=596321 2010-07-14 20:13:55 +0200 Marc-André Lureau * gst/qtdemux/qtdemux_dump.c: qtdemux: make qtdemux_dump_mvhd parse version 1 correctly Versions 0 and 1 of mvhd have different sizes of its values (32bits/64bits). This patch makes it dump them correctly. Also use the right node in the parameter and not the root node. https://bugzilla.gnome.org/show_bug.cgi?id=596321 2010-11-19 12:45:00 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskademux: minor cleanups in setting streamheader on caps 2010-11-02 17:04:04 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: normalize empty Cues to no Cues ... to trigger indexless seeking. 2010-10-26 11:15:49 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: add workaround for buggy list size Fixes truncated extra-data in hdrl/strl/strf due to buggy containing list size not accounting for padding in contained chunks. 2010-12-02 16:11:01 +0100 Mark Nauwelaerts * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: do not hold custom PAD_LOCK when pushing downstream 2010-12-02 16:10:14 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: reset session manager base time when flushing ... as rtpbin uses running time to handle rtpjitterbuffer's buffer mode pauses. 2010-12-01 16:51:33 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: include range request for all streams with non-aggregate control 2010-10-07 14:50:53 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: fix debug statement 2010-12-03 15:38:00 +0100 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Parse more variants of numerical IDIT tag 2010-05-07 17:30:30 +0200 Edward Hervey * ext/libpng/gstpngenc.c: pngenc: Use proper framerate range in caps 2010-12-03 15:04:26 +0100 Edward Hervey * tests/check/pipelines/wavenc.c: tests: Fix previously unbuildable/untested wavenc test 2010-10-24 15:21:08 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Refactor tag pushing logic The logic of when to push was wrong also (resulting in some tags never being pushed). 2010-10-24 15:20:27 +0200 Edward Hervey * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: flvdemux: Use pbutils for codec descriptions 2010-04-13 11:29:30 +0200 Edward Hervey * tests/check/elements/udpsink.c: check: Use fail_unless_equals_int instead of fail_if Makes the error message more interesting 2010-11-30 19:22:11 +0100 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Also extract IDIT tags present too early https://bugzilla.gnome.org/show_bug.cgi?id=636143 2010-11-30 19:21:23 +0100 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Also emit DateTime tag https://bugzilla.gnome.org/show_bug.cgi?id=636143 2010-12-03 00:22:48 +0000 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: detect DTS advertised as PCM correctly in some more cases The DTS typefinder may return a lower probability for frames that start at non-zero offsets and where there's no second frame sync in the first buffer. It's fairly unlikely that we'll acidentally identify PCM data as DTS, so we don't do additional checks for now. https://bugzilla.gnome.org/show_bug.cgi?id=636234 2010-11-08 17:11:42 +0200 Stefan Kost * tests/check/Makefile.am: tests: makefile cleanup Fix indentation. Use $(GST_MAJORMINOR) instead of hardcoded 0.10. 2010-11-08 17:02:56 +0200 Stefan Kost * tests/check/Makefile.am: * tests/check/pipelines/.gitignore: * tests/check/pipelines/wavenc.c: tests: add a test for wav muxing 2010-11-08 16:57:17 +0200 Stefan Kost * tests/check/elements/interleave.c: * tests/check/pipelines/wavpack.c: tests: remove newlines between variable decls (old gst-indent failure) 2010-11-08 14:47:04 +0200 Stefan Kost * ext/libpng/gstpngdec.c: pngdec: use png_error() as recommended by libpng docs to signal an error Without that the element loops endlessly on broekn pngs. Fixes #634314 2010-11-16 17:48:16 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Parse and use creation time tag from mvhd Expose creation time from mvhd as a datetime tag Fixes #634928 2010-10-27 19:15:20 +0200 Andoni Morales Alastruey * gst/icydemux/gsticydemux.c: icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag 2010-10-23 19:34:00 -0400 Tom Janiszewski * gst/flv/gstflvmux.c: flvmux: Fix for nellymoser codecid setting Fixes bug #632897. 2010-10-21 16:15:08 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Add support for E-AC3 2010-10-21 16:14:44 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Add support for DTS 2010-10-31 18:08:17 +0100 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't send seeks behind the end of file to the server Also improve debug output, re-initialize the content size and let the seek handler error out on invalid seek segments. Fixes bug #632977. 2010-12-02 17:53:42 +0100 Wim Taymans * gst/rtp/gstrtpj2kpay.c: j2kpay: use SOP markers to split bitstream When parsing the bitstream, look for SOP markers because we are allowed to split packets on those marker boundaries. Rework the parsing code a little so that we can pack multiple Packetization units in one RTP packet. 2010-11-18 12:49:47 +0100 Wim Taymans * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: rtpj2kpay: use buffer lists Use buffer lists for doing zerocopy payloading. Add property to disable buffer lists. 2010-11-16 16:54:25 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: h264pay: small cleanups Allocate adapter only once. Make some guint8 * const. 2010-11-16 15:39:24 +0100 Tambet Ingo * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: implement full bytestream scan mode. Implement the full bytestream scan mode. Fixes #634910 2010-11-15 10:52:31 +0100 Wim Taymans * tests/examples/rtp/client-H263p-AMR.sh: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H263p.sh: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-H264.sh: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: examples: improve RTP examples Make the examples use autovideosink and ffmpegcolorspace for better compàtibility. Make some more variables for the sink and the decoders. Set zerolatency tuning on x264enc for better realtime results. 2010-11-10 11:04:48 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: select multicast transports in a smarter way When we see a multicast address in the SDP connection, only try to negotiate a multicast transport with the server. Fixes #634093 2010-12-02 18:14:16 +0000 Tim-Philipp Müller * configure.ac: Bump GLib requirement to implicit requirement ie. >= 2.20 while we depend on core/base 0.10.31 2010-12-02 18:13:57 +0000 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development === release 0.10.26 === 2010-12-01 21:15:09 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.26 2010-11-30 15:28:50 -0800 David Schleef * gst/deinterlace/gstdeinterlace.c: deinterlace: analyse RFF fields in correct order Code was repeating the second field, not the first. Fixes: #636179. 2010-11-29 15:32:40 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: handle stale digest authentication session data In particular, handle Unauthorized server response when trying to convey keep-alive. Fixes #635532. 2010-11-26 15:00:29 +0100 Thijs Vermeir * gst/rtp/gstrtph264depay.c: rtph264depay: fix segfault on empty payload https://bugzilla.gnome.org/show_bug.cgi?id=635843 2010-11-25 19:24:56 +0100 Edward Hervey * gst/audiofx/gststereo.c: stereo: Remove dead assignments 2010-11-25 19:06:27 +0100 Edward Hervey * gst/dtmf/gstrtpdtmfdepay.c: dtmf: Remove dead assignments 2010-11-18 00:45:29 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.25.5 pre-release 2010-11-18 00:44:45 +0000 Tim-Philipp Müller * po/bg.po: * po/fi.po: * po/hu.po: * po/sk.po: * po/tr.po: po: update translations 2010-11-14 00:18:16 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix reference leak 2010-11-12 23:59:06 +1100 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: deinterlace: Flush QoS and history before applying segment When handling newsegment, flush out the buffer history in the existing segment, not the new one. Fixes playback in some DVD cases. Partially fixes #633294 2010-11-12 12:20:16 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: improve event logging 2010-11-05 17:00:15 +0100 Robert Swain * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Implement field history flushing In a number of cases it is necessary to flush the field history by performing 'degraded' deinterlacing - that is, using the user-chosen method for as many fields as possible, then using vfir for as long as there are >= 2 fields remaining in the history, then using linear for the last field. This should avoid losing fields being kept for history for example at EOS. This may address part of #633294 2010-11-05 15:44:35 +0100 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Refactor chain function This is needed to be able to output a frame from outside the chain function, i.e. in the following commit that adds flushing of the field history. 2010-11-05 17:17:56 +0000 Tim-Philipp Müller * configure.ac: configure: we still require Gtk+ >= 2.14.0 when compiling against 2.0 The check for the minor version was dropped in the previous commit. 2010-11-05 16:24:42 +0000 Tim-Philipp Müller * configure.ac: configure: add --with-gtk option and default to Gtk+ 2.0 while the 3.0 API is still in flux https://bugzilla.gnome.org/show_bug.cgi?id=634014 2010-11-04 16:42:07 +1000 Jonathan Matthew * gst/icydemux/gsticydemux.c: icydemux: fix use-after-free of taglist Broken by commit 4c2f5333 (bug #630205). https://bugzilla.gnome.org/show_bug.cgi?id=633970 2010-11-01 17:29:01 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.25.4 pre-release 2010-11-01 17:28:36 +0000 Tim-Philipp Müller * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/es.po: * po/fr.po: * po/it.po: * po/nb.po: * po/nl.po: * po/pl.po: * po/sl.po: * po/sv.po: po: update translations 2010-11-01 16:04:20 +0000 Tim-Philipp Müller * configure.ac: configure: fix --disable-external 2010-11-01 14:56:28 +0100 Wim Taymans * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: only set delta unit on all-non-key units Only set the delta flag when all of the units in the packet are delta units. Based on patch from Olivier Crête Fixes #632945 2010-10-26 15:44:37 -0300 Thiago Santos * gst/goom/gstgoom.c: goom: Return not-negotiated when bps is unknown If caps weren't negotiated, goom should return not-negotiated from its chain functions instead of using bps unitialized, which leads to a division by 0 https://bugzilla.gnome.org/show_bug.cgi?id=633212 2010-10-27 13:16:54 +0100 Jan Schmidt * common: Automatic update of common submodule From 7bbd708 to 011bcc8 2010-10-26 16:54:11 +0100 Jan Schmidt * gst/videofilter/gstvideoflip.c: videoflip: Forward src pad events upstream. Fix passing navigation and other events upstream by actually sending them. Fixes: #633205 2010-10-24 18:50:30 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: fix deadlock in error code path GST_ELEMENT_ERROR must not be called with the object lock held, since it will call gst_object_get_parent() internally, which takes the object lock as well. 2010-10-20 10:21:48 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: Remove useless clearing of send_xiph_headers for Dirac This looks like a mistake when copy-pasting the Theora code. https://bugzilla.gnome.org/show_bug.cgi?id=632682 2010-10-20 13:28:28 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: don't crash if vorbis/theora codec data is missing Error out properly in this case instead of crashing. https://bugzilla.gnome.org/show_bug.cgi?id=632682 2010-10-22 18:11:46 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.25.3 pre-release 2010-10-19 16:45:51 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: fix duration reporting Init segment prior to storing duration info in it. Fixes #632548. 2010-10-19 14:21:53 +0100 Bastien Nocera * gconf/Makefile.am: gconf: Don't install schemas when GConf is disabled https://bugzilla.gnome.org/show_bug.cgi?id=632553 2010-10-19 13:43:14 +0300 Stefan Kost * gst/audiofx/gststereo.c: various (gst): add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. 2010-10-19 13:43:14 +0300 Stefan Kost * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: various (gst): add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. 2010-10-19 13:44:25 +0300 Stefan Kost * gst/dtmf/gstdtmfsrc.c: dtmfsrc: remove DEBUG_FUNCPTR from gobject vmethods 2010-10-19 12:20:40 +0300 Stefan Kost * ext/lame/gstlame.c: various: canonicalize property names 2010-10-19 10:06:33 +0300 Stefan Kost * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: various (ext): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-16 15:43:53 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: win32: set GST_PACKAGE_RELEASE_DATETIME also in win32 config.h 2010-10-16 01:33:52 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.25.2 pre-release 2010-10-16 01:26:01 +0100 Tim-Philipp Müller * po/el.po: * po/vi.po: po: update translations 2010-10-15 13:22:03 -0700 David Schleef * tests/check/Makefile.am: tests: Don't dist generated orc files 2010-10-15 14:02:19 -0700 David Schleef * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: Update generated orc code 2010-10-15 18:00:10 +0100 Tim-Philipp Müller * configure.ac: configure: bump Orc requirement to 0.4.11 2010-10-14 17:41:30 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Use the right constant to define the "use-pipeline-clock" property The wrong #define was being used, now use the correct one. 2010-10-14 12:31:48 -0700 David Schleef * common: Automatic update of common submodule From 5a668bf to 7bbd708 2010-10-14 17:26:14 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/qtdemux/qtdemux.c: ac3: demuxers provide framed output 2010-10-14 00:11:27 +0100 Tim-Philipp Müller * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: matroskamux: reduce newsegment event spam and set discont flag where needed Only send newsegment events with new positions downstream when actually needed, instead of sending multiple newsegment events with new seek positions in a row. Also set the discont flag on buffers after a discontinuity. 2010-10-13 23:46:02 +0100 Tim-Philipp Müller * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: matroskamux: set correct buffer offsets after seeks Re-use the existing 'pos' field maintained by ebml writer to set buffer offsets. This also makes sure that we set the right offsets on buffers after a seek (e.g. when writing an index at the end). 2010-10-14 00:22:03 +0100 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: don't forward tag events downstream Don't forward stream-specific tag events downstream (esp. not before any newsegment event).x 2010-10-13 17:15:25 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: handle another mp4v variation ... including the glbl atom containing codec-data. 2010-10-13 17:21:23 +0300 Stefan Kost * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavimux.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/efence.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/negotiation.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/id3demux/gstid3demux.c: * gst/level/gstlevel.c: * gst/matroska/matroska-mux.c: * gst/median/gstmedian.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtsp/gstrtpdec.c: * gst/smpte/gstsmpte.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstvideotemplate.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: various (gst): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 17:13:04 +0300 Stefan Kost * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/ximage/gstximagesrc.c: various (sys): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 16:25:15 +0300 Stefan Kost * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/annodex/gstcmmltag.c: * ext/cairo/gsttextoverlay.c: * ext/dv/gstdvdec.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacenc.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: * ext/hal/gsthalaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/wavpack/gstwavpackenc.c: various (ext): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 16:34:09 +0300 Stefan Kost * ext/aalib/gstaasink.c: * ext/esd/esdmon.c: * gst/median/gstmedian.c: various: wrap property registration and add a single fixme for long desc. 2010-10-13 11:46:58 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: h264depay: always mark the codec_data as keyframe We need to mark the codec_data as a keyframe or else downstream decoders might decide to skip it, waiting for a keyframe. Fixes #631996 2010-10-13 07:16:47 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: matroskamux: make buffer offsets a byte count rather than a buffer count 2010-10-07 21:12:48 +0100 Tim-Philipp Müller * ext/aalib/gstaasink.c: * ext/dv/gstdvdec.c: * ext/esd/esdmon.c: * ext/flac/gstflacenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * gst/debugutils/efence.c: * gst/rtpmanager/gstrtpbin.c: ext, gst: canonicalise property names where this wasn't the case ie. "foo_bar" -> "foo-bar" 2010-10-12 15:02:42 +0200 Thijs Vermeir * gst/rtp/gstrtpmpvpay.c: rtpmpvpay: fix timestamping of rtp buffers Incomming buffer is only pushed on the adapter at the end of the handle_buffer function. But duration/timestamp of this buffer is already taken into account for the current data in the adapter. This leads to wrong rtp timestamps and extra latency. 2010-10-12 11:37:40 +0200 Sebastian Dröge * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: examples: Fix build with GTK+ 3.0 2010-10-11 15:12:00 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: mark as a source Mark the rtspsrc element as a source. Requires 0.10.31.1 now 2010-10-11 14:24:13 +0200 Sebastian Dröge * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosrc.c: autodetect: Set GST_ELEMENT_IS_SOURCE flag on sources 2010-10-11 14:21:07 +0200 Sebastian Dröge * ext/gconf/gstswitchsrc.c: switchsrc: Set the GST_ELEMENT_IS_SOURCE flag 2010-10-11 14:17:33 +0200 Sebastian Dröge * configure.ac: configure: Require core 0.10.30.1 2010-10-10 14:43:58 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: matroskamux: set offsets on outgoing buffers 2010-10-09 14:14:27 +0200 IOhannes m zmölnig * sys/v4l2/gstv4l2sink.c: v4l2sink: Only get/set overlay params if needed it's perfectly ok for a video output device to not have overlay capabilities. this patch removes the need to get/set the overlay parameters if the user does not explicitely request one of the overlay properties 2010-09-30 15:28:23 +0200 IOhannes m zmölnig * sys/v4l2/gstv4l2sink.c: v4l2sink: Protect against NULL-pointer access gst_v4l2sink_change_state() would free the pool without checking whether there was a valid pool... 2010-10-08 12:43:51 -0700 David Schleef * common: Automatic update of common submodule From c4a8adc to 5a668bf 2010-10-08 12:53:33 +0200 Sebastian Dröge * common: Automatic update of common submodule From 5e3c9bf to c4a8adc 2010-10-06 11:29:55 +0200 Robert Swain * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix required fields logic Both history_count and fields_required count from 1. As per the while loop condition that follows this code, to perform the deinterlacing method, we need history_count >= fields_required fields in the history. Therefore if we have history_count < fields_required (not fields_required + 1), we need more fields. 2010-09-20 19:43:45 +0200 Andoni Morales Alastruey * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: resend onMetada tag when tags changes in streamable mode 2010-10-05 19:40:50 +0100 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: AAC codec_data can be > 2 bytes long This fixes the assumption that DecoderSpecificInfo must be 2 bytes long for AAC files. The specification allows HE-AAC to be explicitly signalled in a backward compatible way. This is done by means of an additional information after the regular AAC header. It is expected that decoders that can play AAC but not HE-AAC will parse the header normally and ignore extended bits, much as they do for the HE-AAC specific payload in the actual stream. https://bugzilla.gnome.org/show_bug.cgi?id=612313 2010-10-05 16:01:19 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: only unref buffer when no longer needed for cluster scanning Fixes #629047. 2010-10-05 16:00:45 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: avoid infinite cluster scanning 2010-10-05 12:20:52 +0200 Wim Taymans * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: goom: take duration into account when doing QoS Take the duration of the frames into account so that we don't drop frames that are only partially past the QoS deadline. 2010-10-05 10:40:15 +0200 Wim Taymans * gst/goom/gstgoom.c: * gst/goom/gstgoom.h: * gst/goom2k1/gstgoom.c: * gst/goom2k1/gstgoom.h: goom: use adapter for timestamping Use the adapter timestamp code to get more accurate timestamps. Fix latency calculation, we add our own latency in the worst case. 2010-10-04 22:31:32 +0200 Edward Hervey * configure.ac: * ext/raw1394/Makefile.am: * ext/raw1394/gst1394.c: raw1394: Don't compile hdv1394src if libiec61883 isn't available Fixes #629896 2010-09-20 19:44:09 +0200 Andoni Morales Alastruey * gst/icydemux/gsticydemux.c: icydemux: forward tag events https://bugzilla.gnome.org/show_bug.cgi?id=630205 2010-10-04 19:00:45 +0200 Wim Taymans * gst/goom2k1/gstgoom.c: goom2k1: report our latency correctly Fixes #631303 2010-10-04 18:56:15 +0200 Wim Taymans * gst/goom2k1/gstgoom.c: goom2k1: add defines for default width/height/fps Add some defines for the default width/height/fps instead of using different values in different places. 2010-10-04 18:52:14 +0200 Wim Taymans * gst/goom/gstgoom.c: goom: add latency compensation code. Implement a latency query and report how much latency we will add to the stream. Alse make some defaults for the default width/height/framerate Fixes #631303 2010-10-04 17:56:57 +0200 Wim Taymans * tests/examples/rtp/server-alsasrc-PCMA.py: test: add python version of the audio sender Add a python version of the audio sender pipeline. Ported by Sp4rc on IRC. 2010-10-04 17:52:22 +0200 Wim Taymans * tests/examples/rtp/client-PCMA.py: tests: Add python RTP client example Add a python version of the PCMA client app. Ported by Sp4rc on IRC. 2010-10-04 09:39:59 +0200 Sebastian Dröge * gst/rtp/gstrtpmp4gpay.c: rtp: Fix unitialized compiler warnings on OS X build bot These warnings are wrong though, the variables are only used in the cases where they *are* initialized by the bit reader. 2010-10-03 23:49:08 +0200 Sebastian Dröge * gst/rtp/gstrtpg722pay.c: rtpg722pay: Fix uninitialized variable compiler warning The clock rate is always 8000 Hz according to the RFC and the sampling rate must always be 16000 Hz. 2010-10-01 13:59:10 +0400 Vladimir Eremeev * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: improve article reference in comment block https://bugzilla.gnome.org/show_bug.cgi?id=631082 2010-04-30 21:00:31 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: * gst/qtdemux/quicktime.c: qtdemux: Use pbutils for H.264 profile/level extraction The functions used to extract this data have been moved to gstpbutils to facilitate reuse. https://bugzilla.gnome.org/show_bug.cgi?id=617318 2010-04-30 21:00:31 +0530 Arun Raghavan * gst/matroska/Makefile.am: * gst/matroska/matroska-demux.c: * gst/matroska/matroska.c: matroskademux: Use pbutils for H.264 profile/level extraction The functions used to extract this data have been moved to gstpbutils to facilitate reuse. https://bugzilla.gnome.org/show_bug.cgi?id=617318 2010-04-22 19:39:47 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Export MPEG-4 video profile and level in stream caps This uses gstpbutils to extract the profile and level from the video object sequence and adds this to stream caps. This can be used as metadata and for fine-grained decoder selection. https://bugzilla.gnome.org/show_bug.cgi?id=616521 2010-09-30 12:44:52 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: fix aac channel override based on codec data for 7.1 case 2010-04-30 14:06:27 +0530 Arun Raghavan * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: qtdemux: Export AAC profile and level in caps This exports the AAC profile and level in caps for use as metadata and (eventually) for more fine-grained selection of decoders at caps-negotiation time. (Doesn't work for HE-AAC yet though.) https://bugzilla.gnome.org/show_bug.cgi?id=612313 2010-09-30 18:34:04 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722depay.h: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg722pay.h: rtp: add G722 pay and depayloader 2010-09-30 12:08:49 +0200 Thijs Vermeir * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: update link to documentation 2010-09-30 11:34:56 +0200 Thijs Vermeir * tests/examples/rtp/client-H264.sh: examples: fix indentation on rtp client example 2010-09-30 11:33:24 +0200 Thijs Vermeir * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-H264.sh: examples: fix typo in port of rtp examples 2010-09-29 13:20:22 +0100 Tim-Philipp Müller * gst/wavenc/gstwavenc.c: wavenc: miniscule code clean-up GST_CLOCK_TIME_NONE is not something that should be used in connection with GST_FORMAT_BYTES. 2010-09-29 10:34:36 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: reverse playback; prevent overlap of subsequent fragments 2010-09-28 16:21:48 +0300 René Stadler * gst/rtsp/gstrtspsrc.c: rtspsrc: fix missing null-terminator in protocols array Fixes random crash regression from commit ae84ae. 2010-09-24 16:26:20 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't add /UDP in the transport, it's the default don't add the default UDP lower-transport, some servers don't seem to like it. Fixes #630500 2010-06-25 17:08:03 +0200 Pascal Buhler * gst/rtpmanager/gstrtpjitterbuffer.c: rtpmanager: packet lost should not be a warning. It happens all the time... 2010-09-24 15:33:40 +0200 Pascal Buhler * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe Using _foreach_remove on the hashtable, while releasing the lock protecting that table inside the callback is not a good idea. The hashtable might then change (a source removed or added) while signals like on_timeout are being sent. This solution makes a copy of the table, performs the _foreach without actually removing any sources, but marks them for removal on a second iteration with the real list, but this time not letting go of the lock. Fixes #630452 2010-09-24 15:19:15 +0200 Edward Hervey * gst/id3demux/id3tags.c: id3demux: Sanitize id3 frame names This is similar to what is done in qtdemux. Avoids providing invalid structure/tags names 2010-09-24 14:59:45 +0200 Edward Hervey * gst/apetag/gstapedemux.c: apedemux: Skip empty tags Avoid creating bogus string tags. Also added logging of the string values of the tag name and value. 2010-09-24 08:56:36 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: soup: init debug category before using it 2010-04-12 09:49:14 +0200 Pascal Buhler * gst/rtpmanager/gstrtpbin.c: rtpbin: Handle rysnc of iterator when looking for free pad name If a new pad was added while iterating then a pad could be returned that was already in use. Fixes #630451 2010-09-24 14:09:12 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: fix compilation 2010-04-07 15:31:52 +0200 Trond Andersen * gst/rtpmanager/gstrtpbin.c: rtpbin: Unlock before adding pad in new_payload_found Holding internal locks while potentially calling out is a source of deadlocks, and in this case the application might subscribe to the pad-added signal. Fixes #630449 2009-08-31 18:37:40 +0200 Havard Graff * gst/rtpmanager/rtpsession.c: rtpsession: relax third-party collision detection If the source has been inactive for some time, we assume that it has simply changed its transport source address. Hence, there is no true third-party collision - only a simulated one. Fixes #630447 2010-09-24 13:50:02 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: whitespace fixes 2010-09-24 13:48:50 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: simplify the rate estimation some more 2009-08-31 18:34:08 +0200 Havard Graff * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpmanager: provide additional statistics 2010-09-24 00:01:05 +0100 Tim-Philipp Müller * configure.ac: configure: set plugin release datetime 2010-09-23 21:21:29 +0100 Tim-Philipp Müller * gst/equalizer/gstiirequalizer10bands.h: * gst/equalizer/gstiirequalizer3bands.h: * gst/equalizer/gstiirequalizernbands.h: equalizer: fix class definitions Class structures must be based on the parent class struct, not on the parent instance struct. 2010-09-15 20:36:33 +0100 Tim-Philipp Müller * gst/videomixer/videomixer2.c: videomixer2: pre-register pad class properly with g_type_class_ref Fix code to match the comment. Also, there's no need to register the background enum type again, this is already done via install_property. 2010-09-23 21:57:18 +0200 David Hoyt * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: speex: Fix crashes with MSVC Using the symbols for the different Speex modes results in crashes when using MSVC. Use the library functions to get the modes instead. Fixes bug #630378. 2010-08-24 13:25:02 +0200 Havard Graff * gst/level/gstlevel.c: level: avoid division by zero on silence Fixes bug #630458. 2010-09-23 16:46:31 +0200 Wim Taymans * gst/flv/gstflvdemux.c: flvdemux: parse and use cts For H264, there is an extra header containing the CTS, which is a timestamp offset that should be applied to the PTS. Parse this value and use it to adjust the pts. Fixes #630088 2010-09-23 16:45:41 +0200 Wim Taymans * gst/flv/gstflvdemux.c: flvdemux: improve pts debugging 2010-09-22 19:01:40 +0200 Wim Taymans * configure.ac: * tests/examples/Makefile.am: * tests/examples/pulse/.gitignore: * tests/examples/pulse/Makefile.am: * tests/examples/pulse/pulse.c: pulse: add test app for pulse device probe 2010-09-22 18:50:44 +0200 Wim Taymans * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: fix device_description in READY Make the is_dead check more clear and add an option to check for the status of the stream in addition to the context. We don't need a stream to get the device_description string. Fixes #630317 2010-09-22 12:56:00 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Don't post tags if there are none And make all code go through _post_global_tags. 2010-09-22 12:37:33 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: refactor and simplify AU merging Move the processing of the NALU to a separate method. Simplify the merging of NALU into AU and use common code when possible. 2010-09-21 23:23:07 +0300 Stefan Kost * tests/examples/shapewipe/shapewipe-example.c: shapewipe: add optional border parameter and slowdown animation Allow to play with the border property (sharp/soft edges). 2010-09-21 19:14:40 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: shapewipe: Force format to AYUV in the example pipeline for the same reason 2010-09-21 19:13:07 +0200 Sebastian Dröge * tests/examples/shapewipe/shapewipe-example.c: shapewipe: Force the input to AYUV to prevent negotiation failures in videomixer The second videotestsrc chain might produce YUY2 because everything is accepted downstream before the first shapewipe chain gets negotiated. 2010-09-21 19:12:45 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: shapewipe: Improve debugging and immediately return empty caps from the getcaps functions 2010-09-21 18:33:55 +0200 Edward Hervey * common: Automatic update of common submodule From aa0d1d0 to 5e3c9bf 2010-09-21 12:49:31 +0200 Philippe Normand * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/gstv4l2xoverlay.h: v4l2: use the xoverlay APIs 2010-09-21 12:48:34 +0200 Philippe Normand * configure.ac: * sys/osxvideo/osxvideosink.m: osxvideosink: use the new xoverlay APIs Also bumped -base requirements. 2010-09-21 12:31:59 +0200 Sebastian Dröge * configure.ac: configure: Use -DGST_DISABLE_DEPRECATED again for GIT versions 2010-09-21 11:52:22 +0200 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Fix debug statement 2010-09-20 23:17:35 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Parse uuid atoms in push mode Parses uuid atoms in push mode when they are found, they might contain xmp tags. Also does a minor refactoring to put the global tags posting into a single function instead of repeating it in 3 different places. Fixes #629839 2010-09-16 08:04:02 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Delay tags posting a little Delay tags posting until we've parsed all the headers so that the native and xmp tags get merged before posting https://bugzilla.gnome.org/show_bug.cgi?id=629839 2010-09-15 22:13:43 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Parse xmp packet in uuid atom xmp packet is placed into a top-level uuid atom for isom/mp4 variants. This patch makes qtdemux parse all top-level atoms in pull-mode before starting to push data, making it able to find those tags. https://bugzilla.gnome.org/show_bug.cgi?id=629839 2010-09-17 11:07:52 +0200 Wim Taymans * gst/rtpmanager/rtpstats.c: rtpstats: printf format fixes 2010-09-17 11:07:02 +0200 Wim Taymans * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpg729pay.c: rtppay: some printf format fixes 2010-09-15 18:21:11 +0200 Alessandro Decina * gst/qtdemux/qtdemux.c: qtdemux: fix logic when pushing EOS. Don't check for return values when pushing EOS. Still post an error if EOS is reached and no streams have been found. 2010-09-15 17:02:57 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: docs: add gtk-doc chunks with Since: markers for new v4l2src properties 2010-09-15 18:43:50 +0300 Stefan Kost * tests/examples/v4l2/camctrl.c: camctrl: add license header to demo 2010-09-14 17:41:28 +0200 Alessandro Decina * gst/qtdemux/qtdemux.c: qtdemux: don't send EOS twice on the same pad. 2010-09-14 10:07:58 +0300 Stefan Kost * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: move the shared mainloop from class to static var Just have one static var for the shared mainloop instead of one class variable and copies in the instance. 2010-09-13 17:31:35 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: cleanups for DRI markers Protect against invalid DRI markers. do some cleanups 2010-09-10 11:35:53 -0400 American Dynamics * gst/rtp/gstrtpjpegpay.c: gstrtpjpegpay: Added Define Restart Interval (DRI) Marker Added ability to detect and respond to a JPEG-defined DRI marker 2010-06-19 19:20:18 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Split getting the caps into its own function 2010-09-13 16:03:50 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: small cleanup. 2010-09-13 16:24:26 +0300 Stefan Kost * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: rework context sharing We also need to share the main-loop threads as this owns the context. Thus have a class wide main-loop thread. From this we create a context per client-name. Instead of always looking up the context, we keep this with the instance. The reverse mapping is only needed in pulse singal handlers. This saves a lot of locking. Also one signal handler becomes simpler as ther eis only one mainloop to notify. Now valgind happy - no leaks, no bad reads/writes. This reverts major parts of commit 69a397c32f4baf07a7b2937c610f9e8f383e9ae9. Fixes #628996 2010-09-13 15:44:52 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpstats.c: rtpsession: Small cleanups Make the property description prettier. Actually multiple the bandwidth with the fraction. 2010-06-01 21:35:40 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth Calculate the RTCP bandwidth to be a fraction of the RTP bandwidth if it is specified as a value between 0 and 1. 2010-09-13 15:29:06 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: improve bandwidth recalculation Also recalculate bandwidth when one of the source bandwidths changed. Use the newly calculated bandwidth. 2010-06-01 21:17:26 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: rtpsession: Add the option to auto-discover the RTP bandwidth 2010-09-13 14:38:11 +0200 Thijs Vermeir * gst/rtpmanager/gstrtpbin.c: rtpbin: set use-pipeline-clock on correct GObject 2010-06-02 17:51:12 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Initialise the average scaled by 16 2010-09-13 12:41:11 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: add running_time argument docs 2010-06-23 16:13:01 -0400 Olivier Crête * gst/rtpmanager/rtpstats.h: rtpstats: Rectify description of current_time in RTPArrivalStats It is the current time, it is unrelated to when the packet was actually received. 2010-09-13 12:31:40 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: compute the average correctly scaled 2010-06-01 20:31:18 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Count sent RTCP packets after they have been finished If they are counted before calling gst_rtcp_buffer_end(), then the size is way too big. 2010-06-01 19:51:34 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Don't unref pads in finalize The gstrtpsession object is not holding any reference to them directly 2010-09-12 00:09:09 +0100 Tim-Philipp Müller * po/POTFILES.in: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/gl.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ro.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations for new souphttpsrc messages 2010-09-12 00:08:05 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: soup: hook up i18n bits for plugin Call bindtextdomain() etc. 2010-09-12 00:04:42 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: soup: fix error messages Error messages should be translated. URIs and filenames should not be part of the error message string that's shown to the user. soup_message->reason_phrase is not translated and not suitable as error message for users (see libsoup documentation). Also fix up error codes a bit, as far as possible with the existing codes. 2010-09-10 09:43:24 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: don't post an error message if buffer alloc fails with NOT_LINKED flow This is not fatal, let upstream handle it. 2010-09-10 18:06:48 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't clear sdp when set as uri when we set the SDP with an uri, don't clear it when we go to READY. 2010-09-10 18:01:18 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use sdp uri parse method Use the sdp parse method that does proper uri escaping. 2010-09-10 16:59:10 +0300 Stefan Kost * tests/examples/v4l2/.gitignore: * tests/examples/v4l2/Makefile.am: * tests/examples/v4l2/camctrl.c: example: add v4l2 example, demonstrating the use of gst controller 2010-09-10 16:55:25 +0300 Stefan Kost * sys/v4l2/v4l2src_calls.c: v4l2src: don't skip calculating the duration 2010-06-22 15:48:04 +0300 Stefan Kost * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2src: add controlable colorbalance parameters Expose colorbalance controls as object properties (like we do on xvimagesink). Make them controlable. 2010-09-10 13:25:39 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: fix some mis-implementation Also add some debug. 2010-09-10 13:24:02 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: properly insert dummy buffers 2010-09-10 11:55:26 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add rtsp-sdp protocol support Allow setting an SDP with the rtsp-sdp:// url. Based on patch from Marco Ballesio. See #628214 2010-09-10 11:35:58 +0200 Alessandro Decina * gst/alpha/gstalphacolor.c: alphacolor: make passthrough work. 2010-09-09 21:43:40 +0300 Stefan Kost * gst/rtp/gstrtpmp4adepay.c: mp4adepay: small logging cleanup and addition to debug config parsing 2010-09-09 21:42:46 +0300 Stefan Kost * ext/aalib/gstaasink.c: aasink: fix context initialisation and freeing to not leak 2010-09-09 21:40:51 +0300 Stefan Kost * tests/check/Makefile.am: * tests/check/generic/states.c: tests: allow running state tests for all elements Now one can use GST_NO_STATE_IGNORE_ELEMENTS=1 make generic/states.check to try elements that would normaly be skipped. 2010-09-09 18:47:56 +0200 Wim Taymans * tests/check/elements/rtp-payloading.c: tests: fix rtpjpegpay test Make the data we send to the jpeg payloader be a valid jpeg file because the payloader now expects this. 2010-09-09 18:47:11 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: improve debugging 2010-09-09 16:31:56 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: use valid bitrate for dummy frame 2010-09-08 17:07:53 -0300 Thiago Santos * ext/taglib/gstid3v2mux.cc: id3v2mux: Adds mapping for album artist Maps GST_TAG_ALBUM_ARTIST to TPE2 in id3v2mux 2010-09-08 18:35:08 +0200 Sebastian Dröge * configure.ac: configure: Require orc 0.4.8 The deinterlace plugin apparently fails to compile with older versions. 2010-09-08 17:50:11 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: QoS handling logic only applies to forward playback Fixes #628894. 2010-09-08 17:43:47 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: remove unused code 2010-09-08 14:36:48 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: fixup last commit We need to prevent the eventual leak better. 2010-09-08 14:16:58 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: code cleanups Use g_slist_prepend as we don't care about the order. Check for list == NULL instead of iterating the list to see if it is empty. Move ctx allocation down to prevent leak in case of failure. 2010-09-08 07:13:42 +0200 Sebastian Dröge * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Fix uninitialized variable compiler warning Fixes bug #629018. 2010-09-07 19:02:01 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: simplify clock provide code Don't leak the pulsesink element by having the clock keep a ref to the sink. Create the clock only once in the constructor and use the baseaudiosink clock cleanup code. 2010-09-07 17:49:05 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: move the context table init to _get_type phase This seems to fix the invalid reads on context shutdown better, altough I can't really explain. 2010-09-07 17:06:02 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: use older g_array_free g_array_unref() is only since 2.22 2010-09-07 16:49:16 +0200 Wim Taymans * ext/jpeg/gstjpegdec.c: jpegdec: avoid invalid adapter flush on QoS First store the available data in the adapter in the rem_img_len instance field before trying to flush the adapter with that value on QoS. 2010-09-07 16:40:08 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: do some more sanitity checks Protect some more against invalid input. 2010-09-07 15:20:12 +0200 American Dynamics * gst/rtp/gstrtpjpegpay.c: jpegpay: handle corrupted jpeg better Protect against corrupted jpeg input. 2010-09-07 13:55:04 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: rvawdepay: cleanup unused fields 2010-09-07 13:51:37 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: handle invalid payload better Make sure we don't read more data than available in the input buffer. Clip the input data into the output buffer. 2010-08-16 15:35:51 +0300 Stefan Kost * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: allow setting stream properties Add a "properties" property to the elements to allow setting extra stream properties. Fixes #537544 2010-09-07 12:08:10 +0100 Tim-Philipp Müller * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-gdkpixbuf3.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: remove introspection info for gdkpixbuf3 plugin and update version for others The versions got accidentally reverted to a pre-release version, fix that. 2010-09-07 11:42:10 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From c2e10bf to aa0d1d0 2010-09-07 09:20:03 +0100 Tim-Philipp Müller * ext/annodex/gstcmmldec.c: cmmldec: fix flow return handling Fix buggy GST_FLOW_IS_FATAL substitution, and 'make check': - if (!GST_FLOW_IS_FATAL (dec->flow_return) && !dec->sent_root) { + if (dec->flow_return != GST_FLOW_OK && !dec->sent_root) { 2010-09-07 00:27:07 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: don't free the context multiple times Apparently the close function of the ring-buffer can be called multiple times. 2010-08-12 12:33:06 +0300 Stefan Kost * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: grab the sampling arte and put into caps This is needed to be able to mux the received audio into mp4 (in the case of aac). Fixes #625825. 2010-09-06 14:40:02 +0100 Tim-Philipp Müller * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpqcelpdepay.c: rtp: mark constant tables as const 2010-08-18 14:40:48 +0200 Mark Nauwelaerts * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: rtpamrpay: properly support perfect-rtptime 2010-08-18 11:42:33 +0200 Mark Nauwelaerts * gst/rtp/gstrtpamrpay.c: rtpamrpay: proper duration for multiple frame payload 2010-08-18 11:42:07 +0200 Mark Nauwelaerts * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: rtpamr(de)pay: support AMR-WB SID frame 2010-08-18 11:39:06 +0200 Mark Nauwelaerts * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpg729pay.h: rtpg729pay: properly support perfect-rtptime 2010-08-16 16:08:04 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: improve framerate determining Collect a limited number of starting sample durations and use the median of those to determine caps framerate. 2010-08-17 12:08:10 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: attempt more resync upon (cluster) parse error That is, if parse error occurs in state requiring to move to next cluster, and doing so to the expected next position of cluster fails, then scan for a next cluster from present position and resume from there. Fixes #620790. 2010-08-16 16:05:41 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: not so fatal error handling If some bits out of place in block(group) parsing, forego and move to next. Also skip large blocks in pull mode, but need to give up in push mode. Fixes #626463. Improves #620790. 2010-07-26 15:51:49 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: additional parse recovery In particular, upon parse failure in one cluster, we may forego remaining content and try resuming from next cluster onwards. Fixes #620790. 2010-08-26 02:54:55 -0400 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: dtmfsrc: Make the dtmfsrc accept events sent with gst_element_send_event The doc says to use gst_element_send_event on the pipeline, but if we are to call it on the element itself, it's a noop. This should make it handle the event properly before delegating it to basesrc. 2010-09-06 12:22:11 +0200 American Dynamics * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add property to configure udpsrc buffer size Add a new udp-buffer-size property to configure the buffer-size on the udpsrc elements. Fixes #628058 2010-08-27 17:58:47 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: add ntp-sync property Add an ntp-sync property that will sync the received streams to the server NTP time. This requires synchronized NTP times between the sender and receivers, like with ntpd. Based on patch from Thijs Vermeir. Fixes #627796 2010-08-27 12:14:25 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: rename a variable to avoid confusion 2010-08-27 11:07:34 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: rename some variables for less confusion 2010-08-27 10:41:01 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: move comment where it belongs 2010-08-26 16:00:38 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: session: minor cleanups Make clock snapshots more accurate by only sampling the same clock once. 2010-08-26 10:58:26 +0200 Thijs Vermeir * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: rtpbin: add use-pipeline-clock property With this property RTCP SR NTP times can be based on the system clock (maybe synced with ntpd) or the current pipeline clock. https://bugzilla.gnome.org/show_bug.cgi?id=627796 2010-08-25 09:58:20 +0200 Wim Taymans * gst/rtsp/gstrtspext.c: rtspext: stop configuration on first failure Stop the configuration of a stream as soon as some of the extensions return FALSE. Fixes #581294 2010-08-20 15:35:27 +0200 Wim Taymans * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: multifdsink: use refcount to count host/port duplicates Instead of adding multiple client structures for the same host/port pair, use a refcount. Add a send-duplicates feature that allows you to disable sending multiple copies of the same packet to the same host when it was added multiple times. The send-duplicates property is by default set to TRUE for backwards compatibility although it is very likely that this is not desired behaviour. 2010-08-19 17:06:26 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: implement custom event handler Extend the _push_event() function so that it can also send events to the udp sources when asked. Implement a custum send_event function that correctly dispatches the downstream events in TCP mode. This fixes sending EOS to rtspsrc and have it push the EOS downstream. 2010-08-19 11:37:04 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: use _get_caps_reffed() when we can Use _get_caps_reffed() Add some more debug when opening the server connection. 2010-08-16 11:29:07 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegdepay.h: jpegdepay: handle DISCONT and reset state Put a DISCONT event on the next output buffer when the input buffer had a DISCONT. Make sure we clear our adapter and reset our state before going to PAUSED. Free the qtables. Fixes #626869 2010-08-16 11:27:53 +0200 Wim Taymans * gst/rtp/gstrtpg729pay.h: g729pay: extend from right parent 2010-09-06 09:57:10 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: add since docs for new property. 2010-08-30 16:45:48 +0300 Stefan Kost * gst/qtdemux/qtdemux.c: qtdemux: use GST_BOILERPLATE macro 2010-08-16 17:23:58 +0300 Stefan Kost * gst/videomixer/videomixer.c: videmixer: add a example showing how to use the child properties Show how to position and set the alpho of the videos on gst-launch. 2010-08-16 15:19:38 +0300 Stefan Kost * ext/pulse/pulsesrc.c: pulsesrc: move the property-setter to the getter. 2010-08-11 15:48:18 +0300 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum only aggregate magnitude/phase if user asks for it 2010-08-11 15:45:56 +0300 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: improve performance with local vars Use 'input' instead of 'spectrum->input' which was intende already (variable exists, but not used everywhere). Also use a local version of 'spectrum->input_pos'. 2010-08-11 15:44:03 +0300 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: code cleanup More comments and logging. Extract one complex condition to a variable. Reorder some code for readability. 2010-08-11 15:40:09 +0300 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: improve property setter consistently only update if the property actualy changed the value. Do it without reading the gvalue twice. No need to reset the spectrum analyzer for threshold changes. 2010-08-11 15:38:24 +0300 Stefan Kost * gst/spectrum/gstspectrum.c: spectrum: add helper to only flush ringbuffer data without resetting the fft Reduces some duplicated code as well. 2010-08-11 12:45:53 +0300 Stefan Kost * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: more comments 2010-09-05 22:22:42 -0700 David Schleef * gst/deinterlace/gstdeinterlace.c: deinterlace: Document methods with bad quality 2010-09-05 22:19:56 -0700 David Schleef * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: initialize all deinterlace class members This fixes UYVY deinterlacing. 2010-09-05 18:58:13 -0700 David Schleef * common: Automatic update of common submodule From d3d9acf to c2e10bf 2010-09-05 18:45:21 -0700 David Schleef * gst/videomixer/blend.c: videomixer: orc_init() doesn't need to be called There's no need to call orc_init() unless you're using the Orc API directly. All code created by orcc is guaranteed to work without calling orc_init(). 2010-09-05 18:40:48 -0700 David Schleef * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime.orc: * gst/deinterlace/tvtime/greedy.c: deinterlace: Fix greedyl Orc implementation To agree with the previous C/asm code. 2010-09-05 22:31:34 -0300 Thiago Santos * gst/videomixer/videomixer2.c: videomixer2: Fail when caps are incompatible Do not forget to return false when caps are incompatible. 2010-09-05 20:56:52 -0300 Thiago Santos * gst/videomixer/blend.c: videomixer: Only init orc if it is available Put some ifdef around orc_init to prevent build errors 2010-09-05 12:17:08 +0200 Sebastian Dröge * common: Automatic update of common submodule From ec60217 to d3d9acf 2010-09-04 12:46:31 -0700 David Schleef * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: deinterlace: Update disted Orc files 2009-06-29 11:43:07 -0700 David Schleef * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2src: add decimate property 2010-06-04 12:09:23 -0700 David Schleef * ext/dv/Makefile.am: * ext/dv/gstdvdemux.c: * ext/dv/gstsmptetimecode.h: dvdemux: Parse SMPTE time codes 2010-08-23 02:50:36 -0700 David Schleef * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: deinterlace: remove assembly code in favor of orc 2010-06-08 14:54:49 -0700 David Schleef * gst/deinterlace/tvtime.orc: * gst/deinterlace/tvtime/greedy.c: deinterlace: implement greedy in Orc 2010-09-04 11:43:21 -0700 David Schleef * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: update disted Orc files 2010-09-02 14:34:50 +0200 Thibault Saunier * gst/alpha/gstalphacolor.c: alphacolor: Fix classification This is no effect but a converter. Fixes bug #628608. 2010-09-02 11:19:06 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.types: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-gdkpixbuf3.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst/videomixer/Makefile.am: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixer2pad.h: videomixer2: Add documentation and add to the docs 2010-07-26 16:07:15 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/gstcollectpads2.c: * gst/videomixer/gstcollectpads2.h: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer2: Add videomixer2 element This is based on collectpads2 and is synchronizing all streams based on the running time. New features compared to old videomixer: * Synchronizing frames on the running time * Improved and simplified negotiation * Full QoS support * Variable framerate support Fixes bug #626048, #624905. 2010-09-01 11:11:34 +0200 Pavel Kostyuchenko * gst/matroska/matroska-demux.c: matroskademux: Relax parsing of date tags Before we required a complete date in matroskademux but in id3demux for example only the year or year and month was possible too. Fixes bug #628454. 2010-08-30 19:03:52 +0100 Sjoerd Simons * sys/v4l2/gstv4l2src.c: v4l2src: Use GstBaseSrc::block-size as fallback size 2010-08-30 18:36:54 +0100 Sjoerd Simons * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: v4l2src: Fix using mpegts via the mmap interface MPEG doesn't have a static size per frame, so don't pretend it has one and fail when capturing because it doesn't match. Instead mark the size as unknown and let the read frame grabbing method use a reasonable fallback value (assuming that's only for actual streaming formats) Fixes bug #628349. 2010-08-27 18:15:03 +0200 Sebastian Dröge * ext/wavpack/gstwavpackparse.c: wavpackparse: Don't use GST_FLOW_IS_FATAL() 2010-08-27 18:13:21 +0200 Sebastian Dröge * ext/libpng/gstpngdec.c: pngdec: Don't use GST_FLOW_IS_FATAL() And don't post an error message if downstream returns UNEXPECTED. 2010-08-27 18:09:11 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: dvdemux: Don't use GST_FLOW_IS_FATAL() 2010-08-27 18:05:50 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: jpegdec: Don't use GST_FLOW_IS_FATAL() And don't post an error message if buffer allocation failed because of UNEXPECTED, which only means that downstream wants us to EOS now. 2010-08-27 18:02:57 +0200 Sebastian Dröge * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: flacenc/dec: Don't use GST_FLOW_IS_FATAL() And properly handle UNEXPECTED and WRONG_STATE. 2010-08-27 17:52:18 +0200 Sebastian Dröge * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: cmmldec/enc: Don't use GST_FLOW_IS_FATAL() And as a result, don't ignore WRONG_STATE and NOT_LINKED. Both mean that it's a good idea to pass them upstream instead of pretending that everything is good. 2010-08-27 17:47:22 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Don't use GST_FLOW_IS_FATAL() 2010-08-27 17:45:53 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Don't use GST_FLOW_IS_FATAL() and GST_FLOW_IS_SUCCESS() 2010-08-27 17:39:32 +0200 Sebastian Dröge * gst/qtdemux/qtdemux.c: qtdemux: Don't use GST_FLOW_IS_FATAL() 2010-08-27 17:37:33 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't use GST_FLOW_IS_FATAL() 2010-08-27 17:35:47 +0200 Sebastian Dröge * gst/debugutils/rndbuffersize.c: rndbuffersize: Don't use GST_FLOW_IS_FATAL() 2010-08-27 17:35:38 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Don't use GST_FLOW_IS_FATAL() 2010-08-27 17:32:09 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Don't use GST_FLOW_IS_FATAL() And document why wrong-state doesn't need an error message. 2010-08-26 13:44:49 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Fail gracefully if no threaded PA mainloop can be created Fixes bug #628020. 2010-08-24 15:11:20 +0200 Sebastian Dröge * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: videomixer: Update disted ORC files 2010-08-23 15:44:50 +0200 Sebastian Dröge * configure.ac: * gst/videomixer/Makefile.am: * gst/videomixer/blend.c: * gst/videomixer/blend_mmx.h: * gst/videomixer/blendorc.orc: * gst/videomixer/videomixer.c: videomixer: Optimize ARGB blending and implement BGRA blending with orc This now means, that we have absolutely no handwritten assembly anymore in videomixer and it's also faster now when using SSE. 2010-08-22 01:58:05 -0700 David Schleef * gst/videomixer/blend.c: * gst/videomixer/blendorc.orc: videomixer: Add orc implementation for blending videomixer: Add orc implementation for blending 2010-08-22 01:54:16 -0700 David Schleef * gst/videomixer/videomixer.c: videomixer: Fix example pipelines videomixer: Fix example pipelines 2010-08-20 11:41:55 +0200 Sebastian Dröge * tests/check/elements/imagefreeze.c: imagefreeze: Add test for checking if imagefreeze correctly returns UNEXPECTED after the first buffer 2010-08-20 11:38:09 +0200 Sebastian Dröge * tests/check/elements/imagefreeze.c: imagefreeze: Add test for bufferalloc passthrough 2010-08-20 10:35:15 +0200 Sebastian Dröge * tests/check/elements/imagefreeze.c: imagefreeze: Fix race conditions in the unit test If setting the pipeline to PLAYING before issuing the seek, buffers are already arriving at the sink before the seek is handled and will have the wrong timestamps and everything. Fixes bug #625547. 2010-08-20 10:34:17 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Fix another subtle race condition related to starting the srcpad task Due to a seek the srcpad task could be started in rare circumstances although it shouldn't be started anymore because no upstream buffer is available. 2010-08-20 10:24:33 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Protect the flushing-seek variable by the srcpad's stream lock This fixes a subtle race condition, that caused bufferalloc to fail with wrong-state due to a seek but caused it to be not retried as it should. 2010-08-20 09:14:59 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Always generate a perfectly timestamped stream Before there could be rounding errors when calculating the duration, resulting in timestamp + duration being smaller than the next buffer's timestamp. 2010-08-19 18:38:39 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Only include the server name in the context name if it's not NULL 2010-08-18 16:37:41 +0200 Philippe Normand * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Add "client" property to set the PA client name Allows the application to modify the client name used to connect when connecting to the PulseAudio daemon. Note however that updating the property after the element reached the READY state will have no effect until the next NULL->READY transition. Fixes bug #627174. 2010-08-19 17:59:09 +0200 David Hoyt * ext/soup/gstsouphttpsrc.c: souphttpsrc: Improve error messages Before they contained the URL before the actual failure. The other way around makes more sense and we do the same in other elements like filesrc. Fixes bug #627289. 2010-08-19 12:46:50 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Free the clock on state change failures too 2010-08-17 16:26:41 +0200 Philippe Normand * configure.ac: * ext/pulse/pulseutil.c: * win32/common/config.h: pulseutil: include pid value in gst_pulse_client_name() fallback return value Fixes bug #627162 2010-08-19 12:32:59 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Free the GstPulseContext after usage 2010-08-16 09:12:04 +0200 Philippe Normand * ext/pulse/pulsesink.c: pulsesink: share the PA context between all clients with the same name Avoid to create a new PA context for each new client by using a hash table containing the list of ring-buffers and the shared PA context for each client. Doing this will improve application memory usage in the cases where multiple pipelines involving multiple pulsesink elements are used. Fixes bug #624338. 2010-08-17 13:41:49 +0200 Philippe Normand * ext/pulse/pulsesink.c: pulsesink: clear the PA mainloop if baseaudiosink failed to open the ring_buffer If the application requests a state-change and pulsesink fails to open the ring_buffer device the mainloop attribute of the sink should be cleaned up to avoid future state-change (NULL->READY) failures. 2010-08-19 12:23:16 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Post an error message if EOS happens before valid input is found Fixes bug #627341. 2010-08-12 11:49:47 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Send close newsegment event from the streaming thread 2010-08-11 11:36:31 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Retry bufferalloc if it was aborted with WRONG_STATE because of a flushing seek 2010-08-11 08:46:14 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Return GST_FLOW_UNEXPECTED when getting a second buffer This prevents upstream from pushing many useless buffers and makes it go into EOS state. 2010-08-10 20:11:26 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Passthrough buffer allocations 2010-09-04 13:10:30 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development Temporarily disable -DGST_DISABLE_DEPRECATED for git builds until the code is updated for the GST_FLOW_IS_* macro deprecations. === release 0.10.25 === 2010-09-02 23:44:19 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * gst-plugins-good.doap: * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: * win32/common/config.h: Release 0.10.25 2010-09-02 23:12:48 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update docs for release 2010-09-02 23:07:36 +0100 Tim-Philipp Müller * po/LINGUAS: * po/es.po: * po/gl.po: * po/lt.po: * po/nl.po: * po/ro.po: * po/sv.po: po: update translations 2010-08-25 19:01:50 +0200 Sebastian Dröge * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: 0.10.24.5 pre-release 2010-08-22 21:15:07 -0700 David Schleef * gst/deinterlace/gstdeinterlace.c: deinterlace: use separate buffer metadata for fields Call gst_buffer_make_metadata_writable() on buffers that are duplicated into fields. Fixes #627689. 2010-08-21 21:41:36 +0200 Sebastian Dröge * configure.ac: * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sl.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: 0.10.24.4 pre-release 2010-08-19 18:30:05 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: Prevent crash when reading image with problems Check if we have data on the adapter and fail if not. Fixes #627413 2010-08-13 17:24:01 +0300 Stefan Kost * common: Automatic update of common submodule From 3e8db1d to ec60217 2010-08-11 22:20:25 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Send close segments when seeking only for non-flushing seeks and if we already sent a newsegment event Fixes bug #626619. 2010-08-11 16:50:42 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: * win32/common/gstrtpbin-marshal.c: * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-enumtypes.h: * win32/common/gstudp-marshal.c: 0.10.24.3 pre-release 2010-08-11 11:17:18 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: prevent reading past avc1 atom when parsing ... when one of the subatoms has a large/invalid size. Fixes #626609. 2010-08-10 23:37:23 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: 0.10.24.2 pre-release 2010-08-10 10:57:45 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From bd2054b to 3e8db1d 2010-08-09 00:36:36 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulse: fix printf format in some debugging messages 2010-08-08 23:31:42 +0100 Tim-Philipp Müller * pkgconfig/gstreamer-plugins-good-uninstalled.pc.in: pkgconfig: set pluginsdir to top-level builddir without the pkgconfig/.. bits Removes clutter in plugin dir paths. This is only used to find the -good plugins for unit tests of ugly/bad/ffmpeg/etc. in an uninstalled setup. 2010-08-06 20:04:59 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2src: also log pixel formats in sorted order 2010-08-06 18:07:46 +0100 Sjoerd Simons * sys/v4l2/gstv4l2object.c: v4l2: sort formats in the right order so that non-emulated formats are prefered The format list should be sorted from high ranks to low ranks. In the GSList sorting function this means the compare needs to return a positive value if format a has a lower rank than format b. Among other things this fixes v4l2src to prefer non-emulated formats to emulated formats when built against libv4l. 2010-08-06 19:24:06 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Fix pipeline in the documentation Make sure that we have the same color format on all streams, i.e. AYUV Fixes bug #625452. 2010-08-05 13:56:44 +0300 Stefan Kost * common: Automatic update of common submodule From a519571 to bd2054b 2010-06-14 19:58:11 +1000 Jonathan Matthew * ext/taglib/gstid3v2mux.cc: * tests/check/elements/id3v2mux.c: id3v2mux: write beats-per-minute tag using TBPM frame https://bugzilla.gnome.org/show_bug.cgi?id=621520 2010-07-25 11:47:43 +0200 Sebastian Dröge * gst/videomixer/blend.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Move debug categories into the source files and add debug category for the blend functions 2010-08-04 19:25:31 +0200 Sebastian Dröge * configure.ac: configure: Check if the compiler supports ISO C89 or C99 and which parameters are required This first checks what is required for ISO C99 support and sets the relevant compiler parameters and if no C99 compiler is found, it checks for a C89 compiler. This enables us to check for and use C89/C99 functions that gcc hides from us without the correct compiler parameters. 2010-07-15 10:10:31 +0200 Philippe Normand * ext/pulse/pulsesink.c: pulsesink: use G_TYPE_DEFINE to define ring buffer type The existing get_type() implementation is racy, and the g_type_class_ref() workaround didn't actually work because it was in the wrong function. Since class creation in GObject is thread-safe these days (since 2.16), the class_ref workaround is no longer needed and it is sufficient to ensure the _get_type() function is thread-safe, which G_TYPE_DEFINE does. https://bugzilla.gnome.org/show_bug.cgi?id=624338 2010-08-04 15:20:42 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Post CLOCK-LOST/CLOCK-PROVIDE when going to/from READY Otherwise the clocks are redistributed every time the pipeline goes to PAUSED, which is quite expensive. 2010-07-12 12:35:15 +0200 Wim Taymans * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: rtpmp4gpay: implement perfect timestamps Use bitreader for parsing the config string Reset state variables when going to READY Parse frame length and use it to keep track of the rtptimestamps 2010-07-09 14:07:49 +0200 Wim Taymans * gst/rtp/gstrtph263pdepay.c: rtph263pdepay: allow more clock-rates as input Although the spec says that the clock-rate should always be 90000, some rtsp servers send different clock-rates so we must accept then in order to handle those streams too. 2010-07-06 19:02:14 +0200 Wim Taymans * gst/rtp/gstrtpL16depay.c: L16depay: default to 1 channel When we can't find any channel or encoding-params on the caps for dynamic payload types, set the default number of channels to 1, as the spec says we should. See #623209 2010-07-06 18:22:24 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't reuse udp sockets Don't reuse sockets but make the udpsrc element fail the state change when the socket is already in use. If we don't prevent reuse, we might end up using the same port for different streams in some cases. Fixes #622017 2010-07-06 18:11:21 +0200 Wim Taymans * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: add property to enable port reuse 2010-07-05 10:23:37 +0200 Wim Taymans * gst/rtp/gstrtpL16depay.c: L16depay: use encoding-params for the channels When parsing the number of channels, use the encoding-params property from the RTP caps because that is where we can find the channels according to the spec. Fall back to the channels property in the caps when needed. Fixes #623209 2010-06-29 10:46:41 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: improve error and warning message Improve error and warning message. Fixes #622577 2010-08-02 23:15:56 +0300 Stefan Kost * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: examples: no need to set the color for each frq-band 2010-08-02 12:56:29 +0200 Mark Nauwelaerts * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpg729pay.h: rtpg729pay: avoid basertppayload perfect-rtptime mode G729 packets may only occur intermittently (e.g. cn packets), and as such do not allow for perfect-rtptime calculating rtp times based on frame or byte count. In particular, do not use rtp audio base payloader as base class, but rather base payloader directly. 2010-08-02 12:48:02 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: fix element leak 2010-08-02 12:46:41 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4vdepay.c: rtpmp4vdepay: fix buffer leak 2010-08-02 12:46:20 +0200 Mark Nauwelaerts * tests/check/elements/rtp-payloading.c: tests: rtp payloading: fix pad leak 2010-07-29 17:18:11 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode; use proper movi offset for movi based index Fixes #623357. 2010-07-29 10:00:15 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Correctly parse mvhd atoms Parse mvhd data according to its version to avoid failing on valid files. 2010-07-28 12:21:41 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Fix the max/avg in btrt atom reading According to ISO media base format, the max bitrate is the first one, and the avg comes next. 2010-07-27 15:58:02 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: proper handling of streaming upstream without duration Fixes #625371. 2010-07-26 18:33:09 +0200 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: initialize some variables to fix compiler warnings on OSX build bot 2010-07-26 18:15:25 +0200 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: correctly check what version of gst-plugins-base we're compiling against We need to check the gst-plugins-base version, not the core version (even if both should be the same in any sane setup). 2010-07-26 17:45:42 +0200 Arnaud Vrac * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add port-range property to rtspsrc To support setups with firewall/ipsec, it is useful for an rtsp client to be able to set the range of ports that can be used for rtp/rtcp reception. Allows this by adding a "port-range" property to the rtspsrc element. Fixes #625153 2010-07-26 13:38:31 +0200 Andoni Morales Alastruey * gst/qtdemux/qtdemux.c: qtdemux: set the pixel-aspect-ratio field also for par=1/1 https://bugzilla.gnome.org/show_bug.cgi?id=625302 2010-07-26 15:31:16 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix memory leak in server request reply The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request periodically as a ping. The code in gst_rtspsrc_handle_request forms an OK response and sends, but doesn't call gst_rtsp_message_unset to free the memory after sending the response. This results in a constant slow memory leak. Fixes #624770 2010-07-24 22:39:54 +0100 Zaheer Abbas Merali * gst/debugutils/cpureport.c: cpureport: remove bogus docs 2010-07-24 22:37:11 +0100 Zaheer Abbas Merali * gst/debugutils/Makefile.am: * gst/debugutils/cpureport.c: * gst/debugutils/cpureport.h: * gst/debugutils/gstdebug.c: debugutils: new element cpureport cpureport posts bus messages after every buffer received of cpu used, system clock time, buffer time 2010-07-24 10:29:01 +0200 Sebastian Dröge * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: examples: Destroy the cairo context after usage 2010-07-24 10:21:05 +0200 Sebastian Dröge * configure.ac: * ext/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: Revert "gdkpixbuf: Add a gdkpixbuf3 plugin that uses gdkpixbuf3" This reverts commit b6788153161b4e07fbf3d42a2d8921ea049305d0. There's no gdk-pixbuf3 anymore. gdk-pixbuf was separated from GTK+ and will stay at version 2.0 for GTK+ 3.0. 2010-07-24 10:19:37 +0200 Sebastian Dröge * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: examples: Use cairo instead of to-be-deprecated GDK API Fixes bug #625002. 2010-07-22 16:24:43 +0200 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: fix event leak 2010-07-22 12:05:26 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: pull mode non-cue seeking That is, in files that have no index (Cue), perform seek by scanning for nearest cluster with timecode before requested position. Scanning is done as a combination of interpolation and sequential scan. Fixes #617368. 2010-07-16 12:46:50 +0200 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: streamable files need no _finish Fixes #624455. 2010-07-22 11:46:35 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode; handle 0-size data chunks Fixes #618535. 2010-07-21 08:11:23 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Only reset QoS information and send a NEWSEGMENT event downstream for NEWSEGMENT events on the master pad 2010-07-14 20:31:44 -0700 David Schleef * gst/debugutils/Makefile.am: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapsdebug.h: * gst/debugutils/gstdebug.c: capsdebug: Add new element 2010-07-20 16:11:25 +0100 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: demote WARNING message to LOG level It's not a warning. 2010-07-19 14:47:32 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: Fix regression on markers parsing Fixes a regression introduced when fixing bug #583047 in commit a391bf52cc3c580c7a0a2316ca52eb66da3b85c1 Skip the data when libjpeg asks it to be skipped on one of its callbacks. 2010-07-16 18:04:44 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: add missing argument in debug message 2010-07-16 17:53:55 +0200 Sebastian Dröge * ext/pulse/pulsemixerctrl.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulsesink: Only use gst_audio_clock_new() when compiling against newer base 2010-07-09 17:33:55 +0200 Sebastian Dröge * ext/raw1394/gstdv1394src.c: dv1394src: Post clock-provide and clock-lost messages when going from/to PLAYING In PAUSED and below the clock is not working. 2010-07-04 16:57:55 +0200 Sebastian Dröge * ext/gconf/gstswitchsink.c: * ext/gconf/gstswitchsink.h: * ext/gconf/gstswitchsrc.c: * ext/gconf/gstswitchsrc.h: gconf: Fix ref handling of new child elements and minor cleanup 2010-07-04 09:45:52 +0200 Sebastian Dröge * ext/gconf/gstgconfvideosrc.c: gconfvideosrc: Use correct GConf key 2010-07-03 14:16:42 +0200 Sebastian Dröge * ext/gconf/gstgconfaudiosrc.c: * ext/gconf/gstgconfaudiosrc.h: gconf: Port gconfaudiosrc to GstSwitchSrc 2010-07-03 14:12:12 +0200 Sebastian Dröge * ext/gconf/gstgconfvideosrc.c: * ext/gconf/gstgconfvideosrc.h: gconf: Port gconfvideosrc to GstSwitchSrc 2010-07-03 14:11:55 +0200 Sebastian Dröge * ext/gconf/Makefile.am: * ext/gconf/gstswitchsrc.c: * ext/gconf/gstswitchsrc.h: gconf: Add GstSwitchSrc base class 2010-07-03 13:56:33 +0200 Sebastian Dröge * ext/gconf/gstswitchsink.c: gconf: Create the ghostpad of the switchsink from the template 2010-07-07 10:10:52 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Post clock-provide/clock-lost when going to/from PAUSED Also use gst_audio_clock_new_full() to prevent crashes when the clock is used after the element was destroyed. 2010-07-15 11:49:03 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: remove bogus UNLOCK 2010-07-13 12:34:44 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: also calculate PAR using track width and height for QT files (... as opposed to only for ISO style files). Fixes #624173. 2010-07-12 17:29:12 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: handle bogus files storing ADTS AAC data 2010-07-09 16:57:33 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: do not error out on a block with unknown tracknumber 2010-07-08 18:57:21 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: do not align reverse playback reference stream twice Timestamp rounding issues could lead to going backwards 2 keyframe periods (rather than only 1). While this is not necessarily a problem, it might potentially place additional (buffering) load on downstream and could be avoided (because We Can). Fixes #623629. 2010-07-08 16:07:16 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: convert some more mov format timestamp to gst time 2010-07-07 14:16:59 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: additional verification heuristics for VBR audio stream Check for and override some header field(s) for reasonable values, according to later expected use in calculations. 2010-07-14 15:21:21 +0200 Alessandro Decina * gst/videofilter/gstvideobalance.c: videobalance: Fix wrong lock order that could lead to a deadlock. Fixes #624331. 2010-07-16 11:31:08 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development === release 0.10.24 === 2010-07-15 01:49:04 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.24 2010-07-15 01:35:06 +0100 Tim-Philipp Müller * po/cs.po: * po/lv.po: po: update translations 2010-07-07 00:42:46 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: 0.10.23.4 pre-release 2010-07-07 00:31:17 +0100 Tim-Philipp Müller * po/LINGUAS: * po/da.po: * po/el.po: * po/es.po: * po/fr.po: * po/id.po: * po/pt_BR.po: * po/sl.po: * po/tr.po: * po/zh_CN.po: po: update translations 2010-06-23 11:47:43 +0200 Michael Grzeschik * sys/v4l2/gstv4l2sink.c: v4l2sink: destroy buffer pool when changing state to NULL In the case we change the State from READY_TO_NULL the buffers in the pool still hold an open dup file descriptor to the device, therefore the device release function will not be called and the device will probably answer with -EBUSY when we reopen it in the next NULL_TO_READY transition. Signed-off-by: Michael Grzeschik See bug #622500 and #612244. 2010-07-06 13:21:19 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Fix order of bitrates in 'btrt' atom There seems to be a bug in libmp4v2 that generates a MPEG4BitRateBox as (bufferSizeDB, avgBitrate, maxBitrate) instead of (bufferSizeDB, maxBitrate, avgBitrate), according to the spec. I used the mp4file output while writing this code, so the order is wrong. This patches fixes that. https://bugzilla.gnome.org/show_bug.cgi?id=623654 2010-07-05 12:05:57 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: fix skipping extra 0xff markers Fixes #623585. 2010-06-29 23:18:23 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: fix memory leak Don't leak result of gst_adapter_take(). There are most likely smarter things we can do, but let's keep things simple for the release. Fixes #623172. 2010-07-02 12:31:31 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: strip out bogus tags from XMP atom https://bugzilla.gnome.org/show_bug.cgi?id=623366 2010-07-02 14:25:22 +0200 Andrzej K. Haczewski * gst/flv/gstflvmux.c: flvmux: Write duration at the correct position 2010-06-30 11:12:08 +0200 Thijs Vermeir * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: fix memleak on custom downstream events by not sending custom downstream event twice and fix memleak when not handling the event https://bugzilla.gnome.org/show_bug.cgi?id=623196 2010-06-29 20:18:51 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: 0.10.23.3 pre-release 2010-06-29 20:14:53 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: fix unportable printf format specifiers in commented out code To avoid false positives when grepping for unportable specifiers. 2010-06-29 19:12:36 +0100 Tim-Philipp Müller * configure.ac: configure: fix --disable-external 2010-06-28 15:44:06 +0100 Tim-Philipp Müller * autogen.sh: * configure.ac: Bump automake requirement to 1.10 and autoconf to 2.60 For maintainability reasons and $(builddir). See #622944. 2010-06-28 09:07:58 +0100 Tim-Philipp Müller * gst/goom/plugin_info.c: goom: don't allocate 260kB struct on the stack PluginInfo is quite a sizeable struct, let's not allocate it on the stack, especially not if we're copying it over into another dynamically allocated copy anyway. Fixes #570761. 2010-06-27 10:31:17 +0200 Sebastian Dröge * configure.ac: configure: Require GTK+ >= 2.14 for the examples 2010-06-26 20:12:25 +0200 Guido Günther * tests/examples/equalizer/demo.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: examples: Make demos -DSEAL safe to fix build with GTK+ 3.0 2010-06-26 21:39:34 +0200 Sebastian Dröge * ext/jpeg/Makefile.am: jpeg: Explicitely link with libgstbase 2010-06-26 18:42:29 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.23.2 pre-release 2010-06-26 18:41:49 +0100 Tim-Philipp Müller * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videomixer/blendorc-dist.c: gst: update orc files 2010-06-26 18:41:39 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2010-06-25 19:40:06 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Fix leaking of the streamheader buffers gst_value_set_buffer() increases the refcount and doesn't take ownership of the buffer. 2010-06-24 16:32:23 +0100 Tim-Philipp Müller * gst/matroska/ebml-read.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstvideoflip.c: matroska, videobox, videofilter: fix compiler warnings when debugging is disabled in gstreamer Fixes unused variable warnings when GStreamer's debugging system has been disabled. 2010-06-24 15:17:11 +0100 Tim-Philipp Müller * tests/check/Makefile.am: tests: add plugin loading whitelist to test environment Only want to load core/base/good plugins here. Fixes #619717. 2010-06-24 15:09:16 +0300 Stefan Kost * common: Automatic update of common submodule From 73ff93a to a519571 2010-06-24 13:02:04 +0100 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbuf.c: gdkpixbufdec: bump rank to SECONDARY Bump gdkpixbufdec's rank to SECONDARY to give it an edge over misc. image decoders in gst-ffmpeg that also have a MARGINAL rank. Fixes #620162. 2010-06-23 12:15:13 +0200 Michael Grzeschik * gst/avi/gstavidemux.c: reset the have_index flag at transition PAUSED_TO_READY If we restart the Stream in the case of doing a transition from PAUSED_TO_READY and back with READY_TO_PAUSED aso. the duration of the video will get calculated even if we have a avi header with that information. Signed-off-by: Michael Grzeschik 2010-06-23 20:29:14 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix negotiation for I420/YV12 We don't support conversion into *all* YUV formats for them, only into I420/YV12/AYUV. Fixes bug #622501. 2010-06-22 15:22:44 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: proper closing segment construction Fixes #618982. 2010-06-22 15:46:51 +0300 Stefan Kost * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2src_calls.c: v4l2: precalculate duration Have frame duration in the instance struct and calculate it after changing the caps. 2010-06-21 12:17:39 +0300 Stefan Kost * sys/v4l2/gstv4l2sink.c: v4l2sink: use glib defines in property declarations for readability 2010-06-21 12:15:14 +0300 Stefan Kost * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: v4l2: use G_PARAM_STATIC_STRINGS to save a few bytes and strdups 2010-06-18 20:02:49 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix locking after moving things around 2010-06-18 14:13:58 -0300 Thiago Santos * ext/taglib/gstapev2mux.cc: taglib: Use newly added gst_tag_list_peek_string_index Replace calls to gst_tag_list_get_string_index with gst_tag_list_peek_string_index to avoid a string copy 2010-06-18 16:56:19 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: make some errors as warnings Avoid spamming the testsuite with these error debug lines. 2010-06-18 16:49:08 +0200 Keith Nicholson * gst/udp/gstudpsrc.c: udpsrc: fix multicast support on windows builds On windows builds, sets source address for bind to INADDR_ANY, while maintaining the original multicast group address for subsequent join. Fixes #595978 2010-06-18 16:16:28 +0200 Wim Taymans * gst/udp/gstudpnetutils.c: udp: make url parsing compatible with VLC syntax Skip everything before the @ sign in the url location. VLC uses that as the remote address to connect to (but we ignore it for now). This makes our udp urls compatible with the ones used by VLC. Fixes #597695 2010-06-18 15:08:21 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: factor out the connections Keep a global connection for aggregate control but also keep stream connections for non-aggregate control. Add some helper methods to connect/close/flush the connections. 2010-06-17 13:06:56 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add non-aggregate control Add non-aggregate control. Separate retrieving thr SDP from parsing and setting up the streaming from the SDP. 2010-06-17 22:10:03 +0100 Zaheer Abbas Merali * common: common: update common back to what it was 2010-06-17 17:24:22 +0100 Zaheer Abbas Merali * common: * gst/flv/gstflvmux.c: flvmux: add documentation for streamable property 2010-06-17 16:43:44 +0100 Tim-Philipp Müller * common: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: docs: update introspected plugin docs for gstdoc-scangobj and other changes Update common for latest gstdoc-scangobj, and inspect xml files for escaping and pad template order changes. 2010-06-17 16:41:56 +0100 Tim-Philipp Müller * tests/check/.gitignore: tests: ignore sub-directory with orc tests 2010-06-17 10:44:33 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Fix an uninitialized variable compiler warning 2010-06-16 21:02:13 +0200 Sebastian Dröge * gst/matroska/ebml-read.c: ebml-read: Zero-sized ints/uints/floats have a value of 0 according to the EBML spec 2010-06-16 20:02:58 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Fix possible NULL pointer dereference and assertion that could be caused by invalid files 2010-06-16 19:50:34 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Clean up/fix some minor error handling bugs 2010-06-16 19:30:25 +0200 Sebastian Dröge * sys/ximage/gstximagesrc.c: ximagesrc: Fix NULL pointer dereference when allocation of the ximage fails 2010-06-16 19:28:04 +0200 Sebastian Dröge * ext/flac/gstflactag.c: flactag: Fix possible NULL pointer dereference 2010-06-16 19:24:54 +0200 Sebastian Dröge * gst/audiofx/audioiirfilter.c: audioiirfilter: Fix possible NULL pointer dereference 2010-06-16 19:20:02 +0200 Sebastian Dröge * gst/effectv/gstwarp.c: warptv: Don't use floats as loop counters 2010-06-16 11:21:35 -0400 Havoc Pennington * sys/v4l2/gstv4l2object.c: v4l2src: do not try to change device format if it's already correct This allows set_caps to succeed if caps change in a way that would not modify the format we're getting from the hardware. Otherwise if not in NULL state, setting caps would fail with EBUSY. With this change, in some cases it's OK to go PLAYING->READY->PLAYING rather than PLAYING->NULL->PLAYING to avoid a time-consuming close and reopen of the device. Fixes #621723 2010-06-16 11:09:17 -0400 Havoc Pennington * sys/v4l2/gstv4l2src.c: v4l2src: in negotiate, check for error return from set_caps Fixes #621723 (partially) set_caps can fail if the video device is running, in that case setting its format leads to EBUSY. If set_caps fails then we will not have set up the buffer pool (it will be NULL) which leads to a crash when we try to pull buffers. If we fail the negotiate on set_caps failure, then we won't go to playing state and won't crash. This is a small improvement. Of course, a nicer fix would be to make set_caps work in the case where the format is unchanged. If the format has changed, failing is probably correct because we need to close the device (go to NULL state) in order to set caps. 2010-06-16 15:40:34 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: improve audio vbr detection Subsequent entry time calculations use blockalign value to determine number of frames per chunk, and blockalign == 1 is then most unlikely to result in reasonable values (which also aligns with "spec"). 2010-06-16 15:52:57 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: tweak DELTA_UNIT labeling Consider SPS, PPS and IDR as keyframe, all others as DELTA_UNIT. See #620154. 2010-06-15 20:06:17 +0200 Sebastian Dröge * ext/wavpack/gstwavpackdec.c: wavpackdec: Initialize uninitialized variable and don't unref it if it's NULL 2010-06-15 20:04:35 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Assign variables before printing them 2010-06-15 20:00:28 +0200 Sebastian Dröge * gst/wavparse/gstwavparse.c: wavparse: Initialize uninitialized variable 2010-06-15 19:47:16 +0200 Sebastian Dröge * sys/v4l2/gstv4l2object.c: v4l2: Initialize variable 2010-06-15 19:45:36 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Fix NEWSEGMENT parsing logic and don't use uninitialized variables 2010-06-15 17:20:20 +0200 Edward Hervey * gst/matroska/ebml-read.c: matroska: Fix unitialized variable 2010-06-15 16:49:49 +0200 Edward Hervey * common: Automatic update of common submodule From 9339ccc to 35617c2 2010-06-15 16:54:04 +0300 Stefan Kost * common: Automatic update of common submodule From 5adb1ca to 9339ccc 2010-06-15 16:35:18 +0300 Stefan Kost * common: Automatic update of common submodule From 57c89b7 to 5adb1ca 2010-06-15 14:08:26 +0100 Tim-Philipp Müller * .gitignore: .gitignore: ignore generated tvtime.h file 2010-06-15 15:36:33 +0300 Stefan Kost * common: Automatic update of common submodule From c804988 to 57c89b7 2010-05-17 13:54:03 +0200 Marc-André Lureau * ext/raw1394/gst1394clock.c: * ext/raw1394/gst1394clock.h: raw1394: remove useless last_time It seems to me this code is useless: removing it. https://bugzilla.gnome.org/show_bug.cgi?id=618871 2010-06-14 19:21:22 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: respect aggregate control attributes when the SDP specifies an aggregate control url, use that for playback control. Fixes #619531 2010-06-14 15:36:00 +0200 Sebastian Dröge * gst/goom/gstgoom.c: goom: Call orc_init() before trying to get target flags 2010-06-14 15:35:08 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Call orc_init() before trying to get target flags 2010-06-14 14:26:22 +0100 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: * tests/check/elements/matroskamux.c: matroskamux: revert change that set a reserved flag on the Block. So matroska's Block structure has no keyframe flag, only the SimpleBlock has it. To detect keyframes in Blocks, it is just the BlockGroup container that needs to have a ReferenceBlock attached if it is a delta frame in video. 2010-05-31 12:45:01 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: use libjpeg scatter-gather operation to avoid data copying Fixes #583047 (more). 2010-05-27 15:45:23 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: optimize buffer handling when parsing Use an adapter to collect incoming data, and use adapter API to scan and peek. Fixes #583047. 2010-06-14 13:48:28 +0200 Sebastian Dröge * sys/oss4/oss4-mixer.c: oss4: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp() 2010-06-14 13:27:30 +0200 Sebastian Dröge * configure.ac: configure: Use GLIB_EXTRA_CFLAGS 2010-06-14 13:03:57 +0200 Sebastian Dröge * common: Automatic update of common submodule From 7a0fdf5 to c804988 2010-06-14 11:46:32 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: also consider AU and SEI NALUs as DELTA_UNIT Fixes #620154. 2010-06-14 11:32:43 +0200 Sebastian Dröge * common: Automatic update of common submodule From 6da3bab to 7a0fdf5 2010-06-12 21:26:16 +0300 Stefan Kost * gst/rtp/gstrtpmparobustdepay.c: build: include stdio.h for sscanf 2010-06-12 14:12:50 +0200 Sebastian Dröge * tests/check/Makefile.am: tests: Add clean rule for the orc tests 2010-06-12 14:12:04 +0200 Sebastian Dröge * tests/check/Makefile.am: tests: Add autogenerated orc tests 2010-06-12 08:27:42 +0200 Sebastian Dröge * common: Automatic update of common submodule From 733fca9 to 6da3bab 2010-06-11 16:23:29 -0700 David Schleef * sys/v4l2/gstv4l2src.c: v4l2src: Fix element description 2010-06-11 21:13:59 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmparobustdepay.c: rtpmparobustdepay: don't try to unref NULL buffers Fixes generic/states unit test. 2010-06-11 20:50:23 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: use typefind functions to check if PCM data contains dts stream Use new dts audio typefinder from -base to check if the PCM data contains a dts stream. This way we recognise more varieties more reliably and also detect the dts stream if there isn't a frame sync right at the start of the data. Fixes #413942. 2010-06-11 20:47:22 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: set buffer offsets before using the buffer for the first time gst_type_find_helper_for_buffer() will need the correct offset set on the buffer (ie. 0) and not the byte offset we started pulling the data from. 2010-06-10 16:14:43 +0200 Mark Nauwelaerts * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmparobustdepay.h: rtp: add mpa-robust depayloader Fixes #589997. 2010-06-11 10:57:41 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: fix avi header bytewriting ... by using proper offsets for tag list writing. Also use _reset rather than _free and consistently use bytewriter position. See #619293. 2010-06-10 22:58:41 +0200 Sebastian Dröge * .gitignore: Update .gitignore Add the generated orc source files 2010-06-10 22:55:17 +0200 Sebastian Dröge * tests/check/elements/matroskamux.c: matroskamux: Fix unit test for changed key-frame behaviour All audio frames are marked as keyframe now instead of marking them all as delta unit... 2010-06-10 22:45:13 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend.c: * gst/videomixer/blend_mmx.h: * gst/videomixer/blendorc-dist.c: * gst/videomixer/blendorc-dist.h: * gst/videomixer/blendorc.orc: videomixer: Port most blending related functions to orc Only remaining MMX implementation is the ARGB/BGRA/AYUV blending for which we first need the orc compositing opcodes. 2010-06-10 20:17:07 +0200 Sebastian Dröge * gst/videomixer/blend_mmx.h: videomixer: Replace some tabs by spaces 2010-06-10 11:04:38 +0100 Andoni Morales Alastruey * ext/raw1394/gst1394clock.c: dv1394: Fix the internal clock even more The cycleCount register is 13 bits long and the cycleOffset one is 12 bits long. To read the cycleCount register we need to shift 12 bits and not 13. Fixes #615461 2010-06-09 18:37:29 -0700 David Schleef * configure.ac: configure: use m4 macro to check for Orc 2010-06-09 22:40:23 +0200 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: matroskamux: some non-delta buffers were not marked as keyframes 2010-06-09 22:00:16 +0200 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: change 2 second limit per cluster Start cluster at every keyframe or when we would overflow the previous cluster's relative timestamp field. This would avoid as much as possible starting clusters at non-keyframes. 2010-06-09 12:40:09 -0700 David Schleef * common: Automatic update of common submodule From fad145b to 733fca9 2010-06-09 12:34:01 -0700 David Schleef * common: Automatic update of common submodule From 47683c1 to fad145b 2010-06-09 20:53:06 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Don't request more shared memory than needed 2010-06-09 20:45:04 +0200 Sebastian Dröge * ext/gconf/gstswitchsink.c: switchsink: Set the GST_ELEMENT_IS_SINK flag on the sink 2010-06-09 20:43:50 +0200 Sebastian Dröge * ext/gconf/gstgconfvideosink.c: * ext/gconf/gstgconfvideosink.h: gconfvideosink: Use GstSwitchSink as base class 2010-06-09 20:30:31 +0200 Sebastian Dröge * ext/gconf/gstgconfaudiosink.c: gconfaudiosink: Use G_PARAM_STATIC_STRINGS 2010-06-09 20:29:02 +0200 Sebastian Dröge * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfaudiosink.h: gconfaudiosink: Rename instance variable to be more descriptive 2010-06-09 20:22:30 +0200 Sebastian Dröge * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautovideosink.c: auto{audio,video}sink: Don't lose the GST_ELEMENT_IS_SINK flag after removing the child 2010-06-09 20:07:09 +0200 Julien Moutte * sys/directsound/gstdirectsoundsink.c: directsoundsink: Plug some memleak and support 22050Hz mono sound. Segment size needs to be a multiple of the sample size in bytes. 2010-06-09 16:22:27 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Flush shm buffer immediately if it's full 2010-06-09 16:21:55 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Fix writing of buffers larger than segsize Fixes bug #620540. 2010-06-09 15:42:37 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Fix playback if PA doesn't give us a large enough shared memory buffer 2010-06-09 15:42:19 +0200 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: change indexed property to streamable The property streamable has reverse semantics to indexed. 2010-06-09 09:13:09 -0300 Thiago Santos * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Rename unreleased property 'indexed' to 'streamable' Rename 'indexed' to 'streamable' for a better name while it hasn't been released 2010-06-08 15:23:51 -0700 David Schleef * REQUIREMENTS: * configure.ac: configure: remove liboil check 2010-06-08 14:44:19 -0700 David Schleef * gst/level/gstlevel.c: level: remove unused liboil include 2010-06-04 18:22:42 -0700 David Schleef * gst/videomixer/Makefile.am: * gst/videomixer/blend.c: videomixer: liboil to orc conversion 2010-06-04 18:21:21 -0700 David Schleef * gst/videobox/Makefile.am: * gst/videobox/gstvideobox.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videobox/gstvideoboxorc-dist.h: * gst/videobox/gstvideoboxorc.orc: videobox: liboil to orc conversion 2010-06-04 18:16:25 -0700 David Schleef * gst/goom/Makefile.am: * gst/goom/README: * gst/goom/gstgoom.c: * gst/goom/plugin_info.c: goom: liboil to orc conversion 2010-06-08 16:04:23 -0700 David Schleef * gst/deinterlace/Makefile.am: * gst/deinterlace/tvtime-dist.c: * gst/deinterlace/tvtime-dist.h: * gst/deinterlace/tvtime.orc: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/vfir.c: deinterlace: orcify some deinterlacing methods 2010-06-08 16:03:36 -0700 David Schleef * gst/deinterlace/Makefile.am: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: convert from liboil to orc 2010-06-08 15:23:28 -0700 David Schleef * REQUIREMENTS: * configure.ac: configure: Add orc check 2010-06-08 14:09:00 +0200 Zaheer Abbas Merali * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Add indexed property to replace disabled is-live. Add indexed property to be the negation of what the disabled is-live property was. Fixes bug #613066. 2010-06-08 09:22:30 +0200 Sebastian Dröge * configure.ac: raw1394: Require libraw1394 >= 2.0.0 for raw1394_read_cycle_timer Fixes bug #620929. 2010-06-08 07:35:00 +0200 Sebastian Dröge * ext/annodex/gstcmmlenc.c: cmmlenc: Remove hack to let oggmux start a new page for every CMML buffer oggmux does this for CMML by its own now 2010-06-07 18:32:16 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Don't handle non-TIME seeks Don't send them upstream because for upstream a BYTES seek might make sense but is completely wrong because upstream can't seek to a byte position of the audio or video stream. Also don't build the index in push mode for non-TIME seeks, things will go wrong here otherwise. 2010-06-07 11:15:26 -0400 Olivier Crête * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfdetect.h: dtmfdetect: Only works with rate=8000, fix in caps 2010-06-02 19:16:20 +0100 Sjoerd Simons * gst/rtp/gstrtph264pay.c: Cope with short startcodes in the h264 bytestream 2010-06-06 17:25:16 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulse: log message printf format fixes 2010-06-06 18:00:22 +0200 Sebastian Dröge * ext/dv/gstdvdemux.c: * ext/pulse/pulsemixer.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/speex/gstspeexenc.c: * ext/taglib/gsttaglibmux.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackparse.c: ext: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs 2010-06-06 17:57:03 +0200 Sebastian Dröge * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstossdmabuffer.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxringbuffer.c: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/waveform/gstwaveformsink.c: sys: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs 2010-06-06 17:52:40 +0200 Sebastian Dröge * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/debugutils/breakmydata.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/flv/gstflvdemux.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/videofilter/gstvideobalance.c: * gst/videomixer/videomixer.c: gst: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs 2010-06-06 15:12:16 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: refactor delta unit handling This allows us to skip delta units earlier and is a bit clearer in my opinion. It also makes only video buffers ever be delta units, not just for SimpleBlock as before. 2010-06-06 15:17:00 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Clear adapter on discontinuities 2010-06-06 14:03:53 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: Ignore keyframe flag for non-video streams When the keyframe bit of SimpleBlock Flags wasn't set, the buffer was being marked with GST_BUFFER_FLAG_DELTA_UNIT, causing all buffers to be skipped after a seek. This may be a problem with the Sorenson Squish encoder, but arguably the keyframe bit should only be applied to video. Fixes bug #620358. 2010-06-06 14:56:52 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: First try upstream when handling seek events/queries 2010-06-04 14:54:59 -0400 Tristan Matthews * gst/rtp/gstrtpceltpay.c: gstrtpceltpay: don't always fixate sink caps to 1 channel The getcaps function should not fixate the channels field until we get the encoding-params field from our srcpad's caps. Fixes #620591 2010-06-04 13:57:28 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: try all ranges from the sdp Try all ranges in the SDP before giving up. 2010-06-04 13:56:07 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: make parse_range return result Make the parse_range function return if the parsing succeeded or failed. 2010-06-04 11:44:09 +0200 Edward Hervey * gst/videomixer/videomixer.c: videomixer: if we're not linked downstream, we can do any format Stupid me, assuming _get_allowed_caps() would actually return the pad templates if there was no peer. 2010-05-31 16:26:19 +0100 Sjoerd Simons * gst/rtp/gstrtptheorapay.c: Keep announcing the delivery-method in the capabilities Even though we don't use delivery-method in our payloader, older versions of the theora payloader in gstreamer required it. As such we need to keep this around in the caps for backwards-compatibility. This reverts part of 49463a37cbaa952e1401291f0a2623de6cab3880 Fixes #618940 2010-06-03 17:52:11 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * sys/oss4/oss4-mixer.c: oss4: add some comments for translators to clarify meaning of "Low" "Low" etc. are quality settings here (e.g. for the internal resampler). Some day when we use GLib's i18n functions we might want to use NC_() and g_dpgettext2() here instead of the comments. Fixes #555967. 2010-06-03 19:23:01 +0200 Wim Taymans * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: mp4gdepay: calculate the frame duration correctly When we calculate the frame duration, we need to use the amount of frames in the _previous_ packet, not the current packet. The frame duration is needed to correctly de-interleave interleaved streams. This fixes the case where there are a variable number of frames in a packet. Fixes #620494 2010-06-03 18:58:42 +0200 Edward Hervey * gst/videomixer/videomixer.c: videomixer: Don't return caps in get_caps() that will be rejected This commit basically puts _get_caps() in sync with accept_caps(). If we don't have a master pad OR the master pad caps aren't negotiated then we just return the downstream allowed caps. If we have a master pad with negotiated caps, we return those caps with a free range of width/height/framerate 2010-06-03 13:45:32 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: Revert "pulsesink: Add comments to remove the provide-clock message posting once we depend on base 0.10.30" This reverts commit 8f3708f38aa3839a6a625ca7d1c166101c9fbb7f. The baseaudiosink commit was reverted 2010-06-03 10:27:25 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Add comments to remove the provide-clock message posting once we depend on base 0.10.30 baseaudiosink does all this for us now. 2010-05-07 18:42:06 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: Remove rtpdtmfmux stream-lock code 2010-06-02 16:36:11 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: delayed seek handling also deserves TRUE event response 2010-06-02 15:30:47 +0200 Thijs Vermeir * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix compiler warning unused variable ‘estimated’ 2010-06-02 15:04:00 +0200 Alessandro Decina * common: common: revert the change i did in my previous commit 2010-06-02 13:39:10 +0200 Alessandro Decina * common: * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: stop buffering and emit EOS at the end of a stream When using RTP_JITTER_BUFFER_MODE_BUFFER, make sure that the ringbuffer doesn't get stuck buffering forever when there isn't enough data left to fill the buffer. 2010-06-01 21:52:59 +0200 Benjamin Otte * gst/debugutils/testplugin.c: debugutils: Don't consume preroll buffer twice 2010-06-01 21:32:11 +0200 Benjamin Otte * ext/pulse/pulseutil.c: pulse: Style fix: use g_strdup() instead of printf()ing a simple string 2010-05-27 16:07:31 +0200 Benjamin Otte * gst/debugutils/tests.c: debugutils: Replace md5 implementation with glib's https://bugzilla.gnome.org/show_bug.cgi?id=619824 2010-05-22 11:55:37 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: clean up code for avi header using a bytewriter https://bugzilla.gnome.org/show_bug.cgi?id=619293 2010-06-01 18:54:41 -0500 Pierre-Louis Bossart * configure.ac: * ext/pulse/pulsesink.c: pulsesink: optimize communication with PulseAudio using pa_stream_begin_write 2010-06-02 10:52:56 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Post provide-clock message on the bus if the clock appears/disappears Fixes bug #620277. 2010-06-01 23:49:17 -0700 David Schleef * common: Automatic update of common submodule From 17f89e5 to 47683c1 2010-06-01 22:54:49 -0700 David Schleef * common: Automatic update of common submodule From cdff0fb to 17f89e5 2010-06-01 20:45:29 +0200 Edward Hervey * gst/videomixer/videomixer.c: videomixer: filter caps returned from downstream with our pad template. 2010-06-01 16:56:32 +0100 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: matroskamux: Remove more unneeded warnings 2010-06-01 16:54:03 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: matroskamux: remove unneeded warning 2010-06-01 16:49:14 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: matroskamux: remove unneeded debug statement 2010-06-01 16:24:53 +0100 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: change is-live property to indexed 2010-05-23 13:56:16 +0100 Zaheer Abbas Merali * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: use the uint64 scaling functions In demuxer and muxer use the gst_util_uint64 scaling functions rather than standard integer division. Add warnings (to be changed to debug) for debugging the timestamp and duration. 2010-05-21 14:35:34 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: matroskamux: set delta unit on all buffers except cluster start ones 2010-05-21 13:38:11 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: matroskamux: store caps and set on buffers rather than using pad caps 2010-05-21 13:25:24 +0100 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: matroskamux: make sure pads caps are set before any buffers pushed. 2010-05-21 13:14:04 +0100 Zaheer Abbas Merali * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: matroskamux: add streamheaders 2010-05-21 12:23:08 +0100 Zaheer Abbas Merali * gst/matroska/matroska-mux.c: matroskamux: no need to set cache twice 2010-05-21 01:59:53 +0200 Xavier Queralt * gst/matroska/matroska-mux.c: Do not create a SeekHeader, Cues, .. when doing live 2010-05-20 23:39:59 +0200 Xavier Queralt * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: Add is-live property 2010-06-01 13:22:26 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: fix variable init 2010-05-28 16:37:32 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: matroskademux: improve reverse playback Slightly modify approach to also handle cases where cue entries do not reliably lead to initial keyframes. Fixes #619817. 2010-05-24 16:02:58 +0200 Mark Nauwelaerts * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: avoid gtk-doc confusing comments 2010-05-21 11:21:58 +0200 Mark Nauwelaerts * tests/check/Makefile.am: * tests/check/elements/matroskamux.c: matroskamux: adjust unit test to modified behaviour 2010-05-20 14:33:41 +0200 Mark Nauwelaerts * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: matroskamux: use write caching also when writing buffer data Specifically, this reduces pushing several small buffers for each data buffer and also avoids a seek for each buffer altogether (though a seek is still needed for each cluster). Fixes #619273. 2010-05-20 14:23:07 +0200 Mark Nauwelaerts * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: matroskamux: fix ebml write caching with bytewriter implementation Also cache a bit more during header writing. Fixes #619273. 2010-05-20 14:08:42 +0200 Mark Nauwelaerts * gst/matroska/ebml-write.c: matroskamux: use consistent debug category name for ebmlwrite 2010-05-18 14:44:15 +0200 Mark Nauwelaerts * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: use bytereader based GstEbmlRead as a helper ... rather than basing on it by inheritance. Also use more common code for push and pull mode. Fixes #619198. Fixes #611117. 2010-06-01 15:47:32 +0200 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: _get_pad_template result needs no unref 2010-05-18 19:42:58 -0300 Thiago Santos * ext/libpng/gstpngenc.c: pngenc: Support 8 bit grayscale Adds support to 8 bit grayscale input 2010-05-18 14:46:54 -0300 Thiago Santos * ext/jpeg/gstjpegdec.c: jpegdec: Adds 8bit grayscale support Adds decoding support for jpeg images in 8 bit grayscale format. 2010-05-18 01:57:14 -0300 Thiago Santos * ext/jpeg/gstjpegenc.c: jpegenc: Accept grayscale as input Adds video/x-raw-grayscale (8 bit) support to jpegenc 2010-05-31 13:30:05 +0200 Edward Hervey * gst/videomixer/videomixer.c: videomixer: Implement sinkpad GetCapsFunction. This allows returning only the formats, width, height, framerate and pixel-aspect-ratio that downstream can support. https://bugzilla.gnome.org/show_bug.cgi?id=620148 2010-05-20 11:28:47 -0400 Tristan Matthews * ext/lame/gstlamemp3enc.c: lamemp3enc: implement latency query The encoder's latency is deduced from the framesize. Fixes #618896. 2010-05-31 07:49:21 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't compare running times with stream times when doing QoS 2010-05-27 21:06:43 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Don't reconfigure the caps when changing properties Fixes bug #619848. 2010-05-26 13:13:44 +0200 Sebastian Dröge * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Add property to allow passthrough mode This passthrough mode is used if the alpha method is "set" and the alpha value is 1.0. Fixes bug #617512. 2010-05-25 15:16:06 +1000 Alexander Kojevnikov * gst/spectrum/gstspectrum.c: spectrum: support 24-bit width Fixes #619045 2010-05-24 21:50:58 +1000 Alexander Kojevnikov * gst/spectrum/gstspectrum.c: spectrum: support arbitrary bit depth Partially fixes #619045 2010-05-25 05:36:46 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: fix deadlock introduced by video keyframe QoS 2010-05-23 09:32:08 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: matroskademux: skip buffers before a late keyframe (QoS) Before, vp8dec had no option but to decode all frames even if some/all of them would be late. With this change, performance when keyframes are frequent is helped a great deal. On my Thinkpad X60s, decoding a 20 s 1080p sunflower encode with keyframes every 10 frames went from taking 42 s with 5 frames shown to 21 s with 15 frames shown (still slow enough to count by hand). When keyframes are more sparse, you will still be able to catch up eventually, but the results won't be as noticable. 2010-05-14 17:57:59 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: * gst/videomixer/videomixerpad.h: videomixer: Don't mix input with different pixel aspect ratios Fixes bug #618530. 2010-05-17 19:54:22 +0200 Sebastian Dröge * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: deinterlace: Add MMX/3DNow implementations of greedyh for UYVY 2010-05-17 19:16:43 +0200 Sebastian Dröge * gst/deinterlace/tvtime/greedyh.c: deinterlace: Fix UYVY implementation of greedyh to be actually used 2010-05-11 11:43:07 +0200 Sebastian Dröge * configure.ac: * ext/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: gdkpixbuf: Add a gdkpixbuf3 plugin that uses gdkpixbuf3 2010-06-01 10:06:10 +0100 Tim-Philipp Müller * Makefile.am: * common: * win32/common/gstrtpbin-marshal.c: * win32/common/gstrtpbin-marshal.h: * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-marshal.c: * win32/common/gstudp-marshal.h: win32: add more generated marshal and enumtype files to win32-update 2010-06-01 09:27:00 +0100 Tim-Philipp Müller * gst/matroska/matroska.c: Revert "matroska: add temporary webm typefinder" This reverts commit d148ec0ad2053abb0c38fc681a8953292985388f. We depend on -base git now, which has a webm typefinder in the usual place. 2010-06-01 09:26:11 +0100 Tim-Philipp Müller * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/matroska/matroska-mux.c: Revert "avimux, flvmux, matroskamux: don't crash if tags arrive on multiple input pads at the same time" This reverts commit 6a9983cd20c48b96396229b3f94d0254a05ddf48. Rely on locking done in GstTagSetter in core git. 2010-06-01 09:23:18 +0100 Tim-Philipp Müller * configure.ac: configure: require core/base git For WebM typefinding and GstTagsetter fixes. 2010-06-01 09:17:52 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development === release 0.10.23 === 2010-05-30 14:03:53 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.23 2010-05-30 14:02:04 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2010-05-29 10:23:48 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: flvdemux: Fix position query 2010-05-28 15:14:07 +0100 Tim-Philipp Müller * gst/matroska/webm-mux.c: docs: remove unnecessary videorate element from webmmux example pipeline 2010-05-28 10:43:36 -0300 Thiago Santos * ext/jpeg/gstjpegenc.c: jpegenc: Keep variables in sane state after _reset When reseting, keep 'row' variables at a sane state after freeing to avoid it being freed again on _resync realloc when the element is reused. Fixes #619943 2010-05-27 18:08:17 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix floating point to integer conversion for the alpha values Fixes bug #619835. 2010-05-26 08:54:33 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.22.3 pre-release 2010-05-26 00:33:59 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2010-05-25 15:34:11 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: handle truncated input data at EOS in pull mode Fixes #617733. 2010-05-26 11:55:13 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 357b0db to fd7ca04 2010-05-25 21:14:05 +0200 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Round timestamp up when scaling to mov format Fix timestamp rounding to allow the correct index to be located. The issue was that scaling from GStreamer time format to mov time format was rounding down causing the timestamp of the newsegment event received after a flushing keyframe seek to find the sample index before the one it should causing further backward seeking to the keyframe prior until no rounding error occurred. Rounding up when scaling to mov format has the desired effect, and it is not clear whether just the _round () variant would be sufficient. Fixes bug #619105 2010-05-24 17:26:42 +0100 Tim-Philipp Müller * gst/avi/gstavimux.c: * gst/flv/gstflvmux.c: * gst/matroska/matroska-mux.c: avimux, flvmux, matroskamux: don't crash if tags arrive on multiple input pads at the same time This is a temporary fix for the release only. Fixes #619533. 2010-05-25 17:05:12 +0200 Wim Taymans * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: rtptheora: remove delivery-method from caps We can accept all delivery methods so don't advertise anything on the caps or parse anything, we will handle whatever we receive. Fixes #618940 2010-05-25 15:40:01 +0100 Tim-Philipp Müller * gst/matroska/matroska.c: matroska: add temporary webm typefinder Add webm typefinder just for the release, so webm works for people whose distros don't patch gst-plugins-base as well. We'll remove this again after the release. 2010-05-23 11:17:27 +0100 Tim-Philipp Müller * gst/matroska/webm-mux.c: docs: add some pipeline examples to webmmux docs 2010-05-21 12:27:07 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: add webmmux to docs 2010-05-21 13:01:30 +0100 Tim-Philipp Müller * docs/plugins/inspect/plugin-matroska.xml: * gst/matroska/matroska-demux.c: * gst/matroska/matroska.c: * gst/matroska/webm-mux.c: matroska: fix up plugin and element descriptions a bit 2010-05-21 12:47:03 +0100 Tim-Philipp Müller * gst/matroska/Makefile.am: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska.c: * gst/matroska/webm-mux.c: * gst/matroska/webm-mux.h: matroska: move webmmux into own source files Makes things easier for gtk-doc. 2010-05-21 12:26:05 +0500 Christian Schaller * gst-plugins-good.spec.in: Update spec file with latest changes 2010-05-20 20:01:58 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroska: Remove the doctype enum, it's not needed anymore 2010-05-20 19:57:14 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: webmmux: Add new webmmux element that only supports muxing of WebM ...and remove the doctype property from matroskamux again. 2010-05-20 17:31:59 +0200 Mark Nauwelaerts * tests/check/elements/matroskamux.c: matroskamux: unit test checks version 1 files 2010-05-18 15:27:06 -0400 Tristan Matthews * ext/speex/gstspeexenc.c: speex: fix latency query Speex should report 30 ms latency for narrowband mode, 34 otherwise. Fixes #619018 2010-05-18 21:04:32 +0800 Philip * gst/matroska/ebml-read.c: ebmlread: rm floatcast.h include (not used) 2010-05-17 05:36:00 +0200 Philip Jägenstedt * gst/matroska/matroska-mux.c: matroskamux: bump default doctype version to 2 In this day and age this should be safe. There's otherwise a risk people will be creating unneccessarily big WebM files as they can't use SimpleBlock in v1. 2010-05-17 05:27:44 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: handle matroska and webm doctype versions equally The original plan was to let WebM v1 be the same as Matroska v2 (with extra constraints), but for simplicity it was decided to handle the versions equally, such that e.g. SimpleBlock is only allowed in WebM v2. 2010-05-13 12:10:54 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: Verify lace size in _parse_blockgroup_or_simpleblock Failure to do this for corrupt input can cause a subbuffer bigger than the actual buffer to be created, quickly leading to segfault. Test case: bug_s222005751_r0.001____memcpy.webm 2010-05-13 10:23:10 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: ebml: crude hack to avoid crashing on unexpected metadata The comment says this cannot happen, but it did and I don't know why. This is not the correct fix, needs investigation. Test case: bug_s555010094_r0.0005:0.008____IA__g_assertion_message_expr.webm 2010-05-13 09:18:56 +0200 Philip Jägenstedt * gst/matroska/ebml-read.c: ebml: don't modify out str if returning an error in _read_ascii This is a regression from ASCII validation changes. Test case: bug_s66876390_r0.001____malloc_printerr.webm 2010-05-12 13:16:28 +0200 Philip Jägenstedt * gst/matroska/ebml-read.c: ebml: Validate 7-bit ASCII in gst_ebml_read_ascii This was triggering an UTF-8 assertion in gst_caps_set_simple for corrupt files with garbage as codec id. Test case: gstreamer_error_trying_to_set_invalid_utf8_as_codec_id.webm Old gst_ebml_read_ascii renamed to gst_ebml_read_string, also used by gst_ebml_read_utf8. Unlike for UTF-8, failure to validate is an error, as gst_ebml_read_ascii is used for reading doctype and codec id and we might just as well give up early in those cases. 2010-05-12 14:30:18 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: Ignore unexpected CodecState Because GstMatroskaTrackContext *stream is set up in the first SimpleBlock or Block, a rogue CodecState otherwise causes a segfault on derefencing the NULL pointer. Test case: bug_s5506167_r0.001____gst_matroska_demux_parse_blockgroup_or_simpleblock.webm 2010-05-10 06:00:49 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: Add video/webm sink caps 2010-05-09 19:46:51 +0200 Philip Jägenstedt * gst/matroska/matroska-mux.c: matroskamux: Use SimpleBlock for WebM when possible 2010-05-09 19:28:59 +0200 Philip Jägenstedt * gst/matroska/matroska-demux.c: matroskademux: Support "webm" DocType 2010-05-09 12:35:10 +0200 Philip Jägenstedt * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: rename matroska_version to doctype_version 2010-05-09 12:09:57 +0200 Philip Jägenstedt * gst/matroska/matroska-ids.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: Support "webm" DocType 2010-05-12 18:38:48 -0700 David Schleef * gst/qtdemux/qtdemux.c: qtdemux: Add VP8 2010-04-27 15:26:13 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroskamux: Add support for On2 VP8 ...matroskademux automatically supports it through libgstriff. 2010-04-27 15:25:32 +0200 Sebastian Dröge * gst/avi/gstavimux.c: avimux: Add support for On2 VP8 ...avidemux automatically supports it through libgstriff. 2010-05-17 17:17:01 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: Don't lock the mainloop in NULL 2010-05-15 21:15:52 +0200 Sebastian Dröge * configure.ac: configure: Use = instead of == in shell scripts for equality checks 2010-05-14 18:33:32 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.22.2 pre-release 2010-05-14 18:24:14 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From 4d67bd6 to 357b0db 2010-05-14 18:16:45 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: fix leak in souphttpsrc unit test Unref server objects when done. Fixes check-valgrind. 2010-05-14 17:30:40 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegenc.c: jpegenc: fix two leaks Don't leak othercaps or jpegenc ref. 2010-05-13 13:01:26 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: fix docs Documentation error spotted by tony Fixes #618419 2010-05-11 13:18:42 -0400 Olivier Crête * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: make delivery-method parameter optional It probably will not be in the final RFC as it is not in RFC 5215 for Vorbis. If there is a configuration specified, assume it is in-line and if nothing is specified, assume it is in-band. https://bugzilla.gnome.org/show_bug.cgi?id=618386 2010-05-13 12:16:59 +0200 Wim Taymans * ext/jpeg/gstjpegdec.c: jpegdec: increase acceptable output sizes We can perfectly decode 1x1 images so lower the min width and height to 1. Fixes #618392 2010-05-13 11:30:27 +0200 Wim Taymans * gst/rtp/gstrtpceltpay.c: celtpay: fix queue duration calculations Don't blindly add the durations of incomming buffers to the total queued duration because it might be invalid. Mark the total queued duration invalid when we receive an invalid incomming timestamp because that's when we lose track of the total queued duration. Fixes #618324 2010-05-10 11:14:39 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: extract SPS and PPS from property provided parameter set ... so it can also be regularly inserted into the stream if so configured. Fixes #617164. 2010-05-11 22:28:08 +0200 Alessandro Decina * sys/osxvideo/osxvideosink.m: osxvideosink: allow switching views at runtime. 2010-05-11 20:26:37 +0100 Tim-Philipp Müller * gst/rtp/Makefile.am: rtp: dist missing header file to fix make distcheck 2010-05-11 19:05:08 +0100 Tim-Philipp Müller * sys/oss4/oss4-sink.c: oss4: minor cleanup Remove fixed FIXME, change finalise to finalize for consistency. 2010-05-11 19:01:51 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-oss4.xml: docs: add oss4 elements to docs 2010-05-11 16:09:10 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/ky.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: move oss4 strings from -bad to -good 2010-05-11 16:08:21 +0100 Tim-Philipp Müller * configure.ac: * gst-plugins-good.spec.in: * po/POTFILES.in: * sys/Makefile.am: * tests/icles/.gitignore: * tests/icles/Makefile.am: Move oss4 plugin from -bad to -good Hook up build infrastructure, docs and tests. Fixes #614305. 2010-04-29 13:18:58 +0100 Brian Cameron * sys/oss4/oss4-sink.c: * sys/oss4/oss4-sink.h: oss4sink: implement GstStreamVolume interface and add mute and volume properties OSS4 supports per-stream volume control, so expose this using the right API, so that playbin2 and applications like totem can make use of it (instead of using a volume element for volume control). Fixes #614305. 2010-04-08 10:45:33 +0100 Tim-Philipp Müller * sys/oss4/oss4-audio.c: oss4: 8-bit PCM audio caps don't need an endianness field 2010-04-08 10:40:02 +0100 Tim-Philipp Müller * sys/oss4/oss4-audio.c: oss4: don't iterate the formats table twice for each entry When iterating the formats table, we can just pass the whole entry to our helper function, which avoids iterating the table again to find the entry structure from the passed format id. 2010-03-30 11:43:04 +0100 Tim-Philipp Müller * sys/oss4/oss4-audio.c: oss4: also accept formats not natively supported Also accept formats that are not natively supported by the hardware, OSS4 can convert them internally. List the native formats first in the caps though, to express our preference for the native formats. We need this in order to support the case properly where the audio hardware supports only e.g. little endian PCM, but the host is big endian, since many audio elements only support native endianness and make the reasonable assumption that any audiosink will be able to handle audio in native endianness. Based on patch by Jerry Tan Fixes #614317. 2010-03-30 01:14:58 +0100 Tim-Philipp Müller * sys/oss4/oss4-mixer.c: oss4: add comment for translators Not that that will make these strings much better. Also remove i18n marker where it doesn't make sense. 2010-03-22 16:13:12 +0100 Benjamin Otte * sys/oss4/oss4-mixer.c: oss4: Refactor code to make it look more modern A side effect is that it passes -Wformat-nonliteral and doesn't read invalid memory in some cases, like when the mixer track contains a % sign or there is a number but not a known mixer name. 2010-03-22 14:09:24 +0100 Benjamin Otte * sys/oss4/oss4-mixer.c: oss4: Avoid g_quark_to_string (g_quark_from_string ()) madness We to the strdup inside gst_oss4_mixer_control_get_translated_name() instead of in the only caller. 2010-03-21 21:39:18 +0100 Benjamin Otte * sys/oss4/oss4-mixer.c: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-01-20 13:29:52 +0100 Benjamin Otte * sys/oss4/oss4-mixer.c: Fix compiler warning about unused return value 2009-08-21 01:17:18 +0100 Tim-Philipp Müller * tests/icles/test-oss4.c: tests: fix test-oss4 to treat an empty device name the same as a NULL name 2009-07-16 13:55:14 +0100 Jan Schmidt * sys/oss4/oss4-mixer.c: oss4: Attempt to fix a compiler warning Don't store a const gchar * in a non-const gchar * local var. Also, make the translation string function static since it's only used in the one file. 2009-06-10 19:21:21 +0100 Garrett D'Amore * sys/oss4/oss4-audio.c: * sys/oss4/oss4-mixer-slider.c: * sys/oss4/oss4-mixer-switch.c: * sys/oss4/oss4-mixer.c: oss4: Enhancements to the mixer and audio output Code cleanups, general improvements, support for the new mixer flags in latest gst-plugins-base. Fixes: #584252 Patch By: Brian Cameron Patch By: Garrett D'Amore 2009-06-19 16:21:28 +0100 Tim-Philipp Müller * sys/oss4/oss4-mixer.c: Make build without warnings with debugging disabled 2008-11-04 12:42:30 +0000 Stefan Kost Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-10-12 21:52:27 +0000 Jan Schmidt sys/oss4/: Add some spaces in translateable strings. Original commit message from CVS: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Add some spaces in translateable strings. Fixes: #555969 #555968 #555965 2008-08-07 16:20:30 +0000 Frederic Crozat Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822). Original commit message from CVS: Patch by: Frederic Crozat * ext/sndfile/gstsf.c: (plugin_init): * sys/dvb/gstdvbsrc.c: (gst_dvbsrc_plugin_init): * sys/oss4/oss4-audio.c: (plugin_init): Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822). 2008-06-16 07:30:34 +0000 Stefan Kost Final round of doc updates. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/speed/gstspeed.c: * gst/speexresample/gstspeexresample.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/dvb/gstdvbsrc.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/wininet/gstwininetsrc.c: Final round of doc updates. 2008-06-12 14:49:18 +0000 Stefan Kost Do not use short_description in section docs for elements. We extract them from element details and there will be war... Original commit message from CVS: * ext/dc1394/gstdc1394.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/metadata/gstmetadatademux.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * gst-libs/gst/app/gstappsink.c: * gst/bayer/gstbayer2rgb.c: * gst/deinterlace/gstdeinterlace.c: * gst/rawparse/gstaudioparse.c: * gst/rawparse/gstvideoparse.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/selector/gstinputselector.c: * gst/selector/gstoutputselector.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Do not use short_description in section docs for elements. We extract them from element details and there will be warnings if they differ. Also fixing up the ChangeLog order. 2008-06-12 13:06:37 +0000 Stefan Kost tests/icles/test-oss4.c: Include stdlib.h. Original commit message from CVS: * tests/icles/test-oss4.c: Include stdlib.h. 2008-05-22 16:33:25 +0000 Tim-Philipp Müller tests/icles/: Small oss4 test that probes for available devices and retrieves their caps and mixer tracks and all tha... Original commit message from CVS: * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/test-oss4.c: (opt_show_mixer_messages), (WAIT_TIME), (show_mixer_messages), (probe_mixer_tracks), (probe_pad), (probe_details), (probe_element), (main): Small oss4 test that probes for available devices and retrieves their caps and mixer tracks and all that. Also allows testing of mixer change messages on the bus. 2008-05-22 15:14:26 +0000 Tim-Philipp Müller sys/oss4/: Make device-name probing in NULL state work better (e.g. for the gnome-control-center sound capplet). Original commit message from CVS: * sys/oss4/oss4-mixer.c: (gst_oss4_mixer_open): * sys/oss4/oss4-property-probe.c: (gst_oss4_property_probe_find_device_name), (gst_oss4_property_probe_find_device_name_nofd): * sys/oss4/oss4-property-probe.h: * sys/oss4/oss4-sink.c: (gst_oss4_sink_get_property): * sys/oss4/oss4-source.c: (gst_oss4_source_get_property): Make device-name probing in NULL state work better (e.g. for the gnome-control-center sound capplet). 2008-05-08 19:16:17 +0000 Clive Wright sys/oss4/oss4-mixer-slider.c: Apparently mono sliders have the mono value repeated in the upper bits, so mask those o... Original commit message from CVS: Based on patch by: Clive Wright * sys/oss4/oss4-mixer-slider.c: (gst_oss4_mixer_slider_unpack_volume): Apparently mono sliders have the mono value repeated in the upper bits, so mask those out when reading them. Probably makes the mixer applet work properly in some more cases. 2008-04-11 08:13:22 +0000 Julien Moutte sys/oss4/: Fix arguments format in debug statements. Original commit message from CVS: 2008-04-11 Julien Moutte * sys/oss4/oss4-mixer-enum.c: (gst_oss4_mixer_enum_get_values_locked): * sys/oss4/oss4-source.c: (gst_oss4_source_delay): Fix arguments format in debug statements. 2008-04-02 20:18:58 +0000 Tim-Philipp Müller Add initial support for OSSv4. Mixer still needs a bit more love, but even magic has its limits. Original commit message from CVS: * configure.ac: * sys/Makefile.am: * sys/oss4/Makefile.am: * sys/oss4/oss4-audio.c: * sys/oss4/oss4-audio.h: * sys/oss4/oss4-mixer-enum.c: * sys/oss4/oss4-mixer-enum.h: * sys/oss4/oss4-mixer-slider.c: * sys/oss4/oss4-mixer-slider.h: * sys/oss4/oss4-mixer-switch.c: * sys/oss4/oss4-mixer-switch.h: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-mixer.h: * sys/oss4/oss4-property-probe.c: * sys/oss4/oss4-property-probe.h: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-sink.h: * sys/oss4/oss4-soundcard.h: * sys/oss4/oss4-source.c: * sys/oss4/oss4-source.h: Add initial support for OSSv4. Mixer still needs a bit more love, but even magic has its limits. 2010-05-11 10:52:58 +0200 Alessandro Decina * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: implement the xoverlay interface. Fixes #618349. 2010-05-11 18:42:32 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix push based seeking ... where it comes down to transforming incoming BYTE segment to a corresponding TIME segment. Also fixes #609405. 2010-05-11 14:23:47 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-imagefreeze.xml: * tests/check/Makefile.am: * tests/check/elements/.gitignore: Move imagefreeze plugin from -bad to -good Hook up build infrastructure, docs and unit test for new plugin. Fixes #613786. 2010-05-05 12:23:56 +0200 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Set fixed caps on the correct pad This makes the sink getcaps function actually used instead of using the fixed caps function for it. 2010-03-21 21:39:18 +0100 Benjamin Otte * tests/check/elements/imagefreeze.c: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-03-15 11:54:02 +0100 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Only start the task after a seek if a buffer was received already 2010-02-28 16:08:14 +0100 Sebastian Dröge * tests/check/elements/imagefreeze.c: imagefreeze: Add some unit tests 2010-02-28 16:04:31 +0100 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Set undefined framerate in sink getcaps function 2010-02-28 15:02:02 +0100 Sebastian Dröge * gst/imagefreeze/gstimagefreeze.c: imagefreeze: Implement reverse playback and set buffer offsets 2010-02-27 17:33:05 +0100 Sebastian Dröge * gst/imagefreeze/Makefile.am: * gst/imagefreeze/gstimagefreeze.c: * gst/imagefreeze/gstimagefreeze.h: imagefreeze: Add still frame stream generator element 2010-05-11 13:07:19 +0100 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-debug.xml: * gst/debugutils/Makefile.am: * gst/debugutils/gstdebug.c: * tests/check/Makefile.am: * tests/check/elements/.gitignore: Move capsfilter element from -bad to -good Hook up moved files to the build infrastructure and docs. Fixes #617739. 2010-05-06 13:12:32 +0200 Sebastian Dröge * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstcapssetter.h: capssetter: Some minor cleanup 2010-03-22 16:56:03 +0100 Benjamin Otte * tests/check/elements/capssetter.c: Add -Wold-style-definition and fix the warnings 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/debugutils/gstcapssetter.c: gst_element_class_set_details => gst_element_class_set_details_simple 2009-10-08 19:51:31 +0200 Mark Nauwelaerts * tests/check/elements/capssetter.c: capssetter: add unit test 2009-06-25 16:41:49 +0200 Mark Nauwelaerts * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstcapssetter.h: capssetter: import element into -bad 2010-05-11 12:06:10 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: avimux: check that pads have been negotiated Also set fcc_handler field in audio stream header. Fixes #618351. 2010-05-10 18:33:03 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix partial parsing of ctts table Fixes #616516. 2010-05-10 18:32:15 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: cleanup a comment and add some debug and conditional compilation 2010-05-11 10:01:52 +0200 Sebastian Dröge * configure.ac: configure: Check for GTK+ 3.0 and if it's not available for GTK+ 2.0 2010-05-10 22:11:10 +0200 Jan Urbański * gst/flv/gstflvmux.c: flvmux: only store the last buffer timestamp if it's valid Fixes bug #618305 2010-01-08 22:13:59 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: Re-send SPS/PPS when requested https://bugzilla.gnome.org/show_bug.cgi?id=606689 2010-05-07 17:09:16 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: fix typo in debug message 2010-05-07 15:42:23 +0200 Mark Nauwelaerts * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: rtptheorapay: add config-interval parameter to re-insert config in stream Add a new config-interval property to instruct the payloader to insert configuration headers at periodic intervals in the stream (when a keyframe is countered). 2010-05-07 15:31:03 +0200 Mark Nauwelaerts * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: fix in-band configuration parsing Also make configuration header parsing a bit more relaxed with respect to length field interpretation. 2010-05-07 15:30:30 +0200 Mark Nauwelaerts * gst/rtp/gstrtpvorbisdepay.c: rtpvorbisdepay: fix in-line configuration parsing Also make configuration header parsing a bit more relaxed with respect to length field interpretation. 2010-05-04 16:57:35 +0200 Mark Nauwelaerts * gst/rtp/gstrtptheorapay.c: rtptheorapay: do not discard downstream flow return 2010-05-04 16:57:11 +0200 Mark Nauwelaerts * gst/rtp/gstrtptheorapay.c: rtptheorapay: refactor buffer payloading 2010-05-07 20:41:04 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Add support for UYVY 2010-05-07 19:06:35 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: fix return value 2010-05-07 19:02:21 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't leak the session 2010-05-07 18:59:42 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: configure bandwidth properties in the session 2010-05-07 18:58:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: add properties to configure the bandwidth Add properties to proxy the bandwidth configuration to the session object. 2010-05-07 18:57:13 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: add properties to configure bandwidths Add properties to configure the sender and receiver bandwidths. Configure the bandwidths before calculating the RTCP timeout when we need to. 2010-05-07 18:56:30 +0200 Wim Taymans * gst/rtpmanager/rtpstats.c: rtpstats: add some debug info 2010-05-07 18:55:34 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: small cleanups 2010-05-07 16:55:13 +0200 Wim Taymans * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: rtpstats: make bandwidths more configurable Add a method to configure the various bandwidths in the session. 2010-05-07 13:32:30 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: handle NONE RTCP intervals Prepare for handling RTCP reporting intervals of GST_CLOCK_TIME_NONE, which means don't send RTCP at all. 2010-05-07 12:51:05 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: fall back to SDP ports instead of server_port In multicast, fall back to the ports in the SDP instead of the server_port attribute as this is more in line with the RFC. 2010-05-07 12:24:51 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: refactor collecting the transport info Make a method to collect the ports and destination address. 2010-05-07 11:28:36 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle servers that send broken Transports Handle servers that send their port pairs with the wrong name. Fixes #617537 2010-05-06 16:52:26 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: use the SDP connection info in multicast Parse the connection info from the SDP. When we need to configure the multicast destination, fall back to the SDP connection info when the transport did not specify a destination and ttl. Fixes #617537 2010-05-06 15:42:38 +0300 Stefan Kost * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/monoscope/gstmonoscope.c: goom,monoscope: truncate own caps, instead of copying and using the first only We got the caps from an intersect, it is our own, hence we can truncate it. 2010-05-06 15:40:33 +0300 Stefan Kost * ext/pulse/pulsesrc.c: pulsesrc: reflow to truncate caps just once We get writable cpas from the intersection (unless it failed). As we truncate those anyway, we don't need to manyaly copy the first structure. 2010-05-06 15:39:31 +0300 Stefan Kost * ext/gdk_pixbuf/gstgdkpixbuf.c: gdkpixbuf: don't leak template caps 2010-05-06 15:38:35 +0300 Stefan Kost * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: auto{audio,video}{src,sink}: use can_intersect to avoid a caps copy 2010-04-27 13:36:35 +0300 Stefan Kost * gst/flv/gstflvdemux.c: flvdemux: tell what we can do Any-caps are bad. If apps scan the registry, they'd like to know what we can output. 2010-04-27 13:43:29 +0300 Stefan Kost * ext/jpeg/gstjpegenc.c: jpegenc: also lift the arbitrary restrictions for width and height This was already done for jpegdec. 2010-05-06 14:03:11 +0200 Sebastian Dröge * ext/pulse/pulsesrc.c: pulsesrc: Allocate/free PA mainloop during state changes ...also destroy the stream and context during state changes. 2010-05-06 13:57:01 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Allocate and free the custom clock in NULL<->READY 2010-05-06 13:51:59 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Create and free the PA mainloop in NULL->READY/READY->NULL This fixes a race condition, when stopping the mainloop during finalization is done from a mainloop callback. Fixes bugs #614765 and #590662. 2010-05-05 19:35:48 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Make selection of a sinkpad number threadsafe 2010-05-05 17:39:32 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Add support for all common RGB formats 2010-05-05 16:06:51 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Add support for AYUV 2010-05-04 16:34:27 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: make setup url in a smarter way Make sure we always separate the base and control url parts with a / when creating the setup url. 2010-05-04 16:04:39 +0200 Alessandro Decina * gst/rtsp/gstrtspsrc.c: rtspsrc: handle SEEKING queries. 2010-05-04 11:13:45 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: rtpmp4vpay: add config-interval parameter to re-insert config in stream Add a new config-interval property to instruct the payloader to insert config (VOSH, VOS, etc) at periodic intervals in the stream (when a GOP or VOP-I is encountered). Based on patch by Fixes #607452. 2010-05-03 13:26:32 +0200 Alessandro Decina * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: move some initialization code from change_state to _init. Set ->active to TRUE in _init so it can be set to FALSE after creating the jitterbuffer and it won't be mistakenly reset to TRUE in the change_state function. This is needed to start the jitterbuffer as inactive when rtpbin is buffering. 2010-05-03 11:56:58 +0200 Alessandro Decina * gst/rtpmanager/gstrtpbin.c: rtpbin: fix a bug handling BUFFERING messages. If a session exists but has no streams, set the min buffering percent to 0 since it means that we haven't received anything for that session yet. 2010-05-03 11:51:37 +0200 Alessandro Decina * gst/rtpmanager/gstrtpbin.c: rtpbin: when a stream is created, pause the jitterbuffer if rtpbin is buffering. 2010-05-03 11:23:59 +0200 Alessandro Decina * gst/rtpmanager/gstrtpbin.c: rtpbin: fix a bug calculating stream offsets. 2010-05-01 14:20:59 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: Write previous cluster's size This is useful for backwards playback, which should be implemented in matroskademux at some point. 2010-05-01 14:15:49 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Set interlaced flag in the caps if the flag is set in the Matroska file 2010-05-01 14:12:28 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Write interlaced flag if the input video content is interlaced Unfortunately Matroska has no way to specify TFF and friends... 2010-05-01 11:25:26 +0100 Tim-Philipp Müller * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvorbisdepay.c: rtp: fix printf format of some debug messages 2010-05-01 11:06:53 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska: init variable to avoid compiler warning on OSX Fixes (bogus) "'offset' may be used uninitialized in this function" warning on build bot (also spotted by philn). 2010-04-30 17:19:44 -0700 David Schleef * gst/qtdemux/qtdemux.c: qtdemux: UYVY is 4:2:2, not 4:2:0 2010-04-30 22:22:25 +0200 Sebastian Dröge * ext/pulse/pulseutil.c: pulse: Don't compare values of two different enum types 2010-04-30 22:13:30 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Make automatic detection of interlacing the default Previously "force deinterlacing" was the default, which is a not very sensible default for the normal use case where deinterlace should act in passthrough mode unless interlaced content is present. 2010-04-29 16:26:49 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: optimise buffer scanning Specifically, when needing more data, do not rescan from start next time around, but resume from last position. See also #583047. 2010-04-29 15:38:49 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: disregard superfluous lines when indirect decoding 2010-04-27 15:44:39 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: add support for RGB and grayscale color space Also refactor src caps negotiation and setting. 2010-04-27 12:19:22 +0200 Mark Nauwelaerts * ext/jpeg/Makefile.am: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: support more colour spaces and some cleanups 2010-04-30 12:47:01 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegenc.c: jpegenc: more generic sink getcaps 2010-04-30 12:42:42 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: more sanity checks on input Specifically, verify input components / colour space is as code subsequently expects, thereby avoiding crashes or otherwise bogus output. Presently, that means 3 components YCbCr colour space, and somewhat limited sampling factors. Fixes #600553. 2010-04-22 12:28:22 +0200 Mark Nauwelaerts * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: also accept in-band configuration Fixes #574416 (theora). 2010-04-22 12:27:35 +0200 Mark Nauwelaerts * gst/rtp/gstrtpvorbisdepay.c: rtpvorbisdepay: also accept in-line configuration Fixes #574416 (vorbis). 2010-04-07 17:21:55 -0400 Olivier Crête * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: Ignore packets without a known codebook Don't produce an error if a packet is received without a valid codebook, it's possible that the codebook will just be coming later. See #574416. 2010-04-20 12:17:26 +0200 Mark Nauwelaerts * tests/check/elements/y4menc.c: y4menc: adjust unit test to element behaviour 2010-02-23 22:16:39 -0500 Benjamin M. Schwartz * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: y4menc: add 4:2:2, 4:1:1, and 4:4:4 output support Fixes #610902. 2010-04-15 12:21:56 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: DELTA_UNIT marking of output buffers ... which evidently makes (most) sense if output buffers are actually frames. Partially based on a patch by Miguel Angel Cabrera Fixes #609658. 2010-04-16 17:21:50 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263depay.h: rtph263depay: extra keyframe info from PTYPE header ... as opposed to taking it from h263 payload header, which need not be so reliable. Fixes #610172. 2010-04-16 17:08:47 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263depay.c: rtph263depay: also use Picture Start Code to detect packet loss This ensures a whole frame is dropped if a (start) packet is lost, rather than relying only on the DISCONT flag. 2010-04-16 17:06:11 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263depay.c: rtph263depay: detect frame start using Picture Start Code So we stop dropping fragments as soon as there is a picture start (code). In particular, this prevents dropping the first frame following initial DISCONT. 2010-04-16 16:34:06 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263depay.c: rtph263depay: handle a few FIXMEs 2010-04-16 16:27:25 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263depay.c: rtph263depay: slightly refactor payload dropping 2010-04-16 11:53:17 +0200 Mark Nauwelaerts * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: rtph263pay: use found GOBs to apply Mode A payloading ... rather than falling back to sending the whole frame in one packet if number of GOB startcodes < maximum. One might take this further and still perform Mode B/C payloading, but at least this should cater for decent fragments in typical cases. Fixes #599585. 2010-04-14 11:53:46 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: implement push mode seeking 2010-04-29 20:08:43 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * gst/smpte/gstsmptealpha.c: docs: update for videofilter plugin merge and add gtk-doc blurb for new property 2010-04-26 18:12:46 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Improve segment handling a bit 2010-04-26 18:05:00 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Order caps by amount of contained information 2010-04-26 17:25:38 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Properly set interlaced field in getcaps 2010-04-24 16:28:12 +0200 Sebastian Dröge * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Add planar YUV support to all other simple methods 2010-04-24 16:10:06 +0200 Sebastian Dröge * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: deinterlace: Add planar YUV support to greedyh method 2010-04-24 15:42:07 +0200 Sebastian Dröge * gst/deinterlace/tvtime/greedy.c: deinterlace: Add support for planar YUV formats in greedyl method 2010-04-24 13:58:03 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/vfir.c: deinterlace: Add support for Y444, Y42B, I420, YV12 and Y41B The vfir method supports them and will be used until something else supports it. 2010-04-24 09:16:22 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlacemethod.c: deinterlace: Define deinterlace method base classes as abstract types 2010-04-23 17:40:10 +0200 Sebastian Dröge * gst/deinterlace/Makefile.am: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Move deinterlacing methods to their own file 2010-04-23 17:25:12 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Simplify passthrough mode detection 2010-04-23 14:35:44 +0200 Sebastian Dröge * tests/check/elements/deinterlace.c: deinterlace: Fix unit test that checks caps handling deinterlace now always adds the interlaced field to the output caps, if it wasn't present in the input caps the output caps will still contain interlaced=false. 2010-04-21 17:00:05 +0200 Sebastian Dröge * gst/deinterlace/Makefile.am: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Refactor deinterlacing as preparation for supporting more color formats 2010-04-22 19:05:37 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for Y444, Y42B and Y41B 2010-04-22 15:54:21 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for YVYU and reorder template caps 2010-04-18 21:11:21 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Translate navigation events to make sense again upstream 2010-04-18 20:58:14 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Properly handle ranges/lists of width or height when transforming caps Code partly taken from the videocrop element. 2010-04-22 15:45:15 +0200 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Fix planar YUV->RGB processing 2010-04-22 15:42:03 +0200 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Correctly clamp after YUV->RGB conversion 2010-04-22 15:20:24 +0200 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Add support for YUY2, YVYU and UYVY 2010-04-18 15:02:42 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Sync properties to the controller in before_transform 2010-04-16 17:00:02 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for YUY2 and UYUV 2010-04-21 17:41:43 +0200 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Refactor processing and add support for other planar YUV formats This reduces the generated code size by a factor of 2.5. 2010-04-21 17:15:33 +0200 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Add support for YV12 input 2010-04-22 13:56:58 +0200 Sebastian Dröge * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/videomixer.c: videomixer: Add support for YUY2, YVYU, UYVY 2010-04-20 12:18:18 +0200 Sebastian Dröge * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/videomixer.c: videomixer: Add support for Y444, Y42B, Y41B and YV12 2010-04-21 17:07:10 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: videofilter: Order color formats by their contained amount of information 2010-04-20 18:22:16 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Drop Y41B/Y42B support Rotating 90°/270° with subsampled YUV where horizontal and vertical subsampling are different doesn't really work. 2010-04-19 14:37:54 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Also flip the pixel-aspect-ratio if width/height are exchanged 2010-04-18 23:08:14 +0200 Sebastian Dröge * tests/check/Makefile.am: * tests/check/elements/videofilter.c: videofilter: Extend the unit test to test different color formats 2010-04-18 22:55:36 +0200 Sebastian Dröge * tests/check/elements/videofilter.c: videofilter: Add some more tests These check different property combinations 2010-04-18 22:54:23 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Change the default method to identity 2010-04-18 22:50:20 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: videobalance: Reduce number of allocations per instance 2010-04-18 22:45:58 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: videofilter: Update last-reviewed comments 2010-04-18 22:40:55 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Add support for all RGB formats 2010-04-18 22:28:17 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Add support for YUY2, UYVY, AYUV and YVYU 2010-04-18 22:23:03 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Add debug category 2010-04-18 22:19:55 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Make property access threadsafe 2010-04-18 22:18:24 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Add support for Y41B, Y42B and Y444 2010-04-18 22:17:02 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: videobalance: Use libgstvideo for format specific things 2010-04-18 22:09:06 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Make properties controllable 2010-04-18 22:06:44 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Emit "value-changed" signal of color balance interface when values change 2010-04-18 21:58:13 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: videobalance: Some random cleanup 2010-04-18 21:37:23 +0200 Sebastian Dröge * gst/videofilter/gstvideobalance.c: videobalance: Stop using liboil The used liboil function is deprecated and has no optimized implementation anyway. 2010-04-18 21:14:11 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Make property access threadsafe 2010-04-18 15:00:36 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: gamma: Sync properties to the controller in before_transform 2010-04-18 14:46:09 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Add support for all RGB formats and AYUV 2010-04-18 14:31:36 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Add support for Y41B, Y42B and Y444 2010-04-18 14:29:30 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: videoflip: Make processing more general and use libgstvideo for all format specific things 2010-04-18 13:12:40 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: videoflip: Make method property controllable and improve debug output 2010-04-18 13:03:48 +0200 Sebastian Dröge * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: videoflip: Some random cleanup 2010-04-18 10:17:52 +0200 Sebastian Dröge * Makefile.am: * gst/videofilter/Makefile.am: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/plugin.c: videofilter: Move all elements into a single plugin Having all these small elements in a separate plugin is not very memory effective... 2010-04-18 10:07:24 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: * gst/videofilter/gstgamma.h: gamma: Improve docs a bit 2010-04-18 09:59:43 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: gamma: Add support for all RGB formats 2010-04-18 09:46:15 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: gamma: Add support for many packed YUV formats That is YUY2, UYVY, AYUV and YVYU. 2010-04-18 09:38:36 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: gamma: Add support for all other planar YUV formats That is Y41B, Y42B, Y444, NV12 and NV21. 2010-04-18 09:33:49 +0200 Sebastian Dröge * gst/videofilter/Makefile.am: * gst/videofilter/gstgamma.c: gamma: Stop using liboil The used liboil function is deprecated, only has a reference implementation and is more complex than what's needed here. 2010-04-17 18:13:46 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: * gst/videofilter/gstgamma.h: gamma: Use libgstvideo for format specific values and make gamma processing more generic Allows us to easily add support for new color formats later. 2010-04-17 18:01:06 +0200 Sebastian Dröge * gst/videofilter/Makefile.am: * gst/videofilter/gstgamma.c: gamma: Make gamma property controllable ...and properly use liboil. 2010-04-17 17:55:22 +0200 Sebastian Dröge * gst/videofilter/gstgamma.c: gamma: Some random cleanup 2010-04-19 14:45:33 +0200 Sebastian Dröge * gst/smpte/gstsmptealpha.c: smptealpha: Sync properties to the controller in before_transform 2010-04-17 17:47:05 +0200 Sebastian Dröge * gst/smpte/gstsmptealpha.c: smptealpha: Add support for YV12 (converted to AYUV) 2010-04-17 17:43:51 +0200 Sebastian Dröge * gst/smpte/gstsmptealpha.c: smptealpha: Add support for all 4 ARGB formats ...without format conversion. 2010-04-16 17:27:02 +0200 Sebastian Dröge * gst/smpte/gstsmptealpha.c: * gst/smpte/gstsmptealpha.h: smptealpha: Make color format support more generic This allows easier addition of new formats later. 2010-04-16 17:18:15 +0200 Sebastian Dröge * gst/smpte/gstsmptealpha.c: * gst/smpte/gstsmptealpha.h: smptealpha: Some random cleanup 2010-04-15 22:28:58 +0200 Sebastian Dröge * gst/smpte/gstmask.c: * gst/smpte/gstmask.h: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmpte.h: * gst/smpte/gstsmptealpha.c: * gst/smpte/gstsmptealpha.h: smpte: Add property for inverting the transition mask This converts a left-to-right transition to right-to-left or clock-wise to counter-clock-wise. 2010-04-15 22:27:57 +0200 Sebastian Dröge * gst/smpte/gstsmptealpha.c: smptealpha: Correctly detect property changes and update properties 2010-04-16 19:35:12 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqcelpdepay.h: qcelpdepay: add first version of a QCELP depayloader 2010-04-29 15:18:07 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development. === release 0.10.22 === 2010-04-28 02:58:02 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.22 2010-04-28 02:57:21 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2010-04-25 23:36:29 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.21.3 pre-release 2010-04-25 21:19:33 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: hide is-live property for release At the very least it needs a better/less wrong name. See #613066. 2010-04-25 15:12:20 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: don't crash if jpeg image contains more than three components Our code currently only handles a maximum of 3 components, so error out for now if the image has more components than that. Fixes #604106. 2010-04-20 17:21:29 +0100 Tim-Philipp Müller * gst-plugins-good.doap: doap: update repository info from cvs->git and maintainers 2010-04-23 14:40:20 +0100 Tim-Philipp Müller * common: Automatic update of common submodule From fc85867 to 4d67bd6 2010-04-22 13:30:55 +0200 Sebastian Dröge * gst/videomixer/blend.c: videomixer: Fix byte order for MMX ARGB/AYUV color filling Fixes bug #616409. 2010-04-21 17:53:49 +0200 Sebastian Dröge * gst/videomixer/blend.c: videomixer: Fix AYUV checker/color filling 2010-04-19 16:43:28 +0200 Sebastian Dröge * gst/videomixer/blend_mmx.h: videomixer: Add i387 floating point registers to the clobbered registers list They are the same as the mm0-mm7 MMX registers and will be overwritten by the assembly code if gcc doesn't know about the MMX registers. Note: They're all added to the list of clobbered registers in all cases and not only when __MMX__ is not defined just to make sure that no other bugs happen with this code just because some compiler version gets things wrong. Fixes bug #614466. 2010-04-19 14:09:34 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Use libgstvideo to get the order of RGB 2010-04-17 10:06:41 +0100 Brian Cameron * gst/goom/xmmx.c: goom: add edx to clobber list in inline assembly code mull modifies %edx, so should be mentioned in clobber list. Fixes crash on Solaris (#615998). 2010-04-15 13:39:41 +0100 Tim-Philipp Müller * tests/icles/Makefile.am: tests: don't use GST_PLUGIN_LDFLAGS when building test binaries 2010-04-16 15:27:12 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix I420->I420 copying Fixes bug #615143. 2010-04-13 18:15:50 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix AYUV->I420 copying 2010-04-16 12:14:26 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: profile-level-id is an optional parameter So, if needed, extract the corresponding info from sprop-parameter-sets. Based on patch provided by Fixes #612657. 2010-04-15 07:13:46 -0300 Thiago Santos * configure.ac: configure: Drop -Wcast-align Commit message copied from core's commit from Benjamin Otte: 246f5dba96a5b50bb74621af67b30942cca72af5 Apparently gcc warns that GstMiniObject is not castable to GstEvent/Message/Buffer due to them containing 64bit variables, even though ARM hackers claim that those only need 4byte alignment. And as long as gcc behaves that way, this warning is not very useful. So we'll remove the warning until this problem is fixed. Fixes #615698 2010-04-14 23:46:06 +0100 Tim-Philipp Müller * ext/flac/gstflactag.c: flactag: fix adapter assertion when used directly after flacenc Unlike filesrc, flacenc outputs the flac blocks neatly aligned one in each buffer. This means that when we switch from metadata mode to audio data passthrough mode, there's no data left in the adapter to push out at this point, so check if there's data in the adapter before requesting buffers from it (also needed in case we get input buffers of 0 size). Fixes #615793. 2010-04-14 23:18:27 +0100 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.21.2 pre-release 2010-04-14 20:31:30 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update 2010-04-14 20:06:09 +0100 Tim-Philipp Müller * tests/examples/equalizer/Makefile.am: * tests/examples/shapewipe/Makefile.am: * tests/examples/spectrum/Makefile.am: * tests/examples/v4l2/Makefile.am: * tests/icles/Makefile.am: tests: use LDADD for libs to link to instead of LDFLAGS Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to. This should make sure arguments are passed to the linker in the right order, and makes LDFLAGS usable again. Based on patch by Brian Cameron Fixes #615697. 2010-04-14 18:13:56 +0200 Edward Hervey * gst/videobox/gstvideobox.c: videobox: transform_caps : We can only convert AYUV to xRGB We were previously stating that we could convert AYUV/I420/YV12 to xRGB. 2010-04-13 00:14:46 +0100 Tim-Philipp Müller * configure.ac: configure: also remove -Waggregate-return from warning flags It causes problems with Objective-C code like in osxvideosink. Fixes #613663. 2010-04-12 18:22:39 +0200 Edward Hervey * tests/check/Makefile.am: check: Ignore osx audio/video src/sinks in state change tests And make the line readable for those mere mortals that don't own a 30" screen 2010-04-12 18:03:20 +0200 Edward Hervey * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: * tests/check/elements/level.c: * tests/check/elements/matroskamux.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: * tests/check/elements/spectrum.c: * tests/check/elements/videofilter.c: check: Don't re-declare 'GList *buffers' in the tests It's an external which lives in gstcheck.c. Redeclaring it makes some compilers/architectures think the 'buffers' in the individual tests are a different symbol... and therefore we end up comparing holodecks with oranges. 2010-04-12 14:50:46 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/qtdemux/qtdemux.c: matroskademux, qtdemux: minor code cleanup in avc_level_idc_to_string() Do the same with slightly fewer LOC. 2010-04-12 12:40:11 +0200 Edward Hervey * configure.ac: configure: Remove -Wundef flag Fixes #615161 2010-04-12 11:43:49 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix I420->AYUV copying 2010-04-12 11:25:59 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Correctly clamp frame/background alphas to [0,255] before writing them 2010-04-12 11:16:56 +0200 Edward Hervey * tests/check/elements/.gitignore: check: Ignore jpegenc test 2010-04-11 13:14:30 -0700 David Schleef * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Only check interlaced flag in sink caps Fixes #615460. 2010-04-09 11:21:47 +0200 Sebastian Dröge * common: Automatic update of common submodule From ba33d1f to fc85867 2010-04-08 18:05:46 +0300 Stefan Kost * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/rtpmanager/gstrtpbin.c: docs: do proper escaping for "%" 2010-04-08 17:50:49 +0300 Stefan Kost * gst/rtsp/gstrtspgoogle.c: * gst/rtsp/gstrtspgoogle.h: rtsp: remove obsolete google extension This was not build for a while and can be removed. 2010-04-08 17:42:52 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: move two symbols to private section 2010-04-08 17:36:30 +0300 Stefan Kost * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: add flxdec docs 2010-04-08 17:17:06 +0300 Stefan Kost * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegpay.c: docs: enable the 2 of 65 rtp elements in the docs 2010-04-08 11:54:19 +0200 Benjamin Otte * ext/shout2/gstshout2.c: shout2: Don't wait if we're late In fact, due to signedness issues, a negative delay would be changed to an almost infinite wait causing shout2send to "lock up". Reported by Christopher Montgomery. 2010-04-08 16:56:37 +0300 Stefan Kost * gst/udp/gstmultiudpsink.c: docs: upd -> udp and voila it shows up in the docs 2010-04-08 16:51:27 +0300 Stefan Kost * gst/alpha/gstalpha.h: docs: fix doc blob syntax 2010-04-08 16:51:05 +0300 Stefan Kost * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: add (sparse) docs for auparse element 2010-04-08 14:40:43 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: add videobox symbols 2010-04-08 14:40:19 +0300 Stefan Kost * docs/plugins/Makefile.am: docs: remove dynudpsink until someone documents it 2010-04-08 14:34:59 +0300 Stefan Kost * gst/flv/gstflvdemux.c: flvdemux: make debug category static 2010-04-08 14:29:19 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flxdemux: rename GstFLVDemux for GstFlvDemux 2010-04-08 14:23:19 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: * gst/flv/gstflvparse.h: flvdemux: merge flvparse into the demuxer and make function static No need to hide certain function in the docs. Allows to do more cleanups. 2010-04-08 13:13:34 +0200 Sebastian Dröge * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Add documentation 2010-04-08 14:00:08 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: v4l2buffer pool is now a separate object, remove them from v4l2src docs 2010-04-08 13:58:11 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: remove non existing flags and add two internal methods If someone cares flvparse could be merged into flvdemux. 2010-04-08 13:57:09 +0300 Stefan Kost * gst/rtpmanager/gstrtpsession.h: rtpsession: remove prototype for non existing function There is no function by that name anywhere. 2010-04-08 12:56:50 +0200 Sebastian Dröge * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videomixer.xml: docs: Update inspected plugin information 2010-04-08 12:56:30 +0200 Sebastian Dröge * gst/alpha/gstalphacolor.c: alphacolor: Improve docs a bit 2010-04-08 13:47:42 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: add effecttv defines and reorder list 2010-04-08 13:41:47 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: remove three entries that are not exported from the headers anymore 2010-04-08 13:40:36 +0300 Stefan Kost * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: move macro to c source One less semi public symbol without namespace prefix in the headers. 2010-04-08 13:40:09 +0300 Stefan Kost * ext/speex/gstspeexenc.h: speexenc: remove unused defines 2010-04-08 13:23:38 +0300 Stefan Kost * gst/matroska/matroska-mux.c: matroska-mux: fix last commit Use a local define for WAVEFORMAT_EX based on the size of the struct + 2 bytes for the extension size. 2010-04-08 13:16:53 +0300 Stefan Kost * ext/speex/gstspeexdec.h: speex: remove unused define 2010-04-08 13:03:43 +0300 Stefan Kost * gst/wavenc/Makefile.am: * gst/wavenc/gstwavenc.c: * gst/wavenc/riff.h: wavenc: remove internal copy of riff.h and use riff-library instead. We don't use any function yet, just the structures and defines. 2010-04-08 12:56:09 +0300 Stefan Kost * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: use riff lib more Remove BITMAPINFOHEADER and use the one from riff-lib. Also remove the WAVEFORMATEX_SIZE define and use a sizeof together with the respective struct. Besides better code reuse this lessens the ununsed symbols in the docs. 2010-04-08 12:14:07 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: docs: trim sections file more Rename some defines and move some itesm to *.c files. Add more items to internal subsection. 2010-04-08 11:19:43 +0300 Stefan Kost * docs/plugins/gst-plugins-good-plugins-sections.txt: docsw: trim the section file 2010-04-08 10:26:25 +0300 Stefan Kost * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: add v4l2sink to docs 2010-04-08 10:15:08 +0300 Stefan Kost * gst/audiofx/audioamplify.c: * gst/multifile/gstmultifilesink.c: docs: fix xml The title tag belongs into the refsect2. 2010-04-07 17:43:56 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for YV12, including conversion support for I420/AYUV 2010-04-07 17:27:12 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for grayscale input/output This doesn't do any conversion and is the next step to replacing videocrop by supporting all remaining formats in passthrough mode. 2010-04-07 16:24:38 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: Add support for filling the background with red, yellow and white 2010-04-07 16:11:11 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for direct RGB<->AYUV conversion 2010-04-07 16:11:01 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix RGB24 filling 2010-04-07 16:06:54 +0300 Marco Ballesio * gst/rtp/gstrtph264depay.c: h264depay: handle properly STAPs in rtph264depay.c, lines 577-576, NALU-type 24 (Single-Time Aggregation Packet) is handled in fall-through as NALU-type 26 (unhandled). This leads high quality h264 streams such as: rtsp://stream.yle.mobi/yle/areena/MEDIA_E0342657_p3.mp4 to fail with "NAL unit type 24 not supported yet" (but it's actually supported), and thus to close any stream which contains STAPs. The proposed one-liner patch fixes the issue. Fixes #615051. 2010-04-07 13:47:02 +0200 Thijs Vermeir * gst-libs/gst/gst-i18n-plugin.h: * gst/avi/gstavi.c: build: fix compiler warnings fix warnings for all plugins that use: setlocale (LC_ALL... 2010-04-07 13:31:13 +0200 Thijs Vermeir * gst/avi/gstavi.c: avi: fix compiler warning 2010-03-31 17:54:21 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: restrict resyncing to subtitle tracks This should prevent skipping audio or video in not so well interleaved cases. Fixes #614460. 2010-04-06 13:21:51 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Post avg./max. bitrate tags for H.264 This reads the average and maximum bitrates from the 'btrt' atom if available, and pushes these as tags, https://bugzilla.gnome.org/show_bug.cgi?id=614927 2010-04-03 23:39:20 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: fix racy shutdown Keep a ref of pulsesink for deferred mainloop invocation. Fixes #614765 2010-04-05 15:48:17 -0300 Thiago Santos * tests/check/Makefile.am: * tests/check/elements/jpegenc.c: tests: jpegenc: Adds some getcaps test Adds tests for the jpegenc getcaps function, to avoid having it returning non-subset caps 2010-04-05 14:51:58 -0300 Thiago Santos * ext/jpeg/gstjpegenc.c: jpegenc: Fix getcaps function When creating the caps allowed to upstream using downstream restrictions, use gst_pad_get_allowed_caps as that has the usable formats and puts into it the width, height and framerate fields. This avoids getting errors about getcaps returning non subset caps of its pad template. This error showed up on the metadata plugin unit test in -bad. 2010-04-05 17:31:36 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix conversion from 3 byte RGB to ARGB 2010-04-05 17:08:15 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add support for 3 byte RGB formats and refactor RGB code a bit 2010-04-05 15:51:13 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: Add support for all 32 bit RGB formats ...including conversion between them. 2010-04-05 15:26:03 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add property to control the buffering method Add a property to control how the jitterbuffer performs timestamping and buffering. 2010-04-04 19:02:41 -0300 André Dieb Martins * gst/alpha/gstalphacolor.c: alphacolor: Removing unused variable Fixes bug #614843. 2010-04-04 20:31:38 -0300 André Dieb Martins * ext/jpeg/gstjpegenc.c: jpegenc: should not return caps ANY based on downstream When downstream has a sink pad with ANY caps, jpegenc should treat it the same as NULL and return its template caps. Fixes #614842 2010-04-04 22:28:33 +0300 Stefan Kost * sys/oss/gstosshelper.c: oss: add fixme comment 2010-04-04 22:26:59 +0300 Stefan Kost * gconf/Makefile.am: build: use $(builddir) for installing generated files 2010-04-04 22:07:33 +0300 Stefan Kost * configure.ac: Revert "configure: fix out of source dir builds" This reverts commit ca0bd3a8cea31f9ea0df798a83d3007e696958ba. 2010-04-04 21:36:35 +0300 Stefan Kost * configure.ac: configure: fix out of source dir builds Remove non-existing gst-libs from include and library-paths'. Fixes #614354 even more. 2010-04-01 10:19:00 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Read replaygain peak/gain tags Make qtdemux read tags replaygain tags that are within '----' atoms. Fixes #614471 2010-04-01 18:48:43 +0530 Arun Raghavan * gst/matroska/matroska-demux.c: * gst/qtdemux/qtdemux.c: matroska: Export h.264 profile and level in caps This replicates the code in qtdemux to export the h.264 profile and level in the stream caps. https://bugzilla.gnome.org/show_bug.cgi?id=614651 2010-04-02 18:50:45 +0200 Sebastian Dröge * gst/qtdemux/qtdemux.c: qtdemux: Fix off-by-one introduced in last commit 2010-04-01 18:38:38 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Minor refactor of the code This will make it easier to clump together common code when copying to mastroskademux. https://bugzilla.gnome.org/show_bug.cgi?id=614651 2010-04-01 18:17:09 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Export h.264 level in caps This exports the h.264 level in the stream caps (as a string) which can be used to match a decoder, or as metadata. https://bugzilla.gnome.org/show_bug.cgi?id=614651 2010-04-01 16:58:32 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Export h.264 profile in caps This adds the h.264 profile for a given stream into caps. This can (eventually) be used to select an appropriate decoder and as metadata for certain applications. https://bugzilla.gnome.org/show_bug.cgi?id=614651 2010-03-31 14:43:14 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: remove obsolete reverse playback code path 2010-03-31 14:40:50 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: flvdemux: support (pull mode) negative seek rate 2010-03-29 15:27:37 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: also check for segment stop for non-segment-seek 2010-03-30 16:50:10 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: push correctly sized flac header buffers Fixes #614353. 2010-03-30 07:34:07 -0500 Rob Clark * configure.ac: build: fix compiler warning when srcdir != builddir Fixes '../../gst-libs: No such file or directory' warning/error when the build directory is not the same as the source directory. Fixes #614354. 2010-03-30 01:50:32 +0100 Tim-Philipp Müller * gst/id3demux/id3v2frames.c: id3demux: fix parsing of unsynced frames with data length indicator Fixes bug #614158. 2010-03-29 11:00:04 +0100 Tim-Philipp Müller * common: * ext/Makefile.am: * gst/Makefile.am: * sys/Makefile.am: * tests/examples/Makefile.am: build: build plugins and examples in parallel where possible 2010-03-18 18:49:24 +0000 Tim-Philipp Müller * sys/directsound/gstdirectsoundsink.c: directsoundsink: fix redundant function redeclaration compiler warnings Re-apply this again as well, as it was undone by the previous commit.. 2010-03-18 14:31:35 +0100 Benjamin Otte * sys/directsound/gstdirectsoundsink.c: gst_element_class_set_details => gst_element_class_set_details_simple Apply this again, as it was overwritten by the previous commit. Merging is hard, apparently. 2010-03-26 23:20:10 +0100 Julien Moutte * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: directsoundsink: Implement SPDIF support for AC3. Detect if the sound card supports SPDIF passthru of AC3 and add necessary code to support that like alsasink. 2010-03-26 17:06:57 +0000 Tim-Philipp Müller * Makefile.am: build: add cruft alert for common/shave* 2010-03-26 16:50:22 +0000 Tim-Philipp Müller * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_lang.c: * gst/qtdemux/qtdemux_lang.h: qtdemux: extract stream language in more cases The 16-bit language code can be either a packed ISO-639-2T code or a 'Macintosh language code'. Handle the latter type of language codes as well, and map to the matching ISO code. Lastly, fix language code posting for language #0, which is valid and stands for 'English'. Fixes #614001. 2010-03-26 14:55:53 +0100 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Improve debugging and add some FIXMEs 2010-03-26 14:42:06 +0100 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Sample rate markers 0x01, 0x02 and 0x03 are valid They are for 88.2kHz, 176.4kHz and 192kHz. 2010-03-26 14:16:39 +0100 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Take samplerate, width and number of channels from the STREAMINFO ...and update it from the frame headers if it should change for some reason. This allows playback of files with odd sample rates. 2010-03-26 13:45:46 +0100 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix AYUV->I420 frame copying 2010-03-26 13:34:17 +0100 Raimo Järvi * ext/jpeg/gstjpegenc.c: jpegenc: Set correct getcaps/setcaps functions on srcpads and simplify them This fixes downstream negotiation, upstream negotiation isn't really supported by jpegenc yet. Fixes bug #613789. 2010-03-26 10:31:22 +0100 Sebastian Dröge * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: Always fill the complete frame if borders should be added This makes sure that we don't get any gaps between rectangles because of chroma subsampling for example. 2010-03-18 22:12:40 +0000 Damien Lespiau * autogen.sh: autogen.sh: Don't call configure with --enable-plugin-docs configure gives a nice warning: configure: WARNING: unrecognized options: --enable-plugin-docs and indeed, I could not find anything in the configure.ac or the m4 macros that would allow enabling that option. Remove it then. 2010-03-22 16:58:26 +0100 Sebastian Dröge * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: Refactor boxing to reduce code duplication 2010-03-22 13:13:59 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Simplify caps transformation 2010-03-21 20:14:19 +0100 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Add const qualifier to the source frame data 2010-03-23 17:47:48 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: only seek when in proper state ... and data structures can be thread-safely accessed. See #601617. 2010-03-23 17:34:50 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: matroskademux: support (pull mode) negative seek rate 2010-03-18 15:29:00 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: track clip duration in segment 2010-03-18 13:39:05 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: prefer index of video track to perform seeking 2010-03-25 22:58:47 +0200 Stefan Kost * gst/dtmf/gstdtmfdetect.c: dtmfdetect: if we tell that we handle gap flags, then do so 2010-03-25 22:55:32 +0200 Stefan Kost * gst/dtmf/gstdtmfdetect.c: dtmfdetect: use glib types 2010-03-25 22:54:49 +0200 Stefan Kost * gst/dtmf/gstdtmfdetect.c: dtmfdetect: fix classification 2010-03-25 22:53:20 +0200 Stefan Kost * gst/dtmf/gstdtmfdetect.c: dtmfdetect: reformat message docs Use a list like in other element docs as an untweaked docbook table look ugly. 2010-03-24 16:19:53 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: fix typo in header validation check 2010-03-24 18:53:20 +0100 Edward Hervey * common: Automatic update of common submodule From 55cd514 to c1d07dd 2010-03-24 11:27:40 +0100 Sebastian Dröge * ext/lame/gstlame.h: * ext/lame/gstlamemp3enc.h: * ext/lame/plugin.c: build: Add all kinds of compiler warning flags and fix the resulting warnings 2010-03-23 19:46:43 +0100 Edward Hervey * gst/icydemux/gsticydemux.c: * gst/icydemux/gsticydemux.h: icydemux: Handle upstream Content-Type. Allows us to handle ShoutCast TV (NSV) streams. If the upstream caps have the 'content-type' field set to video/nsv, then we shortcut the typefinding and set video/x-nsv directly. 2010-03-23 19:30:50 +0100 Edward Hervey * ext/soup/gstsouphttpsrc.c: souphttpsrc: Set the Content-Type HTTP header on the caps. First step to fixing ShoutCast (NSV) streaming. 2010-03-23 02:38:43 -0400 Tristan Matthews * sys/osxaudio/gstosxaudioelement.c: * sys/osxvideo/Makefile.am: osx: fix compiler warnings Added void parameter to avoid old-style definition warning. Added -Wno-aggregate-return flag to avoid erroneous aggregate return warning. https://bugzilla.gnome.org/show_bug.cgi?id=613663 2010-03-23 00:15:15 +0000 Tim-Philipp Müller * tests/check/elements/videocrop.c: tests: use loop test for long-running videocrop check This should avoid timeouts on slow machines. Fixes #597739. 2010-03-22 17:26:37 +0200 Stefan Kost * ext/flac/gstflac.c: * ext/pulse/plugin.c: * ext/wavpack/gstwavpack.c: * gst-libs/gst/gettext.h: * gst/multifile/gstmultifilesink.h: i18n: build fixes: #if -> #ifdef for ENABLE_NLS 2010-03-22 17:25:09 +0200 Stefan Kost * gst-libs/gst/gst-i18n-plugin.h: i18n: fix the build Don't inlcude locale.h which we include in gettext.h if needed. Guard the inlcude like we do in the simillar headers in core. 2010-03-22 13:16:33 +0100 Benjamin Otte * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: Add -Wwrite-strings and fix its warnings 2010-03-22 12:02:16 +0100 Benjamin Otte * gst/dtmf/gstrtpdtmfsrc.c: Add -Wredundant-decls flag and fix warnings from it 2010-03-21 21:39:18 +0100 Benjamin Otte * gst/dtmf/gstrtpdtmfdepay.h: Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings 2010-03-21 17:46:06 +0100 Benjamin Otte * configure.ac: -Wold-style-definition is not valid for C++ 2010-03-21 17:36:28 +0100 Benjamin Otte * gst/multifile/gstmultifile.c: multifile: Include headers instead fo defining functions 2010-03-21 17:24:14 +0100 Benjamin Otte * configure.ac: Add a large set of warning flags. None of them trigger warnings anymore, so nothing needed to be fixed. 2010-03-21 17:23:43 +0100 Benjamin Otte * gst/goom/config_param.c: * gst/goom/convolve_fx.c: * gst/goom/filters.c: * gst/goom/flying_stars_fx.c: * gst/goom/goom_config_param.h: * gst/goom/goom_core.c: * gst/goom/goom_filters.h: * gst/goom/goom_fx.h: * gst/goom/ifs.c: * gst/goom/ifs.h: * gst/goom/plugin_info.c: * gst/goom/tentacle3d.c: * gst/goom/tentacle3d.h: Make goom not use aggregate returns 2010-03-21 15:17:46 +0100 Benjamin Otte * configure.ac: * ext/annodex/gstcmmlutils.c: * ext/wavpack/gstwavpackparse.c: * gst/effectv/gstwarp.c: * gst/rtp/gstrtph263pay.c: * gst/udp/gstmultiudpsink.c: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: * tests/check/elements/deinterlace.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rtp-payloading.c: * tests/check/elements/udpsink.c: * tests/check/elements/videofilter.c: * tests/check/elements/wavpackdec.c: * tests/check/generic/states.c: * tests/icles/v4l2src-test.c: Add -Wold-style-definition flag And fix the warnings 2010-03-20 00:54:14 +0100 Benjamin Otte * configure.ac: * ext/hal/hal.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/soup/gstsouphttpsrc.c: * ext/wavpack/gstwavpackcommon.c: * gst/avi/gstavimux.c: * gst/debugutils/gstpushfilesrc.c: * gst/flv/gstflvparse.c: * gst/goom/config_param.c: * gst/goom/goom_config_param.h: * gst/id3demux/id3tags.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/qtdemux/qtdemux.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videofilter/gstvideobalance.c: * sys/oss/gstossmixertrack.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * tests/check/elements/avimux.c: * tests/check/elements/level.c: * tests/check/elements/rtpbin_buffer_list.c: * tests/check/pipelines/simple-launch-lines.c: Add -Wwrite-strings to the configure flags ... and fix all warnings 2010-03-21 11:14:12 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: shapewipe: Add support for the remaining ARGB formats And handle AYUV like ARGB, we need no YUV specific handling. 2010-03-20 21:30:58 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Add support for RGB and xRGB input 2010-03-20 21:13:23 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Add support for ARGB input 2010-03-20 20:46:19 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Add support for generating ARGB output 2010-03-20 10:47:42 +0100 Sebastian Dröge * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/blend_mmx.h: * gst/videomixer/videomixer.c: videomixer: Add support for ABGR and RGBA Now all 4 ARGB variants are supported by videomixer. 2010-03-20 10:24:56 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Move chroma keying parameters into stack variables to prevent multiple pointer dereferences per pixel 2010-03-20 10:20:53 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Move color conversion matrixes into stack variables to speed up processing 2010-03-20 10:18:04 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Use correct matrixes to convert chroma keying color to YUV 2010-03-19 18:51:59 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Add support for different color matrixes 2010-03-19 18:21:19 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Rename and move functions as further preparation for supporting more color formats 2010-03-19 18:18:08 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Remove some unneeded calculations and instance struct fields And document the instance struct fields a bit better 2010-03-19 18:11:12 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Some preparations for supporting more color formats 2010-03-19 17:09:06 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: h264pay: fix config-interval property Use the same units for comparing the elapsed time against the interval. Fixes #613013 2010-03-19 16:44:00 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: alphacolor: Implement color-matrix support and use integer arithmetic only Alphacolor now uses the correct matrixes for SDTV and HDTV and can convert between them. 2010-03-19 15:03:43 +0100 Wim Taymans * configure.ac: * gst/rtsp/gstrtspsrc.c: rtsp: use GType from -base and bump required version Use the transport flags GType from -base and bump the required version of -base because of this. 2010-03-19 00:05:19 +0000 Tim-Philipp Müller * gst/apetag/Makefile.am: apetag: minor Makefile.am surgery -I$(top_srcdir)/gst-libs/ is already in $(GST_CFLAGS) 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/audiofx/gststereo.c: gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 17:30:26 +0100 Benjamin Otte * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-04 22:12:35 +0100 Andoni Morales Alastruey * ext/raw1394/gst1394clock.c: dv1394src: Fix internal clock Fixes #593910. 2010-03-18 21:14:17 +0000 Tim-Philipp Müller * ext/dv/Makefile.am: * ext/esd/Makefile.am: * ext/libcaca/Makefile.am: * ext/pulse/Makefile.am: * ext/shout2/Makefile.am: * ext/speex/Makefile.am: * ext/wavpack/Makefile.am: * gst/auparse/Makefile.am: * gst/avi/Makefile.am: * gst/flx/Makefile.am: * gst/icydemux/Makefile.am: * gst/interleave/Makefile.am: * gst/matroska/Makefile.am: * gst/qtdemux/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/udp/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavparse/Makefile.am: * sys/directsound/Makefile.am: * sys/oss/Makefile.am: * sys/waveform/Makefile.am: * tests/examples/v4l2/Makefile.am: build: Makefile.am cleanups Mostly add $(GST_BASE_CFLAGS) where it was missing, but also fix up order of flags and libs if needed (see docs/random/moving-plugins). 2010-03-18 18:49:24 +0000 Tim-Philipp Müller * sys/directsound/gstdirectsoundsink.c: directsoundsink: fix redundant function redeclaration compiler warnings 2010-03-18 19:00:09 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Remove remaining floating point arithmetic when processing a pixel 2010-03-18 18:55:34 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Refactor chroma keying into a single function This reduces code duplication once we add support for more color formats. 2010-03-18 15:53:14 +0100 Benjamin Otte * ext/lame/gstlame.c: gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:31:35 +0100 Benjamin Otte * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfaudiosrc.c: * ext/gconf/gstgconfvideosink.c: * ext/gconf/gstgconfvideosrc.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: * ext/hal/gsthalaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmng.h: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpng.h: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/efence.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/negotiation.c: * gst/debugutils/progressreport.c: * gst/debugutils/testplugin.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/qtdemux/gstrtpxqtdepay.c: * gst/qtdemux/qtdemux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspgoogle.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:02:30 +0100 Benjamin Otte * gst/oldcore/Makefile.am: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstaggregator.h: * gst/oldcore/gstelements.c: * gst/oldcore/gstfdsink.c: * gst/oldcore/gstfdsink.h: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmd5sink.h: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstpipefilter.h: * gst/oldcore/gstshaper.c: * gst/oldcore/gstshaper.h: * gst/oldcore/gststatistics.c: * gst/oldcore/gststatistics.h: Remove oldcore directory The elements have been unused for ages and all important ones have been replaced or copied elsewhere. 2010-03-18 13:45:08 +0100 Benjamin Otte * gst/avi/gstavidecoder.c: avi: Remove old file Seems to be leftover from the 0.4 days or so. 2010-03-18 12:44:53 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: pulse: use #ifdef rather than #if conditionals 2010-03-18 12:20:17 +0100 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: do not call _push_ts with unneeded (and wrong) time parameter Fixes #613206. 2010-03-18 11:33:59 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: fix typo in header validation check 2010-03-18 01:51:19 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: put more information in the metadata Additional tags are: audiocodecid, videocodecid framerate and (in the non-live case) filesize. While at it, fix index rewriting to update duration and filesize values even if the index is empty. Fixes #613094. 2010-03-17 21:33:28 +0100 Benjamin Otte * configure.ac: * ext/jpeg/gstjpegenc.c: * ext/speex/gstspeexenc.h: * gst/goom/goom_config.h: * gst/goom/mathtools.h: * tests/check/elements/level.c: Add -Wundef to configure flags and fix the resulting warnings 2010-03-17 20:02:16 +0100 Benjamin Otte * configure.ac: -Wmissing-prototypes is not valid for C++ 2010-03-17 19:35:10 +0100 Benjamin Otte * configure.ac: * ext/flac/gstflacdec.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/jpeg/gstjpeg.h: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/soup/gstsouphttpsrc.c: * ext/wavpack/gstwavpackdec.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/equalizer/gstiirequalizer.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtsp/gstrtspsrc.c: * gst/videomixer/videomixer.c: * sys/v4l2/v4l2src_calls.c: Add -Wredundant-decls warning flag Also fix compile issues 2010-03-17 18:49:11 +0100 Benjamin Otte * gst/monoscope/gstmonoscope.h: Fix warnings in experimental plugins, too 2010-03-17 18:23:00 +0100 Benjamin Otte * configure.ac: * ext/annodex/gstannodex.c: * ext/annodex/gstcmmldec.h: * ext/annodex/gstcmmlenc.h: * ext/annodex/gstcmmlparser.c: * ext/annodex/gstcmmlutils.c: * ext/dv/gstdvdec.c: * ext/flac/gstflacenc.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.h: * ext/jpeg/Makefile.am: * ext/jpeg/gstjpeg.c: * ext/jpeg/gstjpeg.h: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/wavpack/gstwavpackstreamreader.c: * ext/wavpack/gstwavpackstreamreader.h: * gst/debugutils/breakmydata.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/mmx.h: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/goom/goom_fx.h: * gst/goom2k1/filters.c: * gst/goom2k1/filters.h: * gst/law/mulaw-conversion.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/multipart/multipart.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: * gst/qtdemux/gstrtpxqtdepay.c: * gst/rtp/fnv1hash.c: * gst/rtp/fnv1hash.h: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpac3depay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpbvdepay.h: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpbvpay.h: * gst/rtp/gstrtpceltdepay.h: * gst/rtp/gstrtpceltpay.h: * gst/rtp/gstrtpdvdepay.h: * gst/rtp/gstrtpdvpay.h: * gst/rtp/gstrtpg723depay.h: * gst/rtp/gstrtpg723pay.h: * gst/rtp/gstrtpg726depay.h: * gst/rtp/gstrtpg726pay.h: * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263depay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.h: * gst/rtp/gstrtpilbcdepay.h: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpj2kdepay.h: * gst/rtp/gstrtpj2kpay.h: * gst/rtp/gstrtpjpegdepay.h: * gst/rtp/gstrtpjpegpay.h: * gst/rtp/gstrtpmp1sdepay.h: * gst/rtp/gstrtpmp2tdepay.h: * gst/rtp/gstrtpmp2tpay.h: * gst/rtp/gstrtpmp4adepay.h: * gst/rtp/gstrtpmp4apay.h: * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpmpvdepay.h: * gst/rtp/gstrtpmpvpay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpqdmdepay.h: * gst/rtp/gstrtpsirendepay.h: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsirenpay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: * gst/rtp/gstrtpsv3vdepay.h: * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstmask.c: * gst/smpte/gstmask.h: * gst/videobox/gstvideobox.h: * gst/videocrop/gstvideocrop.h: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: * gst/wavenc/gstwavenc.h: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2vidorient.h: * sys/ximage/ximageutil.c: * tests/check/elements/aspectratiocrop.c: * tests/check/elements/audioamplify.c: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiodynamic.c: * tests/check/elements/audioecho.c: * tests/check/elements/audioinvert.c: * tests/check/elements/audiopanorama.c: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: * tests/check/elements/avimux.c: * tests/check/elements/avisubtitle.c: * tests/check/elements/cmmldec.c: * tests/check/elements/equalizer.c: * tests/check/elements/level.c: * tests/check/elements/matroskamux.c: * tests/check/elements/multifile.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: * tests/check/elements/shapewipe.c: * tests/check/elements/souphttpsrc.c: * tests/check/elements/spectrum.c: * tests/check/elements/videofilter.c: * tests/check/elements/wavpackdec.c: * tests/check/elements/wavpackenc.c: * tests/check/elements/wavpackparse.c: * tests/check/elements/y4menc.c: * tests/check/generic/states.c: * tests/check/pipelines/simple-launch-lines.c: * tests/check/pipelines/wavpack.c: * tests/examples/equalizer/demo.c: * tests/examples/level/level-example.c: * tests/examples/spectrum/spectrum-example.c: * tests/icles/v4l2src-test.c: Add -Wmissing-declarations -Wmissing-prototypes warning flags And fix all the warnings. 2010-03-17 16:23:24 +0100 Wim Taymans * gst/rtp/gstrtpmp4gdepay.c: mp4gdepay: improve constantDuration guessing When no constantDuration has been given in the caps, try to derive one from the timestamp difference between packets. Also keep doing this for each packet because some broken streams might simply provide wrong timestamps. 2010-03-16 23:43:39 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: Put width and height in the metadata Some players use that info to scale their display. See #613094. 2010-03-16 23:32:45 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: don't put timestamps larger than G_MAXINT32 in the FLV tags For non-live input respond by pushing EOS, for live wrap the timestamps every G_MAXINT32 miliseconds. Fixes #613003. 2010-03-16 23:40:12 +0200 Stefan Kost * ext/soup/gstsouphttpsrc.c: soup: also use g_value_set_static_string() here for static strings 2010-03-16 21:23:11 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: alphacolor: Fix RGBA<->AYUV conversion 2010-03-16 21:16:26 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Remove redundant instance field 2010-03-16 21:10:08 +0100 Sebastian Dröge * gst/alpha/gstalpha.c: alpha: Protect property values from changes during frame processing 2010-03-15 23:29:55 +0300 Руслан Ижбулатов * ext/libpng/gstpngdec.c: pngenc: Use png_get_io_ptr() instead of accessing io_ptr directly Fixes #612700 (for the last time!) 2010-03-15 23:29:06 +0300 Руслан Ижбулатов * configure.ac: png: Check for libpng >= 1.2 instead of libpng12 2010-03-16 01:29:36 +0100 Jan Urbański * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Always put a duration tag in the metadata Some Flash players (for instance JW Player) always expect a duration tag, otherwise they don't start playback. If duration can be queried from the sink pads or is provided as a tag, use it. Otherwise try to determine it from the last seen timestamp of the sink pads after EOS and rewrite it in the header before writing the index. 2010-03-16 00:35:46 +0100 Jan Urbański * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Remove the send_codec_data field from GstFlvPad That field is not used anymore after the changes in 9fdecbc1c11f4e5af6578bba32a9b32771029d33. 2010-03-16 13:53:26 +0100 Wim Taymans * gst/udp/gstmultiudpsink.c: multiudpsink: get family of external sockets too Get the family of externally configured sockets so that we can configure it correctly. 2010-03-15 20:37:51 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: alphacolor: Add support for the remaining ARGB formats 2010-03-15 19:16:18 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: alphacolor: Simplify ARGB<->AYUV conversions by code generation macros 2010-03-15 19:07:28 +0100 Sebastian Dröge * docs/plugins/Makefile.am: * gst/alpha/Makefile.am: * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: Minor cleanups and move declarations into a separate header file 2010-03-15 18:58:51 +0100 Sebastian Dröge * gst/alpha/Makefile.am: * gst/alpha/gstalpha.c: alpha: Use GstVideoFilter as base class for automatic QoS support 2010-03-15 18:50:11 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: alphacolor: Add support for inplace conversions from AYUV to ARGB 2010-03-15 18:14:19 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: alphacolor: Use libgstvideo for caps parsing 2010-03-15 18:09:55 +0100 Sebastian Dröge * gst/alpha/Makefile.am: * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: alphacolor: Use GstVideoFilter as base class for automatic QoS support 2010-03-15 18:07:29 +0100 Sebastian Dröge * gst/alpha/gstalphacolor.c: alphacolor: Some minor cleanup 2010-03-15 14:16:58 +0100 Sebastian Dröge * ext/speex/gstspeexdec.c: * ext/speex/gstspeexdec.h: speexdec: Use speex_stereo_state_init() instead of the deprecated initialization macro Fixes bug #612777. 2010-03-15 01:09:49 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: Correctly mark buffers as delta units Mark video interframes, video codec data buffers and audio buffers (if it's not an audio-only stream) as delta units. 2010-03-14 19:32:20 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: Support streamheaders Put the FLV header, the metadata tag and (if present) codec information in the streamheader to allow the muxer to be used for streaming. 2010-03-14 01:38:21 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: Preallocate index space and fill it after finishing output Make the index appear at the beginning of the file, which is what most players are expecting. Fixes #601236. 2010-03-15 13:47:13 +0100 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Minor coding style fixes and cleanup 2010-03-14 01:34:02 +0100 Jan Urbański * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Add a is-live property If it is set, the muxer will not write the index. Defaults to false. 2010-03-14 01:25:42 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: Only put valid seek points in the index For files containing video only video keyframes are valid points to which a player can seek. For audio-only files any tag start is a valid seek point. See #601236. 2010-03-14 01:09:37 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: Fix index building to make entries point to tag's start offset Previous coding was wrongly incrementing the total byte count before adding an index entry. 2010-03-15 13:40:38 +0100 Sebastian Dröge * ext/cairo/gsttextoverlay.c: cairotextoverlay: Don't render text outside the frame boundaries Fixes bug #611986. 2010-03-15 11:38:23 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't forget to send keepalive messages When we operate in TCP mode, still send keepalive messages when we need to. Fixes #612696 2010-03-13 23:19:35 +0300 Руслан Ижбулатов * ext/libpng/gstpngenc.c: pngenc: Call png_jmpbuf() instead of accessing png_struct_ptr directly Fixes #612700 (again) 2010-03-12 16:44:30 +0300 Руслан Ижбулатов * ext/libpng/gstpngenc.c: pngenc: Call png_error() instead of using longjmp() directly. Fixes #612700 2010-03-12 13:57:28 +0100 Edward Hervey * common: Automatic update of common submodule From e272f71 to 55cd514 2010-03-05 11:06:47 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: add XMP parsing support Use xmp helpers to parse XMP metadata in udta atom. Fixes #609539 2010-03-11 12:32:56 -0800 Michael Smith * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: udp: fix compilation errors on non-windows. 2010-03-10 22:23:43 +0100 Andoni Morales Alastruey * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: multiudpsink: avoid getting the socket family using getsockname() 2010-03-11 17:28:47 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Fix print statements for pointer differences. This fixes it for both 32 and 64 bit 2010-03-11 17:28:35 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Fix unitialized variables 2010-03-11 17:03:47 +0100 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Fix printf formatting for macosx 2010-03-11 17:03:05 +0100 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Fix unitialized variables 2010-03-11 17:02:44 +0100 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Fix unitialized variable. 2010-02-19 13:39:04 +0100 Edward Hervey * gst/flv/gstflvparse.c: flvparse: Make script tag parsing more flexible. * The nb_elements for arrays is just an indication, we can therefore ignore it and carry on parsing metadata items until we reach the end marker. * If type == 3, then the script tag contains a list of object followed by the end marker. Refactor code slightly to handle both cases https://bugzilla.gnome.org/show_bug.cgi?id=610447 2010-03-11 15:51:40 +0000 Tim-Philipp Müller * tests/check/elements/deinterleave.c: * tests/check/elements/interleave.c: tests: fix metadata not writable warnings in interleave and deinterleave tests 2010-03-11 15:38:19 +0000 Tim-Philipp Müller * tests/check/elements/apev2mux.c: * tests/check/elements/id3v2mux.c: tests: fix metadata not writable warnings with apev2mux and id3v2mux tests 2010-03-11 15:24:20 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix metadata writable warnings Set metadata on buffer first, when the refcount is still 1, and only ref again afterwards. 2010-03-11 15:02:48 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: ignore stream with invalid header time metadata 2010-03-08 14:57:17 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Set stream-format=raw on AAC caps Set stream-format=raw for AAC caps, as that is the expected AAC format to be in this container family. Fixes #566250 2010-03-11 12:56:11 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: check for NULL before doing strcmp Check the connection and address type for NULL before doing strcmp and crashing. Fixes #612553 2010-03-11 11:20:59 +0100 Benjamin Otte * common: Automatic update of common submodule From df8a7c8 to e272f71 2010-03-11 11:09:55 +0200 Stefan Kost * gst/udp/gstudpnetutils.c: build: include stdlib.h for atoi() 2010-03-11 10:33:00 +0200 Stefan Kost * gst/audiofx/audiopanorama.c: audiopanorama: move invariant check out of the inner loop Improves performance for simple method. 2010-03-10 22:15:04 +0100 Benjamin Otte * configure.ac: Update CXXFLAGS, too, just like CFLAGS 2010-03-10 21:01:20 +0100 Benjamin Otte * configure.ac: * gst/rtpmanager/Makefile.am: * tests/check/Makefile.am: Update for recent changes to common submodule This just replaces every "$ERROR_CFLAGS" usage with a usage of "$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as previously. Actually using that separation will happen later. 2010-03-10 21:52:09 +0100 Benjamin Otte * common: Automatic update of common submodule From 9720a7d to df8a7c8 2010-03-10 20:43:57 +0100 Benjamin Otte * common: Automatic update of common submodule From 0b6e072 to 9720a7d 2010-03-10 10:51:28 -0800 Andoni Morales Alastruey * gst/udp/gstmultiudpsink.c: multiudpsink: Reset windows error code after getting corresponding error message. 2010-03-09 17:32:27 -0800 Michael Smith * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: put the codec_data blob into the actual data for MPEG4 video, to match other implementations in the wild. 2010-03-10 16:09:56 +0100 Benjamin Otte * common: Automatic update of common submodule From 7cc5eb4 to 0b6e072 2010-02-23 21:06:55 -0300 Thadeu Lima de Souza Cascardo * sys/ximage/gstximagesrc.c: ximagesrc: send new_segment with GST_FORMAT_TIME format Instead of using BaseSrc default format GST_FORMAT_BYTES, send it in GST_FORMAT_TIME. Signed-off-by: Thadeu Lima de Souza Cascardo Fixes #611659 2010-03-10 11:46:06 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode; also report seekable without an element index ... since recent code also seeks around to obtain required data from avi index. 2010-03-09 18:06:52 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: add some check and standardized seek event handling in push mode 2010-03-09 18:05:29 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: fix offset handling in push mode seeking Push mode seeking uses same index data as pull mode, and stores offset to data in chunk, whereas push mode operates in chunks, and as such needs offset consistently corresponding to chunk headers. Also fix determining best matching stream for incoming newsegment event, as well as setting some stream state accordingly. 2010-02-26 21:29:49 +0100 Mark Nauwelaerts * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: conduct index scan in task thread ... rather than in seeking thread, which might then occupy mainloop for some time with possible unresponsive side-effects. 2010-02-26 21:27:33 +0100 Mark Nauwelaerts * gst/flv/gstflvparse.c: flvdemux: avoid indefinite index growth That is, check for and do not add an index entry that has already been added. 2010-02-18 14:57:39 +0100 Mark Nauwelaerts * gst/flv/gstflvparse.c: flvdemux: also collect index info on-the-fly in pull mode 2010-02-18 12:42:31 +0100 Mark Nauwelaerts * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: * gst/flv/gstflvparse.h: flvdemux: incrementally build index in pull mode Scan for needed part upon a seek as opposed to doing a complete scan at startup, which may take some time depending on file and/or platform. Also accept index metadata in pull mode and peek for some metadata at the end of the file when deemed appropriate. 2010-02-18 12:26:46 +0100 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: some more variable cleanup 2010-03-09 18:25:23 +0100 Mark Nauwelaerts * gst/flv/gstflvparse.c: flvdemux: refactor adding index entry 2010-02-17 11:36:13 +0100 Mark Nauwelaerts * gst/flv/gstflvparse.c: flvdemux: fix setting DELTA_UNIT flag on outgoing buffers ... which should not depend on having index available or not. Also refactor resulting collapsed code. 2010-02-11 19:43:47 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: avoid erroneous codec-data overriding of stsd information 2010-02-01 22:37:30 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: adapt to new oggdemux Remove all granulepos hacks and simply use upstream timestamps. 2010-02-01 22:36:02 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: * ext/speex/gstspeexdec.h: speexdec: refactor granulepos hacks 2010-03-10 11:19:46 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: parse connection information Parse the connection information from the SDP and use it to figure out if we are dealing with ipv4 or ipv6 connections. 2010-03-09 17:53:32 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: require a destination for multicast When setting up the multicast sockets, we need a destination address to listen on or else we error. 2010-03-09 17:52:35 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: handle ipv6 listening ports when needed Add some code to make udpsrc listen on an ipv6 address when needed. The detection of IPV6 is not yet implemented. 2010-03-09 17:15:16 +0100 Wim Taymans * gst/udp/gstudpsink.c: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udp: use uri parsing code Use the uri parsing helper functions to manage the host and port pairs. This adds support for IPV6. 2010-03-09 17:13:31 +0100 Wim Taymans * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: udpnetutils: add helper functions for udp uri handling Add some helpers to parse udp uris. Make sure IPV6 is supported too. 2010-03-05 16:08:45 +0100 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsession: Make it possible to favor new sources in case of SSRC conflict Add a "favor-new" property that tells the session to favor new sources when there is a SSRC conflict. This is useful for SIP calls and other such cases where a remote loop is extremely unlikely. Fixes #607615 2010-03-05 15:46:48 +0100 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsession: Move SSRC conflicts lists into RTPSource We will also need to track SSRC conflicts in remote sources. See #607615 2010-02-26 17:13:49 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: send keep alive when paused When we are paused, send keep alive messages to the server so that our session doesn't time out when we go back to playing later. 2010-03-10 01:10:07 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 7aa65b5 to 7cc5eb4 2010-02-23 19:48:10 -0800 David Schleef * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: Add key-frame option to next-file This allows segmenting of MPEG-TS files at key frames, which is exactly what is needed for Apple's HTTP streaming. 2010-03-09 21:32:47 +0000 Sebastian Dröge * common: Automatic update of common submodule From 44ecce7 to 7aa65b5 2010-03-08 20:17:58 +0000 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix autocropping for odd width/height differences 2010-03-08 20:02:19 +0000 Sebastian Dröge * gst/videobox/Makefile.am: * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: Use libgstvideo for format specific stuff 2010-03-08 19:28:47 +0000 Sebastian Dröge * gst/audiofx/audioamplify.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbaseiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: audiofx: Sync properties to the stream time 2010-03-08 19:20:59 +0000 Sebastian Dröge * gst/videobox/Makefile.am: * gst/videobox/gstvideobox.c: videobox: Make properties controllable 2010-03-08 19:09:01 +0000 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Some cleanup 2010-02-28 15:47:50 +0100 Sebastian Dröge * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effectv: Use controller where possible, optimize a bit and make properties threadsafe 2010-02-26 16:35:17 +0100 Sebastian Dröge * pkgconfig/Makefile.am: build: Make some more rules silent if requested 2010-02-26 15:41:52 +0100 Sebastian Dröge * configure.ac: configure: Use automake 1.11 silent rules instead of shave if available This makes sure that we use something that is still maintained and also brings back libtool 1.5 support. 2010-03-08 22:57:34 +0100 Benjamin Otte * ext/libpng/gstpngenc.c: png: fractions don't allow doubles 2010-03-01 12:03:56 +0100 Benjamin Otte * gst/flx/gstflxdec.c: flx: fix description It's video, not audio 2010-03-09 17:45:27 +0000 Tim-Philipp Müller * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * win32/common/config.h: Back to development === release 0.10.21 === 2010-03-09 00:28:16 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.21 2010-03-09 00:24:45 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2010-03-09 00:09:34 +0000 Tim-Philipp Müller * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: Revert "Add 4:2:2, 4:1:1, and 4:4:4 output support" This reverts commit 637c26f61a2bd8d7b01f8b6d081d94da65f74557. === release 0.10.20 === 2010-03-08 23:42:51 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.20 2010-03-08 23:42:06 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2010-03-08 16:47:04 +0000 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: don't send second newsegment event in framed mode, fixes long playback delay Don't send another newsegment event if the upstream muxer/parser has already sent one (otherwise the sink will wait for $duration before starting playback). Fixes long delay until playback starts with flac-in-ogg files. Fixes #610959. 2010-03-05 13:49:31 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: configure multicast correctly Take the transport destination for multicast. Disable loop and autojoin for multicast on the udpsinks. 2010-03-05 13:47:33 +0100 Wim Taymans * gst/udp/gstmultiudpsink.c: multicast: always configure loop and ttl Also configure TTL and loop parameters when we add a client after initializing the sender. 2010-03-08 12:13:32 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: Revert "rtph263depay: baseclass handles timestamps for us" This reverts commit 564581e1b88ecd5ec5da82c3cafb0e7a2d58b302. If we don't call push_ts, there will be no timestamp at all on the outgoing buffer. Fixes #612154 2010-02-23 22:16:39 -0500 Benjamin M. Schwartz * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: Add 4:2:2, 4:1:1, and 4:4:4 output support 2010-03-02 13:21:24 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: use payload size to estimate bitrate Use the length of the payload for estimating the receiver bitrate so that it matches the calculations done on the sender side. Together with the number of packets one can scale the bitrate with the header overhead of the lower transport. 2010-03-02 12:39:20 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpsource: refactor bitrate estimation Don't reuse the same variable we need for stats for the bitrate estimation because we're updating it. Refactor the bitrate estimation code so that both sender and receivers use the same code path. 2010-03-01 16:40:27 -0500 Tristan Matthews * gst/rtpmanager/rtpsource.c: added bitrate estimation to receiver-side stats, fixes #611213 2010-03-01 16:01:24 +0100 Wim Taymans * gst/rtp/gstrtph263pay.c: h263pay: fix typo in debug === release 0.10.19 === 2010-03-06 00:43:03 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.19 2010-03-06 00:42:09 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2010-03-03 20:29:30 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.18.4 pre-release 2010-03-02 18:29:41 +0100 Edward Hervey * gst/matroska/matroska-demux.c: matroskademux: Make sure we don't send invalid newsegments Fixes #611501 2010-03-02 14:09:14 +0100 Edward Hervey * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Mark streams as being EOS at the right time. This allows us to stop streaming only when all streams have gone past the segment.stop and not before. Fixes #611501 2010-02-26 18:10:32 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Advance sparse streams only as much as required to keep the gap smaller than 500ms Changing it to the newest timestamp that was ever pushed will increase the segment start in 500ms jumps, which could be just after the next sparse stream buffer. E.g. Video at 1.0s, sparse stream at 0.5s would jump the sparse stream to 1.0s. Now a new sparse stream buffer could appear that has a timestamp of 0.9s and this would be dropped for no good reason because of bad luck. 2010-02-24 01:36:07 +0000 Tim-Philipp Müller * configure.ac: * po/es.po: * win32/common/config.h: 0.10.18.3 pre-release 2010-02-24 02:05:49 +0100 Alessandro Decina * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: Make sure FLUSH_STOP is sent so not to leave downstream flushing. 2010-02-23 17:25:54 +0100 Volker Grabsch * configure.ac: configure: Use $PKG_CONFIG instead of pkg-config to fix cross compilation Fixes bug #610839. 2010-02-23 17:24:03 +0100 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Reset skew detection after instantiating the jitterbuffer ...not only when going to READY. This sets high_level and friends to a more useful value. 2010-02-23 17:19:14 +0100 Sebastian Dröge * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: Return 100 if high-level is 0 instead of dividing by zero 2010-02-22 12:24:14 +0100 Wim Taymans * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: avoid division by 0 Avoid a division by 0 when no constantDuration was specified and when out two timestamps are equal. Fixes #610265 2010-02-22 18:20:46 +0100 Wim Taymans * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvdepay.h: dvdepay: don't output frames until we have a header Wait for the complete first 6 header DIF packets before outputting a frame. Decoders need this info to correctly decode the data. Fixes #610556 2010-02-22 20:55:29 +0100 David Hoyt * ext/jpeg/gstjpegdec.c: jpegdec: Fix invalid memory access by first checking and then reading Fixes bug #610483. 2010-02-18 09:05:50 +0100 Philippe Normand * ext/pulse/pulsesink.c: pulsesink: gst_pulsesink_get_mute: set result earlier. In the cases where no buffer was process yet or the index is not available, get_pulsesink_get_mute() would unconditionally return FALSE. https://bugzilla.gnome.org/show_bug.cgi?id=610337 2010-02-19 12:35:29 +0000 Tim-Philipp Müller * pkgconfig/gstreamer-plugins-good-uninstalled.pc.in: pkgconfig: fix gstreamer-plugins-good uninstalled .pc file Fix gst-plugins-base reference/requirement. This caused spurious problems with uninstalled -ugly/-bad not finding -good plugins in their unit tests (when distchecking). 2010-02-19 01:03:31 +0000 Tim-Philipp Müller * configure.ac: * po/lv.po: * win32/common/config.h: 0.10.18.2 pre-release 2010-02-19 00:54:13 +0000 Tim-Philipp Müller * tests/check/elements/.gitignore: * tests/examples/shapewipe/.gitignore: Make git ignore shapewipe examples and tests 2010-02-19 00:46:40 +0000 Tim-Philipp Müller * gst/flv/gstflvparse.c: flvdemux: minor micro-optimisation We know these values don't change during the loop, but the compiler doesn't and has to re-check them for every iteration. 2010-02-19 00:39:50 +0000 Tim-Philipp Müller * gst/flv/gstflvparse.c: flvdemux: remove static keyword from variables that shouldn't be static Multiple flvparse/flvdemux instances should be able to operate without trampling over each other by accidentally re-using the same (static) variables. (Spotted by Mark Nauwelaerts) 2010-02-16 02:07:07 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: docs: add Since: markers for new jitterbuffer properties 2010-02-18 18:20:24 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix off-by-one logic error in frame rate cap regression commit 2010-02-17 16:27:33 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Use the correct duration when comparing segments Do not confuse QtDemuxSegments with GstSegments when comparing the total file duration with the segment duration Fixes #610296 2010-02-17 18:06:29 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: add durations modulo 1<<32 For calculating the durations of each sample, we are supposed to add each duration modulo 1<<32 so make the elapsed time counter a uint32. Fixes #610280 2010-02-16 21:05:24 +0100 Anders Skargren * gst/multipart/multipartdemux.c: multipartdemux: improve header mime-type parsing Make the handing of the mime type within the "boundary" a bit less naive. The standard for MIME allows parameters to follow the "type" / "subtype" clause separated from the mime type by ';'. Modifies the multipartdemuxer's header parsing so it doesnt assume the whole line after "content-type:" is the mime type and thus makes it a bit more resilient to finding absurd mime types in the case where parameters are added. Fixes #604711 2010-02-16 19:53:09 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid stopping NULL tasks Check the task for NULL, it could be paused and set to NULL before. 2010-02-16 16:22:28 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix ALAC codec-data handling ALAC codec-data apparently comes in (at least) two flavours (mov, mp4), so use atom based parsing to retrieve required data, rather than aiming for a specific offset. See also #580731. 2010-02-16 15:50:23 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix debug message 2010-02-11 19:39:04 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_types.h: qtdemux: handle signed values in 3GPP location tag 2010-02-08 21:35:53 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: fix typo in debug message 2010-02-16 15:00:13 +0100 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: reset some more stream state after seek In particular, fixes non-flushing seek. 2010-02-16 14:44:11 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix frame rate cap regression Look for a non-zero min_duration during initialisation to avoid incorrect frame rate caps. 2010-02-16 10:13:17 +0200 Stefan Kost * sys/v4l2/gstv4l2bufferpool.c: v4l2: log more details in buffer pool finalize Helps to align with the loggin from libv4l. 2010-02-16 10:11:40 +0200 Stefan Kost * sys/v4l2/gstv4l2object.c: v4l2: init datastructures after pre-conditions checks 2010-02-16 10:10:45 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: jpegenc: add a fixme for handling other YUV variants 2010-02-16 01:40:19 +0000 Brian Cameron * gst/matroska/matroska-demux.c: matroska: fix GST_ELEMENT_ERROR usage Fixes #610053. 2010-02-16 00:50:15 +0000 Tim-Philipp Müller * configure.ac: configure: fix up GST_CXXFLAGS properly We don't want C specific flags in GST_CXXFLAGS, so base it on the GST_CFLAGS that only contains the pkg-config CFLAGS but none of the GST_OPTION_CFLAGS. Also, we only need the local includes once. Fix typo as well (GST_FLAGS -> GST_CFLAGS). 2010-02-15 23:13:46 +0200 Stefan Kost * configure.ac: configure: base GST_CXXFLAGS on --cflags from pkg-config pkg-config sets GST_CFLAGS and GST_LIBS. We need to use CFLAGS as a starting point for for both C and CXX settings. 2010-01-20 18:52:51 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpsession.c: rtpbin: remove use of ntp_ns_base 2010-01-20 18:22:20 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpstats.h: rtpbin: remove more ntpnstime and cleanups Remove some code where we pass ntpnstime around, we can do most things with the running_time just fine. Rename a variable in the ArrivalStats struct so that it's clear that this is the current system time. 2010-01-20 18:19:34 +0100 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: use running_time for jitter Use the running_time to calculate the jitter instead of the ntp time. Part of the plan to get rid of ntpnsbase. 2010-01-20 17:04:03 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpbin: change how NTP time is calculated in RTCP Don't calculate the NTP time based on the running_time of the pipeline but from the systemclock. This allows us to generate more accurate NTP timestamps in case the systemclock is synchronized with NTP or similar. 2010-02-15 12:12:36 +0000 Tim-Philipp Müller * sys/v4l2/v4l2_calls.c: v4l2: printf format string fix The compiler wants a cast here even though the type is already typedefed as 64-bit integer (presumably because glib has typedefed guint64 to unsigned long here). 2010-02-15 10:33:02 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska: fix printf format string 2010-02-15 00:50:10 +0000 Tim-Philipp Müller * ext/raw1394/gst1394clock.h: * gst/matroska/ebml-write.h: * gst/rtpmanager/gstrtpjitterbuffer.h: raw1394, matroska, rtpmanager: remove padding from structures None of these element and class structures are in public headers, so don't need padding. 2010-02-15 00:47:11 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update for new translator comment 2010-02-15 00:45:51 +0000 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: add comment for translators for 'x by y' message Fixes #609724. 2010-02-15 01:28:44 +0100 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Fix leaking of pad templates 2010-02-15 00:50:27 +0100 Sebastian Dröge * tests/check/elements/shapewipe.c: shapewipe: Fix unit test for latest changes Now the alpha is multiplied with the already existing alpha value instead of simply ignoring it and the luma/chroma values are kept, even if the output is 100% transparent. 2010-02-15 00:47:08 +0100 Sebastian Dröge * tests/check/elements/shapewipe.c: shapewipe: Improve unit test output on errors 2010-02-14 23:17:20 +0100 Sebastian Dröge * common: Automatic update of common submodule From 96dc793 to 44ecce7 2010-02-13 23:28:06 +0000 Tim-Philipp Müller * configure.ac: configure: bump -base requirement to git For GST_RIFF_TAG_JUNQ. 2010-02-12 16:11:30 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2.c: v4l2sink: change rank to NONE so it is never autoplugged 2010-02-13 18:18:42 +0100 Edward Hervey * gst/flv/gstflvparse.c: flvdemux: Audio tags without any content are valid. We silently ignore them instead of erroring out. 2010-02-13 18:07:50 +0100 Edward Hervey * gst/flv/gstflvparse.c: flvdemux: Fix GST_CLOCK_DIFF usage. It was previously checking for DIFF(a, b > 6 * GST_SECOND) instead of the proper DIFF(a,b) > 6 * GST_SECOND 2010-02-13 16:27:07 +0100 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Don't forget to reset the indexed variable when cleaning up 2010-02-13 11:01:53 +0100 Edward Hervey * gst/flv/gstflvparse.c: flvdemux: Speedup GstIndex usage Used the _add_associationv variant of GstIndex since we know how many associations we're adding. Trims up to 50% from index generation time. Note : It would be great if the index could be generated on the fly or on request as opposed to being fully created at startup. 2010-02-12 19:32:27 +0100 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: don't resync to invalid timestamps If we detect backward timestamps on the server, don't try to resync when we don't have an input timestamp (such as when using RTSP over TCP) instead, do nothing but assume the timestamp was ok, it will correct itself when time goes forwards. 2010-02-12 17:21:43 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: fix typo 2010-02-12 16:47:29 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: start out active and not buffering There is no need to set the latency in the jittebuffer in _init, we will set that later when going to PAUSED. Set the jitterbuffer active and not buffering when starting. 2010-01-27 17:57:55 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpbin: more buffering work When deactivating jitterbuffers when the buffering starts, keep the current percent of the jitterbuffer and also set the jitterbuffer in the buffering state so that we know when it's filled again. Add property to get the buffering percentage of the jitterbuffer. 2009-10-14 16:29:35 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: adjust latency in buffer mode When we are in buffer mode, adjust the buffering low/high thresholds based on the total configured latency. If we don't and there is a huge queue or element with a big latency downstream we might drain the complete queue immediately and start buffering again. 2009-10-12 11:54:07 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add ts-offset to timestamp Add the ts-offset to the buffer timestamp to get the final output timestamp of the buffer. 2009-10-08 19:23:53 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/rtpjitterbuffer.c: rtpbin: do more accurate buffer offsets Return the next timestamp in the jitterbuffer. Use the min-timestamp of the jitterbuffers to calculate an offset so that the next timestamp is pushed with a timestamp equal to running_time. Start producing timestamps from 0 in the buffering case too. 2009-10-08 18:42:11 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: only start buffering when < 100% Only start buffering when the percentage message is < 100 %. 2009-10-06 13:34:34 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: keep track of elapsed pause time Keep track of the time we spend pausing the jitterbuffers when they were buffering and distribute this elapsed time to the jitterbuffers. Also keep the latency in nanosecond precision. 2009-10-06 13:33:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: jitterbuffer: keep track of offset Keep track of an outgoing offset that we add to each outgoing buffer to compensate for PAUSE when buffering. Adjust the offset when activating. 2009-10-06 13:30:54 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: report level using high watermark 2009-10-05 21:31:59 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtsp/gstrtspsrc.c: rtpbin: pass running_time to jitterbuffer pause Pass the current running time to the jitterbuffer when pausing or resuming so that it calculate the right offsets. Small cleanups and comments. Set the default rtspsrc latency to 2 seconds. 2009-10-05 20:09:30 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpjitterbuffer.c: rtpbin: add some comments 2009-10-05 19:45:35 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpbin: more buffering updates Add signal to pause the jitterbuffer. This will be emitted from gstrtpbin when one of the jitterbuffers is buffering. Make rtpbin collect the buffering messages and post a new buffering message with the min value. Remove the stats callback from jitterbuffer but pass a percent integer to functions that affect the buffering state of the jitterbuffer. This allows us then to post buffering messages from outside of the jitterbuffer lock. 2009-10-05 13:32:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: rtpbin: propagate buffer-mode property Propagate buffer-mode property to the jitterbuffers. Intercept BUFFERING messages in rtpbin 2009-10-01 17:14:09 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: do more buffering implementation Add callback for buffering stats. Configure the latency in the jitterbuffer instead of passing it with _insert. Calculate buffering levels when pushing and popping Post buffering messages. 2009-10-01 12:46:21 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: flesh out buffering mode some more Add a buffering state to the jitterbuffer and wait until buffering ends before pushing out packets. 2009-10-01 12:09:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: hook up the mode property Expose a mode property on the jitterbuffer. Fix the case where timestamps are -1 in the check for outgoing timestamps. 2009-10-01 11:20:08 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: add buffering mode options Add getters and setters for different buffering modes that the jitterbuffer will support. Default to the current slave mode. 2010-02-12 15:54:37 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2.c: v4lsink: lower rank to MARGINAL 2010-02-12 16:06:45 +0100 Robert Swain * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: flvdemux: Obtain the index from the end of an flv file in push mode Allows for better support of seeking in flv files when in push mode 2010-01-21 11:55:15 +0100 Robert Swain * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Drop video frames up to the desired keyframe after a seek The audio packets in AVI are generally muxed ~0.5s before the corresponding video packet. This changes causes downstream to only receive packets with roughly corresponding timestamps. 2010-01-19 18:35:49 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: more DISCONT handling Add some debug in the DISCONT handling code. When we receive a DISCONT in push mode, mark all streams as DISCONT. 2010-01-19 10:51:08 +0100 Robert Swain * gst/avi/gstavidemux.c: avidemux: Fix _handle_seek_push () and new segement behaviour 2010-01-18 17:13:06 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: cleanups Make sure we reset the demuxer correctly wrt parsing the index. Don't leak pending seek events. Rename some methods to reflect what they do and to avoid confusion with similar method names. Try to make the seeking threadsafe by protecting the setup code with a lock. Make sure we post errors when a seek fails. 2010-01-18 11:45:38 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: rename some variables seek_event -> seg_event event_seek -> seek_event 2010-01-15 18:00:46 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: take fallback duration from avih When we have not parsed any indexes yet, we don't know the length of the streams and we must take the length given in the avih as a fallback. Avoid some typechecking. 2009-12-04 15:13:12 +0100 Robert Swain * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Push mode seeking support 2010-02-01 16:04:41 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: cleanup properties Use more default constants. Use static strings param flag. Init properties explicitly instead of letting gobject do this. 2010-02-12 15:34:38 +0200 Stefan Kost * ext/speex/gstspeexdec.c: speex: add missing include 2010-02-05 13:28:53 +0200 Stefan Kost * gst/debugutils/gsttaginject.c: taginject: fix multi-value tag example We need to use {} to specify a list. 2010-02-01 14:43:04 +0200 Stefan Kost * gst/avi/gstavidemux.c: * gst/wavparse/gstwavparse.c: avi,wav: also handle JUNQ chunk in addition to JUNK 2010-02-04 15:59:25 +0100 Wim Taymans * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp2tpay.c: rtppay: don't ignore result from set_outcaps set_outcaps can fail and we need to propagate the result upstream. 2010-02-04 15:36:24 +0100 Wim Taymans * gst/flv/gstflvparse.c: flvparse: fix confusing debug messages 2010-01-27 13:28:13 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add some more debug info 2010-01-27 13:26:46 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: avoid segfault when shutting down when we are shutting down, we might still receive state updates from pulseaudio but since we are unparented we should not do anything with the NULL parent anymore. 2010-01-26 18:33:27 +0100 Wim Taymans * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: fix timestamp problems When the pad with the highest framerate goes EOS, instead of not timestamping output buffers, intepollate timestamps and durations from the last seen ones. Fixes #608026 2010-02-12 11:32:40 +0100 Sebastian Dröge * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: Update documentation 2010-02-12 11:18:26 +0100 Sebastian Dröge * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-shapewipe.xml: * tests/check/Makefile.am: * tests/examples/Makefile.am: Moved 'shapewipe' from -bad to -good Fixes bug #584536. 2010-02-10 10:52:53 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 29/29] shapewipe: Preserve the input color values in all cases 2010-02-10 10:50:49 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 28/29] shapewipe: Scale mask alpha values by the source alpha values 2010-02-10 10:42:32 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 27/29] shapewipe: Fix ARGB processing 2010-02-10 10:34:24 +0100 Sebastian Dröge * tests/examples/shapewipe/shapewipe-example.c: [MOVED FROM BAD 26/29] shapewipe: Print some more details on error/warning messages 2010-02-08 08:26:33 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 25/29] shapewipe: Improve/add debug output 2010-02-08 08:20:44 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 24/29] shapewipe: Always hold the mask mutex before signalling the GCond 2010-02-08 08:19:48 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 23/29] shapewipe: Move chain function error cases at the end of the function and add useful debug output 2010-02-08 08:12:11 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: [MOVED FROM BAD 22/29] shapewipe: Fix race condition during shutdown that can lead to a deadlock 2010-02-08 08:11:33 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 21/29] shapewipe: Drop mask buffer on FLUSH events 2010-02-08 08:09:55 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: [MOVED FROM BAD 20/29] shapewipe: Update copyright year 2010-02-08 08:08:44 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 19/29] shapewipe: Don't reset properties when going PAUSED->READY Also use defines for the default values of the properties. 2010-01-16 16:52:11 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 18/29] shapewipe: Replace floating point arithmetic in the inner processing loops by integer arithmetic 2009-12-10 10:40:10 +0100 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 17/29] shapewipe: Don't do pointer dereferences in the processing loop Lowers the time taken there in my testcase from 6.91% to 6.20% as measured by callgrind. 2009-07-08 17:59:29 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 16/29] shapewipe: Add BGRA support for video in/output 2009-07-02 11:24:48 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: [MOVED FROM BAD 15/29] shapewipe: Add support for ARGB video input/output 2009-06-23 18:23:13 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 14/29] shapewipe: Correctly handle 0/1 fps 2009-06-09 19:14:41 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: [MOVED FROM BAD 13/29] shapewipe: Implement basic QoS This change is based on Tim's QoS implementation for jpegdec. 2009-06-09 18:45:19 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 12/29] shapewipe: Proxy queries on the video pads to the correct peers 2009-06-09 18:37:43 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 11/29] shapewipe: Proxy bufferalloc on the video sinkpad 2009-06-09 18:25:13 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 10/29] shapewipe: Try to work inplace if possible This saves one new, large allocation per frame for the most cases. 2009-06-04 08:56:14 +0200 Sebastian Dröge * tests/check/elements/shapewipe.c: [MOVED FROM BAD 09/29] shapewipe: Increase timeout of the unit test 2009-06-01 21:24:27 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 08/29] shapewipe: Fix some issues that were exposed by the new unit test 2009-06-01 21:24:12 +0200 Sebastian Dröge * tests/check/elements/shapewipe.c: [MOVED FROM BAD 07/29] shapewipe: Add unit test for shapewipe 2009-05-31 21:33:01 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 06/29] shapewipe: Add documentation and integrate into the build system 2009-05-29 21:07:26 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: [MOVED FROM BAD 05/29] shapewipe: Adjust border to still have everything transparent at 1.0 and the other way around 2009-05-29 16:55:25 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: * tests/examples/shapewipe/shapewipe-example.c: [MOVED FROM BAD 04/29] shapewipe: Divide the border value by two, otherwise we use a twice a wide border 2009-05-29 16:51:50 +0200 Sebastian Dröge * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: * tests/examples/shapewipe/shapewipe-example.c: [MOVED FROM BAD 03/29] shapewipe: Add border property to allow smooth borders ...and use a border of 0.01 in the example application. 2009-05-29 16:00:16 +0200 Sebastian Dröge * tests/examples/shapewipe/Makefile.am: [MOVED FROM BAD 02/29] shapewipe: Fix Makefile of the example application 2009-05-29 15:32:24 +0200 Sebastian Dröge * gst/shapewipe/Makefile.am: * gst/shapewipe/gstshapewipe.c: * gst/shapewipe/gstshapewipe.h: * tests/examples/shapewipe/Makefile.am: * tests/examples/shapewipe/shapewipe-example.c: [MOVED FROM BAD 01/29] shapewipe: Add a simple shapewipe transition filter & example application 2010-02-06 18:19:27 +0100 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Only flush the FLAC decoder if it wasn't created right before If the FLAC decoder is flushed, its state will be set to frame-sync mode, which will sync to the next *audio* frame and makes it ignore all headers. This prevented tags and everything else to show up when using flacdec in push mode. Fixes bug #608843. 2010-02-11 01:12:15 +0000 Tim-Philipp Müller * MAINTAINERS: Update MAINTAINERS 2010-02-12 00:03:09 +0000 Tim-Philipp Müller * configure.ac: configure: back to development Slushy freeze remains in effect. === release 0.10.18 === 2010-02-10 23:18:22 +0000 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.18 2010-02-10 23:17:21 +0000 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2010-02-10 20:36:56 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: temporary safety check to avoid crashes with a certain file Add temporary check to avoid crashes with a certain file when seeking until the real cause of this is figured out. See #609405. 2010-02-05 18:05:39 +0100 Robert Swain * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: skip unknown atoms when looking for moov Fixes bug #609107 2010-02-05 02:13:33 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.17.3 pre-release 2010-02-04 19:10:36 +0000 Tim-Philipp Müller * po/bg.po: * po/hu.po: po: update translations 2010-02-04 14:46:56 +0100 Robert Swain * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Set the segment start time to the requested seek time for non-keyframe seeks 2010-02-04 12:00:03 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix time returned for index at a byte offset The logic for searching forwards/backwards was swapped 2010-02-01 19:22:24 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: initialize stereo decoding state 2010-01-28 18:58:08 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: improve stream synchronization In particular, do not make it send newsegment updates that sort-of contradict the indented playback segment (e.g. start time). 2010-01-28 18:53:18 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: fix bridging (time) gaps in streams As a side effect, avoid sending newsegment updates with start times that go back and forth, which leads to bogus downstream running_time. Also fixes seeking in bug #606744. 2010-01-28 18:49:57 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: fix stream synchronization .. by initializing streams starting at 0, as that is basically where we 'seek to' at the start and assume streams to start elsewhere. Also enables newsegment update events for subtitle streams. 2010-02-02 13:41:03 +0200 Stefan Kost * ext/jpeg/gstjpegdec.c: jpeg: don't directly access message, some message have args This caused bogus messages, such as reported in bug #607471. 2010-02-02 00:02:34 +0000 David Hoyt * ext/libpng/gstpngdec.c: png: fix compilation with libpng 1.4 png_set_gray_1_2_4_to_8() has been deprecated for a while and was finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8() instead. Fixes #608629. 2010-02-01 16:46:36 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: free transports on errors See #608564 2010-02-01 09:18:53 +0000 Tim-Philipp Müller * sys/v4l2/v4l2_calls.c: v4l2: fix unportable printf format 2010-01-30 15:18:48 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 15d47a6 to 96dc793 2010-01-27 17:53:07 +0100 Robert Swain * gst/flv/gstflvmux.c: flvmux: index timestamps should be in seconds, not milliseconds 2010-01-27 15:24:52 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: free some more when resetting Fixes #608255. 2010-01-27 15:24:24 +0100 Mark Nauwelaerts * gst/rtp/gstrtpspeexpay.c: rtpspeexpay: fix occasional buffer leak Fixes #608255. 2010-01-27 15:22:46 +0100 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: prevent invalid arithmetic if not setup yet Fixes #608255. 2010-01-27 16:34:21 +0100 Sebastian Dröge * gst/videomixer/blend_mmx.h: videomixer: Fix assembly register constraints Fixes bug #608209. 2010-01-27 01:56:03 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.17.2 pre-release 2010-01-27 01:52:59 +0000 Tim-Philipp Müller * po/LINGUAS: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2010-01-27 01:49:49 +0000 Tim-Philipp Müller * tests/check/elements/.gitignore: checks: ignore deinterlace check binary 2010-01-27 01:18:51 +0000 Tim-Philipp Müller * configure.ac: configure: purge all mention of CVS 2010-01-26 11:18:28 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: ignore streams that finished When we receive an UNEXPECTED from a stream, move to the next stream and only go EOS when all streams are EOS. When selecting a stream to push, ignore streams that went EOS. Fixes #607949 2010-01-25 17:23:43 +0200 Stefan Kost * sys/v4l2/v4l2src_calls.c: v4l2src: don't deref NULL Error out when the pool gets shutdown. 2010-01-25 17:21:13 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: * sys/v4l2/v4l2src_calls.c: * tests/check/Makefile.am: Revert "v4l2src: don't deref NULL" This reverts commit 3d9d34bd60faeb940b36d992a47168fc895036ba. 2010-01-25 14:16:22 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: * sys/v4l2/v4l2src_calls.c: * tests/check/Makefile.am: v4l2src: don't deref NULL Error out when the pool gets shutdown. 2010-01-23 15:32:48 -0800 Michael Smith * ext/jpeg/gstjpegenc.c: jpegenc: when creating an overflow buffer, copy timestamps. 2010-01-23 14:47:55 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: dmb1 is a valid fourcc for Motion-JPEG 2010-01-23 14:20:02 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdeux: IV32 is also used for Indeo 3 video streams 2010-01-22 16:48:01 +0200 Stefan Kost * tests/icles/ximagesrc-test.c: build: no unused variables when disabling asserts 2010-01-21 23:17:40 -0300 Roland Krikava * gst/qtdemux/qtdemux.c: qtdemux: Avoid negative overflow on keyframe search Do not overflow negatively when searching a previous "keyframe" on audio streams. Could cause infinite loops on backwards playback Fixes #607718 2010-01-21 17:22:38 -0800 Peter van Hardenberg * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: enlarge buffer if libjpeg tells us it's out of space. Fixes buffer overflow on some high-quality, low-resolution jpeg encodes. 2010-01-21 19:24:22 +0100 Alessandro Decina * gst/qtdemux/qtdemux.c: qtdemux: fix compiler warnings under OS X. 2010-01-21 17:57:36 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: don't parse NULL indexes for some streams we might fail to fetch the index offsets. Don't try to parse NULL indexes in those cases. 2010-01-18 21:15:51 -0500 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: ptime should is in nanoseconds https://bugzilla.gnome.org/show_bug.cgi?id=607403 2010-01-20 15:11:15 -0300 Thiago Santos * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: Post warning if file isnt finished properly When the pipeline is shut down and the file isn't finished properly, wavenc should post a warning. Fixes #607440 2009-05-27 13:51:44 +0200 Arnout Vandecappelle * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: make index size configurable. Added the 'min-index-interval' property to matroskamux, which determines how much time (nanoseconds) is left between keyframes stored in the index. Fixes #583985. 2010-01-20 16:28:31 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: scale spspps_interval to milliseconds The spspps_interval is kept in seconds. Convert it to milliseconds before comparing it to another value in milliseconds. 2010-01-20 15:18:47 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: always keep media segments within total duration ... as opposed to only doing so following a seek. 2010-01-20 15:44:40 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: rename spspps-interval property Rename the spspps-interval property to config-interval because it is nicer. 2010-01-19 18:37:31 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: skip RIFF and index in push mode When we are in push mode, we can encounter RIFF and idx tags in the data chunk when we are dealing with ODML files. In these cases, simply skip the chunks and continue streaming instead of going EOS. 2010-01-20 11:27:23 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: more DISCONT handling Add some debug in the DISCONT handling code. When we receive a DISCONT in push mode, mark all streams as DISCONT. 2010-01-20 11:26:34 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: reset on flush events When we receive a flush event on the sinkpad, reset the EOS state and the flowreturn of all streams. Also mark the streams with a DISCONT. 2010-01-20 11:22:04 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: rename some variable Rename the seek_event variable to seg_event because it really contains the newsegment event that needs to be pushed. 2010-01-20 00:54:03 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 14cec89 to 15d47a6 2010-01-18 14:49:26 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Don't set profile-level-id in out caps The profile-level-id represents restrictions on what can be sent, it does not describe the stream. So it should be reflected in the sink caps of the payloader, not the src caps. https://bugzilla.gnome.org/show_bug.cgi?id=607353 2010-01-18 14:41:10 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: Don't ignore the return value from set_outcaps https://bugzilla.gnome.org/show_bug.cgi?id=607353 2010-01-18 17:43:41 +0100 Sebastian Dröge * gst/deinterlace/tvtime/greedyhmacros.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Fix license and copyright headers 2010-01-18 14:57:42 +0200 Stefan Kost * sys/v4l2/gstv4l2bufferpool.h: v4l2: move G_END_DECLS to the end 2010-01-18 14:55:38 +0200 Stefan Kost * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: fix bufferpool file names in header comment 2010-01-15 18:15:14 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: avoid some typecasting 2010-01-15 18:13:24 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: avoid some type checks 2010-01-15 18:09:15 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: fallback to avih duration when we have not yet parsed the indexes (in push mode, for example) use the duration as given in the avih header instead of -1. 2010-01-15 13:32:32 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: g_free is NULL safe 2010-01-15 13:27:40 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: use DEMUX errors, instead of DECODE qtdemux should use DEMUX errors, and not DECODE Conflicts: gst/qtdemux/qtdemux.c 2010-01-14 19:16:19 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Minor refactor Replace repeated code with a function call 2010-01-14 17:11:13 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Handle another kind of redirect trak Some traks might contain a redirect rtsp uri inside hndl atom (which is a dref atom entry). This commit makes qtdemux post a message when it finds one of these traks and there are no other traks. Fixes #597497 2010-01-14 16:13:08 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Post error when reaching EOS without pads Post an error when EOS is reached and there are no src pads 2010-01-14 14:13:50 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Do not post empty redirect messages Some misinterpreted data could result in posting redirect messages with empty redirect strings. It is better not to post them. An example is the file on bug #597497 2010-01-14 18:19:25 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: polish last buffer end time usage That is, reset it upon seek, and note that (rarely) last pushed buffer time might precede segment start. 2010-01-13 16:48:46 +0200 Stefan Kost * gst/videomixer/blend_mmx.h: videomixer: use 'q' constraint instead of 'r' This avoids the "bad register name `%dil'" compilation errors on 32bit where because of 'r' gcc puts the value in a general purpose register and then tries to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests a-d registers 2010-01-13 16:44:58 +0200 Stefan Kost * gst/avi/gstavidemux.c: avi: add missing include for sscanf 2010-01-13 09:36:03 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer10bands.c: equalizer: Fix property description for the 3rd band of the 10band equalizer The frequency is actually 237 Hz, not 227 Hz. Fixes bug #606692. 2010-01-13 09:22:20 +0100 Kipp Cannon * gst/audiofx/audioamplify.c: audioamplify: Allow negative amplifications Fixes bug #606807. 2010-01-13 09:17:05 +0100 Sebastian Dröge * ext/taglib/gstapev2mux.cc: apev2mux: Don't call constructors directly, this leads to compiler errors with gcc 4.5 2010-01-12 17:39:05 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier Fixes build on macosx 2010-01-11 19:02:34 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: refactor eos sending when pausing loop Also, prevent hanging if no pads yet on which to send eos by posting a message instead. 2010-01-11 17:50:35 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: standardize seek handling ... which implies fixing some corner cases. 2010-01-11 15:14:06 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: use more generic xiphN_streamheader_to_codecdata helper 2010-01-11 17:50:04 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: reflow audio and video setcaps and improve logging Also ensure width and height are available as they are mandatory in matroska specs. 2010-01-11 11:42:43 -0800 Michael Smith * gst/qtdemux/qtdemux.c: qtdemux: fix offset for type 2 mp4a sound sample descriptions. Allows us to correctly find the esds (and thus the codec data) for such mp4a files. 2010-01-11 15:45:49 -0300 Thiago Santos * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: rtpmp4g(de)pay: Only handle raw aac rtpmp4g(de)pay should only handle raw AAC streams 2010-01-11 18:59:43 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Implement basic QoS This drops frames if they're too late anyway before blending and all that starts but QoS events are not forwarded upstream. In the future the QoS events should be transformed somehow and forwarded upstream. 2010-01-11 14:48:26 -0300 Thiago Santos * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: rtpmp4a(de)pay: Only accept raw aac rtpmp4a(de)pay should only handle raw aac to conform to the RFC 2010-01-11 18:35:47 +0100 Sebastian Dröge * gst/videomixer/blend.c: * gst/videomixer/blend_mmx.h: videomixer: Add MMX implementations for I420 and all non-alpha RGB formats 2010-01-04 10:24:45 +0100 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/blend_mmx.h: * gst/videomixer/blend_rgb.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Refactor processing functions This allows easier plugging of optimized processing functions in the future, like for SSE or AltiVec. 2010-01-11 13:26:32 -0300 Thiago Santos * gst/avi/gstavimux.c: * gst/matroska/matroska-mux.c: avimux: matroskamux: rename aac's stream-format to raw AAC's none stream-format has been renamed to raw, rename on avimux and matroskamux as well 2010-01-11 12:07:29 -0300 Thiago Santos * gst/matroska/matroska-mux.c: matroskamux: Only accept raw aac makes matroskamux reject aac streams that are not in raw format (stream-format=none) Fixes #598350 2010-01-11 12:08:55 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: Only accept raw aac makes avimux reject aac streams that are not in raw format (stream-format=none) Fixes #598350 2010-01-11 10:38:10 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Oops. The gpointer cast is needed because of the const qualifiers on the data elements 2010-01-11 10:17:54 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Debug -> info level for a message for benchmarking index parsing The extra message output at higher levels affects the accuracy of the benchmark. 2010-01-11 10:05:10 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Don't check for NULL pointers or cast to gpointer as this is not needed 2010-01-08 13:55:05 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Refactor stbl sub-atom freeing. Free when index has been completely parsed. 2010-01-08 14:32:06 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Avoid whitespace commits due to inconsistent GNU indent behaviour 2010-01-11 00:10:34 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: remove newline at end of debug statement 2010-01-08 19:26:21 +0100 Havard Graff * gst/udp/gstmultiudpsink.c: multiudpsink: Compiler warning fixes for Windows Just simple missing casts Fixes bug #606438. 2010-01-08 18:04:14 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: fix seekpoints property copy-and-paste documentation 2010-01-06 17:06:53 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: optionally add a seek table API: GstFlacEnc:seekpoints Fixes #351595. 2010-01-08 11:33:02 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: Use more glib and be safer Be safer on sscanf by limiting string format sizes. Remove useless parameter and use g_strndup. 2010-01-08 10:44:44 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: Simplifying code Greatly simplify the IDIT chunk handling by using sscanf instead of 'manually' parsing. Also replaces strncasecmp and is_alpha/is_digit with glib versions. 2010-01-08 10:18:30 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: it's feb for february Fix typo in last commit. 2010-01-08 09:17:22 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: Parse and post IDIT dates Parses and post date tags contained in IDIT chunks. Fixes #503582 2010-01-07 17:25:05 +0100 Sebastian Dröge * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Add property for not draining the history on kernel changes Currently this only works if the kernel size doesn't change, in the future it will be possible to change the kernel size too without draining the complete history and without loosing anything. Partially based on a patch by Thiago Santos 2010-01-07 16:58:55 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: remove weird memcmp code Use plain memcmp for comparing memory instead of the custom buggy one. Fixes #606198 2010-01-07 15:38:36 +0100 Edward Hervey * gst/level/gstlevel.c: level: fix typo in 'message' property description 2010-01-06 14:06:14 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: really use upstream timestamp if there is one See/fixes #603471. 2010-01-06 13:45:59 +0100 Wim Taymans * gst/rtp/gstrtpg729pay.c: rtpg728pay: remove unused adapter peek 2010-01-05 19:00:35 -0300 Thiago Santos * tests/check/elements/deinterlace.c: deinterlace: Improve passthrough tests Improve passthrough tests by forcing more specific interlaced/deinterlaced caps to be tested 2010-01-05 18:22:49 -0300 Thiago Santos * tests/check/elements/deinterlace.c: deinterlace: Adds some docs to the new tests Adds some docs explaining the utility functions of the check tests of deinterlace 2010-01-05 18:14:08 -0300 Thiago Santos * tests/check/elements/deinterlace.c: deinterlace: Adds tests for passthrough Adds tests for checking if the element really does passthrough in disabled mode and in auto (if the input is not interlaced) 2010-01-05 07:50:51 -0300 Thiago Santos * tests/check/Makefile.am: * tests/check/elements/deinterlace.c: deinterlace: Adds tests for caps acceptance Adds check unit tests for deinterlace for validating caps accepting and the expected caps output on the other pad 2010-01-04 13:43:00 -0300 Thiago Santos * tests/check/Makefile.am: * tests/check/elements/deinterlace.c: deinterlace: Adds basic check test Adds a basic check test for deinterlace element 2010-01-04 15:44:28 -0800 Michael Smith * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: qtdemux: Add support for wave-style audio in qt. Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM content. 2009-12-31 17:09:03 -0500 Olivier Crête * tests/check/elements/rtp-payloading.c: tests: Add G.729 RTP payloader/depayloader test https://bugzilla.gnome.org/show_bug.cgi?id=606050 2009-12-31 16:52:30 -0500 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Simplify adapter usage https://bugzilla.gnome.org/show_bug.cgi?id=606050 2009-12-31 16:27:30 -0500 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Support ptime from caps https://bugzilla.gnome.org/show_bug.cgi?id=606050 2009-12-02 19:35:21 +0530 Olivier Crête * gst/rtp/README: rtp: Add maxptime to the README https://bugzilla.gnome.org/show_bug.cgi?id=606050 2010-01-05 19:03:06 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723depay.h: rtpg723depay: add G723 depayloader 2010-01-05 19:02:39 +0100 Wim Taymans * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729depay.h: rtpg729depay: remove unused variable 2010-01-05 18:33:25 +0100 Wim Taymans * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: rtpg723pay: rewrite payloader Handle all 3 packet sizes according to RFC 3551. Totally untested, we don't have a G723 encoder. Fixes #605882 2010-01-05 11:47:20 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: fix chunk counter 2010-01-04 19:44:53 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: more work at reducing loop overhead Try to avoid derefs when parsing the index. Save the state into the structures when we exit the loop instead of for each iteration. 2010-01-04 16:33:30 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: cleanups and make duration more accurate Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset as their 32 bit values. Make some macros to calculate PTS, DTS and duration of a sample. Deref the sample index less often by keeping a ref to the sample we're dealing with. 2010-01-04 13:41:18 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: simplify logic to calculate duration Since we no longer store the timestamp and duration in nanoseconds, we can now simply store the duration as-is. 2010-01-01 16:42:57 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Store timestamps in mov format in the index This allows faster building of the index upon seeks so that scaling of timestamps only occurs when actually needed. 2009-12-18 13:54:46 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: make seeking in push mode work Move sample position checks into qtdemux_parse_samples where we can protect it with a lock. Refactor and make an qtdemux_ensure_index function. Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion with gst_qtdemux_do_push_seek. 2009-12-18 12:44:27 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: move error code out of normal flow 2009-11-24 16:27:26 +0100 Robert Swain * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Add push mode seek support for seeking to obtain the moov atom 2010-01-05 12:22:09 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix on-npt-stop signal warnings for RDT The RDT manager does not implement this signal so we need to check for it before trying to connect to it. 2010-01-05 09:47:00 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2src.c: v4l2src: fix memory leak in new uri handler code Don't leak a string everytime get_uri() is called and a device has been set. There's a limited number of devices, so just intern the string instead of doing more elaborate housekeeping and storing it in the instance struct or so. 2010-01-01 14:10:49 +0200 Stefan Kost * gst/avi/gstavimux.c: avimux: fix typo in warning message 2010-01-04 09:28:36 -0300 Robert Weidlich * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2send: Add 'public' property Adds a property to set 'public' flag on libshout, making the stream listed on the server's stream directory. Fixes #605269 2009-12-30 14:14:55 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Add tags for average and maximum bitrate Fixes #599300. 2009-12-26 16:59:14 -0300 Thiago Santos * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: do not try to alloc really large buffers When nsamples_out is larger than nsamples_in, using unsigned ints lead to a overflow and the resulting value is wrong and way too large for allocating a buffer. Use signed integers and returning immediatelly when that happens. 2009-12-25 12:38:35 +0100 Wim Taymans * gst/videomixer/blend_ayuv.c: videomixer: optimize blend code some more Use more efficient formula that uses less multiplies. Reduce the amount of scalar code, use MMX to calculate the desired alpha value. Unroll and handle 2 pixels in one iteration for improved pairing. 2009-12-24 22:59:09 +0100 Wim Taymans * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/blend_rgb.c: videomixer: scale and clamp Scale and clamp to the max alpha values. 2009-12-24 22:50:31 +0100 Wim Taymans * gst/alpha/gstalpha.c: alpha: scale and clamp alpha to its full extend Convert the alpha value to 0->255 when setting and to 0->256 when using as a scaling factor. This makes sure we can reach the full opacity value of 0xff in all cases. 2009-12-24 22:23:01 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix some comments, remove property check Fix some comments, clarify some FIXMEs Remove the on-ntp-stop signal check now that the jitterbuffer is in -good and we know that it supports this signal. 2009-12-24 20:27:57 +0100 Wim Taymans * gst/videomixer/videomixer.c: videomixer: some trivial cleanups 2009-12-24 17:04:28 -0300 Thiago Santos * gst/rtsp/gstrtspsrc.c: rtspsrc: Parse all rtpinfo entries Do not forget to parse all rtp-info entries, instead of parsing the first one only. Fixes #605222 2009-12-22 12:44:50 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: perf tag should map to GST_TAG_ARTIST 2009-12-24 17:03:02 +0100 Wim Taymans * gst/interleave/interleave.c: interleave: fix weird indentation 2009-12-24 17:01:54 +0100 Wim Taymans * gst/rtp/gstrtph263ppay.c: rtph263ppay: use faster _adapter_copy() whem possible 2009-12-24 17:01:15 +0100 Wim Taymans * tests/examples/audiofx/firfilter-example.c: tests: use right type when passing vararg value 2009-12-23 17:50:34 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: use a single decoder field for both push and pull mode 2009-12-23 17:03:32 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: fix possible hanging in pull mode seeking A seek in multi-sink pipeline typically leads to several seek events in a row, which could lead to sending several newsegments in a row without intermediate flushing. These would then accumulate, distort rendering times and as such lead to 'hanging'. 2009-12-23 19:39:05 +0100 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: fix uninitialized variable 2009-12-23 13:09:54 +0100 Wim Taymans * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: use boilerplate 2009-12-23 00:38:05 +0100 Wim Taymans * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: rtpL16pay: convert to baseaudiopayload Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes a bunch of problems that were already solved in the base class. Fixes #853367 2009-12-23 00:30:49 +0100 Wim Taymans * gst/rtp/gstrtppcmapay.c: rtppcmapay: the boilerplate macro sets parent_class 2009-12-22 22:27:21 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpbin: avoid some structure copies Don't make copied in the getter and setter for SDES in the RTPSource. This avoids a couple of copies of the SDES structure when generating RTCP packets. 2009-08-31 18:42:25 +0200 Pascal Buhler * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpmanager: improve SDES handling Store SDES internally as a struct to support multiple PRIV values. Include all values set in SDES struct when sending RTCP SDES. 2009-12-22 14:41:35 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: add some fixmes 2009-12-22 14:35:13 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: baseclass handles timestamps for us 2009-12-22 14:27:40 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: reset start variable properly 2009-05-29 15:49:27 +0300 Marco Ballesio * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263depay.h: Drop the whole frame if a packet is lost. Fixes #582575 2009-12-21 20:39:53 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: add option to insert PPS/SPS in streams Add a new spspps-interval property to instruct the payloader to insert SPS and PPS at periodic intervals in the stream. Rework the SPS/PPS handling so that bytestream and AVC sample code both use the same code paths to handle sprop-parameter-sets. This also allows to have the AVC code to insert SPS/PPS like the bytestream code. Fixes #604913 2009-12-21 19:12:22 +0100 Mark Nauwelaerts * common: Automatic update of common submodule From 47cb23a to 14cec89 2009-12-21 12:01:53 -0300 Jonathan Conder * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: qtdemux: Adds new tags Adds some new tags mapping to qtdemux. Fixes #599759 2009-12-21 15:05:09 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: add property to remove pads automatically Add a property called autoremove to automatically remove the pads of sources that timed out. Fixes #554839 2009-12-21 14:55:16 +0100 Wim Taymans * gst/rtpmanager/gstrtpssrcdemux.c: ssrcdemux: fix comparison A NULL means no pad was found. 2009-11-08 11:49:14 +0100 Edward Hervey * sys/v4l2/gstv4l2src.c: v4l2src: Add GstURIHandler interface. Fixes #601143 This allows using v4l2://[] 2009-12-20 17:24:47 -0800 Michael Smith * gst/udp/gstmultiudpsink.c: multiudpsink: pass length parameter to g_convert 2009-12-18 12:44:50 +0100 Edward Hervey * gst/matroska/matroska-demux.c: matroska: Fix unitialized variable. Yes, it's stupid, but macosx compilers are even more stupid. 2009-12-17 16:01:25 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: videomixer: Fix assembly compilation on x86 Fixes bug #604814. 2009-12-17 17:37:03 +0100 Branko Čibej * gst/replaygain/rganalysis.c: rganalysis: fix timestamp rounding Use scaling function to round and avoid overflows. Fixes #604352 2009-12-17 17:27:42 +0100 Tiago Katcipis * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: rtp: add G723 payloader Fixes #597823 2009-12-17 16:22:56 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_types.c: qtdemux: Fix ALAC codec_data parsing Fixes #604611 2009-12-16 17:28:30 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Remove cpp style coments Removes // comments and replace them with /* */ comments 2009-12-16 12:48:02 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: also consider BlockNumber indicated in index when seeking 2009-12-16 12:43:27 +0100 Mark Nauwelaerts * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: support push based mode Fixes #598610. 2009-12-16 12:44:36 +0100 Mark Nauwelaerts * gst/matroska/ebml-read.c: matroskademux: fix ebml read cache usage 2009-12-16 10:50:32 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: videomixer: Use movzbl instead of movzxb for moving one byte to a l register For some reason latest gcc/binutils accept movzxb here while movzbl would be correct and is the only thing accepted by older gcc/binutils. Fixes bug #604679. 2009-12-16 06:59:01 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: videomixer: src/dest are input and output of the AYUV blending MMX assembler 2009-12-15 18:18:54 +0100 Sebastian Dröge * gst/audiofx/audiowsincband.c: audiowsincband: Use the same upper length limit as audiowsinclimit 2009-12-12 17:00:50 +0100 Sebastian Dröge * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: audiowsinc{limit,band}: Allow much larger filter lengths now 2009-12-11 12:27:32 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Fix frequency response calculation 2009-12-08 14:57:02 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Remove dead assignments 2009-12-06 16:58:51 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Add special processing functions for Mono/Stereo This provides another 7% speedup for the time domain convolution and 1.5% speedup for the FFT convolution on Mono input. This optimization assumes that the compiler simplifies calculations and conditions on constant numbers and unrolls loops with a constant number of repeats. 2009-12-04 09:25:49 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Add a "low-latency" mode This will always use time-domain convolution, which lowers the latency. With FFT convolution it's always a multiple of the kernel length, with time domain convolution it's only the pre-latency of the filter kernel. 2009-12-04 09:00:22 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Remove obsolete TODO comments 2009-12-03 20:12:01 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes 2009-12-03 17:27:13 +0100 Sebastian Dröge * gst/audiofx/Makefile.am: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: FFT convolution implementation This provides a great speedup, especially the relationship between kernel length and processing size is now logarithmic instead of linear. Below a kernel size of 32 it's a bit slower, afterwards it's much faster: 17 0.788000 -> 0.950000 33 1.208000 -> 1.146000 65 2.166000 -> 1.146000 ... 4097 107.444000 -> 1.508000 For sizes smaller 32 the normal time-domain convolution is chosen, for larger sizes the FFT convolution is automatically used. Fixes bug #594381. 2009-11-27 20:33:14 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm Only remaining part is the residue pushing, which will be fixed later. 2009-11-26 15:17:27 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Optimize time-domain convolution Remove some redundant calculations, move comparisions out of inner loops, etc. This makes the convolution about 3 (!) times faster but processing time is of course still proportional to the filter size. 2009-11-26 10:45:37 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once 2009-11-25 18:12:05 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Rewrite timestamp tracking It's much simpler now and doesn't introduce accumulating rounding errors. 2009-11-25 17:39:53 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Rename some variables and change comments 2009-11-24 20:06:25 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Add const qualifier to the source data array 2009-12-14 20:08:06 +0100 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend_ayuv.c: * gst/videomixer/videomixer.c: videomixer: Add MMX implementations of the AYUV blending and color filling functions This provides a 20% speedup for blending and 100% for color filling. The blending can probably be optimized even more. 2009-12-13 13:19:43 +0000 Tim-Philipp Müller * gst/id3demux/id3v2frames.c: id3demux: prefer two letter ISO 639-1 code for extended comment 2009-12-13 13:10:12 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: fix up language code extraction some more Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE is supposed to hold a ISO 639-1 code, so convert as needed using the new API from -base. See #602126. 2009-12-13 12:45:22 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: fix language code writing and extraction Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is supposed to contain two-letter ISO 639-1 codes, so use new language code mapping functions in -base to convert between those two as needed. Fixes #505823. 2009-12-07 20:54:07 +0000 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: minor debug message changes Fix up a few debug messages so that it's clearer what they mean. 2009-12-12 17:44:04 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: Revert "qtdemux: Correctly parse classification tags" This reverts commit cd883aa60c1133196a6ae052884d15c295c37dde. Previous code was correct, 4 is due to table and language code, not only language code 2009-12-12 16:28:36 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Correctly parse classification tags In clsf atoms, the language code is 2 bytes long, not 4. 2009-12-12 16:55:13 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers ... NULL buffers shouldn't really happen anymore when popping the buffer from GstCollectPads but better check for this and print a warning. 2009-12-11 13:11:12 +0100 Sebastian Dröge * gst/videomixer/blend_i420.c: videomixer: Fix stupid mistake in last commit 2009-12-11 12:35:59 +0100 Sebastian Dröge * gst/videomixer/blend_i420.c: videomixer: Don't do floating point math in the inner processing loop for I420 blending 2009-12-10 18:43:44 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle NULL and empty transport strings When an RTSP extension returns NULL or an empty transport string, just ignore it and try to get the next possible transport. Fixes playback of RealMedia streams. 2009-12-10 18:42:51 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: install event function on internal RTCP pad Install a custom event function on the internal RTCP pad so that we can reply TRUE to a latency event. 2009-12-10 10:48:49 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_rgb.c: videomixer: Remove wrong comments, copied from the I420 blend function 2009-12-09 21:15:07 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: The queued duration is a signed integer ...and it will really be negative sometimes. 2009-12-09 21:03:57 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Only pop buffers from collectpads after they're fully consumed This decreases latency and memory usage because new buffers are only accepted by collectpads if there's no queued buffer. 2009-12-09 20:42:44 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: Clean up position/duration handling Also use the last end time for closing the segment, not the start time of the last buffer. 2009-12-09 16:50:02 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Close the segment on EOS if the real duration is known 2009-12-09 16:46:18 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Update duration if current buffer is already after the old duration 2009-12-09 16:43:41 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Drop buffers that are after segment stop ...and if this happened for all streams go EOS. 2009-12-09 16:41:04 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Fix position tracking and sending of filler segments 2009-12-09 16:15:09 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations 2009-12-08 17:34:15 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Keep the segment stop position for update newsegment events 2009-12-04 14:42:49 +0100 Sebastian Dröge * configure.ac: * ext/Makefile.am: * ext/ladspa/Makefile.am: * ext/ladspa/gstladspa.c: * ext/ladspa/gstladspa.h: * ext/ladspa/gstsignalprocessor.c: * ext/ladspa/gstsignalprocessor.h: * ext/ladspa/load.c: * ext/ladspa/search.c: * ext/ladspa/utils.h: ladspa: Remove the sources from gst-plugins-good It's disabled anyway and the latest version of it is in gst-plugins-bad. Fixes bug #603779. 2009-12-04 13:50:59 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: init current_entry in push mode Set the current_entry to 0 (instead of -1) in push mode so that we correctly calculate the current frame number and timestamp. Add some more debug info and fic the duration debug. 2009-12-04 11:14:03 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: fix major memory leak when playing back rtsp video streams Don't forget to unref QoS, navigation and latency events when dropping them. 2009-12-03 08:58:08 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: only send pending tags with newsegment events Send pending tags only from the streaming thread, just after we've sent the newsegment event, not with e.g. flush-start. This not only does the right thing, but also makes sure we're not trampling over variables set up in the streaming thread from the seeking thread in case someone tries to issue a seek just as the demuxer is parsing the headers. Fixes #601617. Spotted by Ognyan Tonchev. 2009-12-03 17:49:55 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: fix debug message printf args Fixes debug message printf format to make it build in mac's gcc 2009-12-02 13:33:20 -0300 Thiago Santos * ext/shout2/gstshout2.c: shout2: Convert delay correctly Use GST_MSECOND to convert delay in msecs to nanosecs Fixes #603547 2009-12-02 11:21:22 -0300 Thiago Santos * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: Avoid crash when seeking before negotiating lame's 'lgv' variable is only initialized when the caps is negotiated, whenever a seek happens before that, it would attempt to call a function on an empty pointer, causing the crash. Fixes #603515 2009-12-01 19:24:02 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: jpegdec: reset segment info after flush Reset the segment info after a flush. We use the segment for handling QoS and if we don't reset the segment, QoS is basically disabled after a flushing seek. 2009-12-01 15:07:06 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 87bf428 to 47cb23a 2009-12-01 14:15:46 +0100 Sebastian Dröge * common: Automatic update of common submodule From da4c75c to 87bf428 2009-11-30 15:59:50 +0100 Aurelien Grimaud * gst/rtpmanager/rtpsession.c: rtpsession: avoid buffer ref/unref pairs for CSRCs We ref the buffer before pushing it downstream in order to get the CSRCs of it after pushing. This causes performance problems when downstream elements want to change the metadata because the buffer needs to be subbuffered. Instead, read and store the CSRCs of the buffer in an array before pushing it and process the array after pushing the buffer. This allows us to remove the ref/unref pair. Fixes #603376 2009-11-28 19:23:26 +0100 Wim Taymans * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2: use gstpoll for timeouts Use our own GstPoll based timeout instead of the shout sleep so that we can interrupt when doing a state change and shutting down. Fixes #602887 2009-11-28 12:25:06 +0100 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: check: fix jitterbuffer check Make sure we set a base_time on the element. Fix the timeout to at least twice the jitterbuffer latency. Enable previously failing tests. Remove impossible checks. 2009-11-27 18:55:20 +0100 Edward Hervey * common: Automatic update of common submodule From 53a2485 to da4c75c 2009-11-26 16:14:30 +0100 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: optionally merge NALUs into Access Units ... which may be expected/desired by some downstream decoders (and spec-wise highly recommended for at least non-bytestream mode). 2009-11-26 17:29:03 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix timestamp datatype 2009-11-25 10:38:23 -0600 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: avoid using wrong clock-rate Check for a valid clock-rate before attempting to estimate the npt stop time. 2009-11-25 10:37:30 -0600 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: fix typo in comments 2009-11-25 16:05:10 +0200 Stefan Kost * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffertest: add one more test and file a bug now CHange the backwards test to always send first buffer first to have a define basetime. Add another test that sends buffers backwards to assert that only first sent buffer is keep and used as basetime. Disabled those tests still, as its not passing/failing consitently and file a bug for jitterbuffer. 2009-11-25 10:17:34 +0200 Stefan Kost * tests/check/elements/rtpjitterbuffer.c: jitterbuffertest: improve the test the tests are a bit more solid now but still not produce reliable results. Wonder if they are still flawky or if its a bug in jitterbuffer. 2009-11-24 11:13:06 -0800 Michael Smith * gst/udp/gstmultiudpsink.c: multiudpsink: return error message on windows too. 2009-11-24 10:58:49 -0800 Michael Smith * gst/udp/gstmultiudpsink.c: multiudpsink: first phase of fixing up error reporting for windows. 2009-10-30 03:13:54 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: also set the suggested buf size for audio We were only setting the suggested buf size for video, we can set it for audio as well. This and 195e14529d80ef318ce3a778c1995efb11f266cd fix an issue that prevented seeking on large avi files on WMP (non-recent versions). 2009-11-04 16:10:23 -0300 Thiago Santos * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: fix indx duration for PCM audio GstBuffers for PCM audio usually contains more than 1 sample, we need to get the total number of samples to set the indx duration. 2009-11-04 16:04:10 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: Audio buffers should be picked earlier Adds a 0.5s advantage for audio buffers to being picked earlier for muxing. 2009-11-24 16:40:19 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix push mode by making sure stbl information is available in next_entry_size () 2009-11-24 16:35:20 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix order of arguments in log message 2009-11-24 15:51:21 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: jpegenc: fix spelling in comment 2009-11-23 17:58:17 +0100 Robert Swain * common: build system: Fix wrongly committed change to common/ 2009-11-10 10:26:07 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Ease debugging by removing a goto for an error message 2009-11-14 15:52:09 +0100 Robert Swain * common: * gst/qtdemux/qtdemux.c: qtdemux: Parse per sample rather than all at once but build complete index when seeking 2009-11-04 17:31:15 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Save atom data for later use so it doesn't get freed after initial parsing 2009-11-06 11:00:04 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Parse from the previously parsed sample up to sample n 2009-11-04 17:04:22 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Make qtdemux_parse_samples () parse up to n samples 2009-10-28 17:49:02 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Separate off stbl sub-atom initialisation 2009-10-26 22:42:36 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Move variables into context in preparation for refactorisation 2009-10-26 20:36:08 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix bug where stps is never parsed due to logic error 2009-11-04 17:31:15 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Port ctts from Gnode * to GstByteReader 2009-10-23 13:06:44 +0100 Robert Swain * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_types.h: qtdemux: Switch from QtAtomParser to GstByteReader 2009-11-23 12:53:50 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: fix typo and grammar 2009-11-22 19:30:58 +0000 Tim-Philipp Müller * gst/dtmf/Makefile.am: Clean up LDFLAGS, LIBS, CFLAGS Fix order, fix variables that don't exist, like GST_LIBS_LIBS, use $(LIBM) instead of -lm, and move _LIBS from LDFLAGS to LIBADD. Spotted by Havard Graff. 2009-11-20 10:31:47 -0500 Olivier Crête * gst/dtmf/tone_detect.h: dtmf: Use _stdint.h from configure https://bugzilla.gnome.org/show_bug.cgi?id=602465 2009-11-20 10:30:00 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix typo in mode enum description 2009-11-20 11:25:49 +0200 Stefan Kost * gst/rtpmanager/gstrtpbin.c: docs: more links and better short description Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change the short description to be more meaningful. 2009-11-20 09:58:26 +0100 Sebastian Dröge * tests/check/elements/wavpackparse.c: wavpackparse: Fix unit test for recent position reporting changes 2009-11-19 20:33:07 -0500 Olivier Crête * gst/dtmf/tone_detect.c: * gst/dtmf/tone_detect.h: dtmf: Update dtmfdetect to make it MSVC friendly https://bugzilla.gnome.org/show_bug.cgi?id=602465 2009-11-19 16:09:38 +0100 Sebastian Dröge * ext/wavpack/gstwavpackparse.c: wavpackparse: After pushing a frame, update last_stop to the end of the frame This improves position reporting, especially because of the fact that WavPack frames are usually 0.5-1.0 seconds long. 2009-11-19 16:08:33 +0100 Sebastian Dröge * ext/wavpack/gstwavpackparse.c: wavpackparse: Allow pulling the last WavPack frame of a file Because of a >= instead of a >, that last frame of a WavPack file would never be parsed in pull mode. 2009-11-19 10:30:43 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 0702fe1 to 53a2485 2009-10-29 08:29:38 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Add more fields to SVQ3 caps qtdemux only added the whole stsd atom as 'codec_data' in its output caps for SVQ3. This patch makes it add the SEQH (inside a SMI atom) and a gamma field (taken from the gama atom) if available. Fixes #587922 2009-11-18 17:55:42 +0100 Edward Hervey * gst/wavenc/gstwavenc.c: wavenc: Raise rank of muxer to PRIMARY 2009-11-18 17:54:16 +0100 Edward Hervey * gst/y4m/gsty4mencode.c: y4m: Raise rank of encoder to PRIMARY 2009-11-18 17:54:02 +0100 Edward Hervey * gst/law/alaw.c: * gst/law/mulaw.c: law: Raise rank of encoders to PRIMARY 2009-11-12 19:11:18 +0000 Bastien Nocera * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: Add user-id and user-pw properties So that one doesn't need to modify the URL to have access to authenticated RTSP streams. fixes #601728 2009-11-18 12:22:10 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: use acquired flag when checking valid state Use the acquired field of the ringbuffer in get_time to know when we are in an invalid state. We don't clear the rate flag when releasing the ringbuffer so this values is not usable. Avoids some error messages being posted because the pulseaudio connection is down. 2009-11-18 10:17:02 +0000 Tim-Philipp Müller * configure.ac: configure: bump core requirement to 0.10.25.1 as well Make implicit requirement explicit. 2009-11-18 12:53:44 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix bogus memory chunk size check 2009-11-18 12:01:52 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: implement some more callbacks Implement some more callbacks for debugging purposes. 2009-11-11 15:50:19 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: release lock before emiting signals Release the jbuf lock before emiting the request-pt-map signal to avoid deadlocks. We also need to catch the shutdown case when locking again. Fixes #593354 2009-11-11 11:59:16 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvdepay.h: rtp: add BroadcomVoice depayloader 2009-11-11 11:38:36 +0100 Wim Taymans * gst/rtp/gstrtpbvpay.c: rtpbvpay: add rfc reference 2009-11-11 11:37:07 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpbvpay.h: rtp: add BroadcomVoice payloader 2009-11-09 12:17:34 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: properly finish the ECMA array The ECMA array with the file index was missing a mandatory end marker. Fixes bug #601242. 2009-11-18 02:15:15 +0000 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: Use new still-frame API from gst-plugins-base 2009-11-18 02:14:46 +0000 Jan Schmidt * configure.ac: Bump gst-plugins-base requirement to 0.10.25.1 2009-11-17 17:59:13 -0800 Michael Smith * gst/qtdemux/qtdemux.c: qtdemux: identify IMA adpcm in qt properly. 2009-11-18 01:27:37 +0000 Jan Schmidt * configure.ac: * win32/common/config.h: Back to development -> 0.10.17.1 2009-11-17 01:53:08 +0000 Jan Schmidt * gst-plugins-good.doap: Add release 0.10.17 to the doap file === release 0.10.17 === 2009-11-17 01:25:30 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: Release 0.10.17 2009-11-17 00:18:22 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2009-11-13 02:07:25 +0000 Jan Schmidt * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: 0.10.16.3 pre-release 2009-11-10 11:52:24 +0100 Sebastian Dröge * sys/v4l2/gstv4l2object.c: v4l2: Make sure to initialize variables before using them 2009-11-09 20:06:03 +0000 Jan Schmidt * ChangeLog: * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: 0.10.16.2 pre-release 2009-11-09 15:20:00 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: free temporary buffer when changing state to NULL Free temporary allocations in the state change function and not only when the object is finalised. 2009-11-09 11:40:25 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: only allocate as much temporary memory as needed for indirect decoding When we can't decode directly into the output buffer, make our temp buffers only as big as needed instead of allocating for the worst case scenario (well, we still alloc more than strictly needed for some cases, but significantly less than before). 2009-11-05 23:46:58 +0000 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: printf format fix 2009-11-05 23:44:27 +0000 Tim-Philipp Müller * ext/raw1394/gst1394clock.c: * ext/raw1394/gsthdv1394src.c: raw1394: printf format fixes 2009-11-05 23:40:15 +0000 Tim-Philipp Müller * gst/equalizer/gstiirequalizer.c: equalizer: printf format fix 2009-11-04 22:19:58 -0500 Olivier Crête * gst/dtmf/Makefile.am: * gst/dtmf/gstdtmf.c: * gst/dtmf/gstdtmfdetect.c: * gst/dtmf/gstdtmfdetect.h: * gst/dtmf/tone_detect.c: * gst/dtmf/tone_detect.h: dtmfdetect: Add DTMF tone detector It looks at raw audio data and emits messages when DTMF is detected. The dtmf detector is the same Goertzel implementation used in FreeSwitch and Asterisk. It is in the public domain. 2009-11-05 12:13:44 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: do not write empty INFO list avoid writing an empty INFO list chunk, both because it is useless and because vlc refuses to play the resulting file. 2009-11-05 10:54:12 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: equalizer: Notify about band property changes caused by changing number of bands 2009-11-05 10:45:59 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer.h: * gst/equalizer/gstiirequalizernbands.c: equalizer: Make changes to band properties and the number of bands threadsafe 2009-11-05 10:30:46 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: equalizer: Fix stupid off by two bug 2009-11-05 08:18:05 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: equalizer: Add band property to select the band filter type This allows per band configuration of a peak, low shelf or high shelf filter, which can be very useful if the band frequencies and widths are manually configured. 2009-11-05 08:17:53 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: equalizer: Fix code style 2009-11-05 08:03:13 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: equalizer: Some cleanup 2009-11-04 22:21:35 -0500 Olivier Crête * gst/dtmf/gstdtmfsrc.c: dtmfsrc: Reject empty caps 2009-11-04 22:21:22 -0500 Olivier Crête * gst/dtmf/gstdtmfsrc.c: dtmfsrc: Use log level for repeated debug messages 2009-11-04 20:05:17 -0500 Olivier Crête * gst/dtmf/gstdtmfsrc.c: dtmfsrc: Allow for any samplerate 2009-10-07 09:31:19 -0400 Gabriel Millaire * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: celtpay/depay : change GST_DEBUG_OBJECT to GST_LOG_OBJECT in pay_handle_buffer and depay_process 2009-10-02 17:04:43 -0400 Gabriel Millaire * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltdepay.h: * gst/rtp/gstrtpceltpay.c: celtpay/depay: Negotiate parameters through caps celtdepay : added default framesize(480) channels(1) and clockrate(32000) depay_setcaps : now gets channels and framesize from string with default value depay_process : now adds timestamp to outbuf Added frame_size to GstRtpCeltDepay Changed some GST_DEBUG to GST_DEBUG_OBJECT or GST_LOG_OBJECT celtpay : getcaps : gets channel and framesize and sets caps Added frame-size to static caps for audio/x-celt 2009-11-04 15:58:34 +0000 Jan Schmidt * gst/deinterlace/Makefile.am: deinterlace: Pull in CFLAGS and LIBS flags from -base before core before system. 2009-10-15 16:33:24 +0100 Jan Schmidt * po/Makevars: po: Don't create backup .po files As well as preventing creation of useless backup files, it works around a bug in gettext 0.17 on OS/X 2009-11-04 16:47:42 +0100 Edward Hervey * gst/qtdemux/qtdemux_dump.c: qtdemux: init variables to make compiler on osx build bot happy 2009-11-03 16:04:37 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux_dump.c: qtdemux: init variables to make compiler on osx build bot happy 2009-11-03 17:35:15 +0200 Stefan Kost * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: don't allocate big arrays on the stack Add the arrays to the instance data and allocate on first use. 2009-11-01 15:57:44 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: remove pointless call to gst_element_no_more_pads() 2009-11-01 00:29:57 +0200 Stefan Kost * gst/level/gstlevel.c: level: fix decay to be smooth The length not having any fractional part as it was promoted to gdouble after dividing two guint64. 2009-11-01 00:29:24 +0200 Stefan Kost * gst/level/gstlevel.c: * gst/level/gstlevel.h: level: calculate the message-intervall when it changes 2009-11-01 00:14:08 +0200 Stefan Kost * gst/level/gstlevel.c: level: clocktime is a guint64, use right macro to init fields 2009-11-01 00:10:01 +0200 Stefan Kost * gst/level/gstlevel.c: level: use more g-style types 2009-10-30 09:27:59 +0100 Sebastian Dröge * configure.ac: * ext/pulse/pulsesink.c: pulsesink: Only set the volume on stream connection if pulse >= 0.9.20 is available In older versions the volume set during stream connection had no defined sematic and usually it was a relative volume. What was needed for our use case is an absolute volume though, otherwise the volume will be always decreased on stream connection if it's less than 100%. Since pulse 0.9.20 that volume is always an absolute volume if flat volumes are used and relative otherwise, which is the same as for pa_context_set_sink_input_volume(). Relevant pulse changesets: http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=f27a50691c8fe45bac7dd6b21fac91a359def3a1 http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=2501687579e359d5032a4d165b2ffc8f5b1b8ba6 2009-10-27 18:07:18 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: use segment_full when we can Use segment_full so that we can pass the applied rate to the segment values. We will change the applied rate when we implement skip mode. 2009-10-18 00:16:06 +0100 Robert Swain * gst/wavenc/gstwavenc.c: wavenc: Fix buffer offset by moving length incrementation 2009-10-23 18:31:14 -0700 Michael Smith * sys/osxvideo/osxvideosink.m: osxvideosink: Create the video NSView in READY->PAUSED rather than NULL->READY 2009-10-23 18:28:22 -0700 Michael Smith * sys/osxvideo/Makefile.am: osxvideo: explicitly link to GST_LIBS 2009-10-23 18:09:43 -0700 Michael Smith * gst/avi/Makefile.am: * gst/matroska/Makefile.am: * gst/wavparse/Makefile.am: Add dependencies of gstriff to things that link to gstriff, needed on Win32. 2009-10-23 17:25:17 -0700 Michael Smith * tests/examples/rtp/client-PCMA.c: * tests/examples/rtp/server-alsasrc-PCMA.c: rtp examples: remove executable bits from C files. 2009-10-23 11:21:44 +0100 Tim-Philipp Müller * tests/check/elements/rtpjitterbuffer.c: tests: disable all jitterbuffer tests for now Since even the one enabled seems to fail. 2009-10-22 13:39:58 +0300 Stefan Kost * tests/check/elements/rtpjitterbuffer.c: tests: also include the new test for prev commit 2009-10-22 13:19:07 +0300 Stefan Kost * gst/rtpmanager/gstrtpjitterbuffer.c: * tests/check/Makefile.am: * tests/check/elements/.gitignore: tests: add a jitterbuffer test Tests pushing a few buffers in various order and asserting the order sent by the jitterbuffer. Contains two disabled tests that need more work. 2009-10-22 12:30:14 +0200 Sebastian Dröge * gst/matroska/matroska-mux.c: matroskamux: Dirac "muxing" units end on EOS too A Dirac muxing unit are all non-picture, non-end-of-sequence packets up to and including the first picture or eos packet. See http://www.diracvideo.org/wiki/index.php/ContainerFormatMappingGuidelines 2009-10-22 02:09:08 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: fix compilation with debugging disabled total_idx is always evaluated. 2009-10-19 21:59:46 +0300 Priit Laes * ext/libcaca/gstcacasink.h: cacasink: minor cleanups for header. Use G_BEGIN_DECLS macros, remove unused variables and fix typo. See #599018. 2009-10-19 21:59:23 +0300 Priit Laes * ext/libcaca/gstcacasink.c: cacasink: exit properly when invalid driver has been selected. See #599018. 2009-10-20 18:23:28 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Stop scanning at the last entry... and not the one before :) This ensures we actually push out everything 2009-10-20 17:20:55 +0200 Andy Wingo qtdemux: unpack more information into image/x-j2c caps * gst/qtdemux/qtdemux_fourcc.h: Add new fourccs for use by the mj2 unpacker. * gst/qtdemux/qtdemux.c (qtdemux_parse_trak): Unpack JPEG2000 component mapping and channel definitions from the jp2h header. Will add component-map and channel-definitions elements to the caps if the component maps or channel definitions are nonstandard, where standard order means RGB, 444 packed YUV, or greyscale, with no alpha channel. Fixes #598915. 2009-10-20 17:33:41 +0300 Stefan Kost * tests/check/elements/deinterleave.c: tests: include stdio.h for sscanf 2009-10-19 15:21:57 +0100 Bastien Nocera * ext/pulse/pulsesink.c: Fix the StreamVolume interface not being advertised gst_pulsesink_interface_supported() was missing a check for it. https://bugzilla.gnome.org/show_bug.cgi?id=598933 2009-10-16 21:14:14 +0300 Stefan Kost * gst/level/gstlevel.c: level: code cleanup Use gdouble instead of double. Calculate falloff_time once instead of twice. 2009-10-18 15:52:02 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: MEMDUMP the junk blobs It will only actually pull the junk blobs from upstream if the memdump level is activated 2009-10-18 15:51:34 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Some avi files have INFO lists in the headers. 2009-10-18 16:02:01 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Don't seek on empty streams 2009-10-18 15:50:39 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Ensure _calculate_durations_from_index only uses valid streams 2009-10-18 15:49:29 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Only call convert function if we have strf.auds 2009-10-18 15:48:06 +0200 Edward Hervey * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Use first indexed stream for seeking. In the future, main_stream can be adjusted to contain the optimal stream as mentionned in the FIXME line 3440 2009-10-18 15:46:48 +0200 Edward Hervey * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Only expose streams that actually have something in it. This guarantees that in pull-mode, all streams have a valid index to work with. 2009-10-18 15:40:37 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Properly mark presence of index. Instead of blindly saying we have an index, only do so if we have a non-empty index. 2009-10-17 02:18:53 +0200 Lennart Poettering * ext/pulse/pulsesink.c: pulse: never apply volume more than once Generally decisions on the volume of the stream should be done inside of PA, not inside of Gst. Only PA knows how volumes translate between devices and s on. This patch makes sure that all volumes set via the volume property are only applied *once* to the underlying stream. After applying them the client side will not store them anymore. This should make sure that really only user-triggered volume changes are forwarded to server, but the client never tries to save/restore the volume internally. Fixes bug #595231. 2009-10-17 08:55:16 +0200 Sebastian Dröge * ext/pulse/plugin.c: pulsesink: Initialize gettext for the translated strings in plugin_init() 2009-10-17 00:10:30 +0200 Lennart Poettering * ext/pulse/pulsesink.c: pulse: use 'performer' as a fallback for 'artist' tag 2009-10-17 00:09:36 +0200 Lennart Poettering * ext/pulse/pulsesink.c: * po/POTFILES.in: pulse: when constructing a stream title from tag data make sure it is translatable 2009-10-17 00:06:15 +0200 Lennart Poettering * ext/pulse/pulsemixerctrl.c: pulse: loop while connecting to server pthread does not guarantee that there are no spurious condition variable wakeups, neither does pa_threaded_mainloop_xxx() which is a wrapper around it. So we need to loop around the _wait() function to make sure we get the right wakeup. Also, unify the order of the wait loops across the file. 2009-10-17 00:05:10 +0200 Lennart Poettering * ext/pulse/pulsemixerctrl.c: * ext/pulse/pulseprobe.c: pulse: mainloop creation can fail too, so handle that 2009-10-17 00:03:06 +0200 Lennart Poettering * ext/pulse/pulsemixerctrl.c: pulse: adjust CHECK_DEAD_GOTO macro to glib style 2009-10-16 17:28:42 +0200 Lennart Poettering * ext/pulse/pulsemixerctrl.c: * ext/pulse/pulsemixerctrl.h: * ext/pulse/pulseprobe.c: * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.h: pulse: make a few things smaller by making them bitfields 2009-10-16 17:26:41 +0200 Lennart Poettering * configure.ac: pulse: bump minimum libpulse version to 0.9.10 Older versions than 0.9.10 are really really old and buggy. Drop compatibility with them. Nobody should run anything that old. Also see: https://bugzilla.gnome.org/show_bug.cgi?id=595029 2009-10-16 18:18:31 +0200 Mark Nauwelaerts * gst/debugutils/gstdebug.c: debugutils: register pushfilesrc element 2009-10-16 17:28:09 +0200 Mark Nauwelaerts * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: support (some) VBR audio muxing AVI format can handle VBR audio provided audio chunks are of fixed duration (cfr fixed duration video frames). Apply this approach to (always) parsed raw AAC and (if parsed) to MPEG-1/2 audio. See #368681. 2009-10-16 13:41:45 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix branch hints Remove inappropriate branching hints and add some new ones. 2009-10-16 12:33:04 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix regression in indirect decode path Revert variable name back to what it was before the G_LIKELY was added (in commit 69c24fb9). The code works better that way. 2009-10-16 02:47:38 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix regression with certain formats Fix regression introduced by previous commit (#598517). 2009-10-15 19:49:55 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: don't use decompress structure members we shouldn't be using 2009-10-14 17:53:52 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.h: jpegdec: remove some unused members from jpegdec instance structure 2009-10-16 11:53:38 +0300 Stefan Kost * gst/rtpmanager/Makefile.am: * gst/udp/Makefile.am: build: use gst-glib-gen.mak to fix the glib build rules. The build rules in glib-gen.mak were using pattern rules in a non save way. 2009-10-16 10:15:35 +0300 Stefan Kost * common: Automatic update of common submodule From 85d1530 to 0702fe1 2009-10-15 21:04:02 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: adjust flow return aggregation to updated loop_data In particular, each stream is now treated separately, and one stream's EOS should not lead to overall EOS. 2009-10-15 11:52:35 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: check some more atom sizes prior to parsing 2009-10-15 13:19:13 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtsp: handle events in TCP mode We need to handle events in TCP mode so that we can reply to the LATENCY event with TRUE. 2009-10-15 11:24:45 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: add missing argument in debug message 2009-10-14 18:58:06 +0200 Marvin Schmidt * tests/check/elements/flvmux.c: flvmux: Use loop test to prevent timeout on slow machines Partially fixes bug #597739. 2009-10-14 16:15:48 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: forward events into the rtpbin Only catch the SEEK event on the srcpad and let other events enter the rtpbin. 2009-10-14 11:33:24 -0300 Thiago Santos * gst/matroska/matroska-demux.c: matroskademux: Fix late tags finding Use the correct taglist variable when notifying of late tags. 2009-10-14 13:09:03 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: use GstIndex for (limited) seeking in push mode ... but disable this for now. Although it basically works fine, user experience might be shaky (depending on taste), since there is no keyframe info in push mode. 2009-10-14 13:08:47 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: add GstIndex support 2009-10-14 11:55:33 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: also determine duration in push mode 2009-10-14 11:54:44 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: add GstIndex support 2009-10-14 07:38:26 -0300 Thiago Santos * sys/v4l2/gstv4l2src.c: v4l2src: Set duration on buffers Use framerate to estimate duration of buffers. Fixes #590362 2009-10-14 12:28:55 +0200 Håvard Graff * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: only forward the lost-event to the last seen pt-number forward all events on all pads except for the PacketLost event, which we want to forward to the last seen pt pad. Fixes #598377 2009-10-06 22:28:50 +0300 René Stadler * ext/pulse/pulsesink.c: pulsesink: set desired minreq value to segsize/latency-time If we let the daemon decide freely by passing -1, we end up always getting 20ms. We want to set this value because in some cases we want to select a higher latency-time in order to save power. Fixes #597601 2009-10-14 10:41:21 +0200 Edward Hervey * common: Automatic update of common submodule From a3e3ce4 to 85d1530 2009-10-13 18:33:34 +0200 Edward Hervey * tests/check/pipelines/flacdec.c: tests/pipeline/flac: Fix build on macosx 10.5 2009-10-13 18:19:32 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: demote some warnings to debug 2009-10-13 17:47:40 +0200 Wim Taymans * gst/avi/avi-ids.h: avi: add new avi flag we might want to use 2009-10-13 17:46:22 +0200 Wim Taymans * gst/avi/gstavimux.c: avimux: calculate suggested buffer size Calculate the suggested buffer size based on the largest chunk in the file. See #597847 2009-10-13 17:45:14 +0200 Wim Taymans * gst/avi/gstavimux.c: avimux: add jpeg2000 to allowed caps 2009-10-13 17:41:13 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: add debug for the superindex offsets 2009-10-13 16:02:37 +0100 Jan Schmidt * gst/qtdemux/qtdemux.c: qtdemux: Fix uninitialized variable warning Fix another bogus may-be-used-uninitialized warning in qtdemux 2009-10-13 13:08:33 +0200 Wim Taymans * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avi: lower max file size Make a constant of the max file size and lower the value to what ffmpeg does, hopefully improving compatibility with windows media player. See #597847 2009-10-13 01:02:15 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix bogus warning about discont flag on first buffer The very first buffer should always have the DISCONT flag set, no need to warn about that. Only warn if we get a DISCONT buffer in non-packetised mode and we already have some data. 2009-10-13 00:41:57 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix crash for unusual vertical chroma subsampling factors Fixes #597351. 2009-10-13 00:12:42 +0100 Jan Schmidt * gst/qtdemux/qtdemux.c: qtdemux: Fix uninitialized variable warnings The gcc on the OS/X buildbot complains about these variables not being initialized, even though they can't possibly actually be used uninitialized. 2009-10-11 11:35:23 +0200 Josep Torra * gst/dtmf/gstrtpdtmfdepay.c: dtmf: fix warnings in macosx snow leopard 2009-10-10 00:37:08 +0200 Josep Torra * ext/jpeg/gstjpegdec.c: jpegdec: fixes warning building in snow leopard 2009-10-09 17:12:46 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: also consider Quicktime text subtitles 2009-10-09 17:02:57 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: provide language tag for stream 2009-10-09 16:30:57 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: refactor common parts in track parsing 2009-10-09 16:21:03 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: refactor buffer processing and sending ... so it can be used in both pull and push based mode. 2009-10-08 13:39:25 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: extract palette data for dvd subpicture streams ... and send it downstream using custom dvd event 2009-10-07 14:03:17 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: support 3GPP timed text subtitles In particular, also make subtitle support less subp(icture)-centric. 2009-10-07 16:15:55 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: NULL is not a valid taglist 2009-09-23 17:20:25 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: recognize some more encypted track cases 2009-10-09 15:59:25 +0200 Josep Torra * gst/id3demux/id3tags.c: id3: fixes warnings building on macosx Another round on the formating of that debug line. 2009-10-09 14:44:02 +0300 Stefan Kost * gst/id3demux/id3tags.c: id3: cast pointer math results to glong 2009-10-09 14:37:32 +0300 Stefan Kost * ext/flac/gstflacdec.c: flac: apparently on some platforms a FLAC__uint64!=guint64 2009-10-09 14:21:09 +0300 Stefan Kost * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtpvrawpay.c: buikd: explicitely cast, to tell some compilers that this is not long int 2009-10-09 13:38:17 +0300 Stefan Kost * ext/flac/gstflacdec.c: * gst/id3demux/id3tags.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtpvrawpay.c: build: don't cast, but use the right format specified instead This correct some of the previous macos fixes. 2009-10-09 12:40:47 +0200 Josep Torra * ext/dv/gstdvdemux.c: dv: fix warnings on macosx 2009-10-09 12:25:19 +0200 Josep Torra * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: flac: fix warnings on macosx 2009-10-09 12:19:35 +0200 Josep Torra * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: annodex: fix warnings in macosx 2009-10-09 12:14:22 +0200 Josep Torra * sys/osxvideo/cocoawindow.m: osxvideo: fix a warning doing a cast 2009-10-09 12:11:12 +0200 Josep Torra * sys/osxaudio/gstosxringbuffer.c: osxaudio: fix warnings on macosx 2009-10-09 12:01:10 +0200 Josep Torra * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: fix warning on macosx 2009-10-09 11:57:59 +0200 Josep Torra * gst/rtp/gstrtph263pay.c: rtph263pay: fix warning on macosx 2009-10-09 11:54:03 +0200 Josep Torra * gst/qtdemux/qtdemux.c: qtdemux: fix warnings building on macosx 2009-10-09 11:42:36 +0200 Josep Torra * gst/id3demux/id3tags.c: id3demux: fix printf warnings on macosx 2009-10-09 11:30:00 +0200 Josep Torra * gst/avi/gstavidemux.c: avidemux: fix warning in macosx making the format portable 2009-10-09 10:51:29 +0200 Josep Torra * gst/audiofx/audiofxbasefirfilter.c: audiofx: use G_GUINT64_FORMAT to fix warnings on OSX 2009-10-09 10:11:38 +0200 Josep Torra * sys/osxaudio/gstosxringbuffer.c: osxaudio: Fixes build on macosx snow leopard. 2009-10-09 11:34:16 +0200 Pau Garcia i Quiles * sys/v4l2/gstv4l2object.h: v4l2: Include sys/ioctl.h for the V4L ioctl requests Old videodevice2.h kernel headers used ioctl stuff without including ioctl.h, making compilation fail on older systems. Note: Including ioctl.h here is only a workaround for old kernel headers, should be removed once everybody has new enough headers. Fixes bug #597867. 2009-10-09 00:14:07 +0100 Jan Schmidt * configure.ac: * tests/check/elements/level.c: check: Make the level unit test succeed on Solaris 10 Add a configure check for functional isinf() and fpclass(), and use fpclass() where possible when isinf() is not available. 2009-05-16 13:52:50 +0300 René Stadler * gst/matroska/matroska-demux.c: matroskademux: fix strstr() usage on possibly unterminated string 2009-10-08 16:16:14 +0100 Jan Schmidt * tests/check/Makefile.am: * tests/check/elements/level.c: check: Link against LIBM and include math.h for isinf() 2009-10-07 21:51:38 +0100 Sjoerd Simons * sys/oss/gstossaudio.c: oss: Downgrade the rank of osssrc to SECONDARY which is the same rank as osssink has. Fixes bug #597730. 2009-10-08 10:59:53 +0100 Jan Schmidt * common: Automatic update of common submodule From 19fa4f3 to a3e3ce4 2009-10-08 10:20:09 +0100 Jan Schmidt * gst/avi/gstavidemux.c: * gst/wavparse/gstwavparse.c: avi/wav: Fix some compiler warnings about incompatible pointers. 2009-10-05 17:36:55 +0100 Jan Schmidt * gst/multifile/gstmultifile.c: multifile: Fix plugin description 2009-10-07 14:03:20 +0300 Stefan Kost * ext/annodex/gstcmmlutils.c: * ext/jpeg/gstjpegdec.h: * ext/jpeg/gstjpegenc.h: * gst/apetag/gstapedemux.c: * gst/debugutils/tests.c: * gst/id3demux/id3v2frames.c: * gst/qtdemux/qtdemux.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtsp/gstrtpdec.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: * tests/examples/spectrum/spectrum-example.c: build: fprintf, sprintf, sscanf need stdio.h 2009-10-07 00:33:49 +0300 Stefan Kost * gst/equalizer/gstiirequalizer.c: equalizer: use shelfing filters for first and last band Refactor the filter setup. Add two new filters with shelf characteristics for first and last band. Change gain calculation as recommended in the quoted document (no qrt needed). Rename variables to match the formulas in the document. 2009-10-02 23:51:29 +0300 René Stadler * ext/pulse/pulsesrc.c: pulsesrc: guard fragment size with a lower limit based on latency-time In case that the pulse daemon runs the source device at a relatively low fixed fragment size compared to the requested latency-time, configure the ring buffer segsize to the largest integer multiple of the fragment size that is still smaller than or equal to the requested latency-time. Fixes bug #597463. 2009-10-06 17:40:47 +0300 Stefan Kost * ext/jpeg/gstjpegdec.c: jpegdec: comment/logging cleanups and more branch guides 2009-10-05 22:43:11 +0300 Stefan Kost * gst/equalizer/gstiirequalizer.c: equalizer: fix filter history usage. Fixes #597397 The process functions where overwriting the history for each channel. Also pull some static things out of the inner loop. 2009-10-05 16:07:24 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: use locking around the sessions 2009-10-05 11:46:08 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: make sure compatible brands buffer exists before dereferencing it 2009-10-04 21:59:24 +0200 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: fix printf warnings on OSX Cast variables passed to printf to avoid warnings about incorrect formats (most likely caused by sizeof returning a size_t). Fixes #597348. 2009-10-02 00:23:34 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: remove internal genre table No need to maintain our own genre table in qtdemux. The genres are identical to the ID3 genres, so we can just use libgsttag's gst_tag_id3_genre_get() to look them up. 2009-10-03 17:18:28 +0200 Robert Swain * gst/avi/gstavidemux.c: Fix printf formats to avoid warnings in avidemux. Fixes #597214 https://bugzilla.gnome.org/show_bug.cgi?id=597214 2009-10-03 09:52:57 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Change one GST_WARNING to a GST_DEBUG 2009-10-02 14:37:54 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: flvdemux: If there's no audio stream after 6 seconds of video signal no-more-pads ...and the other way around. Also ignore any audio/video streams that appear after no-more-pads. Fixes bug #597091. 2009-10-02 14:37:40 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: flvdemux: Make sure to only signal no-more-pads a single time 2009-10-02 22:55:45 +0300 René Stadler * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: rename pa_buffer_attr variables Makes it much easier to see what is going on and is a lot less error prone. 2009-10-02 18:25:16 +0300 Stefan Kost * gst/rtp/gstrtpjpegdepay.c: rtp: add missing include to fix the build 2009-10-02 13:15:59 +0300 Stefan Kost * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: videofilter: add G_OBJECT_WARN_INVALID_PROPERTY_ID to property setter 2009-10-02 13:10:44 +0300 Stefan Kost * gst/level/gstlevel.c: level: don't give wrong number of fields in the message docs 2009-10-01 12:52:40 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: cache latency in nanoseconds Cache the latency in nanoseconds units to avoid having to convert the milliseconds value to nanoseconds all the time. 2009-10-01 12:12:09 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: handle -1 input timestamps Don't try to check a -1 timestamp against the max delay. 2009-10-01 10:54:55 +0300 Stefan Kost * gst/avi/gstavidemux.c: avi: don't misues perf-category and remove unused ext category The performance category is meant to be used to audit codepaths that lead to bad performance (e.g. copies, conversion that can be avoided). Remove the event category which is not used. 2009-09-16 14:23:24 -0400 Olivier Crête * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: rtpg729pay/depay: Demote per-buffer debug messages to log level 2009-09-16 14:16:27 -0400 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Don't leak incoming buffers after subbuffering them 2009-09-16 13:57:05 -0400 Olivier Crête * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: rtpg729pay/depay: Add debug categories 2009-09-16 13:55:19 -0400 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Remove long unneeded define replacement 2009-09-30 18:06:07 +0100 Christian F.K. Schaller * ext/dv/Makefile.am: Update makefile with missing header file 2009-09-30 18:45:17 +0200 Sebastian Dröge * tests/examples/rtp/client-H263p-AMR.sh: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: rtp: Use autoaudio{sink,src} instead of alsa in the examples 2009-09-29 17:51:04 +0300 Stefan Kost * ext/jpeg/gstjpegdec.c: jpegdec: don't leak output buffers on decoding errors The setjmp handles libjpeg error. Free the outputbffer if we don't need it. 2009-09-29 00:01:59 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix 'unused variable' compiler warning when compiling with GST_DISABLE_GST_DEBUG 2009-09-23 14:25:08 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: small cleanups 2009-09-23 13:57:02 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: fix timestamping in some audio streams For vbr audio streams we need to use the number of blocks to calculate the timestamps. When the allocation of additional index memory fails, don't throw away what we had before. Various cleanups. 2009-09-23 12:56:07 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: add support for ODML indexes again 2009-09-22 22:12:58 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avi: implement index scanning Implement scanning of the file when we can parse the index. Some refactoring of common code. Cleanups and comments. Remove some reimplemented code. Remove index massage code and put a FIXME where we should do something equivalent later. 2009-09-22 18:18:20 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: fix reverse playback 2009-09-22 17:42:48 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: fix prev keyframe search and cleanups 2009-09-22 14:51:30 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: remove code that got converted 2009-09-22 14:44:42 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avi: more cleanups Remove some duplicate counters. Be smarter when updateing the current the timestamp and offset in the stream because we can reuse previously calculated values when simply go forward one step. Correctly set metadata on outgoing buffers. 2009-09-22 12:35:30 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: small cleanups 2009-09-22 01:28:54 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: fix read offset and cleanups 2009-09-21 18:04:25 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avi: rewrite index playback disable code, start on reimplementing loop based operation. Rewrite the index handling so that all streams use their own index for decoding media. 2009-09-21 15:35:55 +0200 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: add new index parsing code Add a new function and datastructure to parse and hold the index entries on a per stream base. Also avoid doing too much work trying to figure out the timestamps and durations as we can trivially do that later. Less information in the entries makes them 2 times smaller and not doing too much work makes this code about 12 times faster than the regular case. Hook in the new function alongside the existing function for comparison until the rest of the code is updated to handle the new index datastructure. 2009-09-28 16:29:45 +0300 Stefan Kost * ext/jpeg/gstjpegdec.c: jpeg: handle more libjpeg return values, add some more branch hints Also remove unused size variable in _chain(). 2009-09-25 19:21:32 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: some optional QT specified stsd MPEG-4 atoms also apply to H264 Fixes #596319. 2009-09-25 16:40:31 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: only send tag events downstream after newsegment 2009-09-25 14:14:03 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: if transport protocol unsupported, try another one Also change error message to more accurately reflect cases in which it can occur. 2009-09-25 11:54:06 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: add durations modulo 1<<32 For calculating the durations of each sample, we are supposed to add each duration modulo 1<<32 so make the elapsed time counter a uint32. Fixes #595942 2009-09-24 20:38:54 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: small cleanup 2009-09-24 19:33:39 +0100 Tim-Philipp Müller * gst/qtdemux/qtatomparser.h: qtdemux: don't use core API that doesn't exist yet There's no gst_byte_reader_has_remaining() yet. Fixes build. 2009-09-24 13:20:50 +0100 Tim-Philipp Müller * gst/qtdemux/qtatomparser.h: qtdemux: map some atomparser functions to their new bytereader equivalents Now that GstByteReader has unchecked and inlined variants as well, map atomparser functions to their respective bytereader equivalents. 2009-08-25 12:11:28 +0100 Tim-Philipp Müller * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: qtdemux: add qt_atom_parser_has_chunks() and fix indentation 2009-08-20 18:21:59 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: bail out instead of trying to alloc silly index sizes If it looks like we would be allocating a silly size for our sample index, just bail out instead of trying to allocate it. Helps with broken or fuzzed files where we might end up trying to malloc a couple of hundred MBs otherwise. 2009-08-20 16:47:25 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: error out correctly if we don't even have enough bytes for an atom header 2009-08-20 15:39:00 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: init fourcc to 0 as well to avoid invalid reads when printf'ing error message 2009-08-20 01:39:17 +0100 Tim-Philipp Müller * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: qtdemux: add qt_atom_parse_has_remaining() to avoid overflows with _get_remaining() 2009-08-20 01:21:04 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: use GstByteReader when parsing tkhd atom 2009-08-19 19:13:38 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: use unsigned ints for node length and do more sanity checking of the atom length 2009-08-19 01:36:33 +0100 Tim-Philipp Müller * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux_dump.c: * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_types.h: qtdemux: use GstByteReader for atom dumping and fix a few bugs 2009-08-21 14:21:08 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: move stco, stts, stss and stps atom parsing over to GstByteReader Make sure we don't read beyond the atom boundary. Note that the code behaves slightly differently in the corner case where there is not enough atom data for the specified number of samples (n_samples_time) in the atom, but still enough data to fill the pre-allocated index of n_samples entries: before we would just stop parsing the stts data and continue, whereas now we will likely error out. This should not be a problem in practice though. We could maintain the old behaviour by doing reads with a size check inside the loop if needed. 2009-06-30 19:51:15 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: use bytereader to parse stsz and stsc atoms Use GstByteReader to parse stsz and stsc chunks, and check size of available data before parsing it, instead of blindly assuming there will be enough data. Fixes crashes with some fuzzed/broken files. 2009-08-15 20:38:40 +0100 Tim-Philipp Müller * gst/qtdemux/qtatomparser.h: qtdemux: add qt_atom_parser_get_offset() and optimise _peek_sub() 2009-07-01 13:49:57 +0100 Tim-Philipp Müller * gst/qtdemux/Makefile.am: * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux.c: qtdemux: add QtAtomParser, an inlined GstByteReader variant 2009-09-23 17:19:34 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: use proper order for no-more-pads and newsegment and tag sending 2009-09-23 09:50:37 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: sprinkle a few branch prediction macros 2009-09-22 15:03:20 +0200 Alessandro Decina * ext/jpeg/gstjpegdec.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvparse.c: * gst/id3demux/id3v2frames.c: Fix compile warnings with gcc 4.0.1. 2009-09-22 11:48:50 +0100 Jan Schmidt * gst/matroska/matroska-mux.c: matroskamux: Don't get stuck in an infinite loop with Dirac At the end, Dirac streams have an EOS packet with 0 length. Don't ever sit in an infinite loop when processing one. Allows muxing Dirac into mkv to complete successfully. 2009-09-22 11:03:46 +0100 Tim-Philipp Müller * .gitignore: Update .gitignore 2009-09-22 11:02:02 +0100 Tim-Philipp Müller * gst/videomixer/Makefile.am: videomixer: fix up Makefile some more Remove CFLAGS from LIBADD and make order of the various CFLAGS and LIBS at least consistent with each other. 2009-09-22 08:02:48 +0200 Brian Cameron * gst/videomixer/Makefile.am: videomixer: Add $(GST_PLUGINS_BASE_LIBS) to LDFLAGS for linking libgstvideo Fixes bug #595897. 2009-09-21 18:09:12 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: fix timestamps in push mode 2009-09-18 17:26:42 +0300 Stefan Kost * ext/jpeg/gstjpegdec.c: jpegdec: add a G_UNLIKELY and put perf-cat log to code path that copies 2009-09-21 12:32:51 +0200 Wim Taymans * gst/avi/gstavidemux.c: avi: add some performance measurements Measure the performance of various index and header parsing steps to the PERFORMANCE debug category. 2009-09-18 11:53:12 +0200 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: allow for unknown varying number of frames per buffer In particular, this caters for RTP payloads with multiple frames per packet. 2009-09-18 11:45:06 +0200 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: use correct sample size in conversions 2009-09-18 11:43:46 +0200 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: fix buffer time and duration for multiple frames per packet 2009-09-18 14:22:02 +0300 Stefan Kost * gst/avi/gstavidemux.c: avidemux: some logging cleanup to help understanding the index parsing overhead 2009-09-16 13:28:27 -0700 David Schleef * sys/osxaudio/Makefile.am: osxaudio: link against GST_BASE_LIBS 2009-09-15 17:24:24 -0400 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Fix adapter leak The adapter would be leaked if it was empty and the data could be pushed out directly. 2009-09-15 10:04:30 +0200 Sebastian Dröge * ext/pulse/pulsesrc.c: pulsesrc: Don't dereference NULL pointers pa_stream_get_timing_info() can return NULL. Fixes bug #595220. 2009-09-15 10:01:54 +0200 David Henningsson * ext/pulse/pulsesink.c: pulsesink: Don't dereference NULL pointers pa_stream_get_timing_info() can return NULL. Fixes bug #595220. 2009-09-14 16:05:30 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: handle stream events Handle stream events and request a PAUSE/PLAY state change from the application when we receive a CORK/UNCORK event. 2009-09-13 12:30:34 -0700 David Schleef * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: Add next-file property Add a property to allow control over what event causes a file to finish being written and a new file start. The default is the same as before -- each buffer causes a new file to be written. Added is a case where buffers are written to the same file until a discontinuity in the stream. 2009-09-13 15:55:02 -0700 David Schleef * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Use values from decoder structure directly Don't store the same values in the GstDvDemux. This fixes a bug where dvdemux would detect a stream as PAL instead of NTSC, and silently parse it wrong. 2009-09-13 12:20:23 -0700 David Schleef * ext/dv/Makefile.am: * ext/dv/gstsmptetimecode.c: * ext/dv/gstsmptetimecode.h: * ext/dv/smpte_test.c: dvdemux: Add code to parse SMPTE time codes Code to convert time codes to/from timestamps and frame numbers. 2009-09-13 12:01:27 -0700 David Schleef * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Fix detection of new media There are 5 or 6 AAUX source control packs in a frame, and any of them could have REC_ST cleared, indicating a recording start point. libdv only checks the first. 2009-09-12 19:25:36 +0200 Edward Hervey * ext/dv/gstdvdemux.c: dvdemux: Set DISCONT flag on buffers when REC_ST flag is set. Also add a few branch prediction macros 2009-09-12 00:13:04 +0100 Jan Schmidt * tests/check/elements/souphttpsrc.c: * tests/check/elements/y4menc.c: check: Fix a couple of tests. The souphttpsrc test wasn't compiling. The soup-misc.h header is needed for soup_ssl_supported. Fix the y4menc test to use a 'progressive' header for the test data now that the element outputs correct interlacing info. 2009-09-11 13:32:39 -0700 Michael Smith * gst/wavparse/gstwavparse.c: wavparse: treat a zero-sized data chunk as extending to the end of the file. This fixes playback of some files that don't have a valid data chunk length, apparently some program creates these. 2009-09-11 22:24:47 +0300 Stefan Kost * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2src: add a function pointer for get_frame function and optimize a bit Use a function-pointer for mmap/read, as this can't change during capture. Also sprinkle a few G_LIKELY/UNLIKELY to improve the error-less code path. 2009-09-11 22:15:01 +0300 Stefan Kost * sys/v4l2/gstv4l2.c: * sys/v4l2/v4l2src_calls.c: v4l2: log buffer copies on queue underrun in perf category v4l2src has a slow path where it does buffer-copies when it runs out of queued buffers. Log this to performance category to help monitoring it. 2009-09-11 15:14:15 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: pulsesink: Implement GstStreamVolume interface 2009-09-11 16:09:40 +0200 Sebastian Dröge * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Implement mute property 2009-09-11 13:33:31 +0200 Wim Taymans * ext/gdk_pixbuf/gstgdkpixbufsink.c: gdkpixbufsink: fix docs refering to send-messages 2009-09-11 13:28:35 +0200 Wim Taymans * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: spectrum: add post-messages property Add a post-messages property and deprecate the less descriptive message property. 2009-09-11 13:20:06 +0200 Wim Taymans * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/gstgdkpixbufsink.h: pixbufsink: add post-messages property Add post-messages and deprecate send-messages as the former is more descriptive of what actually happens. 2009-09-11 13:12:54 +0200 Wim Taymans * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: rename silent to post-messages Use the post-messages property name instead of silent as it is more descriptive. 2009-09-11 12:16:18 +0200 Wim Taymans * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: post messages for each buffer Add a silent property that can be set to FALSE to post messages on the bus for each written file. Do some more cleanups. Add some docs. Fixes #594663 2009-09-09 18:13:29 -0400 Olivier Crête * gst/rtp/gstrtph263pay.c: rtph263pay: Allocate Boundry structs on the stack instead of the heap to avoid leaks Fixes bug #594691. 2009-09-10 10:28:48 +0300 Stefan Kost * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: docs: fix gtk-doc warnings 2009-09-10 10:26:23 +0300 Stefan Kost * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: docs: fix gtk-doc warnings 2009-09-09 17:51:19 -0700 David Schleef * ext/raw1394/Makefile.am: * ext/raw1394/gst1394clock.c: * ext/raw1394/gst1394clock.h: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gstdv1394src.h: dv1394src: Add a clock based on isochronous cycle counter Partial fix for #169383. 2009-09-09 16:02:03 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Fix AYUV->I420 conversion For this fix the averaging of the chroma values. It should't be (a/2 + b)/2 but just (a + b)/2. Fixes bug #594599. 2009-09-09 16:25:06 +0200 Wim Taymans * configure.ac: * ext/pulse/pulsesink.c: pulsesink: remove ringbuffer reset compensation Remove the code to deal with a ringbuffer reset as this code is now in the base class. Bump the -base requirement as we need the new baseaudiosink code to function properly. 2009-09-09 16:24:49 +0200 Wim Taymans * ext/pulse/pulsesink.h: pulsesink: whitespace fixes 2009-09-09 10:27:55 +0200 Wim Taymans * sys/v4l2/gstv4l2colorbalance.h: whitespace fixes 2009-09-08 19:34:09 +0200 Wim Taymans * ext/pulse/pulsemixer.c: * ext/pulse/pulsemixerctrl.c: * ext/pulse/pulseprobe.c: pulse: small cleanups Add some debug info Fix the state changes 2009-09-08 18:29:35 +0200 Marc-André Lureau * gst/multipart/multipartmux.c: multipartmux: mark data buffer as delta-unit So that multifdsink always start sending header buffer first Fixes #594520 2009-09-08 17:37:15 +0200 Marc Leeman * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: add ignore-pt parameter Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux. Fixes #594490 2009-09-04 13:51:37 +0200 Marvin Schmidt * tests/check/elements/souphttpsrc.c: checks: only run HTTPS test if libsoup has SSL support 2009-09-08 13:59:56 +0200 Håvard Graff * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: propagate payload-type-change signal from demuxer fixes #594254 2009-08-31 18:46:25 +0200 Havard Graff * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: change severity of clock-rate change debug Make log GST_DEBUG under normal circumstances, GST_WARNING otherwise. Fixes #594253 2009-09-08 13:39:31 +0200 Håvard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: avoid throwing reordered buffers with same timestamps When we receive a reordered packet with the same timestamp as the previous one (which can happen for fragmented packets) don't consider the packet as lost but instead wait for the reordered packet to arrive. Switch the warning-level, so that a reordering does not get a warning, only an actual produced lost-packet. Fixes #594251 2009-08-31 21:16:54 +0200 Havard Graff * gst/rtp/gstrtpjpegdepay.c: rtpjpegdepay: add missing math.h include Fixes #594247 2009-09-08 13:30:29 +0200 Arnout Vandecappelle * gst/rtsp/gstrtspsrc.c: rtspsrc: fix memory leak In gst_rtspsrc_parse_digest_challenge(), rtspsrc does a g_strndup of the auth header items and then passes them to gst_rtsp_connection_set_auth_param() without freeing. Fixes #594133 2009-09-08 13:18:29 +0200 Stig Sandnes * gst/rtpmanager/gstrtpbin.c: rtpbin: make free_session() remove stream references When receiving a sync-packet, all sessions with the same cname will be compared and synced together. In this process, there could still be references to a session that has been shut down in the meanwhile. This patch makes sure that these references are removed when shutting down a session, so that the syncing can be done safely. Fixes #594283 2009-08-31 18:46:51 +0200 Havard Graff * gst/rtpmanager/gstrtpbin.c: rtpbin: use locked state on internal bins Set the locked state on internal elements to make sure that they don't change back to another state when shutting down. Fixes #594248 2009-09-07 18:28:51 +0200 Wim Taymans * sys/v4l2/gstv4l2object.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: v4l2src: add support for mpeg formats 2009-09-05 20:51:14 -0700 Zaheer Merali * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: y4menc: Add interlaced support Fixes #591713 Signed-off-by: David Schleef 2009-08-24 13:42:42 -0700 David Schleef * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfaudiosrc.c: * ext/gconf/gstgconfvideosink.c: * ext/gconf/gstgconfvideosrc.c: * gst/apetag/gstapedemux.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * sys/v4l2/gstv4l2src.c: Remove Ronald Bultje from Authors field Replaced with "GStreamer maintainers " or just removed, depending on the number of other authors. 2009-09-05 10:21:31 +0200 Sebastian Dröge * common: Automatic update of common submodule From 00a859e to 19fa4f3 2009-09-04 13:42:43 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: prevent a spurious debug warning 2009-09-04 09:32:42 +0200 Sebastian Dröge * sys/v4l2/gstv4l2object.c: v4l2: Define V4L2_FMT_FLAG_EMULATED if it's not defined yet libv4l2 already uses this flag, even on Linux kernel versions before 2.6.32. 2009-09-04 07:10:03 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Correctly handle NULL GstIndex 2009-09-03 20:40:17 +0200 Sebastian Dröge * sys/v4l2/gstv4l2object.c: v4l2: Fix stupid typo in last commit 2009-09-03 20:38:50 +0200 Sebastian Dröge * sys/v4l2/gstv4l2object.c: v4l2: Put emulated formats behind native formats Fixes bug #593764. 2009-09-03 19:37:10 +0200 Laurent Glayal * gst/rtpmanager/rtpsource.c: rtpsource: fix memleak Don't leak the input buffer when the received and expected seqnum are different when in probation. fixes #594039 2009-09-02 15:21:02 -0400 Olivier Crête * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: Lock clock_rate variable The priv->clock_rate variable could become -1 between when its checked to not be -1 and when its used, causing an assertion. Fixed by taking the mutex earlier in the chain() function. Fixes #593955 2009-09-03 19:12:39 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: whitespace fixes 2009-09-03 19:09:12 +0200 Wim Taymans * gst/rtp/gstrtpmpapay.c: rtpmpapay: whitespace fixes 2009-09-03 19:08:53 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: whitespace fixes 2009-09-03 17:33:28 +0200 Edward Hervey * ext/jpeg/gstjpegdec.c: jpegdec: Avoid unnecessary processing until we have a full picture. This is for non-packetized mode, when we know the upstream size in bytes. 2009-09-03 14:40:20 +0300 Stefan Kost * gst/flv/gstflvmux.c: flvmux: fully use tagsetter to manage the tags. Fixes #563221 There is no need to manage a separate taglist. 2009-09-03 14:13:43 +0300 Stefan Kost * ext/speex/gstspeexenc.c: speexenc: small taglist handling cleanup Don't eventualy leak the list and instead assert (like in other elements). 2009-09-02 23:12:41 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: also guard reseting subscribe callback with ifdefs It is conditionaly set, so do the same when unsetting. 2009-09-01 15:06:46 +0200 Peter Kjellerstedt * gst/rtpmanager/gstrtpsession.c: rtpmanager: Fixed a copy & paste error 2009-09-01 13:21:23 +0200 Peter Kjellerstedt * gst/rtpmanager/gstrtpsession.c: rtpmanager: Removed unused variable priv The variable priv was initialized in a lot of functions but then never used for anything. 2009-09-01 13:03:57 +0200 Peter Kjellerstedt * gst/rtpmanager/gstrtpsession.c: rtpmanager: A little clean up Make the code flow of gst_rtp_session_send_rtcp() and gst_rtp_session_sync_rtcp() identical. 2009-09-01 12:47:51 +0200 Peter Kjellerstedt * gst/rtpmanager/gstrtpsession.c: rtpmanager: Make sure that used caps are not freed already (take 2) This reintroduces the fix for bug #593391. It also applies it in gst_rtp_session_sync_rtcp() which has very similar code to gst_rtp_session_send_rtcp(). 2009-09-01 12:41:36 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: jitterbuffer: make sure time does not go backwards When we construct a timestamp that would result in a timestamp that is earlier than when the packet was received, reset the skew calculation as this is probably a sign that the sender restarted or paused. Fixes #593354 2009-09-01 11:32:41 +0200 Peter Kjellerstedt * gst/rtpmanager/gstrtpsession.c: rtpmanager: Set caps in gst_rtp_session_send_rtcp() correctly again The test for when to set an RTCP caps on the output pad in gst_rtp_session_send_rtcp() accidentally got inverted in the last commit. 2009-09-01 10:26:46 +0200 Sebastian Dröge * gst/qtdemux/qtdemux.c: qtdemux: Add support for QCELP audio Fixes bug #593757. 2009-08-31 18:10:11 +0200 Peter Kjellerstedt * gst/effectv/gstaging.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: effectv: Fix compilation with gcc 3 Recent changes in gst-plugins-good/gst/effectv prevents it from being compiled with gcc 3. The problem is that the new code uses preprocessor conditionals within a macro call which does not work with older versions of gcc. Fixes bug #593688. 2009-08-31 16:20:59 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins-sections.txt: docs: small clean-ups in -sections.txt Remove duplicate entry for warptv; there is no taglibmux element. 2009-08-27 15:46:52 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: consider (optional) auxiliary data when parsing 2009-08-27 15:46:15 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gdepay.h: rtpmp4gdepay: handle broken AU-Index in non-interleaved streams In case of non-interleaved (= sequentially payloaded) streams, the AU-Index serves little purpose (that is not already covered by RTP fields). (Broken) Payloaders might consider this field then to be disregarded and have non spec compliant values, e.g. each RTP packet having AU-Index 2 (rather than 0). As such, ensure/force simple sequential sending of non-interleaved streams. 2009-08-18 17:17:28 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: also extract ftyp info in push mode 2009-08-13 16:11:59 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: consider 3gpp style tag parsing in some more cases 3GPP specs define a number of tags along with precise layout. While these are normally expected to be found in a container whose major brand is a 3GPP brand, this may also happen when a 3GPP brand is only mentioned as a compatible brand. Apply some checks, heuristic and fallbacks to extract such tags as well. 2009-08-11 13:56:43 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: reflow exit, and fix some leaks 2009-08-11 13:54:56 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: push mode; add pad if needed so downstream gets EOS 2009-08-10 16:19:03 +0200 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: wavparse: push mode; fix/improve chunk handling Handle large, invalid or otherwise unusual chunk sizes. Verify some chunk sizes to be at least the size they are expected to be and round up some sizes to even number for e.g. offset administration, which must also be properly tracked in push mode. 2009-08-08 21:54:00 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: push mode; cater for unusual chunk sizes 2009-08-31 16:34:14 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: use proper locking for pads and caps Use the sesion lock and shotdown variable to protect and ref the pads we are going to push on. fixes #561825 2009-08-31 16:33:26 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: whitespace fixes 2009-08-31 13:38:08 +0100 Tim-Philipp Müller * gst/wavparse/gstwavparse.c: wavparse: clean up adapter properly Reflow code so we don't try to clear or re-use an already-freed adapter. 2009-08-31 13:07:53 +0100 Tim-Philipp Müller * ext/flac/gstflactag.c: * gst/wavparse/gstwavparse.c: flactag, wavparse: GstAdapter is not a GstObject 2009-08-31 12:28:52 +0100 Tim-Philipp Müller * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update plugin docs to git version 2009-08-31 11:32:39 +0100 Jan Schmidt * gst/flv/gstflvdemux.c: flvdemux: Fix tests warning from setting a NULL index Setting a null index in the tests was causing warnings by unreffing NULL pointers. This is a bug exposed by a recent change in core, it seems. 2009-08-31 13:02:16 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: add slope estimation code and debug Add some code to measure the sender speed vs the receiver speed. This can be used to detect bursts. 2009-08-31 12:57:32 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: reset skew when timestamps change Refactor the jitterbuffer resync code. Reset the skew correction when we detect a big timestamp discont. See #593354 2009-08-31 12:47:15 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: make sure time never goes invalid Since the skew can be negative, we might end up with invalid timestamps. Check for negative results and clamp to 0. See #593354 2009-08-31 12:16:01 +0200 Jarkko Palviainen * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpnetutils.c: udpsink: Add ttl multicast property Add a new ttl-mc property to control the TTL on multicast addresses. Fixes #588245 2009-08-31 12:13:07 +0200 Jarkko Palviainen * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: udp: split out TTL and loop options Split setting the TTL and loop parameters in 2 methods as they are not related. Fix setting the TTL correctly for multicast streams. See #588245 2009-08-27 12:36:37 +0200 Wim Taymans * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: rtp: whitespace fixes 2009-08-14 13:45:22 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins.args: videobox: Correctly add to the docs 2009-08-14 13:40:21 +0200 Sebastian Dröge * gst/videobox/Makefile.am: * gst/videobox/gstvideobox.c: * gst/videobox/gstvideobox.h: videobox: Split declarations into a header file and add autocrop stuff to the docs 2009-08-14 13:26:36 +0200 Sebastian Dröge * gst/videobox/gstvideobox.c: videobox: Reconfigure basetransform if something changes again For this invent a new lock and don't abuse the basetransform lock, otherwise we'll end up in deadlocks. 2009-08-14 13:15:57 +0200 Stephen Jungels * gst/videobox/gstvideobox.c: videobox: Add support for autocropping according to the caps Fixes bug #582238. 2009-08-30 21:57:57 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: rtpsession: Make sure that used caps are not freed already Fixes bug #593391. 2009-08-26 17:02:45 +0200 Sebastian Dröge * configure.ac: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpstats.c: rtp: Use new gst_iterator_new_single() for the internal linked pads iteration 2009-08-19 16:57:05 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpsession.c: rtpsession: Use iterate internal links instead of deprecated get internal links 2009-08-19 16:48:25 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: Use iterate internal links instead of deprecated get internal links 2009-08-19 16:37:11 +0200 Sebastian Dröge * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Use iterate internal links instead of deprecated get internal links 2009-08-30 23:27:09 +0100 Tim-Philipp Müller * common: Update common 2009-08-30 23:26:48 +0100 Tim-Philipp Müller * configure.ac: Back to hacking -> 0.10.16.1 === release 0.10.16 === 2009-08-29 12:05:40 +0100 Tim-Philipp Müller * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Release 0.10.16 2009-08-26 00:58:45 +0100 Tim-Philipp Müller * configure.ac: 0.10.15.5 pre-release 2009-08-25 16:53:29 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: don't use relative seeks Don't use relative seeks, it's too hard to track where we are after a flush etc. fixes #593015 2009-08-24 17:50:29 +0100 Tim-Philipp Müller * configure.ac: * po/LINGUAS: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: 0.10.15.4 pre-release 2009-08-24 16:22:47 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: don't discard the result of _set_caps() Use the result of gst_pad_set_caps() instead of assuming success. See #590678 2009-08-21 11:44:43 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: add support for agsm Fixes #592530 2009-08-18 17:16:11 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix qt style string tag extraction QT style tags are tested on starting with (C) symbol using >>, and (unsigned) int (may) have different >> behaviour. Fixes #592232. 2009-08-17 15:48:20 +0100 Tim-Philipp Müller * ext/jpeg/smokecodec.c: smokeenc: don't crash when compiled against libjpeg7 Set parameters so that we don't crash with libjpeg7. Based on Stefan Kost's fix for jpegenc. Fixes #591951. 2009-08-14 20:18:04 +0100 Tim-Philipp Müller * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: 0.10.15.3 pre-release 2009-08-14 13:45:08 +0100 Tim-Philipp Müller * tests/check/elements/rtpbin.c: checks: add test for leak to rtpbin unit test See #591476. 2009-08-11 14:47:12 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Fix reference leak Fixes #591476. 2009-08-14 13:34:53 +0100 Zaheer Merali * ext/dv/gstdvdec.c: dvdec: set bottom field first on PAL interlaced content, not top field first DV interlaced content is always bottom field first. Fixes #591712. 2009-08-14 12:44:06 +0100 Hans de Goede * sys/v4l2/gstv4l2src.c: v4l2src: fix 'hang' with some cameras caused by bad timestamping if no framerate is available For cameras/drivers that don't support e.g. VIDIOC_G_PARM we'd end up without a framerate and would try to divide by 0, causing run-time warnings and all frames to be timestamped with 0, which makes sinks that sync against the clock drop them, causing 'hangs' (observed with the pwc driver and a Logitech QuickCam Pro 4000). So if we do not know the framerate, simply don't adjust the timestamps. Fixes #591451. 2009-08-14 10:11:25 +0200 Filippo Argiolas * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: v4l2src: clear format list in READY->NULL Clear format list and probed caps when going to NULL so if a new device is set we'll probe the formats again instead of using previously detected ones. Fixes bug #591747. 2009-08-11 16:42:51 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: dtmfsrc: Empty event queue on finalize 2009-08-11 16:39:42 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: dtmf: Use GSlice for internal event structures 2009-08-11 16:23:20 -0400 Tim-Philipp Müller * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Cleanup events on finalize Problem found by Laurent Glayal Fixes bug #591440 2009-08-11 16:23:20 -0400 Tim-Philipp Müller * gst/dtmf/gstrtpdtmfsrc.c: rtpdtmfsrc: Cleanup events on finalize Problem found by Laurent Glayal Fixes bug #591440 2009-08-11 17:30:41 +0100 Tim-Philipp Müller * configure.ac: * po/LINGUAS: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: 0.10.15.2 pre-release 2009-08-11 15:25:39 +0100 Tim-Philipp Müller * MAINTAINERS: Add myself to MAINTAINERS file and update Wim's e-mail. 2009-08-11 03:08:01 +0100 Tim-Philipp Müller * sys/v4l2/Makefile.am: v4l2: fix make distcheck by disting some more headers 2009-08-11 02:42:16 +0100 Tim-Philipp Müller * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-wavparse.xml: docs: update 2009-08-11 02:31:44 +0100 Tim-Philipp Müller * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * gst-plugins-good.spec.in: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/pipelines/.gitignore: Move rtpmanager from -bad to -good. Hook up build infrastructure (autotools, docs, unit test). 2009-08-06 19:26:21 +0200 ric * gst/rtpmanager/rtpsource.c: rtpsource: avoid buffer leak on bad seqnum Fixes #590797 2009-07-28 18:18:20 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: allow for NULL caps on buffers Add the NULL caps check where it matters and also cover another case of potential NULL caps. Fixes #590030 2009-07-28 11:59:56 -0400 Olivier Crête * gst/rtpmanager/rtpsource.c: rtpsource: Incoming buffers do not always have caps 2009-07-27 15:46:23 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: avoid doing lip-sync in BYE When we get a BYE packet, don't do lip-sync with the SR inside because some senders have trouble constructing valid SR packets after BYE. 2009-07-27 13:17:20 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpbin: don't do lip-sync after a BYE After a BYE packet from a source, stop forwarding the SR packets for lip-sync to rtpbin. Some senders don't update their SR packets correctly after sending a BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with the current lip-sync instead. 2009-07-27 12:43:02 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpbin: only reconsider once for BYE When iterating the sources of a BYE packet, don't signal a reconsideration for each of them but signal after we handled all sources. 2009-07-21 15:33:41 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Free conflicting addresses on finalize 2009-07-01 12:55:03 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpbin: use new method for netaddress to string 2009-06-29 18:48:33 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * tests/check/elements/rtpbin.c: rtpbin: do better cleanup of the src ghostpads Connect to the pad-removed signal of the ptdemux elements so that we remove the ghostpads for them. Fixes cleanup when going to NULL as well as when releasing the sinkpads. Fixes #561752 2009-05-28 19:08:40 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: add a comment 2009-06-29 16:37:54 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: rtpbin: add SDES property Remove all individual SDES properties and use one sdes property that takes a GstStructure instead. This will allow us to add more custom stuff to the SDES messages later. 2009-06-29 16:21:05 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpbin: add SDES property that takes GstStructure Remove all individual SDES properties and use one sdes property that takes a GstStructure instead. This will allow us to add more custom stuff to the SDES messages later. 2009-06-02 17:46:08 +0200 Wim Taymans * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpmanager.c: rtpbin: removed old gstrtpclient 2009-06-19 19:09:19 +0200 Branko Subasic * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * tests/check/elements/rtpbin_buffer_list.c: rtpbin: add support for buffer-list Add support for sending buffer-lists. Add unit test for testing that the buffer-list passed through rtpbin. fixes #585839 2009-06-19 16:21:28 +0100 Tim-Philipp Müller * gst/rtpmanager/gstrtpjitterbuffer.c: Make build without warnings with debugging disabled 2009-05-28 17:37:44 -0400 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Transform the right session sdes message Fixes #584165 2009-05-28 17:33:10 -0400 Olivier Crête * gst/rtpmanager/rtpsource.c: Add ssrc to application/x-rtp-source-sdes structure 2009-05-27 11:03:14 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsouce: the network address is in network order Bring the network address in netowkr byte order to the host order. 2009-05-26 15:40:52 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: byteswap the port from GstNetAddress Since the port in GstNetAddress is in network order we might need to byteswap it before adding it to the source statistics. 2009-05-25 13:46:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: remove ptdemux ghostpads 2009-05-25 13:33:20 +0200 Wim Taymans * tests/check/elements/rtpbin.c: tests: add receive rtpbin unit test 2009-05-22 16:41:19 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: add to new signal to remove SSRC pads 2009-05-22 16:35:20 +0200 Ali Sabil * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: ssrcdemux: emit signal when pads are removed Add action signal to clear an SSRC in the ssrc demuxer. Add signal to notify of removed ssrc. See #554839 2009-05-22 15:45:19 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: use our ghostpads instead of its target Since we keep a reference to our ghostpads, we can use them to track sessions. This avoid us having to mess with the target of the ghostpad. 2009-05-22 15:37:29 +0200 Wim Taymans * tests/check/elements/rtpbin.c: tests: more rtpbin checks 2009-05-22 15:36:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: don't warn when getting request pads twice Allow getting the request pads multiple times, just return the previously created pads. 2009-05-22 13:47:30 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: rtpsource: add RTP and RTCP source address Add the RTP and RTCP sender addresses in the stats structure. 2009-05-22 13:45:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: reuse source code for SDES Reuse the RTPSource object property instead of duplicating code. 2009-05-22 13:44:17 +0200 Wim Taymans * tests/check/elements/rtpbin.c: tests: add more rtpbin tests 2009-05-22 12:23:27 +0200 Wim Taymans * tests/check/elements/rtpbin.c: tests: add rtpbin unit test Add the beginnings of an rtpbin unit test Add some more stuff to .gitignore 2009-05-22 12:20:13 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: set target state on new elements Set the state on newly added elements to the state of the parent. Add some debug info and do some cleanups 2009-05-22 11:59:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: unref requests pads after releasing 2009-05-22 01:43:50 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Implement releasing the streams See #561752 2009-05-22 01:16:11 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Keep jb signals handler Keep the signal handlers so they can be disconnected at release time See #561752 2009-05-22 01:12:57 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: use the right lock for the sessions Use the right lock when iterating the sessions. 2009-05-22 01:03:55 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Free session if request pads are released Free the session when all the request pads are released. Don't mess with the session list in free_session as it is called from a foreach on that list. Set the state of the upstream element to NULL first. See #561752 2009-05-22 00:51:53 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Implement relasing of the rtp recv pad 2009-05-22 00:44:51 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Implement releasing of rtp send pads 2009-05-22 00:34:36 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Implement release of the recv rtcp pad See #561752 2009-05-22 00:16:19 +0200 Olivier Crête * gst/rtpmanager/gstrtpbin.c: rtpbin: Implement releasing of rtcp src pad See #561752 2009-05-05 16:48:37 +0200 Wim Taymans * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: drop unexpected RTCP packets We usually only get SR packets in our chain function but if an invalid packet contains the SR packet after the RR packet, we must not fail but simply ignore the malformed packet. Fixes #581375 2009-04-27 11:09:08 +0200 Olivier Crete * gst/rtpmanager/rtpsource.c: rtpsouce: make WARNING into LOG Since neither rtpmanager nor any of the payloaders properly implement pad allocation, there is no way for the rtpmanager to inform downstream elements of the new SSRC if there is an SSRC collision. So the warning is emitted all the time and it is confusing. Fixes #580144 2009-04-27 11:06:01 +0200 Olivier Crete * gst/rtpmanager/rtpsession.c: rtpsession: notify when SSRC changes Emit a g_object_notify when the SSRc changes because of a collision. Fixes #580144 2009-04-17 16:16:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpsession: join the RTCP thread Avoid a case where a joinable thread would be left unjoined, which leaked the thread structure. Fixes #577318. 2009-04-15 18:14:48 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: prevent overflow in EOS estimation Use a guint64 instead of a guint to hold a 64bit value to prevent completely bogues EOS estimation values due to overflows. 2009-04-15 17:44:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: we should not provide a clock There is no need to provide a clock. 2009-04-15 17:28:56 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: more estimated EOS fixes Do more accurate EOS estimate and guard against backward timestamps. 2009-04-15 17:25:02 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: release lock before pushing EOS Make sure we release the jitterbuffer lock before we start pushing out data because else we might deadlock. 2009-03-27 17:44:57 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: rtpbin: add on_npt_stop signal Add the on_npt_stop signal to rtpbin and rtpjitterbuffer to notify the application that the NPT stop position has been reached. 2009-03-13 15:59:37 +0100 Wim Taymans * gst/rtpmanager/gstrtpsession.c: rtpbin: don't return FALSE on seek events Silently ignore the seek event instead of returning FALSE. 2009-02-26 13:10:29 +0100 Olivier Crête * gst/rtpmanager/gstrtpsession.c: gstrtpbin: Don't forward revc events to sender Don't send events from the receiver to the sender side. Fixes #572900. 2009-02-25 11:45:05 +0200 Stefan Kost * gst/rtpmanager/rtpjitterbuffer.c: docs: various doc fixes No short-desc as we have them in the element details. Also keep things (Makefile.am and sections.txt) sorted. Reword ambigous returns. No text after since please. 2009-01-23 12:13:00 +0100 Wim Taymans * gst/rtpmanager/rtpstats.c: Send BYE packets immediatly for small sessions When the number of participants is less than 50, the RFC allows for sending the BYE packet immediatly instead of using the regular BYE timeout. Fixes #567828. 2009-01-22 13:33:14 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: Unlock the jitterbuffer before pushing out the packet-lost events. Move some code before we do the unlock to make the jitterbuffer state consistent while we are unlocked. 2009-01-02 17:40:06 +0000 Olivier Crete gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink): * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc): When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910. 2009-01-02 16:50:53 +0000 Wim Taymans gst/rtpmanager/: Rename a method to better reflect what it really does. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_getcaps_send_rtp): * gst/rtpmanager/rtpsession.c: (check_collision), (rtp_session_schedule_bye_locked), (rtp_session_schedule_bye): * gst/rtpmanager/rtpsession.h: Rename a method to better reflect what it really does. 2008-12-29 15:49:37 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_getcaps_send_rtp): Use method to get the internal SSRC. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_set_property), (rtp_session_get_property): Add property to congiure the internal SSRC of the session. Fixes #565910. 2008-12-29 15:21:58 +0000 Wim Taymans gst/rtpmanager/rtpsession.c: Only change the SSRC of the session and reset the internal source when the SSRC actually... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc): Only change the SSRC of the session and reset the internal source when the SSRC actually changed. See #565910. 2008-12-29 14:21:47 +0000 Wim Taymans gst/rtpmanager/rtpsource.*: When no payload was specified on the caps but there was a clock-rate, assume the clock-ra... Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (get_clock_rate): * gst/rtpmanager/rtpsource.h: When no payload was specified on the caps but there was a clock-rate, assume the clock-rate corresponds to the first payload type found in the RTP packets. Fixes #565509. 2008-12-23 11:39:59 +0000 Arnout Vandecappelle gst/rtpmanager/rtpjitterbuffer.*: Keep track of the last outgoing timestamp and of the last sender-side time. Timest... Original commit message from CVS: Patch by: Arnout Vandecappelle * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of the last outgoing timestamp and of the last sender-side time. Timestamps can only go forward if they do at the sender side, can only go back if they do at the sender side, and remain the same if they remain the same at the sender side. Fixes #565319. 2008-11-26 12:40:18 +0000 Wim Taymans gst/rtpmanager/rtpsession.c: Make obtain_source return an aditional ref so that we don't lose our ref to it when a se... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (obtain_source), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye): Make obtain_source return an aditional ref so that we don't lose our ref to it when a session cleanup occurs when we are emiting a signal. Emit the on_new_ssrc signal for the CSRC, not the SSRC. Fixes #562319. 2008-11-26 12:02:21 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Reset the sync parameters when clearing the payload type map too. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync), (gst_rtp_bin_clear_pt_map): Reset the sync parameters when clearing the payload type map too. Fixes #562312. 2008-11-26 11:44:37 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Remove a lot of per stream state that is not needed and pass new info in the method call. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (get_client), (gst_rtp_bin_reset_sync), (gst_rtp_bin_associate), (gst_rtp_bin_handle_sync), (create_stream), (gst_rtp_bin_class_init), (new_ssrc_pad_found): * gst/rtpmanager/gstrtpbin.h: Remove a lot of per stream state that is not needed and pass new info in the method call. Add signal to reset sync parameters. Avoid parsing the caps to get a clock_base, we get this from the sync signal now. 2008-11-25 15:12:06 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Fix event leak. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtcp_src): Fix event leak. 2008-11-22 15:31:36 +0000 Wim Taymans gst/rtpmanager/rtpsession.c: Add property to configure the RTCP MTU. Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_set_property), (rtp_session_get_property): Add property to configure the RTCP MTU. 2008-11-22 15:24:47 +0000 Wim Taymans gst/rtpmanager/rtpsession.c: Add G_PARAM_STATIC_STRINGS. Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (copy_source), (rtp_session_create_sources), (rtp_session_get_property): Add G_PARAM_STATIC_STRINGS. Add property to return a GValueArray of all known RTPSources in the session. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_create_sdes), (rtp_source_set_property), (rtp_source_get_property): Remove properties to set the various SDES items, an application is never supposed to change the RTPSource data. Change the SDES getter properties to one SDES property that returns all SDES items in a GstStructure. 2008-11-22 13:17:24 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Also unref the target pad for unknown pads. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad): Also unref the target pad for unknown pads. 2008-11-21 16:17:22 +0000 Olivier Crete gst/rtpmanager/gstrtpbin.c: Release the right pads on rtpbin. Fixes #561752. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad): Release the right pads on rtpbin. Fixes #561752. 2008-11-20 18:41:34 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (get_current_times), (rtcp_thread), (gst_rtp_session_chain_recv_rtp): Pass the running time to the session when processing RTP packets. Improve the time function to provide more info. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sdes), (rtp_session_process_rtcp), (session_start_rtcp), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Mark the internal source with a flag. Use running_time instead of the more useless timestamp. Validate a source when a valid SDES has been received. Pass the current system time when processing SR packets. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_create_stats), (rtp_source_get_property), (rtp_source_send_rtp), (rtp_source_process_rb), (rtp_source_get_new_rb), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: Add property to get source stats. Mark params as STATIC_STRINGS. Calculate the bitrate at the sender SSRC. Avoid negative values in the round trip time calculations. * gst/rtpmanager/rtpstats.h: Update some docs and change some variable name to more closely reflect what it contains. 2008-11-20 08:19:15 +0000 Sebastian Dröge gst/rtpmanager/gstrtpjitterbuffer.c: Initialize return value to fix compiler warning about uninitialized variable. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain_rtcp): Initialize return value to fix compiler warning about uninitialized variable. 2008-11-19 16:48:38 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Mark signal arg as static scope. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init): Mark signal arg as static scope. 2008-11-19 09:06:29 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to get lip-sync notifications. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_handle_sync), (create_stream), (free_stream), (new_ssrc_pad_found): Remove internal sync pad, use signals instead to get lip-sync notifications. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_base_init), (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink), (remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad), (gst_rtp_jitter_buffer_release_pad), (gst_rtp_jitter_buffer_sink_rtcp_event), (gst_rtp_jitter_buffer_chain_rtcp), (gst_rtp_jitter_buffer_get_property): * gst/rtpmanager/gstrtpjitterbuffer.h: Make it possible to send SR packets to the jitterbuffer. Check if the SR timestamps are valid by comparing them to the RTP timestamps. Signal the SR packet and the timing information to listeners. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query): Remove some unused code. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of the last seen RTP timestamp so that we can filter out invalid SR packets. 2008-11-17 19:47:32 +0000 Sebastian Dröge gst/rtpmanager/rtpsource.c: Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes... Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (get_clock_rate): Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes a compiler warning. 2008-11-17 15:17:52 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji... Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found): Do not try to keep track of the clock-rate ourselves but simply get the value from the jitterbuffer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add some debug info. Pass the clock-rate to the jitterbuffer. Also pass the clock-rate along with the rtp timestamp when getting the sync parameters. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Fix some debug. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of clock-rate changes and return the clock-rate together with the rtp timestamps used for sync. Don't try to construct timestamps when we have no base_time. * gst/rtpmanager/rtpsource.c: (get_clock_rate): Request a new clock-rate when the payload type changes. Reset the jitter calculation when the clock-rate changes. 2008-11-13 15:48:54 +0000 Wim Taymans gst/rtpmanager/: Small cleanups and some more debug info. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain): * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew): Small cleanups and some more debug info. 2008-11-10 15:26:40 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Also configure the next expected output seqnum when we get a seqnum-base on the ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain): Also configure the next expected output seqnum when we get a seqnum-base on the caps. 2008-11-04 12:42:30 +0000 Stefan Kost Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-10-16 13:05:37 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Fix problem with using the output seqnum counter to check for input seqnum disco... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Fix problem with using the output seqnum counter to check for input seqnum discontinuities. Improve gap detection and recovery, reset and flush the jitterbuffer on seqnum restart. Fixes #556520. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert): Fix wrong G_LIKELY. 2008-10-16 09:51:28 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, make LATENCY event return Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src): Install event handler on the rtcp_src pad, make LATENCY event return TRUE. 2008-10-07 18:54:41 +0000 Håvard Graff gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal. Original commit message from CVS: Patch by: Håvard Graff * gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: Add action signal to retrieve the internal RTPSession object. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_get_property), (gst_rtp_session_release_pad): Add property to access the internal RTPSession object. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (check_collision): * gst/rtpmanager/rtpsession.h: Add action signal to retrieve an RTPSource object by SSRC. See #555396. 2008-10-07 11:33:10 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Release pads of the session manager. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (find_session_by_pad), (free_session), (gst_rtp_bin_dispose), (remove_recv_rtp), (remove_recv_rtcp), (remove_send_rtp), (remove_rtcp), (gst_rtp_bin_release_pad): Release pads of the session manager. Start implementing releasing pads of gstrtpbin. * gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink), (remove_recv_rtcp_sink), (remove_send_rtp_sink), (remove_send_rtcp_src), (gst_rtp_session_release_pad): Implement releasing pads in gstrtpsession. 2008-10-07 10:02:20 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Only update the seqnum-base when it was not already configured for the streams. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps): Only update the seqnum-base when it was not already configured for the streams. 2008-09-30 15:08:52 +0000 Wim Taymans gst/rtpmanager/rtpsession.c: Ref the rtpsource object before we release the session lock when we emit the signals. Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout): Ref the rtpsource object before we release the session lock when we emit the signals. 2008-09-23 18:13:31 +0000 Wim Taymans gst/rtpmanager/: Fix some docs. Original commit message from CVS: * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpsession.c: (on_sender_timeout), (session_cleanup): * gst/rtpmanager/rtpsource.c: Fix some docs. 2008-09-17 13:59:21 +0000 Jan Schmidt Fix compiler warnings on OS/X Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (jack_process_cb): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Fix compiler warnings on OS/X 2008-09-13 01:37:50 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A... Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain): Do not try to adjust the offset of streams for which we have not yet seen an SR packet. Avoids large ts-offsets in some cases. 2008-09-05 13:52:34 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout), (create_session), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: Add signal to notify listeners when a sender becomes a receiver. Tweak lip-sync code, don't store our own copy of the ts-offset of the jitterbuffer, don't adjust sync if the change is less than 4msec. Get the RTP timestamp <-> GStreamer timestamp relation directly from the jitterbuffer instead of our inaccurate version from the source. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add G_LIKELY macros, use global defines for max packet reorder and dropouts. Reset the jitterbuffer clock skew detection when packets seqnums are changed unexpectedly. * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: Add sender timeout signal. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Add some G_LIKELY macros. Keep track of the extended RTP timestamp so that we can report the RTP timestamp <-> GStreamer timestamp relation for lip-sync. Remove server timestamp gap detection code, the server can sometimes make a huge gap in timestamps (talk spurts,...) see #549774. Detect timetamp weirdness instead by observing the sender/receiver timestamp relation and resync if it changes more than 1 second. Add method to report about the current rtp <-> gst timestamp relation which is needed for lip-sync. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_sender_timeout), (check_collision), (rtp_session_process_sr), (session_cleanup): * gst/rtpmanager/rtpsession.h: Add sender timeout signal. Remove inaccurate rtp <-> gst timestamp relation code, the jitterbuffer can now do an accurate reporting about this. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (calculate_jitter), (rtp_source_process_rtp): * gst/rtpmanager/rtpsource.h: Remove inaccurate rtp <-> gst timestamp relation code. * gst/rtpmanager/rtpstats.h: Define global max-reorder and max-dropout constants for use in various subsystems. 2008-08-28 15:21:45 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp), (gst_rtp_session_event_send_rtp_sink): Send EOS when the session object instructs us to. * gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Make it possible for the session manager to instruct us to send EOS. We currently will EOS when the session is a sender and when the sender part goes EOS. This is not entirely correct behaviour because the session could still participate as a receiver. Fixes #549409. 2008-08-13 14:31:02 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (new_ssrc_pad_found): Reset rtp timestamp interpollation when we detect a gap when the clock_base changed. Don't try to adjust the ts-offset when it's too big (> 3seconds) * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc): * gst/rtpmanager/gstrtpsession.h: Add method to set session SSRC. * gst/rtpmanager/rtpsession.c: (check_collision), (rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Added debugging for the collision checks. Add method to change the internal SSRC of the session. * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp): Reset the clock base when we detect large jumps in the seqnums. 2008-08-11 07:20:15 +0000 Stefan Kost gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log. * sys/dshowsrcwrapper/gstdshowaudiosrc.c: * sys/dshowsrcwrapper/gstdshowvideosrc.c: Use "-" instead of "_" in property names. Can we call them just "device" like everywhere else? 2008-08-05 09:42:53 +0000 Olivier Crete gst/rtpmanager/gstrtpjitterbuffer.c: Make the buffer metadata writable before inserting it in the jitterbuffer becaus... Original commit message from CVS: Based on patch by: Olivier Crete * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Make the buffer metadata writable before inserting it in the jitterbuffer because the jitterbuffer will modify the timestamps. * gst/rtpmanager/rtpjitterbuffer.c: Update method comment about requiring writable metadata on buffers. * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr), (rtp_session_process_rtcp): Make the RTCP buffer metadata writable because we want to modify the metadata. Fixes #546312. 2008-08-05 09:00:50 +0000 Håvard Graff gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum. Original commit message from CVS: Patch by: Håvard Graff * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Fix debug by logging the right seqnum. 2008-08-05 08:58:27 +0000 Olivier Crete gst/rtpmanager/gstrtpbin.c: Release lock before emitting the request-pt-map signal. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpbin.c: (get_pt_map): Release lock before emitting the request-pt-map signal. Fixes #543480. 2008-07-03 14:44:51 +0000 Peter Kjellerstedt gst/rtpmanager/: Corrected a typo (interpollate -> interpolate). Original commit message from CVS: * ChangeLog: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr): Corrected a typo (interpollate -> interpolate). 2008-07-03 14:31:10 +0000 Peter Kjellerstedt gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_send_rtp): * gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp): Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally. 2008-07-03 13:47:19 +0000 Peter Kjellerstedt gst/rtpmanager/: Do not mix the use of g_get_current_time() with gst_clock_get_time(). Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init), (gst_rtp_session_finalize), (rtcp_thread), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsession.c: (check_collision), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_send_bye_locked), (rtp_session_send_bye), (rtp_session_next_timeout), (session_report_blocks), (session_cleanup), (is_rtcp_time), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Do not mix the use of g_get_current_time() with gst_clock_get_time(). 2008-06-16 07:30:34 +0000 Stefan Kost Final round of doc updates. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/speed/gstspeed.c: * gst/speexresample/gstspeexresample.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/dvb/gstdvbsrc.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/wininet/gstwininetsrc.c: Final round of doc updates. 2008-06-16 07:03:58 +0000 Stefan Kost gst/: More doc updates. More xrefs. Original commit message from CVS: * gst/deinterlace/gstdeinterlace.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/sdp/gstsdpdemux.c: More doc updates. More xrefs. 2008-06-12 14:49:18 +0000 Stefan Kost Do not use short_description in section docs for elements. We extract them from element details and there will be war... Original commit message from CVS: * ext/dc1394/gstdc1394.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/metadata/gstmetadatademux.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * gst-libs/gst/app/gstappsink.c: * gst/bayer/gstbayer2rgb.c: * gst/deinterlace/gstdeinterlace.c: * gst/rawparse/gstaudioparse.c: * gst/rawparse/gstvideoparse.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/selector/gstinputselector.c: * gst/selector/gstoutputselector.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Do not use short_description in section docs for elements. We extract them from element details and there will be warnings if they differ. Also fixing up the ChangeLog order. 2008-06-06 13:01:05 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Fix deadlock when shutting down, use a new lock instead to properly shutdown. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init), (gst_rtp_bin_finalize), (gst_rtp_bin_change_state): Fix deadlock when shutting down, use a new lock instead to properly shutdown. 2008-05-27 16:48:10 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Break out of callbacks when we are shutting down. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_propagate_property_to_jitterbuffer), (gst_rtp_bin_change_state), (new_payload_found), (new_ssrc_pad_found): Break out of callbacks when we are shutting down. Make sure no state changes can happen when we reconfigure. 2008-05-26 10:09:29 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the jitterbuffer if the gap is too big, we need ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): When checking the seqnum, reset the jitterbuffer if the gap is too big, we need to do this so that we can better handle a restarted source. Fix some comments. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): Tweak the skew resync diff. Use our working seqnum compare function in -base. Rework the jitterbuffer insert code to make it clearer and more performant by only retrieving the seqnum of the input buffer once and by adding some G_LIKELY compiler hints. Improve debugging for duplicate packets. * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp): Fix a comment, we don't do skew correction here.. 2008-05-26 10:00:24 +0000 Håvard Graff gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to the jitterbuffers when they are changed o... Original commit message from CVS: Patch by: Håvard Graff * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_propagate_property_to_jitterbuffer), (gst_rtp_bin_set_property): Propagate the do-lost and latency properties to the jitterbuffers when they are changed on rtpbin. 2008-05-26 09:57:40 +0000 Wim Taymans Don't use _gst_pad(). Original commit message from CVS: * examples/switch/switcher.c: (switch_timer): * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init): * gst/rtpmanager/gstrtpclient.c: (create_stream): * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp), (gst_sdp_demux_stream_configure_udp_sink): * tests/check/elements/deinterleave.c: (GST_START_TEST), (pad_added_setup_data_check_float32_8ch_cb): * tests/check/elements/rganalysis.c: (send_eos_event), (send_tag_event): Don't use _gst_pad(). 2008-05-16 19:56:30 +0000 Jan Schmidt docs/Makefile.am: Don't attempt to build plugin docs when they're disabled. Original commit message from CVS: * docs/Makefile.am: Don't attempt to build plugin docs when they're disabled. * gst/bayer/Makefile.am: Add libgstvideo to the link. * gst/rtpmanager/Makefile.am: Fix link order, and move LIBS things to _LIBS 2008-05-14 21:02:19 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a warning instead of just posting an error and ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Simply drop bad RTP packets with a warning instead of just posting an error and stopping. This is a perfectly recoverable event and we don't force people to use an rtpbin to filter out bad packets first. 2008-05-13 09:06:51 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): Actually add the do-lost property to the object. 2008-05-12 18:43:41 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) duration when the last packet has a lower ti... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Avoid waiting for a negative (huge) duration when the last packet has a lower timestamp than the current packet. 2008-05-12 14:28:09 +0000 Peter Kjellerstedt gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memor... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src): Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memory leak. 2008-05-12 14:12:08 +0000 Jan Schmidt gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning. 2008-05-09 07:41:58 +0000 Peter Kjellerstedt gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to prevent a memory leak. Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Make sure to unref the caps used by RTPSource to prevent a memory leak. 2008-05-08 09:43:33 +0000 Olivier Crete gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our callbacks. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/rtpsession.c: (source_clock_rate), (rtp_session_process_bye), (rtp_session_send_bye_locked): Unlock the session lock when calling one of our callbacks. Fixes #532011. 2008-05-08 06:23:39 +0000 Sjoerd Simons gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955. Original commit message from CVS: Patch by: Sjoerd Simons * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink): Send RTP BYE command on EOS. Fixes bug #531955. 2008-04-25 11:32:09 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Expose new jitterbuffer property in rtpbin too. 2008-04-25 11:22:13 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost events by default and make a property to ena... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Disable sending out rtp packet lost events by default and make a property to enabe it. We will likely enable it by default when the base depayloaders have a default handler for them so that we don't send these events all through the pipeline for now. 2008-04-25 09:35:43 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function that is in -base now. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Remove private version of a function that is in -base now. Add src event handler. Rework the jitterbuffer pushing loop so that it can quickly react to lost packets and instruct the depayloader of them. This can then be used to implement error concealment data. 2008-04-25 08:21:06 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the RTCP and sync pads because the defaults ... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink), (create_send_rtcp_src): Set up some internal links functions for the RTCP and sync pads because the defaults are really not correct. Implement a query handler for the RTCP src pad, mostly to correctly report about the latency. 2008-04-25 08:15:58 +0000 Wim Taymans gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain): * gst/rtpmanager/rtpsession.c: (update_arrival_stats), (rtp_session_process_sr), (rtp_session_on_timeout): * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Also keep track of the first buffer timestamp together with the first RTP timestamp as they both are needed to construct the timing of outgoing packets in the jitterbuffer and are therefore also needed to manage lip-sync. This fixes lip-sync if the first RTP packets arrive with a wildly different gap. 2008-04-21 08:26:37 +0000 Olivier Crete gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (new_ssrc_pad_found): Ref caps when inserting into the cache. Don't leak pads. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_get_clock_rate), (gst_rtp_jitter_buffer_query): Avoid a caps leak. Don't leak refcount in query. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps), (gst_rtp_pt_demux_chain): Avoid caps leaks. * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure), (gst_rtp_session_init), (return_true), (gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate): Ref caps when inserting into the cache. Fix some more caps leaks. Fixes #528245. 2008-04-17 07:31:44 +0000 Wim Taymans gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a refcount leak. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client), (gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_get_clock_rate): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate): Unset GValues after g_signal_emitv so that we avoid a refcount leak. Don't leak a padname. Don't leak client streams list. Lock rtpbin when associating streams. Fixes #528245. 2008-04-09 22:27:50 +0000 Peter Kjellerstedt gst/rtpmanager/: Avoid leaking pads in the RTP manager. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (free_session): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize): Avoid leaking pads in the RTP manager. 2008-03-11 12:40:58 +0000 Olivier Crete gst/rtpmanager/rtpsession.*: Implement collision and loop detection in rtpmanager. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/rtpsession.c: (find_add_conflicting_addresses), (check_collision), (obtain_source), (rtp_session_create_new_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_send_bye_locked), (rtp_session_send_bye), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Implement collision and loop detection in rtpmanager. Fixes #520626. * gst/rtpmanager/rtpsource.c: (rtp_source_reset), (rtp_source_init): * gst/rtpmanager/rtpsource.h: Add method to reset stats. 2008-03-11 11:36:03 +0000 Ole André Vadla Ravnås gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d... Original commit message from CVS: Based on patch by: Ole André Vadla Ravnås * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init), (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), (join_rtcp_thread), (gst_rtp_session_change_state): Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked downstream. Also avoid spawning multiple rtcp threads. Fixes #520894. 2008-03-11 10:43:32 +0000 Stefan Kost gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we have no timestamps. Original commit message from CVS: Patch by: Stefan Kost * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Don't try to reset the clock skew when we have no timestamps. Fixes #519005. 2008-02-20 09:33:25 +0000 Olivier Crete gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug 517571. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Fix small memory leak, leaking caps. Fixes #bug 517571. 2008-02-14 16:25:51 +0000 Olivier Crete gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate): Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160. 2008-01-29 18:57:27 +0000 Thijs Vermeir gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the buffer caps when we receive a new payload... Original commit message from CVS: Patch by: Thijs Vermeir * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Try to get the new clock-rate from the buffer caps when we receive a new payload type instead of always firing the signal. Fixes #512774. 2008-01-25 16:58:00 +0000 Olivier Crete gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not provided with caps but with a signal. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (create_stream), (payload_type_change), (new_ssrc_pad_found): Also handle lip-sync when the clock-rate is not provided with caps but with a signal. 2008-01-25 16:00:52 +0000 Olivier Crete gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided... Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided with each buffer instead. Fixes #511686. 2008-01-25 15:49:55 +0000 Olivier Crete gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable. Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Remove old unused variable. Track pt on input buffers and get the clock-rate when it changes. Ignore packets with unknown clock-rate. See #511686. 2008-01-25 01:44:27 +0000 Olivier Crete gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920 Original commit message from CVS: Patch by: Olivier Crete * gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920 2008-01-11 17:02:30 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to parse the clock-rate instead of returning... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate): If we find the caps in the cache, use it to parse the clock-rate instead of returning an error. Fixes a TODO as found by Youness Alaoui. 2008-01-11 16:45:57 +0000 Youness Alaoui gst/rtpmanager/: Make it possible to use different user_data for each of the callbacks. Original commit message from CVS: Patch by: Youness Alaoui * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate): * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks), (rtp_session_set_process_rtp_callback), (rtp_session_set_send_rtp_callback), (rtp_session_set_send_rtcp_callback), (rtp_session_set_sync_rtcp_callback), (rtp_session_set_clock_rate_callback), (rtp_session_set_reconsider_callback), (source_push_rtp), (source_clock_rate), (rtp_session_process_bye), (rtp_session_process_rtcp), (rtp_session_send_bye), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Make it possible to use different user_data for each of the callbacks. Fixes #508587. 2008-01-10 20:57:17 +0000 Thijs Vermeir gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch 2008-01-10 14:34:30 +0000 Thijs Vermeir gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515) Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515) 2008-01-09 14:39:44 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do everything the upsteam peer pad can renegot... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink): Don't set fixed caps, we can basically do everything the upsteam peer pad can renegotiate to. Fixes #507940. 2008-01-04 18:47:57 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we don't have ownership. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Don't unref the popped buffer when we don't have ownership. Fixes #507020. 2007-12-31 13:12:06 +0000 Wim Taymans gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED. Original commit message from CVS: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_change_state): Don't clean up pads when going to PAUSED. 2007-12-12 16:59:03 +0000 Wim Taymans gst/rtpmanager/: Clean up the dynamic pads when going to READY. Original commit message from CVS: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset), (gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query), (gst_rtp_ssrc_demux_change_state): Clean up the dynamic pads when going to READY. 2007-12-12 12:11:53 +0000 Wim Taymans gst/rtpmanager/: Fix some leaks. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize), (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string), (gst_rtp_bin_handle_message): * gst/rtpmanager/rtpsession.c: (rtp_session_finalize), (rtp_session_send_bye): * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Fix some leaks. 2007-12-10 18:36:04 +0000 Wim Taymans gst/rtpmanager/: Post a message when the SDES infor changes for a source. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init), (gst_rtp_bin_handle_message): * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure), (on_ssrc_sdes): Post a message when the SDES infor changes for a source. * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: Update some comments. 2007-12-10 15:34:19 +0000 Wim Taymans gst/rtpmanager/: Add signal to notify of an SDES change. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: * gst/rtpmanager/rtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.h: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_ssrc_sdes), (rtp_session_process_sdes): * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.c: * gst/rtpmanager/rtpstats.h: Add signal to notify of an SDES change. Fix object type in the signal callbacks. 2007-12-10 14:03:32 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Expose SDES items as properties and configure the session managers with them. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name), (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Expose SDES items as properties and configure the session managers with them. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_set_property): Fix SSRC property. 2007-12-10 11:08:11 +0000 Wim Taymans gst/rtpmanager/: Update comment. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session): * gst/rtpmanager/rtpjitterbuffer.c: Update comment. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property): Define some GObject properties to set SDES and other configuration. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (rtp_session_set_property), (rtp_session_get_property), (on_ssrc_sdes), (rtp_session_set_bandwidth), (rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction), (rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string), (rtp_session_get_sdes_string), (obtain_source), (rtp_session_get_internal_source), (rtp_session_process_sdes), (rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes), (is_rtcp_time): * gst/rtpmanager/rtpsession.h: Add signal when new SDES infor has been found for a source. Create properties for SDES and other info. Simplify the SDES API. Add method for getting the internal source object of the session. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_finalize), (rtp_source_set_property), (rtp_source_get_property), (rtp_source_set_callbacks), (rtp_source_get_ssrc), (rtp_source_set_as_csrc), (rtp_source_is_as_csrc), (rtp_source_is_active), (rtp_source_is_validated), (rtp_source_is_sender), (rtp_source_received_bye), (rtp_source_get_bye_reason), (rtp_source_set_sdes), (rtp_source_set_sdes_string), (rtp_source_get_sdes), (rtp_source_get_sdes_string), (rtp_source_get_new_sr), (rtp_source_get_new_rb): * gst/rtpmanager/rtpsource.h: Add GObject properties for various things. Don't leak the bye reason. 2007-11-22 09:08:27 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited amount of time and thus has no max_latency ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): jitterbuffer can buffer an unlimited amount of time and thus has no max_latency requirements. 2007-11-02 21:45:38 +0000 Ole André Vadla Ravnås gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798). Original commit message from CVS: Patch by: Ole André Vadla Ravnås * gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798). 2007-10-09 10:01:39 +0000 Laurent Glayal gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990. Original commit message from CVS: Patch by: Laurent Glayal * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_class_init): Fix memleak. Fixes #484990. 2007-10-08 17:46:45 +0000 Jan Schmidt gst/: Fix compiler warnings shown by Forte. Original commit message from CVS: * gst/librfb/rfbbuffer.c: (rfb_buffer_new_and_alloc): * gst/librfb/rfbbuffer.h: * gst/librfb/rfbdecoder.c: (rfb_socket_get_buffer): * gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_chain): * gst/nsf/nes6502.c: (nes6502_execute): * gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps): * gst/real/gstrealvideodec.c: (open_library): * gst/real/gstrealvideodec.h: * gst/rtpmanager/gstrtpsession.c: (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink): Fix compiler warnings shown by Forte. 2007-10-08 10:39:35 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (get_pt_map), (gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init): Fix caps refcounting for payload maps. When clearing payload maps, also clear sessions and streams payload maps. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps), (gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain), (find_pad_for_pt): Implement clearing the payload map. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink): Forward flush events instead of leaking them. * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_rtcp_sink_event): Correctly refcount events before pushing them. 2007-10-05 17:26:14 +0000 Wim Taymans gst/rtpmanager/rtpsession.c: When reconsidering RTCP timeouts, set the next timeout against the last report time inst... Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout), When reconsidering RTCP timeouts, set the next timeout against the last report time instead of the current clock time so that we don't end up reconsidering forever. 2007-10-05 12:07:37 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead of popping it off, which allows us to grea... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Only peek at the tail element instead of popping it off, which allows us to greatly simplify things when the tail element changes. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_recv_rtp_sink): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_sink_event): Forward FLUSH events instead of leaking them. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Remove the tail-changed callback in favour of a simple boolean when we insert a buffer in the queue. Add method to peek the tail of the buffer. 2007-10-02 10:27:45 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Remove some old unused variables. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (apply_offset), (gst_rtp_jitter_buffer_loop): Remove some old unused variables. Don't add the latency to the skew corrected timestamp, latency is only used to sync against the clock. Improve debugging. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_reset_skew), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Handle case where server timestamp goes backwards or wildly jumps by temporarily pausing the skew correction. Improve debugging. 2007-09-28 14:51:58 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (free_client): Fix crasher in dispose. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Handle cases where input buffers have no timestamps so that no clock skew can be calculated, in this case interpollate timestamps based on rtp timestamp and assume a 0 clock skew. 2007-09-28 11:17:35 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now in the lower level object. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query): Remove jitter correction code, it's now in the lower level object. Use new -core method for doing a peer query. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Move jitter correction to the lowlevel jitterbuffer. Increase the max window size. When filling the window, already start estimating the skew using a parabolic weighting factor so that we have a much better startup behaviour that gets more accurate with the more samples we have. Increase the default weighting factor for the steady state to get smoother timestamps. 2007-09-26 20:08:28 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose), (gst_rtp_bin_finalize): Fix cleanup crasher. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Dynamically adjust the skew calculation window so that we calculate it over a period of around 2 seconds. 2007-09-20 14:34:57 +0000 Wim Taymans gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. Fixes #478566. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: (on_ssrc_active), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_event_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_ssrc_active), (rtp_session_process_rb): * gst/rtpmanager/rtpsession.h: Add notification of active SSRCs to various RTP elements. Fixes #478566. 2007-09-17 02:01:41 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Link to the right pads regardless of which one was created first in the ssrc demuxer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsource.c: (calculate_jitter): Improve debugging. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links): * gst/rtpmanager/gstrtpssrcdemux.h: Fix race in creating the RTP and RTCP pads when a new SSRC is detected. 2007-09-16 19:40:31 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Use lock to protect variable. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): Use lock to protect variable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop): Reconstruct GST timestamp from RTP timestamps based on measured clock skew and sync offset. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_set_tail_changed), (rtp_jitter_buffer_set_clock_rate), (rtp_jitter_buffer_get_clock_rate), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek): * gst/rtpmanager/rtpjitterbuffer.h: Measure clock skew. Add callback to be notfied when a new packet was inserted at the tail. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Remove clock skew detection, it's move to the jitterbuffer now. 2007-09-15 18:48:03 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session): Also set NTP base time on new sessions. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Use the right lock to protect our variables. Fix some comment. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_getcaps_send_rtp), (gst_rtp_session_chain_send_rtp), (create_send_rtp_sink): Implement getcaps on the sender sinkpad so that payloaders can negotiate the right SSRC. 2007-09-12 21:23:47 +0000 Wim Taymans gst/rtpmanager/: Various leak fixes. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (free_session), (get_client), (free_client), (gst_rtp_bin_associate), (free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose), (gst_rtp_bin_finalize): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_finalize): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose): * gst/rtpmanager/rtpsession.c: (rtp_session_finalize): * gst/rtpmanager/rtpsession.h: Various leak fixes. 2007-09-12 18:04:32 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so that we can generate better Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base), (gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp): Calculate and configure the NTP base time so that we can generate better NTP times in SR packets. Set caps on new ghostpad. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Clean debug statement. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: Add ntp-ns-base property to convert running_time to NTP time. Handle NEWSEGMENT events on send and recv RTP pads so that we can calculate the running time and thus NTP time of the packets. Simplify getting the current NTP time using the pipeline clock. Implement internal links functions. Use the buffer timestamp to calculate the NTP time instead of the clock. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links), (gst_rtp_ssrc_demux_src_query): * gst/rtpmanager/gstrtpssrcdemux.h: Implement internal links function. Calculate the diff between different streams, this might be used later to get the inter stream latency. * gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp): Simple cleanup. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr): Make the clock skew window a little bigger. Apply the clock skew to all buffers, not just one with a new timestamp. Calculate and debug sender clock drift. Use extended last timestamp to interpollate for SR reports. 2007-09-04 15:23:34 +0000 Tim-Philipp Müller gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562). Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562). 2007-09-03 21:19:34 +0000 Wim Taymans gst/rtpmanager/: Updated example pipelines in docs. Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp): * gst/rtpmanager/gstrtpbin.h: Updated example pipelines in docs. Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync. Set the default latency correctly. Add some more points where we can get caps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Add ts-offset property to control timestamping. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Various cleanups. Feed rtpsession manager with NTP time based on pipeline clock when handling RTP packets and RTCP timeouts. Perform all RTCP with the system clock. Set caps on RTCP outgoing buffers. * gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc), (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain): * gst/rtpmanager/gstrtpssrcdemux.h: Also demux RTCP messages. * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_rb), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_send_bye), (session_start_rtcp), (session_report_blocks), (session_cleanup), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Remove the get_time callback, the GStreamer part will feed us with enough timing information. Split sync timing and RTCP timing information. Factor out common RB handling for SR and RR. Send out SR RTCP packets for lip-sync. Move SR and RR packet info generation to the source. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_new_sr), (rtp_source_get_new_rb), (rtp_source_get_last_sr): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Use caps on incomming buffers to get timing information when they are there. Calculate clock scew of the receiver compared to the sender and adjust the rtp timestamps. Calculate the round trip in sources. Do SR and RR calculations in the source. 2007-08-31 15:26:14 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release buffers from the jitterbuffer so that we can h... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop): Use extended timestamp to release buffers from the jitterbuffer so that we can handle the rtp wraparound correctly. 2007-08-29 16:56:27 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Improve Comments. * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_parse_caps), (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps), (gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink), (create_send_rtp_sink): Also parse the sink caps for clock-rate instead of only relying on the result of the signal. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): Make sure we fetch the clock rate for payloads we are sending out so that we can use it for SR reports. 2007-08-29 01:22:43 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the session manager so that it can generate ... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_event_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: Distribute synchronisation parameters to the session manager so that it can generate correct SR packets for lip-sync. * gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time), (rtp_session_set_timestamp_sync), (session_start_rtcp): * gst/rtpmanager/rtpsession.h: Add methods for setting sync parameters. Set correct RTP time in SR packets using the sync params. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Record last RTP <-> GST timestamp so that we can use them to convert NTP to RTP timestamps in SR packets. 2007-08-28 20:30:16 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map): Add some more advanced example pipelines. * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_send_rtcp): Add some debug and FIXME. Release LOCK when performing session cleanup. * gst/rtpmanager/rtpsession.c: (session_report_blocks): Add some debug. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (rtp_source_send_rtp): Make sure we always send RTP packets with the session SSRC. 2007-08-27 21:17:21 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer latency into account. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query): When synchronizing buffers, take peer latency into account. Don't try to add our latency to invalid peer max latency values. 2007-08-23 21:39:58 +0000 Tim-Philipp Müller Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPF... Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.signals: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPFoo types that farsight registers (luckily GType names are case sensitive). Should finally fix #430664. 2007-08-21 17:18:29 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: When drop-on-latency is set but we have no latency configured, just push the buf... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_set_property): When drop-on-latency is set but we have no latency configured, just push the buffer as fast as possible. Fix typo in comment. 2007-08-21 16:04:47 +0000 Wim Taymans gst/rtpmanager/rtpjitterbuffer.*: Fix undefined overflow prone ts_diff handling. Original commit message from CVS: * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_get_ts_diff): * gst/rtpmanager/rtpjitterbuffer.h: Fix undefined overflow prone ts_diff handling. 2007-08-16 11:40:16 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Fix EOS handling. Convert some DEBUG into WARNINGs. Pause task when flushing. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink): Use system clock for RTCP session management timeouts. * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout): Release the session lock when emiting signals. 2007-08-13 06:16:40 +0000 Stefan Kost gst/rtpmanager/rtpjitterbuffer.c: Include stdlib. Original commit message from CVS: * gst/rtpmanager/rtpjitterbuffer.c: Include stdlib. 2007-08-10 17:16:53 +0000 Wim Taymans gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some... Original commit message from CVS: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/async_jitter_queue.c: * gst/rtpmanager/async_jitter_queue.h: * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init), (rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize), (rtp_jitter_buffer_new), (compare_seqnum), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop), (rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets), (rtp_jitter_buffer_get_ts_diff): * gst/rtpmanager/rtpjitterbuffer.h: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some bugs. * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (create_session), (gst_rtp_bin_class_init), (create_recv_rtp), (create_send_rtp): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property): * gst/rtpmanager/gstrtpsession.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup): Use new jitterbuffer code. Expose some new signals in preparation for handling EOS. 2007-07-18 07:35:32 +0000 Stefan Kost Add stdlib include (free, atoi, exit). Original commit message from CVS: * examples/app/appsrc_ex.c: * examples/switch/switcher.c: * ext/neon/gstneonhttpsrc.c: * ext/timidity/gstwildmidi.c: * ext/x264/gstx264enc.c: * gst/mve/mveaudioenc.c: (mve_compress_audio): * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/spectrum/demo-audiotest.c: * gst/spectrum/demo-osssrc.c: * sys/dvb/gstdvbsrc.c: Add stdlib include (free, atoi, exit). 2007-06-22 20:23:18 +0000 Jens Granseuer gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.). Original commit message from CVS: Patch by: Jens Granseuer * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_push_sorted): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): * gst/switch/gstswitch.c: (gst_switch_chain): Build fixes for gcc-2.9x (no mid-block variable declarations etc.). Fixes #450185. 2007-05-28 16:37:47 +0000 Wim Taymans Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream), (gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpclient.c: (create_stream), (gst_rtp_client_request_new_pad): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpssrcdemux.c: Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664. 2007-05-23 13:08:52 +0000 Wim Taymans Document stuff. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_clear_pt_map): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (rtcp_thread), (gst_rtp_session_clear_pt_map): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init): Document stuff. Add clear-pt-map action signal where needed. 2007-05-15 13:29:53 +0000 Wim Taymans gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps. Original commit message from CVS: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): We always use fixed caps. 2007-05-15 03:45:45 +0000 David Schleef gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around. 2007-05-14 15:28:36 +0000 Wim Taymans gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing. Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_set_flushing_unlocked): Fix leak when flushing. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: Add clear-pt-map signal. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop): Init clock-rate to -1 to mark unknow clock rate. Fix flushing. 2007-05-10 14:02:07 +0000 Stefan Kost gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde... Original commit message from CVS: * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows, gst_qtdemux_loop_state_movie, gst_qtdemux_loop, qtdemux_parse_segments, qtdemux_parse_trak): * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth, rtp_session_get_rtcp_bandwidth, rtp_session_get_cname, rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone, rtp_session_get_location, rtp_session_get_tool, rtp_session_process_bye, session_report_blocks): * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp, rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb): More format arg fixing (spotted by Ali Sabil ). * gst/switch/Makefile.am: Add require libraries(spotted by Ali Sabil ). 2007-05-10 12:38:49 +0000 Stefan Kost * gst/rtpmanager/async_jitter_queue.c: gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a... Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, async_jitter_queue_set_low_threshold, async_jitter_queue_length_ts_units_unlocked, async_jitter_queue_unref_and_unlock, async_jitter_queue_unref, async_jitter_queue_lock, async_jitter_queue_push, async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted, async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop, async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked, async_jitter_queue_set_flushing_unlocked, async_jitter_queue_unset_flushing_unlocked): Format arg fix (spotted by Ali Sabil ) 2007-05-09 11:24:22 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): Pass queries upstream. 2007-05-04 12:32:27 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): Add some debug info. * gst/rtpmanager/rtpsession.c: (rtp_session_init), (rtp_session_send_rtp): Store real user name in the session. 2007-04-30 13:41:30 +0000 Wim Taymans gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block. Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads), (async_jitter_queue_pop_intern_unlocked): Fix the case where the buffer underruns and does not block. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): Rename RTCP send pad, like in the session manager. Allow getting an RTCP pad for receiving even if we don't receive RTP. fix handling of send_rtp_src pad. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): When no pt map could be found, fall back to the sinkpad caps. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Fix pad names. * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_create_source), (rtp_session_process_sr), (rtp_session_send_rtp), (session_start_rtcp): * gst/rtpmanager/rtpsession.h: Unlock session when performing a callback. Add callbacks for the internal session object. Fix sending of RTP packets. first attempt at adding NTP times in the SR packets. Small debug and doc improvements. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): Update stats for SR reports. 2007-04-29 14:46:27 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Remove debug. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp): Remove debug. * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr), (rtp_session_process_sdes), (calculate_rtcp_interval), (rtp_session_next_timeout), (session_report_blocks): * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval): Improve debugging Fix interval for BYE/RTCP packets. 2007-04-27 15:09:12 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider): Move reconsideration code to the rtpsession object. Simplify timout handling and add reconsideration. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (rtp_session_set_callbacks), (obtain_source), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_bye), (rtp_session_process_rtcp), (calculate_rtcp_interval), (rtp_session_send_bye), (rtp_session_next_timeout), (session_start_rtcp), (session_report_blocks), (session_cleanup), (session_sdes), (session_bye), (is_rtcp_time), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Handle timeout of inactive sources and senders. Implement BYE scheduling. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (rtp_source_process_sr), (rtp_source_get_last_sr), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: Add members to check for timeouts. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter), (rtp_stats_calculate_bye_interval): * gst/rtpmanager/rtpstats.h: Use RFC algorithm for calculating the reporting interval. 2007-04-25 16:38:03 +0000 Wim Taymans gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (rtcp_thread): Implement forward and reverse reconsideration. * gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources), (rtp_session_get_num_active_sources), (rtp_session_process_sr), (session_report_blocks): * gst/rtpmanager/rtpsession.h: Small cleanups. 2007-04-25 15:48:46 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable. Original commit message from CVS: reviewed by: * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Make default jitterbuffer latency configurable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Debuging cleanups. 2007-04-25 13:19:36 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_change_state): Report NO_PREROLL when going to PAUSED. * gst/rtpmanager/gstrtpsession.c: (rtcp_thread): Don't send RTCP right before we are shutting down. * gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp), (rtp_session_process_sr), (session_report_blocks), (rtp_session_perform_reporting): Improve report blocks. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_last_sr), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Cleanups, add methods to access stats. 2007-04-25 08:30:48 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: fix for pad name change Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_rtcp): fix for pad name change * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate): Fix for renamed methods. * gst/rtpmanager/rtpsession.c: (rtp_session_init), (rtp_session_finalize), (rtp_session_set_cname), (rtp_session_get_cname), (rtp_session_set_name), (rtp_session_get_name), (rtp_session_set_email), (rtp_session_get_email), (rtp_session_set_phone), (rtp_session_get_phone), (rtp_session_set_location), (rtp_session_get_location), (rtp_session_set_tool), (rtp_session_get_tool), (rtp_session_set_note), (rtp_session_get_note), (source_push_rtp), (obtain_source), (rtp_session_add_source), (rtp_session_get_source_by_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_sdes), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_reporting_interval), (session_report_blocks), (session_sdes), (rtp_session_perform_reporting): * gst/rtpmanager/rtpsession.h: Prepare for implementing SSRC sampling. Create SSRC for the session. Add methods to set the SDES entries. fix accounting of senders/receivers. Implement SR/RR/SDES RTCP reporting. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr): * gst/rtpmanager/rtpsource.h: Implement extended sequence number. * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval): * gst/rtpmanager/rtpstats.h: Rename some fields. 2007-04-21 19:21:49 +0000 Tim-Philipp Müller gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment. Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_finalize): Don't use GLib-2.10 API, we only require GLib 2.8 at the moment. 2007-04-18 18:58:53 +0000 Wim Taymans configure.ac: Disable rtpmanager for now because it depends on CVS -base. Original commit message from CVS: * configure.ac: Disable rtpmanager for now because it depends on CVS -base. * gst/rtpmanager/Makefile.am: Added new files for session manager. * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (pt_map_requested), (new_ssrc_pad_found): Some cleanups. the session manager can now also request a pt-map. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate), (gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_send_rtcp_src), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpsession.h: We can ask for pt-map now too when the session manager needs it. Hook up to the new session manager, implement the needed callbacks for pushing data, getting clock time and requesting clock-rates. Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to be send to clients. Add code to start and stop the thread that will schedule RTCP through the session manager. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (rtp_session_set_property), (rtp_session_get_property), (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks), (rtp_session_set_bandwidth), (rtp_session_get_bandwidth), (rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth), (source_push_rtp), (source_clock_rate), (check_collision), (obtain_source), (rtp_session_add_source), (rtp_session_get_num_sources), (rtp_session_get_num_active_sources), (rtp_session_get_source_by_ssrc), (rtp_session_get_source_by_cname), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_process_app), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_rtcp_interval), (rtp_session_produce_rtcp): * gst/rtpmanager/rtpsession.h: The advanced beginnings of the main session manager that handles the participant database of RTPSources, SSRC probation, SSRC collisions, parse RTCP to update source stats. etc.. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_finalize), (rtp_source_new), (rtp_source_set_callbacks), (rtp_source_set_as_csrc), (rtp_source_set_rtp_from), (rtp_source_set_rtcp_from), (push_packet), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_process_bye), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb): * gst/rtpmanager/rtpsource.h: Object that encapsulates an SSRC and its state in the database. Calculates the jitter and transit times of data packets. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter): * gst/rtpmanager/rtpstats.h: Various stats regarding the session and sources. Used to calculate the RTCP interval. 2007-04-13 09:20:55 +0000 Wim Taymans gst/rtpmanager/: Protect lists and structures with locks. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found), (create_recv_rtp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_request_new_pad): Protect lists and structures with locks. Return FLOW_OK from RTCP messages for now. 2007-04-12 08:18:32 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (gst_rtp_bin_class_init), (pt_map_requested): Emit pt map requests and cache results. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_get_clock_rate), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Emit request-pt-map signals. 2007-04-11 13:49:54 +0000 Wim Taymans gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers. Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers. * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (clock_rate_request), (create_stream), (gst_rtp_bin_class_init), (pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp): * gst/rtpmanager/gstrtpbin.h: Prepare for caching pt maps. Connect to signals to collect pt maps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: Add request_clock_rate signal. Use scale insteat of scale_int because the later does not deal with negative numbers. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_chain): * gst/rtpmanager/gstrtpptdemux.h: Implement request-pt-map signal. 2007-04-10 09:14:07 +0000 Wim Taymans gst/rtpmanager/: Added custom marshallers for signals. Original commit message from CVS: * gst/rtpmanager/.cvsignore: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpbin-marshal.list: Added custom marshallers for signals. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: Prepare for emiting pt map signals. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init): Fix signals. 2007-04-06 12:28:29 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Provide a clock. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_provide_clock): * gst/rtpmanager/gstrtpbin.h: Provide a clock. 2007-04-06 12:07:30 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_rtcp): Fix pad template name parsing. 2007-04-05 16:10:24 +0000 Wim Taymans gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Add some debug and comments. Fix double unref() in error cases. 2007-04-05 13:54:23 +0000 Wim Taymans gst/rtpmanager/gstrtpbin.*: Add debugging category. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (find_session_by_id), (create_session), (find_stream_by_ssrc), (create_stream), (gst_rtp_bin_class_init), (new_payload_found), (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp): * gst/rtpmanager/gstrtpbin.h: Add debugging category. Added RTPStream to manage stream per SSRC, each with its own jitterbuffer and ptdemux. Added SSRCDemux. Connect to various SSRC and PT signals and create ghostpads, link stuff. * gst/rtpmanager/gstrtpmanager.c: (plugin_init): Added rtpbin to elements. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Fix caps and forward GstFlowReturn * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src), (gst_rtp_session_request_new_pad): Add debug category. Add event handling * gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc), (create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.h: Add debug category. Add new-pt-pad signal. 2007-04-04 10:23:15 +0000 Wim Taymans gst/rtpmanager/: Added simple SSRC demuxer. Original commit message from CVS: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc), (create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init), (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event), (gst_rtp_ssrc_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.h: Added simple SSRC demuxer. 2007-04-03 11:35:39 +0000 Wim Taymans gst/rtpmanager/: Some more ghostpad magic. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (find_session_by_id), (create_session), (gst_rtp_bin_base_init), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: Some more ghostpad magic. 2007-04-03 09:51:13 +0000 Wim Taymans gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly. Original commit message from CVS: * gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly. 2007-04-03 09:13:17 +0000 Wim Taymans Add RTP session management elements. Still in progress. Original commit message from CVS: * configure.ac: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new), (signal_waiting_threads), (async_jitter_queue_ref), (async_jitter_queue_ref_unlocked), (async_jitter_queue_set_low_threshold), (async_jitter_queue_set_high_threshold), (async_jitter_queue_set_max_queue_length), (async_jitter_queue_get_g_queue), (calculate_ts_diff), (async_jitter_queue_length_ts_units_unlocked), (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref), (async_jitter_queue_lock), (async_jitter_queue_unlock), (async_jitter_queue_push), (async_jitter_queue_push_unlocked), (async_jitter_queue_push_sorted), (async_jitter_queue_push_sorted_unlocked), (async_jitter_queue_insert_after_unlocked), (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop), (async_jitter_queue_pop_unlocked), (async_jitter_queue_length), (async_jitter_queue_length_unlocked), (async_jitter_queue_set_flushing_unlocked), (async_jitter_queue_unset_flushing_unlocked), (async_jitter_queue_set_blocking_unlocked): * gst/rtpmanager/async_jitter_queue.h: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_finalize), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property), (gst_rtp_bin_change_state), (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream), (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init), (gst_rtp_client_class_init), (gst_rtp_client_init), (gst_rtp_client_finalize), (gst_rtp_client_set_property), (gst_rtp_client_get_property), (gst_rtp_client_change_state), (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad): * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_base_init), (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps), (gst_jitter_buffer_sink_setcaps), (free_func), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_activate_push), (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt), (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init), (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init), (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain), (gst_rtp_pt_demux_getcaps), (find_pad_for_pt), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (gst_rtp_session_change_state), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src), (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad): * gst/rtpmanager/gstrtpsession.h: Add RTP session management elements. Still in progress. 2009-08-10 13:30:23 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: push mode; cater for chunk padding 2009-08-04 19:45:43 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: only use stream's pad after having checked it exists 2009-08-04 13:38:09 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: sprinkle some more GST_DEBUG_FUNCPTR 2009-08-04 13:36:36 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: post error message if no pads to push EOS event on 2009-08-04 11:39:59 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: fix typo in warning message 2009-08-04 11:39:39 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: fix some buffer ref handling 2009-08-04 11:37:16 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: do not exceed maximum number of supported streams 2009-08-04 11:35:18 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs 2009-08-04 11:32:27 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: verify size of INFO LIST to satisfy subsequent expectations 2009-07-29 15:25:38 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: check video stream framerate against avi header frame duration The former might be bogus in silly cases, and the latter seems to carry more weight. 2009-08-04 12:16:13 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: streamline stream duration calculation 2009-07-03 14:04:13 +0200 Edward Hervey * ext/raw1394/gstdv1394src.c: dv1394src: Fix element for live usage... which has been broken for 2 years :( This is a live source, therefore: * Use GST_FORMAT_TIME as the default format * set_timestamp to True * properly implement query latency. This allows expected live usage like : playbin2 uri=dv:// 2009-08-09 09:43:41 +0200 Edward Hervey * ext/raw1394/gstdv1394src.c: raw1394: Remove unneeded variable 2009-08-09 09:43:29 +0200 Edward Hervey * gst/matroska/matroska-demux.c: matroska: remove dead assignments 2009-08-09 09:43:00 +0200 Edward Hervey * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: rtp: Remove dead assignments and resulting unneeded variables. 2009-08-10 09:53:28 +0200 Sebastian Dröge * configure.ac: * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/md5.c: * ext/wavpack/md5.h: wavpack: Use GLib GChecksum instead of our own MD5 implementation This requires GLib 2.16 but that version is already required by core anyway. 2009-08-08 00:47:48 -0300 Thiago Santos * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroska: Adds support to muxing/demuxing WMA Adds support for muxing wma audio family and fixes demuxing of wma family in matroskademux. matroskademux was broken because it missed codec_data. 2009-08-06 20:15:17 -0300 Thiago Santos * gst/matroska/matroska-mux.c: matroskamux: adds support for wmv family Adds support to WMV1, WMV2, WMV3 and other family formats that are signaled by the 'format' field in the caps (i.e. WVC1). Partially fixes #576378 2009-08-09 14:19:42 +0100 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2src: if max == min width/height put an int in the probed caps, not an int range Fixes #560033. 2009-08-09 13:58:07 +0100 Tim-Philipp Müller * sys/osxaudio/gstosxaudiosrc.c: osxaudiosrc: if max_channels == min_channels, use an int instead of an int range in the caps 2009-08-09 12:52:17 +0200 LoneStar * gst/id3demux/id3v2frames.c: id3demux: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8 Fixes bug #499242. 2009-08-09 01:29:50 +0100 Tim-Philipp Müller * configure.ac: configure: bump core/base requirements to latest release To avoid confusion. 2009-08-09 01:27:01 +0100 Tim-Philipp Müller * tests/check/elements/flvmux.c: check: fix flvmux unit test on big endian machines flvmux only accepts raw audio in little endian, but audiotestsrc produces audio in the native endianness, which makes linking between audiotestsrc and flvmux fail on big endian machines. Add an audioconvert element in between the two to fix this. 2009-02-15 18:49:44 +0000 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: matroska: add kate subtitle support to matroska muxer and demuxer See #525743. 2009-08-07 16:51:45 +0100 Tim-Philipp Müller * gst/id3demux/id3v2.3.0.html: id3demux: add ID3 v2.3 spec as well 2009-08-07 16:42:39 +0100 Tim-Philipp Müller * gst/id3demux/id3v2frames.c: id3demux: sizes in ID3 v2.3 are unlikely to be sync-safe integers In ID3 v2.3 compressed frames will have a 4-byte data length indicator after the frame header to indicate the size of the decompressed data. This integer is unlikely to be a sync-safe integer for v2.3 tags, only in v2.4 it's sync-safe. 2009-08-07 16:36:55 +0100 Tim-Philipp Müller * gst/id3demux/id3tags.c: id3demux: fix typo in debug message 2009-08-07 16:02:23 +0100 Tim-Philipp Müller * gst/id3demux/id3tags.c: * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: * tests/check/elements/id3demux.c: * tests/files/Makefile.am: * tests/files/id3-588148-unsynced-v24.tag: id3demux: fix parsing of unsync'ed ID3 v2.4 tags and frames Reversing the unsynchronisation seems to work slightly differently for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame sizes in the frame header, so the unsynchronisation is applied to the whole frame data including all the frame headers. v2.4 frames have sync-safe sizes, however, so the unsynchronisation only needs to be applied to the actual frame data, and it seems that's what's being done as well. So we need to undo the unsynchronisation on a per-frame basis for v2.4 tags for things to work properly. Fixes extraction of coverart/images from APIC frames in ID3 v2.4 tags (#588148). Add unit test for this as well. 2009-08-06 21:24:14 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Use SOUP_METHOD_GET instead of "GET" string Fixes bug #590970. 2009-08-06 13:00:59 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: set the default slave method to skew Set the default slave method to the much better skew algorithm. This is the default in the new base class but we override this here as well for the upcomming release. 2009-08-06 10:20:34 +0100 Tim-Philipp Müller * ext/pulse/pulsesrc.c: pulsesrc: fix compilation with --disable-gst-debug 2009-08-03 18:59:32 +0200 Wim Taymans * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: use array instead of queue 2009-08-03 18:55:19 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: push NALs only after SPS/PPS parse complete (bytestream) buffer for SPS/PPS before pushing NALs. Fixes #564501. 2009-08-04 14:44:36 +0200 Sebastian Dröge * sys/v4l2/v4l2_calls.h: v4l2: Directly use GST_PTR_FORMAT for printing caps with the LOG_CAPS macro 2009-08-04 11:17:17 +0200 Edward Hervey * gst/rtp/gstrtpqdmdepay.c: rtpqdm2depay: Fix debug statement. 2009-08-04 09:32:07 +0200 Sebastian Dröge * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2_calls.h: v4l2: Remove some OMAP specific hacks They require special build flags and are not useful in general. 2009-08-04 09:22:29 +0200 Rob Clark * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/v4l2src_calls.c: v4l2sink: change where buffers get dequeued It seems to cause strange occasional high latencies (almost 200ms) when dequeuing buffers from _buffer_alloc(). It is simpler and seems to work much better to dqbuf from the same thread that is queuing the next buffer. 2009-08-04 09:14:20 +0200 Rob Clark * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: v4l2: Add v4l2sink element This also does the following changes: (1) pull the bufferpool code out into gstv4l2bufferpool.c, and make a bit more generic so it can be used both for v4l2src and v4l2sink (2) move some of the device probing/configuration/caps stuff into gstv4l2object.c so it does not have to be duplicated between v4l2src and v4l2sink Fixes bug #590280. 2009-08-04 07:07:45 +0200 Sebastian Dröge * tests/check/Makefile.am: flvmux: Enable unit test now that it passes 2009-08-03 21:21:39 +0200 Edward Hervey * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsv3vdepay.c: rtpqdm2depay,rtpsv3vdepay: Add debugging category. 2009-08-03 21:22:48 +0200 Edward Hervey * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpqdmdepay.h: rtpqdm2depay: Handle gaps in incoming packets. Whenever we see a gap, we flush the temporary packets (but not the adapter). If we had some data temporarily stored it will be outputted (the sound will sound a bit garbled... but that's how it sounds on MacOSX :) 2009-08-03 19:01:07 +0200 Edward Hervey * gst/rtp/gstrtpqdmdepay.c: rtpqdmdepay: Fix CRC calculation and remove commented code. 2009-08-02 13:42:12 +0200 Edward Hervey * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpqdmdepay.h: rtp: New QDM2 rtp depayloader. Reverse-engineered by comparing: * A rtp hinted file provided by DarwinStreamingServer * The output procued by DSS for that same file Also used various streaming sources available on the internet to fine-tune the code. The header/codec_data extraction methods are from FFMpeg (LGPL). 2009-08-03 21:24:44 +0200 Edward Hervey * gst/rtp/gstrtpsv3vdepay.c: rtpsv3vdepay: Properly fill codec_data and cleanup code a bite more. 2009-08-03 19:02:17 +0200 Edward Hervey * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtpsv3vdepay.h: rtpsv3vdepay: Only output buffers once we're configured. 2009-08-03 19:02:00 +0200 Edward Hervey * gst/rtp/gstrtpsv3vdepay.c: rtpsv3vdepay: Add more encoding-name variants 2009-08-03 20:08:33 +0200 Sebastian Dröge * tests/check/elements/flvmux.c: flvmux: Fix unit test to correctly handle request pads Request pads are removed by the element instance in PAUSED->READY so we need to re-request pads for every run and link them again. Last fix for bug #590447. 2009-08-03 20:08:00 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Fix writing of the index for < 128 buffers Partially fixes bug #590447. 2009-08-03 20:07:00 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Fix resetting of the element Reset the have_video/have_audio flags and make sure to properly release the request pads. Partially fixes bug #590447. 2009-08-03 18:13:46 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't add non-utf8 chars to structures 2009-08-03 18:02:31 +0200 Luc Deschenaux * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegdepay.h: jpegdepay: use attributes for extra properties Use some of the SDP attributes when they are present to specify the output dimension and framerate. This allows us to receive jpeg frames larger than 2040 width/height. Fixes #564437 2009-08-03 18:01:27 +0200 Wim Taymans * gst/rtp/README: RTP docs: update with attributes in caps 2009-08-03 17:21:44 +0200 Luc Deschenaux * gst/rtsp/gstrtspsrc.c: rtspsrc: put all SDP attributes on caps Put the SDP attributes on the caps too so that they can be used by depayloaders. See #564437 2009-08-03 13:32:12 +0200 Jonathan Tellier * ext/pulse/pulsesrc.c: pulsesrc: initialize the probe with the server When creating a new probe, pass the server instead of the device string. fixes #590401 2009-08-02 11:44:03 +0100 Tim-Philipp Müller * gst/udp/gstmultiudpsink.c: multiudpsink: don't do things with side-effects inside g_return_val_if_fail() Someone might compile this code with -DG_DISABLE_ASSERT some day. 2009-08-01 21:39:30 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: don't do logic within g_assert() statements Otherwise that code will just be expanded to nothing when compiled -DG_DISABLE_ASSERT (PS: why is mainloop_start() called in the init function and not when changing state to READY?) 2009-08-01 17:07:42 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: send newsegment event when operating push-based and unframed For some reason flac doesn't call our metadata callback when we operate in push mode with unframed input, but that's where we set up the newsegment event (since that's where we'd get the duration from the stream info header), so we didn't send a newsegment event at all in this case. Hack around this by storing a generic newsegment event for now which will be used if we don't replace it with a better one that includes the duration. 2009-08-01 16:48:36 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: small cleanups Remove some callback indirections which are no longer needed because there's only one decoder object type now. Also remove unused variable. 2009-08-01 15:22:49 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: use gst_adapter_copy() to avoid unnecessary buffer merges gst_adapter_peek() will merge buffers as needed, which we can avoid here since we're doing a memcpy anyway and then flush the copied data from the adapter right away. 2009-08-01 00:00:41 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: repair some broken indenting 2009-08-01 12:19:41 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/flvmux.c: checks: add basic unit test for flvmux, but disable it for now Basic unit test for flvmux. Fails miserably, hence disabled for now. 2009-07-31 23:28:12 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/flvdemux.c: * tests/files/Makefile.am: * tests/files/pcm16sine.flv: check: add basic unit test for flvdemux In particular, test re-use of flvdemux in both pull and push mode (see #583030). 2009-07-31 20:25:17 +0100 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: fix invalid write caused by using sizeof("string") as length sizeof("foo") includes the string's NUL-terminator in the size returned, but we're writing strings here with an explicit size at the beginning and no NUL-terminator. In most cases using sizeof("foo") as length in memcpy is not harmful, but it is where the string goes right at the end of our buffer to write, since we don't allocate space for that NUL terminator. 2009-07-27 18:44:45 +0200 Edward Hervey * ext/soup/gstsouphttpsrc.c: soup: Use "GET" instead of SOUP_METHOD_GET. Fixes build with libsoup-2.7.* This is due to a quality API change in libsoup 2.7. SOUP_METHOD_* are now integers and not strings... they could have changed the names. 2009-07-30 17:57:53 +0300 Stefan Kost * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: jpeg: use longer macro names to not clash with some stupid windows defines libjpeg headers pull some windows system inlcudes (on windows) that contain a define for DEFAULT_QUALITY. 2009-07-29 14:31:48 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Fix last commit and improve readability 2009-07-24 19:04:31 +0400 Руслан Ижбулатов * gst/avi/gstavidemux.c: Fixed the fix for TIME->DEFAULT conversion. Fixes bug #578052 again. 2009-07-29 13:38:03 +0200 Edward Hervey * gst/rtp/gstrtpsv3vdepay.c: rtpsv3depay: Fix width/height calculation, bring up to marginal rank. Based on documentation found on http://wiki.multimedia.cx/ 2009-07-29 12:13:20 +0200 Wim Taymans * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: pulse: conditionally compile newer stuff configured_sink/source_usec in the timing_info is only since 0.9.11 so conditionally compile this information. fixes #590038 2009-07-28 18:29:07 +0200 Wim Taymans * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: cleanups Keep track of the paused state of the source and leave the read function when paused. don't wait for a latency update when the delay is not yet known but simply return 0 instead of blocking. Keep track of the corked state of the stream. Fix the state changes. 2009-07-28 16:11:18 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulsesrc: set maxlength always to -1 2009-07-28 15:53:57 +0200 Wim Taymans * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc; cleanups, report real latency Add some more debug info Avoid some type casts Report the real latency to the application. 2009-07-28 16:11:36 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: when scanning for 0xff marker ends, ensure desired result Otherwise, any non 0xff byte at end of data would be mistaken for a tag byte, and in case of a frame_len 0 tag subsequently lead to an infinite loop. 2009-07-28 00:30:43 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: adds support to wma 2009-07-28 00:07:15 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: adds support to wmv 2009-07-27 21:34:22 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Downgrade warning message to debug 2009-07-27 11:51:39 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: avoid using ivalid stream indexes when we get an invalid stream index from pulse because we were just starting, avoid using it for getting and setting the volume. Fixes #589365 2009-07-24 19:38:07 +0200 Sebastian Dröge * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effectv: Don't allow caps changes for some effectv filters These filters use information from previous frames to generate the current frame and a caps change will make the effect start from the beginning again. 2009-07-24 19:37:09 +0200 Sebastian Dröge * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: warptv: Make the sine table global instead of having it in every instance 2009-07-24 10:47:44 +0300 Stefan Kost * ext/jpeg/gstjpegenc.c: jpeg: make encoder work with libjpeg v7 We have to specify do_fancy_downsampling = FALSE in the encoder with did not exist before. 2009-07-24 00:42:33 +0300 Stefan Kost * common: Automatic update of common submodule From fedaaee to 94f95e3 2009-07-23 12:06:27 +0200 Sebastian Dröge * ext/flac/gstflacdec.c: flacdec: Implement SEEKING query Fixes bug #589423. 2009-07-22 11:16:06 +0100 Colin Guthrie * ext/pulse/pulsesink.c: pulsesink: Fix a couple error messages that mentioned incorrect function names. Fixes #589459. 2009-07-23 11:50:16 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: * gst/flv/gstflvparse.c: flvdemux: Implement SEEKING query Also add some more query types to the answer of the query type function. Fixes bug #589424. 2009-07-21 19:46:55 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: fix intermittent FLAC__STREAM_DECODER_ABORTED errors when seeking When seeking in a local flac file (ie. operating pull-based), the decoder would often just error out after the loop function sees a DECODER_ABORTED status. This, however, is the read callback's way of telling our loop function that pull_range failed and streaming should stop, in this case because of the flush-start event that the seek handler pushed upstream from the seeking thread. Handle this slightly better by storing the last flow return from pull_range, so the loop function can evaluate it properly when it encounters a DECODER_ABORTED and take the right action. Fixes #578612. 2009-07-21 10:07:00 +0300 Stefan Kost * gst/interleave/interleave.c: interleave: fix indenting and upgrade two debugs to warnings. Fix newlines in variable decls. Change two debugs to become warnings as they indicate that things will not work. 2009-07-21 10:04:36 +0300 Stefan Kost * ext/jpeg/gstjpeg.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpeg: code cleanups for encoder Remove some disabled code in encoder. Try #if 0'ed code and add comments about why it is disabled. Move idct-method enum to jpeg.c and use in both encoder and decoder. Add idct-method property to encoder. 2009-07-21 07:50:46 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Answer SEEKING queries in the original format 2009-07-21 01:12:44 +0200 Josep Torra * gst/udp/gstudpnetutils.c: udputils: initialize struct content with 0. Fixes some random crashes. 2009-07-20 19:09:19 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: set some values to their defaults Set the minreq and maxlength buffer attributes to -1 to let puleseaudio select a sensible value. 2009-07-20 19:04:09 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: don't wait for posted message We can't wait for the ENTER/LEAVE messages to be be posted because the base class sometimes calls the start method with the object lock, which would block the message posting. Instead, just assume that the message will be posted soon and continue. We'll have to fix this in the base class. 2009-07-20 18:11:33 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: use relative seeks Use relative seeks because I was told that absolute seeks don't work. 2009-07-20 16:52:19 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Implement SEEKING query 2009-07-20 08:07:13 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Add support for ARGB/BGRA input Note that videotestsrc outputs 100% transparent video which will result in white output from cairorender. 2009-07-17 13:22:57 +0100 Elaine Xiong * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: v4l2: Fix v4l2src on OpenSolaris The v4l2 driver for USB webcams on OpenSolaris does not support select() calls. Detect when select() fails, and skip polling the device afterward, which restores the pre 0.10.14 behaviour on OpenSolaris. Signed-off-by: Jan Schmidt 2009-07-17 11:22:06 +0100 Jan Schmidt * tests/check/elements/.gitignore: * tests/examples/v4l2/.gitignore: gitignore: Ignore some new binaries 2009-07-17 13:49:21 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-cairo.xml: * ext/cairo/gstcairorender.c: cairorender: Add to the documentation 2009-07-17 13:42:49 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Return not-negotiated if we have no caps 2009-07-17 13:41:19 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: * ext/cairo/gstcairorender.h: cairorender: Fix caps and colorspace handling 2009-07-17 13:30:02 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Use correct mimetypes for PDF and SVG 2009-07-17 13:24:28 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Remove pull mode, it only adds complexity but not advantages 2009-07-16 21:55:31 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Fix caps negotiation and cairo surface creation 2009-07-16 21:42:21 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: cairorender: Correctly set srccaps 2009-07-16 21:31:43 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: * ext/cairo/gstcairorender.h: cairorender: Move instance/class struct definitions to the header 2009-07-16 21:30:00 +0200 Sebastian Dröge * ext/cairo/gstcairorender.c: * ext/cairo/gstcairorender.h: cairorender: Add Lutz' copyright to the file header 2009-07-16 21:27:45 +0200 Lutz Mueller * ext/cairo/Makefile.am: * ext/cairo/gstcairo.c: * ext/cairo/gstcairorender.c: * ext/cairo/gstcairorender.h: cairo: Add cairo-based PDF/PS/SVG encoder element Fixes bug #331420. 2009-07-16 20:44:40 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: Optionally write a PADDING block The size of the PADDING block is specified by a new "padding" property. Fixes bug #588483. 2009-07-16 19:35:44 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Only assume seekability if the server provides Content-Length Previously seekability way always assumed until the first seek actually failed. Now we assume that all servers are not seekable unless they provide a Content-Length header. If a seek fails after that we continue to assume no seekability. Fixes bug #585576. 2009-07-16 15:14:43 +0200 Arnout Vandecappelle * ext/soup/gstsouphttpsrc.c: souphttpsrc: don't try to authenticate if no username/password is set. 2009-07-16 17:10:21 +0200 Sebastian Dröge * gst/effectv/gstwarp.c: effectv: Chain up finalize to the parent class in warptv Fixes a memory leak. 2009-07-16 12:55:49 +0200 Sebastian Dröge * tests/check/Makefile.am: * tests/check/pipelines/effectv.c: effectv: Add unit test for all effectv elements 2009-07-16 12:17:32 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-videomixer.xml: effectv: Add new effectv elements to the docs 2009-07-15 14:37:19 +0200 Sebastian Dröge * gst/effectv/Makefile.am: * gst/effectv/gsteffectv.c: * gst/effectv/gstripple.c: * gst/effectv/gstripple.h: effectv: Add rippletv element This produces a water ripple effect on the video input, based on motion or a rain drop algorithm. Kindly relicensed to LGPL2+ by Kentaro Fukuchi . Fixes bug #588695. 2009-07-12 15:42:35 +0200 Sebastian Dröge * gst/effectv/Makefile.am: * gst/effectv/gsteffectv.c: * gst/effectv/gststreak.c: * gst/effectv/gststreak.h: effectv: Add streaktv effect filter element This combines the StreakTV and BaltanTV filters from the effectv project. Kindly relicensed to LGPL2+ by Kentaro Fukuchi . Fixes bug #588368. 2009-07-12 12:31:15 +0200 Sebastian Dröge * gst/effectv/gstaging.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: effectv: Fix processing on big endian architectures 2009-07-12 11:52:04 +0200 Sebastian Dröge * gst/effectv/Makefile.am: * gst/effectv/gsteffectv.c: * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: effectv: Add radioactv effect filter This filter adds a radiation-like motion blur effect to the video stream. Kindly relicensed to LGPL2+ by Kentaro Fukuchi . Fixes bug #588359. 2009-07-12 11:26:57 +0200 Sebastian Dröge * gst/effectv/gstop.c: * gst/effectv/gstop.h: effectv: Make the optv threshold property an uint 2009-07-12 10:39:02 +0200 Sebastian Dröge * gst/effectv/Makefile.am: * gst/effectv/gsteffectv.c: * gst/effectv/gstop.c: * gst/effectv/gstop.h: effect: Add optv effect filter from the effectv project This filter binarizes input frames and combines them with various optical pattern. Kindly relicensed to LGPL2+ by Kentaro Fukuchi . Fixes bug #588349. 2009-07-03 05:11:26 -0400 Olivier Crête * ext/pulse/pulsesink.c: pulsesink: Emit stream-status leave message Fixes #587695 2009-07-03 05:06:45 -0400 Olivier Crête * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Emit stream-status enter message Emit stream-status messages for the pulse thread. Don't use our own GCond for signaling but simply use the pulse mainloop mechanisms for synchronisation. See #587695 2009-07-14 18:15:59 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: debug the latency update values 2009-07-14 16:12:55 +0200 Wim Taymans * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: pulsesink: add 24bit sample formats Add check for pulseaudio 0.9.15 and enable 24bits samples in that case. 2009-07-13 12:23:37 -0400 Olivier Crête * common: Automatic update of common submodule From 5845b63 to fedaaee 2009-07-13 17:53:25 +0200 Marc Leeman * gst/rtp/gstrtpmpvpay.c: mpvpay: Rework the timestamping Rework the timestamping in the mpv payloader so that the timestamps are more accurate. Fixes #587680 2009-07-03 08:47:12 +0200 Filippo Argiolas * configure.ac: * tests/examples/Makefile.am: * tests/examples/v4l2/Makefile.am: * tests/examples/v4l2/probe.c: v4l2src: add a simple test case for device probing 2009-07-03 08:38:43 +0200 Filippo Argiolas * configure.ac: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2object.c: v4l2src: optional support for device probing with gudev Enumerate v4l2 devices using gudev if available. Fixes bug #583640. 2009-07-10 19:54:25 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Random cleanup 2009-07-10 19:54:13 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Send queries to the master pad by default instead of all pads 2009-07-10 19:34:41 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend_rgb.c: * gst/videomixer/videomixer.c: videomixer: Add RGB, BGR, xRGB, RGBx, xBGR, BGRx support 2009-07-10 17:43:07 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Clean up debugging a bit 2009-07-10 17:25:48 +0200 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Remove some redundant checks and error out immediately if not negotiated Also stop leaking the output buffer in some error cases. 2009-07-10 17:23:03 +0200 Sebastian Dröge * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Remove the calculate_frame_size() function and use libgstvideo instead 2009-06-30 15:13:44 +0200 Edward Hervey * gst/videomixer/videomixer.c: videomixer: Remove unused link/unlink pad methods 2009-06-30 12:43:04 +0200 Edward Hervey * gst/videomixer/blend_i420.c: videomixer: I420 mode: Add fast path for 0.0 and 1.0 alpha If the source alpha is 0.0, we take nothing. If the source alpha is 1.0, we overwrite everything. 2009-06-30 12:40:02 +0200 Edward Hervey * gst/videomixer/blend_i420.c: videomixer: I420 blending : Fix main algorithm. When blending a source layer with an alpha of 'a' on top of another destination layer we take the sum of: * 'a' percent of the source layer * (100 - 'a') percent of the destination layer (the remainder) 2009-06-30 12:39:19 +0200 Edward Hervey * gst/videomixer/blend_i420.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: * gst/videomixer/videomixerpad.h: videomixer: Make debugging category global to all the code. 2009-06-29 19:23:41 +0200 Edward Hervey * gst/videomixer/videomixer.c: videomixer: improve readability of debugging statements. 2009-07-08 13:38:53 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: do not leak timeout message 2009-07-09 07:14:23 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avi: Don't forward NEWSEGMENT events from upstream New ones are generated later and simply forwarding them can result in NEWSEGMENT events of different format going downstream. Fixes bug #587983. 2009-07-08 18:19:45 +0200 Sebastian Dröge * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_i420.c: videomixer: Make checker pattern lookup table constant 2009-07-08 18:17:48 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend_bgra.c: * gst/videomixer/videomixer.c: videomixer: Add support for ARGB And clean up the caps parsing. 2009-07-08 15:17:41 +0200 Benjamin Gaignard * gst/udp/gstudpnetutils.c: udp: Initialize pointer to NULL Otherwise we're calling free() with some random memory address in error cases. Fixes bug #587982. 2009-07-07 16:35:24 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: sprinkle some more const 2009-07-07 15:57:55 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: perform some more (careful) data buffering Once buffering has started (with an mdat atom), continue buffering until moov atom is reached, which handles cases with multiple mdat atoms. Also keep adapter/offset better in sync with upstream and fix some debug statements. Fixes #587426. 2009-07-06 10:40:31 +0200 Philip J�genstedt * gst/avi/gstavidemux.c: avidemux: Replace deprecated GST_DISABLE_DEBUG with correct macro. Fixes #587826 2009-07-01 13:07:48 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: error out instead of dividing by 0 Error out if timescale is 0. 2009-07-01 09:32:42 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: Revert "qtdemux: Make sure we don't blacklist streams by wrongly comparing their" This reverts commit 5503a59a5779b67451d8a271000181790ee76bc7. Reverting this since it causes regressions with a lot of sample files I have, all of which worked fine with the last -good release (#586891). 2009-06-30 15:54:47 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: comment out unused structure 2009-06-30 13:12:09 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: more size checks, and use g_try_new0() instead of g_new0() Whenever we alloc something based on a user-supplied size, we should really use g_try_new(), otherwise we can easily be made to abort by passing a ridiculously large number to us for allocing. Fixes problems with some fuzzed files. 2009-06-29 18:58:33 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: guard against bogus atom sizes and short reads Check the possibly 64-bit atom size more carefully before casting it to an int and passing it to gst_pad_pull_range(), otherwise we might end up pulling 0 bytes, getting an empty buffer as requested and dereferencing not available data whilst thinking we actually asked for and got 0x1000000000000 bytes. Similar fix for push mode operation where neededbytes ends up being 0 bytes, which makes us assert. Fixes crash with broken or fuzzed file (NB #122378). 2009-06-29 16:52:41 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: use 0x prefix when logging numbers in hex 2009-07-01 08:40:40 +0200 Edward Hervey * ext/flac/gstflacdec.c: flacdec: Don't send empty string tags 2009-06-30 21:35:37 +0400 LRN * gst/udp/gstmultiudpsink.c: Don't use sendmsg()-dependent code on Windows Fixes #585842 2009-06-30 15:59:20 +0200 Wim Taymans * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/alaw.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/law/mulaw.c: law: fix caps and negotiation Fix the caps to include the depth (instead of width twice) in the caps of audio/x-raw-int. Fix negotiation to not only copy the rate/channels of the first structure. 2009-06-30 14:48:09 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: include "1.0=100%" in volume and change upper limit Upper volume limmit was 1000. That appear unneceasrily high. It would also cause sever distortion if accidentialy used. Now its 10 (~ +15db) which is also in sync with volume and playbin2. 2009-06-29 15:39:43 +0200 Wim Taymans * ext/pulse/pulsesrc.c: pulse: some more trivial cleanups 2009-06-29 15:38:49 +0200 Wim Taymans * ext/pulse/pulsemixer.c: pulse: trivial cleanups 2009-06-29 15:20:31 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: clear ringbuffer when asked to Since we map the ringbuffer to the pulseaudio internal ringbuffer, flush the pulseaudio buffer when we are asked to clear the ringbuffer. This avoids some leftover audio after a seek. 2009-06-26 15:00:14 +0100 Jan Schmidt * autogen.sh: autogen.sh: Actually do the 'echo -n' -> printf change. 2009-06-26 14:40:14 +0100 Jan Schmidt * autogen.sh: autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01] Check for more automake command variants. Use printf instead of 'echo -n' for portability 2009-06-26 13:42:09 +0100 Jan Schmidt * common: Automatic update of common submodule From f810030 to 5845b63 2009-06-26 13:19:04 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: don't process track_num/track_count tags with a 0 value Number/count values of 0 mean they're not set. Don't put those in the taglist. 2009-06-25 18:51:12 +0100 Tim-Philipp Müller * sys/waveform/gstwaveformsink.c: waveformsink: use 'guint8' instead of 'byte' to fix compilation with MSVC8 We need a cast here for pointer arithmetic to work correctly, but some MSVC versions don't seem to like 'byte', so use guint8 here. Hopefully fixes #585361. 2009-06-25 19:39:37 +0300 Stefan Kost * sys/v4l2/v4l2_calls.c: v4l2src: set structs to zero before using them in ioctls This fixes valgrind warnings. 2009-06-25 13:23:40 +0200 Julien Moutte * gst/qtdemux/qtdemux.c: qtdemux: Make sure we don't blacklist streams by wrongly comparing their duration with entire clip duration. 2009-06-25 13:18:14 +0200 Krzysztof Błaszkowski * gst/rtsp/gstrtpdec.c: rtpdec: fix some buffer leaks 2009-06-25 08:11:09 +0200 Edward Hervey * gst/flv/gstflvparse.c: flvparse: Add missing break in switch/case. 2009-06-25 08:10:38 +0200 Edward Hervey * gst/flv/gstflvdemux.c: flvdemux: Remove unused variable, hint branch likeliness, add comments. 2009-06-25 08:09:57 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Removed unused variable 2009-06-25 07:41:07 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Remove dead assignments and unused variables. Also add branch likeliness macros. 2009-06-25 07:40:26 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Fix uninitialized variables. Fixes build on macosx 2009-06-24 17:43:25 +0300 Stefan Kost * ext/soup/gstsouphttpsrc.c: souphttpsrc: free memory in finalize finalize is called only once. no need to clear pointers there. dispose is for unreffing. 2009-06-24 15:14:14 +0100 Jan Schmidt * common: Automatic update of common submodule From 6ab11d1 to f810030 2009-06-08 14:46:48 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: short-circuit gst_avi_demux_src_convert() when parsing the index Don't call gst_avi_demux_src_convert() for each single index entry. Not only do we already have the pointer to the stream context, we also know the formats we want to convert from and to already, so we may just as well use optimised conversion routines that bypass some of the checks and lookups made in gst_avi_demux_src_convert(). 2009-06-17 16:39:36 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Another round of G_*LIKELY micro-optimisations. 2009-06-17 16:20:25 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Take last sample duration for dummy segment calculation. This fixes the cases where files without EDL wouldn't output their last buffer. 2009-06-24 12:36:31 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Sprinkle branch likeliness macros over the code. 2009-06-23 16:54:32 +0200 Edward Hervey * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: raw1394: sprinkle branch likeliness macros accross the code. 2009-06-14 10:36:17 +0200 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: Add GST_MEMDUMP statements for unknown atoms. This is to help developers track down and implement unhandled atoms faster. 2009-06-23 17:51:32 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Remove the interlaced field from the output caps if deinterlacing is enabled 2009-06-23 17:48:47 +0200 Sebastian Dröge * gst/deinterlace/tvtime/greedyh.c: deinterlace: Copy the correct line from correct place in the history 2009-06-23 16:35:36 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: use same protocols after redirect After a redirect we want to use the same protocols that we were using for the current url. 2009-06-23 15:35:37 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: don't leak cover art 2009-06-23 14:10:10 +0100 Tim-Philipp Müller * gst/udp/gstudpnetutils.c: udp: fix compiler warning about EAI_ADDRFAMILY getting redefined in some cases Include the header from where we include all the system headers with the socket stuff before we try to define EAI_ADDRFAMILY ourselves, otherwise we define it ourselves and then get a compiler warning if a system header defines it as well without guarding against it being defined already. 2009-06-23 14:39:56 +0200 Wim Taymans * gst/matroska/matroska-ids.h: matroska: and the new headers too 2009-06-23 14:32:43 +0200 Wim Taymans * gst/matroska/matroska-demux.c: matroske: fix compiler error change gpointer to guint8 * for codec_state and codec_priv as some functions operate on those types and it avoids breaking strict-aliasing rules. 2009-06-23 12:42:33 +0200 Wim Taymans * gst/matroska/matroska-demux.c: matroskademux: avoid leaking buffers Don't leak buffers when resyncing to a keyframe. Avoid leaking buffers when exiting the loop on error conditions. Add some more debug info. Fixes #585911 2009-06-22 15:56:58 +0300 Stefan Kost * sys/v4l2/gstv4l2src.c: v4l2: open/close the device in READY This allows to query the device in READY. Before one need to switch it to PAUSED and that also starts streaming. 2009-06-20 15:41:44 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: qtdemux: use GST_MEMDUMP 2009-06-19 00:16:41 +0100 Tim-Philipp Müller * gst/apetag/Makefile.am: * gst/apetag/gstapedemux.c: apedemux: add container-format tag Use pbutils here because the string is translated. 2009-06-19 00:15:51 +0100 Tim-Philipp Müller * gst/id3demux/Makefile.am: * gst/id3demux/gstid3demux.c: id3demux: add container-format tag Using pbutils here because the string is translated. 2009-06-18 23:51:52 +0100 Tim-Philipp Müller * ext/dv/gstdvdemux.c: dvdemux: post container-format tag Also merge the two almost identical _add_*_pad() functions into one. 2009-06-18 23:43:49 +0100 Tim-Philipp Müller * ext/dv/gstdvdemux.c: dvdemux: don't screw up first audio buffer Query the audio format, esp. dvdemux->num_channels, before we use that variable to allocate the initial buffer. That way we don't accidentally push a zero-sized buffer as first audio buffer. 2009-06-18 23:38:30 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: multipartdemux: post container-format tag 2009-06-18 23:37:11 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: post container-format tags 2009-06-18 23:36:28 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: post container-format tag 2009-06-18 23:35:29 +0100 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: post container-format tags 2009-06-21 17:13:43 +0200 Sebastian Dröge * gst/audiofx/audioamplify.c: audioamplify: Fix integer overflows on 32 bit architectures 2009-06-21 09:50:54 +0200 Kipp Cannon * gst/audiofx/audioamplify.c: audioamplify: Don't declare a loop index static The previous patch to add support for additional sample formats possibly introduced a reentrancy bug: a variable used for a loop index was declared static. This patch fixes that, and also adds a "/* *INDENT-ON* */" annotation following the macro block. (I don't know what the annotation is for, but the adder, where I copied this from, has it). 2009-06-19 22:37:27 +0200 Sebastian Dröge * gst/audiofx/audioamplify.c: audioamplify: Fix off-by-one in wrap-positive mode 2009-06-19 22:20:45 +0200 Kipp Cannon * gst/audiofx/audioamplify.c: * gst/audiofx/audioamplify.h: audioamplify: Add noclip method and support for more formats Fixes bug #585828 and #585831. 2009-06-19 21:46:41 +0200 Koop Mast * gst/udp/gstudpnetutils.h: udp: Fix build on FreeBSD Fixes bug #586397. 2009-06-19 18:12:27 +0200 Ognyan Tonchev * tests/check/elements/rtp-payloading.c: tests: add unit tests for buffer-list payloaders See #585559 2009-06-19 18:00:35 +0200 Ognyan Tonchev * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: rtpmp4vpay: add support for buffer-list See #585559 2009-06-19 17:57:12 +0200 Ognyan Tonchev * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpjpegpay.h: rtpjpegpay: add support for buffer-lists See #585559 2009-06-19 17:53:32 +0200 Ognyan Tonchev * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: add support for buffer-lists See #585559 2009-06-18 11:54:22 +0200 Wim Taymans * gst/udp/gstudpnetutils.c: udputils: don't free invalid memory As spotted by benjiG in IRC. don't free invalid memory when getaddrinfo failed. 2009-06-17 17:48:31 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulseink: don't leak device_description don't leak the device_description. some cleanups. 2009-06-19 14:44:40 +0100 Tim-Philipp Müller * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: update .po files for sunaudiomixer string changes 2009-06-18 16:58:26 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: streaming; adjust sizes to cater for padding in chunks 2009-06-17 11:54:53 +0200 Mark Nauwelaerts * gst/avi/gstavidemux.c: avidemux: streaming mode; handle data chunks grouped in rec lists. Fixes #567983. 2009-06-10 12:36:50 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: map some tags to COMPOSER rather than ARTIST 2009-06-10 12:34:43 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix some 3GP tag extraction (keywords, genre, location) 2009-06-09 15:36:50 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: extract pixel-aspect-ratio information 2009-06-17 07:14:09 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Fix leaking of the Matroska TITLE element 2009-06-16 20:38:42 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst/effectv/gstaging.c: * gst/effectv/gstaging.h: * gst/effectv/gstdice.c: * gst/effectv/gstdice.h: * gst/effectv/gstedge.c: * gst/effectv/gstedge.h: * gst/effectv/gstquark.c: * gst/effectv/gstquark.h: * gst/effectv/gstrev.c: * gst/effectv/gstrev.h: * gst/effectv/gstshagadelic.c: * gst/effectv/gstshagadelic.h: * gst/effectv/gstvertigo.c: * gst/effectv/gstvertigo.h: * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: effectv: Add basic documentation for the effectv elements 2009-06-16 20:16:13 +0200 Sebastian Dröge * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gsteffectv.h: * gst/effectv/gstquark.c: * gst/effectv/gstshagadelic.c: effectv: Define the fast PRNG function at a central place 2009-06-16 20:13:35 +0200 Sebastian Dröge * gst/effectv/Makefile.am: * gst/effectv/gstaging.c: * gst/effectv/gstaging.h: * gst/effectv/gstdice.c: * gst/effectv/gstdice.h: * gst/effectv/gstedge.c: * gst/effectv/gstedge.h: * gst/effectv/gsteffectv.c: * gst/effectv/gsteffectv.h: * gst/effectv/gstquark.c: * gst/effectv/gstquark.h: * gst/effectv/gstrev.c: * gst/effectv/gstrev.h: * gst/effectv/gstshagadelic.c: * gst/effectv/gstshagadelic.h: * gst/effectv/gstvertigo.c: * gst/effectv/gstvertigo.h: * gst/effectv/gstwarp.c: * gst/effectv/gstwarp.h: effectv: Move type definitions into separate headers This is needed for the docs later. 2009-06-16 19:41:02 +0200 Sebastian Dröge * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effectv: Remove get_unit_size implementations The default on from GstVideoFilter handles this already. 2009-06-16 14:54:34 +0100 Tim-Philipp Müller * configure.ac: configure: bump core/base requirements to git Need git core for basesink bufferlist additions; -base requirement bumped gratuitously. 2009-06-16 15:25:54 +0200 Wim Taymans * tests/check/elements/udpsink.c: tests: add some debug, send newsegment 2009-06-16 15:06:50 +0200 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: add debug line for the socket 2009-06-16 15:06:25 +0200 Wim Taymans * tests/check/pipelines/flacdec.c: tests: turn g_print into debug 2009-06-16 15:04:15 +0200 Ognyan Tonchev * gst/udp/gstmultiudpsink.c: * tests/check/Makefile.am: * tests/check/elements/udpsink.c: multiudpsink: add support for buffer lists Add support for BufferList and add a unit test. Fixes #585842 2009-06-16 00:02:42 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: souphttpsrc: reset session state when stopping Increases the chances that the element is actually reusable. 2009-06-15 23:49:48 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: souphttpsrc: log response and request headers and fix some broken indenting 2009-06-15 22:40:35 +0200 Wim Taymans * gst/rtp/gstrtpmp4gdepay.c: mp4gdepay: guess constantDuration better Do a better job at guessing the constantDuration parameter when it is not present in the caps. Fixes #585205 2009-06-15 21:09:47 +0200 Sebastian Dröge * gst/effectv/gstwarp.c: warptv: Clean up warptv element and fix some minor bugs and leaks 2009-06-15 20:53:23 +0200 Sebastian Dröge * gst/effectv/gstvertigo.c: vertigotv: Clean up vertigotv element and fix some minor bugs and leaks 2009-06-15 20:38:01 +0200 Sebastian Dröge * gst/effectv/gstdice.c: dicetv: Use guint8 instead of char (which can be signed or unsigned) 2009-06-15 20:36:39 +0200 Sebastian Dröge * gst/effectv/gstshagadelic.c: shagadelictv: Use guint8/gint8 instead of char (which can be signed or unsigned) 2009-06-15 20:31:30 +0200 Sebastian Dröge * gst/effectv/gstshagadelic.c: shagadelictv: Clean up element and free all memory in finalize 2009-06-15 20:21:58 +0200 Sebastian Dröge * gst/effectv/gstrev.c: revtv: Clean up revtv element 2009-06-15 20:07:42 +0200 Sebastian Dröge * gst/effectv/gstquark.c: quarktv: Simplify some code 2009-06-15 20:07:10 +0200 Sebastian Dröge * gst/effectv/gstquark.c: quarktv: Use the input data if a NULL buffer is chosen instead of the value 0 2009-06-15 20:00:43 +0200 Sebastian Dröge * gst/effectv/gstquark.c: quarktv: Fix setting the planes property of quarktv Setting it to a value<16 would cause crashes before because current_plane was set to the old number of planes-1. Also fix calculations for non-2^n planes values. 2009-06-15 17:50:41 +0200 Sebastian Dröge * gst/effectv/gstquark.c: quarktv: Clean up the quarktv element 2009-06-15 17:39:20 +0200 Sebastian Dröge * gst/effectv/gsteffectv.c: effectv: Make elements list constant 2009-06-15 17:37:53 +0200 Sebastian Dröge * gst/effectv/gstedge.c: edgetv: Clean up edgetv element and fix memory leak 2009-06-15 17:21:58 +0200 Sebastian Dröge * gst/effectv/gstdice.c: dicetv: Clean up dicetv element and fix some smaller issues This fixes a memory leak (the dice map) and a crash when setting the square-bits property before caps are set. 2009-06-15 17:20:21 +0200 Sebastian Dröge * gst/effectv/Makefile.am: * gst/effectv/gstaging.c: agingtv: Actually use GstController for syncing the properties to timestamps 2009-06-15 17:03:38 +0200 Sebastian Dröge * gst/effectv/gstaging.c: agingtv: Export some more agingtv properties via GObject properties 2009-06-15 15:06:56 +0200 Sebastian Dröge * gst/effectv/gstaging.c: agingtv: General cleanup and updating of copyright Also make the scratch-lines property exported via a GObject property and initialize/reset the internal state correctly. 2009-06-15 15:05:58 +0200 Sebastian Dröge * gst/effectv/gstaging.c: agingtv: Store and update state inside the instance struct This makes the coloraging effect and pits effect visible. 2009-06-15 15:51:32 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: ref custom ring buffer class and type in class_init Hack around thread-safety issues in GObject and our racy _get_type() functions (we could easily fix the _get_type() functions, but we still need to hack around the GObject class races until we require a newer GLib version, I think). 2009-06-14 19:19:19 +0100 Tim-Philipp Müller * ext/dv/demo-play.c: * tests/old/examples/Makefile.am: * tests/old/examples/level/Makefile.am: * tests/old/examples/level/README: * tests/old/examples/level/demo.c: * tests/old/examples/level/plot.c: * tests/old/examples/switch/.gitignore: * tests/old/examples/switch/Makefile.am: * tests/old/examples/switch/switcher.c: Remove a few old example apps from the 0.8 days Some have been replaced by newer ones, others are demoing elements that don't exist any longer (not in -good anyway), and others have not been touched in many years and it seem pointless to keep them around. Removing these files makes sure we don't have any code in our repository that uses Gtk+ symbols which are to be removed for GNOME3, and as such will make some script that greps for this kind of stuff give us a clean bill of code health. Fixes #585757. 2009-06-13 21:02:45 -0400 Olivier Crête * common: * gst/rtp/gstrtpsirenpay.c: rtpsirenpay: Remove deprecated symbol Patch by: Luis Menina 2009-06-13 10:43:55 +0200 Marvin Schmidt * tests/check/Makefile.am: tests: Don't run the flacdec test if the plugin isn't built. Fixes #585630 2009-06-12 16:06:28 +0200 Patrick Radizi * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add RTP blocksize functionality Add property to make the client suggest a blocksize to the server. Fixes #585549 2009-06-11 22:30:06 +0200 Wim Taymans * gst/rtp/README: rtp: update README, fix some typos, mention gstrtpbin 2009-06-11 19:10:53 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: handle border cases in resampler 2009-06-11 13:32:22 +0100 Jan Schmidt * common: * docs/Makefile.am: * docs/plugins/Makefile.am: * docs/upload.mak: docs: Bump common. Use upload-doc.mak instead of upload.mak Remove the local copy of upload.mak in favour of using the shared upload-doc.make in common/ 2009-06-11 11:39:25 +0100 Jan Schmidt * gst/goom/goom_config_param.h: * gst/videomixer/videomixer.c: docs: Quieten a couple more docs warnings 2009-06-11 11:27:26 +0100 Jan Schmidt * gst/matroska/lzo.c: docs: Remove gtk-doc comment marker These comment blocks aren't gtk-doc comments and cause annoying noise in the docs build. 2009-06-11 10:05:32 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Implement upstream negotation 2009-06-10 21:47:40 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Improve debugging and clean up some code 2009-06-10 14:55:18 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Clip buffers to the current segment if possible 2009-06-10 14:45:06 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Clean up includes and clean up order of instance struct fields 2009-06-10 16:09:56 -0400 Olivier Crête * gst/rtp/gstrtph263pay.h: rtph263pay: Default to doing A, B and C modes, not only A 2009-06-10 09:56:11 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix QoS calculations The diff is a signed integer, not an unsigned one of course. In modes other than GST_DEINTERLACE_ALL every frame has twice the duration of the field duration. 2009-06-09 14:13:31 -0400 Olivier Crête * gst/rtp/gstrtpsirenpay.c: rtpsirenpay: Put the bitrate in the RTP caps The MS code seems to require the bitrate to interoperate and draft-ietf-avt-rtp-g7221-00 also has it. 2009-06-09 19:55:36 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: Implement basic QoS This change is based on Tim's QoS implementation for jpegdec. 2009-06-09 19:29:51 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: Directly proxy events/queries to the peer pads This removes some overhead introduced by the default handlers that need to iterate over the other pads. 2009-06-09 10:38:52 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: debug_memdump() unknown tags. Refactor junk parsing code. This makes life slightly easier when debugging avi files. 2009-06-08 08:21:43 +0200 Edward Hervey * gst/rtp/Makefile.am: rtp: Don't forget to dist the headers for the CELT (de)payloaders. 2009-06-07 20:54:06 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: Revert "Revert "qtdemux: fill timestamp table completely"" This reverts commit 9f022c8a8503c2ce0fa617fdb50e41706dd412f5. Sorry, I was thinking about the wrong module. 2009-06-07 20:49:50 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: Revert "qtdemux: fill timestamp table completely" This reverts commit 790b050fc5302cae89cddcd23b258093967d05a9. I forgot we were frozen. 2009-06-07 20:46:45 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: fill timestamp table completely When there are less timestamps that there are samples, fill up the sample table with the last know timestamp. This situation can happen when the last sample does not decode and doesn't need a timestamp. We however calculate the total track length using the last sample timestamp so we need to have something sensible in there. Fixes #585056 2009-06-07 13:37:04 +0200 Wim Taymans * gst/wavparse/gstwavparse.c: wavparse: handle LIST INFO of 0 size Handle LIST INFO chunks of 0 size instead of causing errors. Fixes #584981 2009-06-07 13:24:45 +0200 Wim Taymans * gst/wavparse/gstwavparse.c: Revert "wavparse: Remove dead assignments, move variable to where it's needed." Reverts commit 44256a78f8dd79a91f3bb2ab7c3aa623c097bb8a and use the result in error reporting so that we can see what's going on. 2009-06-05 18:55:02 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltdepay.h: celtdepay: add CELT depayloader 2009-06-05 15:30:51 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpceltpay.h: rtpceltpay: add CELT RTP payloader 2009-06-05 16:54:48 +0100 Jan Schmidt * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixeroptions.c: * sys/sunaudio/gstsunaudiomixertrack.c: sunaudio: Fix switch setting on some devices. Add debug. Fix a FIXME. Fix the setting of toggle switches on some broken audio drivers which report that no audio ports are settable by ignoring the mod_port field there. Add some debug statements. Fix a FIXME now that Good relies on a new enough gst-plugins-base. 2009-06-04 12:27:19 +0100 Jan Schmidt * sys/sunaudio/Makefile.am: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixeroptions.c: * sys/sunaudio/gstsunaudiomixeroptions.h: * sys/sunaudio/gstsunaudiomixertrack.c: * sys/sunaudio/gstsunaudiomixertrack.h: sunaudio: Support new flags for options and actions Use new audio mixer flags added in Base 0.10.23 to expose flags and options on the SunAudio devices. Fixes: #583593 Patch By: Brian Cameron Patch By: Garrett D'Amore 2009-05-15 11:50:38 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: deinterlace: First try to handle DVD still frames correctly This helps a bit with bug #582740 but still doesn't make it work. 2009-06-04 17:37:03 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: only notify if all checks passed Replace goto done: with return, as those are checks when we don't want to flag a pending notify. 2009-06-04 15:19:05 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: set the right state on rtpbin We need to set the state of gstrtpbin to the same state as our source elements. This fixes fallback to TCP again. 2009-06-03 18:23:53 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: check pointer before accessing Move existing check a few lines up, so that we check before accessing fields. 2009-06-03 18:21:12 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: rename gst_pulse_sink_get_time to gst_pulsesink_get_time Rename internal method for consistency. 2009-06-03 18:19:22 +0300 Stefan Kost * ext/pulse/pulsesink.c: pulsesink: use values from pa_stream_get_buffer_attr() We were putting the requested values back into ringbuffer spec, instead of using the queried values. 2009-06-02 19:32:21 +0200 Wim Taymans * gst/rtp/gstrtpvrawpay.c: vrawpay: trim output buffers Remove the leftover unused bytes in the output buffer. Fixes #584613 2009-06-02 19:30:30 +0200 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: vrawdepay: fix parsing of sampling field commit a12d9a80f225be97b3674b1a0506ac66544dbf49 broke the parsing of the sampling. 2009-05-27 17:06:34 +0100 Jan Schmidt * ext/libpng/gstpngdec.c: pngdec: Avoid possible overflow in calculations A malformed (or simply huge) PNG file can lead to integer overflow in calculating the size of the output buffer, leading to crashes or buffer overflows later. Fixes SA35205 security advisory. 2009-06-02 00:48:00 +0100 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: some more logging - dump header packets Also, the final fixing up of the headers is expected and not something we should warn about. 2009-06-02 00:37:15 +0100 Tim-Philipp Müller * ext/flac/gstflacenc.c: flacenc: never ever pass values >36bits to _set_total_samples_estimate() Let's be paranoid and make sure we never pass a number that takes up more than 36 bits to _set_total_samples_estimate(), since libFLAC expects all the other bits to be zero, and if this is not the case neighbouring fields in the global stream info header may get messed up inadvertently, so that flac -d refuses to decode the stream. See #584455. 2009-06-01 22:33:02 +0200 Thomas Vander Stichele * ext/flac/gstflacenc.c: Address bad FLAC sample length encoding of #5844455 Commit df707c666433a78d3878af6f055698d5756226c4 introduced an obvious bug in the sample length calculation, using the wrong macro for conversion. 2009-06-01 11:58:21 -0700 Brian Cameron * gst/deinterlace/tvtime/mmx.h: deinterlace: Fix spurious colons in asm code Fixes #584174. Signed-off-by: David Schleef 2009-06-01 00:40:55 +0100 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: skip JUNK chunks in data section in streaming mode Skip JUNK tags in streaming mode as well instead of EOSing prematurely. Fixes #564100. 2009-05-28 14:01:17 +0200 Sebastian Dröge * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/videomixer.c: videomixer: Don't use // comments 2009-05-28 13:56:15 +0200 Sebastian Dröge * gst/videomixer/blend_bgra.c: videomixer: Fix background blitting when a color mode is selected with BGRA 2009-05-28 13:54:14 +0200 Sebastian Dröge * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Some cleanup and fix the calculation of the frame size in bytes 2009-05-28 13:35:52 +0200 Sebastian Dröge * gst/videomixer/blend_i420.c: videomixer: Fix I420 blending to actually do something For this we a) implement the checkers filling and b) actually blend the src/dest by using the src alpha value from the pad. 2009-05-28 13:14:13 +0200 Sebastian Dröge * gst/videomixer/blend_bgra.c: videomixer: Fix ARGB blending to actually work 2009-05-28 13:04:51 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend_bgra.c: videomixer: Blend BGRA ourselves instead of using Cairo 2009-05-28 12:55:16 +0200 Alex Ugarte * gst/videomixer/Makefile.am: * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Add support for blending BGRA and AYUV Fixes bug #577017. 2009-05-28 12:39:46 +0200 Ghislain 'Aus' Lacroix * gst/equalizer/gstiirequalizer.c: equalizer: Use floating point arithmetic internally for the int16 mode By using int32 arithmetic we will introduce distortions as the IIR filter is very sensitive to rounding errors. Fixes bug #580214. 2009-05-28 10:55:16 +0100 Christian Schaller * gst-plugins-good.spec.in: Update spec file with latest plugins 2009-05-26 17:19:08 +0100 Jan Schmidt * common: Automatic update of common submodule From 888e0a2 to c572721 2009-05-26 16:20:35 +0300 Stefan Kost * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: cleanup and commenting Remove newlines inserted by gst-indent once. Remove unused var from instance struct. Add comments. Add another #define for default property value. 2009-05-06 12:43:35 +0300 Stefan Kost * tests/check/Makefile.am: makefile: idea about makeing more sources/sinks testable again 2009-05-25 16:33:35 +0200 John Keeping * ext/libpng/gstpngdec.c: pngdec: match g_malloc() with g_free() Matching g_malloc() with a g_free() is important when a custom allocator is installed. Fixes #583803 2009-05-12 18:39:28 +0200 Wim Taymans * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: rtpmp4vpay: don't look for headers in some cases In some streams (starting with 00000100) don't look for the headers but push data as it is. Fixes #582153 2009-05-13 11:50:22 +0200 Patrick Radizi * gst/rtsp/gstrtspsrc.c: rtspsrc: fix memory leak of messages Free messages correctly. Fixes #577318 2009-05-24 19:32:17 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: make fakesrc silent Make the fakesrc that is responsible for sending dummy packets silent. 2009-05-24 16:33:42 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't send teardown before setup Don't send a TEARDOWN request when we did not manage to successfully setup a stream. 2009-05-14 14:46:14 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: matroskademux: Populate a GstIndex that is set on matroskademux 2009-05-14 10:35:22 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: flvmux: Get the max duration from upstream if there's no duration tag 2009-05-14 10:29:49 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: flvmux: Write an index table to the end of the file 2009-05-22 01:12:35 +0100 Tim-Philipp Müller * autogen.sh: * configure.ac: autotools: move the -Wno-portability from autogen.sh to configure.ac If we're lucky it'll get used on automatic rebuilds as well that way. 2009-05-22 01:10:12 +0100 Tim-Philipp Müller * common: * configure.ac: * m4/gst-fionread.m4: m4: fix 'suspicious cache id' warnings and update common to pull in a similar fix. Also check in configure whether the compiler supports do while macros (GLib wants this defined and it is needed to avoid warnings with some c++ compilers apparently). 2009-05-22 01:39:33 +0300 Zeeshan Ali (Khattak) * configure.ac: souphttpsrc: Bump-up libsoup-2.24 dep to >= 2.26 The helper function soup_message_headers_get_content_type that we now use was added in 2.26. 2009-05-20 17:57:59 +0300 Zeeshan Ali (Khattak) * ext/soup/gstsouphttpsrc.c: souphttpsrc: Set caps for audio/L16 content-type When "Content-Type" header is "audio/L16", we need to set the caps on the outgoing buffers so that downstream elements can have means to detect the stream type and handle it appropriately. Tested with HTTP stream provided by pulse-audio's http module (git master). 2009-05-20 15:06:25 +0300 Zeeshan Ali (Khattak) * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Rename icy_caps to src_caps 2009-05-21 23:39:13 +0200 Philippe Normand * ext/jpeg/gstjpegdec.c: jpegdec: bump max size to 65535x65535 Remove artificial jpeg image limits. Fixes #583048. 2009-05-21 21:36:02 +0100 Jan Schmidt * win32/common/config.h: win32: Update the win32 config.h 2009-05-19 15:12:09 +0100 Jan Schmidt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Recognise PGS subpicture streams - the bluray format. Recognise and apply appropriate caps to PGS (Presentation Graphic Stream) subpicture streams. 2009-05-15 10:42:19 +0100 Jan Schmidt * ext/pulse/pulsesink.c: pulsesink: Convert an erroneous assertion Occasionally, we get a change callback for an old stream, triggering the assertion unnecessarily. Just ignore such callbacks. 2009-05-20 16:14:40 -0400 Olivier Crête * ext/pulse/pulsesink.c: pulse: Print a warning on under/overflows 2009-05-20 18:45:45 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: parse in24 boxes to get endianness in24 samples are normally big-endian but an enda box can change this to little-endian. Recurse into the in24 box and find the enda box so that we get the endianness right. Fixes #582515 2009-05-20 14:14:31 +0200 Wim Taymans * gst/multipart/multipartdemux.c: multipartdemux: add proper padtemplate 2009-05-20 14:02:43 +0200 Wim Taymans * gst/multipart/multipartdemux.c: multipartdemux: add more mime types Add mime-type for Panasonic g726 and add more required caps properties for other G726 mime-types. Make mime-types case insensitive. See #582169 2009-05-20 13:47:52 +0200 Wim Taymans * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: add flow aggregation 2009-05-20 13:29:02 +0200 Arnout Vandecappelle * gst/multipart/multipartdemux.c: multipartdemux: allow content to be empty. gst_adapter_take_buffer doesn't allow buffer to be empty. Simply skip any part where the content is empty. Don't create a pad for it either. See #582169 2009-05-18 22:19:39 +0200 Wim Taymans * gst/rtp/gstrtpchannels.h: rtp: fix channel positions for mono 2009-05-21 21:02:11 +0100 Jan Schmidt * configure.ac: Back to hacking -> 0.10.15.1 === release 0.10.15 === 2009-05-20 22:34:18 +0100 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.15 2009-05-20 22:03:21 +0100 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2009-05-16 02:59:14 +0100 Jan Schmidt * ChangeLog: * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: 0.10.14.3 pre-release 2009-05-16 02:37:06 +0100 Jan Schmidt * tests/check/pipelines/flacdec.c: check: Don't change directory in the test Changing directory invalidates the paths the registry has picked up for our plugins, because the test environment specifies relative paths. Fixing that is a separate problem, in the meantime, build a path to the test files instead of changing directory. Fixes the distcheck. 2009-05-16 01:53:46 +0100 Jan Schmidt * win32/MANIFEST: win32: Remove directdraw project files from the win32 manifest 2009-05-16 01:21:34 +0100 Jan Schmidt * tests/check/elements/rganalysis.c: check: Remove assertion that breaks check again git master Remove the assertion that the sender of the tags message is the element until we decide whether that's going to be true or not. 2009-05-16 01:11:33 +0100 Jan Schmidt * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-directdraw.xml: * sys/Makefile.am: * sys/directdraw/Makefile.am: * sys/directdraw/gstdirectdrawplugin.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: * win32/vs6/libgstdirectdraw.dsp: * win32/vs7/libgstdirectdraw.vcproj: * win32/vs8/libgstdirectdraw.vcproj: Moved 'directdraw' from -good to -bad 2009-05-16 00:18:34 +0100 Jan Schmidt * tests/check/pipelines/.gitignore: ignores: Ignore the flacdec check binary 2009-05-16 00:17:57 +0100 Jan Schmidt * docs/plugins/inspect/plugin-avi.xml: docs: Update inspection details for the avi plugin 2009-05-16 00:00:07 +0100 Jan Schmidt * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-y4menc.xml: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/y4menc.c: Moved 'y4menc' from -bad to -good 2009-05-13 17:55:46 +0200 Wim Taymans * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] y4menc: change my email change my email to something more current See #580783 2009-05-13 17:54:47 +0200 Wim Taymans * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] y4menc: don't strip timestamps Fixes #582483 2008-11-04 12:42:30 +0000 Stefan Kost [MOVED FROM BAD] Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-06-26 15:52:40 +0000 Mark Nauwelaerts [MOVED FROM BAD] Add documentation for YUV4MPEG2 encoder element. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/y4m/gsty4mencode.c: Add documentation for YUV4MPEG2 encoder element. 2007-04-24 15:49:18 +0000 Tim-Philipp Müller [MOVED FROM BAD] Plug some leaks; try to make build bot happy again. Original commit message from CVS: * gst/y4m/gsty4mencode.c: (gst_y4m_encode_init), (gst_y4m_encode_setcaps): * tests/check/elements/y4menc.c: (GST_START_TEST): Plug some leaks; try to make build bot happy again. 2006-11-13 18:55:57 +0000 Mark Nauwelaerts [MOVED FROM BAD] configure.ac: Enable cdaudio and y4m. Original commit message from CVS: Patch by: Mark Nauwelaerts * configure.ac: Enable cdaudio and y4m. * gst/y4m/Makefile.am: * gst/y4m/gsty4mencode.c: (gst_y4m_encode_base_init), (gst_y4m_encode_class_init), (gst_y4m_encode_init), (gst_y4m_encode_reset), (gst_y4m_encode_setcaps), (gst_y4m_encode_get_stream_header), (gst_y4m_encode_get_frame_header), (gst_y4m_encode_chain), (gst_y4m_encode_set_property), (gst_y4m_encode_get_property), (gst_y4m_encode_change_state), (plugin_init): * gst/y4m/gsty4mencode.h: Port of y4mencode to 0.10. 2006-04-25 21:56:38 +0000 Stefan Kost [MOVED FROM BAD] Define GstElementDetails as const and also static (when defined as global) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/arts/gst_arts.c: * ext/artsd/gstartsdsink.c: * ext/audiofile/gstafparse.c: * ext/audiofile/gstafsink.c: * ext/audiofile/gstafsrc.c: * ext/audioresample/gstaudioresample.c: * ext/bz2/gstbz2dec.c: * ext/bz2/gstbz2enc.c: * ext/cdaudio/gstcdaudio.c: * ext/directfb/dfbvideosink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/libfame/gstlibfame.c: * ext/libmms/gstmms.c: (gst_mms_base_init): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/nas/nassink.c: (gst_nassink_base_init): * ext/neon/gstneonhttpsrc.c: * ext/sdl/sdlaudiosink.c: * ext/sdl/sdlvideosink.c: * ext/shout/gstshout.c: * ext/snapshot/gstsnapshot.c: * ext/sndfile/gstsf.c: * ext/swfdec/gstswfdec.c: * ext/tarkin/gsttarkindec.c: * ext/tarkin/gsttarkinenc.c: * ext/theora/theoradec.c: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init): * gst/chart/gstchart.c: * gst/colorspace/gstcolorspace.c: * gst/deinterlace/gstdeinterlace.c: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init): * gst/festival/gstfestival.c: * gst/filter/gstbpwsinc.c: * gst/filter/gstiir.c: * gst/filter/gstlpwsinc.c: * gst/freeze/gstfreeze.c: * gst/games/gstpuzzle.c: (gst_puzzle_base_init): * gst/librfb/gstrfbsrc.c: * gst/mixmatrix/mixmatrix.c: * gst/mpeg1sys/gstmpeg1systemencode.c: * gst/mpeg1videoparse/gstmp1videoparse.c: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/multifilesink/gstmultifilesink.c: * gst/overlay/gstoverlay.c: * gst/passthrough/gstpassthrough.c: * gst/playondemand/gstplayondemand.c: * gst/qtdemux/qtdemux.c: * gst/rtjpeg/gstrtjpegdec.c: * gst/rtjpeg/gstrtjpegenc.c: * gst/smooth/gstsmooth.c: * gst/smoothwave/gstsmoothwave.c: * gst/spectrum/gstspectrum.c: * gst/speed/gstspeed.c: * gst/stereo/gststereo.c: * gst/switch/gstswitch.c: * gst/tta/gstttadec.c: (gst_tta_dec_base_init): * gst/tta/gstttaparse.c: (gst_tta_parse_base_init): * gst/vbidec/gstvbidec.c: * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/virtualdub/gstxsharpen.c: * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init): * gst/y4m/gsty4mencode.c: * sys/cdrom/gstcdplayer.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: * sys/glsink/glimagesink.c: * sys/qcam/gstqcamsrc.c: * sys/v4l2/gstv4l2src.c: * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init): * sys/ximagesrc/ximagesrc.c: Define GstElementDetails as const and also static (when defined as global) 2006-04-08 21:48:01 +0000 Stefan Kost [MOVED FROM BAD] Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): * ext/arts/gst_arts.c: (gst_arts_class_init): * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): * ext/audiofile/gstafsink.c: (gst_afsink_class_init): * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): * ext/audioresample/gstaudioresample.c: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_class_init): * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): * ext/jack/gstjack.c: (gst_jack_class_init): * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): * ext/nas/nassink.c: (gst_nassink_class_init): * ext/shout/gstshout.c: (gst_icecastsend_class_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): * ext/sndfile/gstsf.c: (gst_sf_class_init): * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), (gst_swfdec_class_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): * gst/chart/gstchart.c: (gst_chart_class_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): * gst/festival/gstfestival.c: (gst_festival_class_init): * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstiir.c: (gst_iir_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_class_init): * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_class_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init): * gst/overlay/gstoverlay.c: (gst_overlay_class_init): * gst/passthrough/gstpassthrough.c: (passthrough_class_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): * gst/smooth/gstsmooth.c: (gst_smooth_class_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): * gst/stereo/gststereo.c: (gst_stereo_class_init): * gst/switch/gstswitch.c: (gst_switch_class_init): * gst/tta/gstttadec.c: (gst_tta_dec_class_init): * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_class_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): * sys/v4l2/gstv4l2colorbalance.c: (gst_v4l2_color_balance_channel_class_init): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), (gst_v4l2_tuner_norm_class_init): * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 2006-04-08 19:04:01 +0000 Stefan Kost [MOVED FROM BAD] gst/: Fix more broken GObject macros Original commit message from CVS: * gst/colorspace/gstcolorspace.h: * gst/deinterlace/gstdeinterlace.h: * gst/passthrough/gstpassthrough.h: * gst/y4m/gsty4mencode.h: Fix more broken GObject macros 2006-04-06 11:35:26 +0000 j@bootlab.org [MOVED FROM BAD] Unify the long descriptions in the plugin details (#337263). Original commit message from CVS: Patch by: j^ * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/arts/gst_arts.c: * ext/artsd/gstartsdsink.c: * ext/audiofile/gstafparse.c: * ext/audiofile/gstafsink.c: * ext/audiofile/gstafsrc.c: * ext/cdaudio/gstcdaudio.c: * ext/directfb/dfbvideosink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/libfame/gstlibfame.c: * ext/libmms/gstmms.c: (gst_mms_base_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/nas/nassink.c: (gst_nassink_base_init): * ext/neon/gstneonhttpsrc.c: * ext/polyp/polypsink.c: (gst_polypsink_base_init): * ext/sdl/sdlaudiosink.c: * ext/sdl/sdlvideosink.c: * ext/shout/gstshout.c: * ext/snapshot/gstsnapshot.c: * ext/sndfile/gstsf.c: * ext/tarkin/gsttarkindec.c: * ext/tarkin/gsttarkinenc.c: * ext/theora/theoradec.c: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init): * gst/chart/gstchart.c: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init): * gst/festival/gstfestival.c: * gst/filter/gstiir.c: * gst/filter/gstlpwsinc.c: * gst/freeze/gstfreeze.c: * gst/games/gstpuzzle.c: (gst_puzzle_base_init): * gst/mixmatrix/mixmatrix.c: * gst/mpeg1sys/gstmpeg1systemencode.c: * gst/mpeg1videoparse/gstmp1videoparse.c: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/multifilesink/gstmultifilesink.c: * gst/overlay/gstoverlay.c: * gst/passthrough/gstpassthrough.c: * gst/playondemand/gstplayondemand.c: * gst/qtdemux/qtdemux.c: * gst/rtjpeg/gstrtjpegdec.c: * gst/rtjpeg/gstrtjpegenc.c: * gst/smooth/gstsmooth.c: * gst/tta/gstttadec.c: (gst_tta_dec_base_init): * gst/tta/gstttaparse.c: (gst_tta_parse_base_init): * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/virtualdub/gstxsharpen.c: * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init): * gst/y4m/gsty4mencode.c: Unify the long descriptions in the plugin details (#337263). 2006-04-01 10:09:11 +0000 Thomas Vander Stichele * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] rework build; add translations for v4l2 Original commit message from CVS: rework build; add translations for v4l2 2005-09-05 17:20:29 +0000 Jan Schmidt * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] Fix up all the state change functions. Original commit message from CVS: Fix up all the state change functions. 2005-07-05 10:51:49 +0000 Andy Wingo [MOVED FROM BAD] Way, way, way too many files: Remove crack comment from the 2000 era. Original commit message from CVS: 2005-07-05 Andy Wingo * Way, way, way too many files: Remove crack comment from the 2000 era. 2005-01-14 18:36:42 +0000 Stéphane Loeuillet [MOVED FROM BAD] I'm a bad boy. using /1001. to force C to do float division and not integer division (as it did in my last commit) Original commit message from CVS: * ext/dv/gstdvdec.c: * gst/subparse/gstsubparse.c: (parse_mdvdsub): * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect): I'm a bad boy. using /1001. to force C to do float division and not integer division (as it did in my last commit) Thanks to David I. Lehn for pointing this mistake. 2005-01-14 12:27:22 +0000 Stéphane Loeuillet [MOVED FROM BAD] replace framerate aproximations by their real value (24000/1001, 30000/1001, 60000/1001) Original commit message from CVS: * ext/dv/gstdvdec.c: * ext/libfame/gstlibfame.c: * gst/subparse/gstsubparse.c: (parse_mdvdsub): * gst/y4m/gsty4mencode.c: (gst_y4mencode_sinkconnect): replace framerate aproximations by their real value (24000/1001, 30000/1001, 60000/1001) Finish fixing bug #164049 2004-07-27 21:41:30 +0000 Steve Lhomme * gst/y4m/y4menc.vcproj: [MOVED FROM BAD] more working plugins Original commit message from CVS: more working plugins 2004-07-27 09:57:33 +0000 Steve Lhomme * gst/y4m/y4menc.vcproj: [MOVED FROM BAD] rename GStreamer-0.8.lib to libgstreamer.lib Original commit message from CVS: rename GStreamer-0.8.lib to libgstreamer.lib 2004-07-27 09:48:51 +0000 Steve Lhomme * gst/y4m/y4menc.vcproj: [MOVED FROM BAD] avoid problems with math.h, fix release dependancy Original commit message from CVS: avoid problems with math.h, fix release dependancy 2004-07-26 13:20:11 +0000 Steve Lhomme * gst/y4m/y4menc.vcproj: [MOVED FROM BAD] more plugins supported under windows Original commit message from CVS: more plugins supported under windows 2004-04-01 11:48:27 +0000 Jan Schmidt * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] a52dec: Use a debug category, Output timestamps correctly Original commit message from CVS: a52dec: Use a debug category, Output timestamps correctly Emit tag info, Handle events, tell liba52dec about cpu capabilities so it can use MMX etc. dvdec: Fix a crasher accessing invalid memory dvdnavsrc:Some support for byte-format seeking. Small fixes for still frames and menu button overlays mpeg2dec: Use a debug category. Adjust the report level of several items to LOG. Call mpeg2_custom_fbuf to mark our buffers as 'custom buffers' so it doesn't lose the GstBuffer pointer navseek: Add the navseek debug element for seeking back and forth in a video stream using arrow keys. mpeg2subt:Pretty much a complete rewrite. Now a loopbased element. May still require work to properly synchronise subtitle buffers. mpegdemux: dvddemux: Don't attempt to create subbuffers of size 0 Reduce a couple of error outputs to warnings. y4mencode:Output the y4m frame header correctly 2004-03-15 19:32:27 +0000 Thomas Vander Stichele * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] don't mix tabs and spaces Original commit message from CVS: don't mix tabs and spaces 2004-03-15 16:32:54 +0000 Johan Dahlin [MOVED FROM BAD] *.h: Revert indenting Original commit message from CVS: * *.h: Revert indenting 2004-03-14 22:34:33 +0000 Thomas Vander Stichele * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: [MOVED FROM BAD] gst-indent Original commit message from CVS: gst-indent 2004-01-12 02:01:52 +0000 Benjamin Otte [MOVED FROM BAD] gst-libs/gst/video/video.h: Fix caps template names to be understandable. Original commit message from CVS: 2004-01-12 Benjamin Otte * gst-libs/gst/video/video.h: Fix caps template names to be understandable. Prefix everything with GST_VIDEO. * ext/aalib/gstaasink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/hermes/gstcolorspace.c: (gst_colorspace_base_init): * ext/jpeg/gstjpegdec.c: (raw_caps_factory): * ext/jpeg/gstjpegenc.c: (raw_caps_factory): * ext/libcaca/gstcacasink.c: * ext/libpng/gstpngenc.c: (raw_caps_factory): * ext/snapshot/gstsnapshot.c: * ext/swfdec/gstswfdec.c: * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/chart/gstchart.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gsteffectv.c: * gst/flx/gstflxdec.c: (gst_flxdec_loop): * gst/goom/gstgoom.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: (gst_monoscope_init), (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/overlay/gstoverlay.c: * gst/smooth/gstsmooth.c: * gst/smpte/gstsmpte.c: * gst/synaesthesia/gstsynaesthesia.c: * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/y4m/gsty4mencode.c: * sys/qcam/gstqcamsrc.c: * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps): Make them work with new video.h file. * sys/ximage/ximagesink.c: (gst_ximagesink_chain), (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain), (gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc): Make it work with new buffer allocation system. 2003-12-22 01:47:09 +0000 David Schleef * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] Merge CAPS branch Original commit message from CVS: Merge CAPS branch 2003-12-04 10:37:38 +0000 Andy Wingo * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] remove copyright field from plugins Original commit message from CVS: remove copyright field from plugins 2003-11-16 22:02:23 +0000 Leif Johnson * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] + checking in plugin category changes Original commit message from CVS: + checking in plugin category changes 2003-11-07 12:47:02 +0000 Ronald S. Bultje * gst/y4m/gsty4mencode.h: [MOVED FROM BAD] Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... Original commit message from CVS: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files 2003-11-02 19:17:27 +0000 Benjamin Otte * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] fix to new plugin system Original commit message from CVS: fix to new plugin system 2003-10-08 16:08:19 +0000 Andy Wingo * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. Original commit message from CVS: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. 2003-08-10 00:01:58 +0000 David Schleef * gst/y4m/Makefile.am: [MOVED FROM BAD] Remove redundant plugindir definition Original commit message from CVS: Remove redundant plugindir definition 2003-07-06 20:49:52 +0000 Ronald S. Bultje * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: [MOVED FROM BAD] New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri... Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs 2003-06-29 19:46:13 +0000 Benjamin Otte * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] compatibility fix for new GST_DEBUG stuff. Original commit message from CVS: compatibility fix for new GST_DEBUG stuff. Includes fixes for missing includes for config.h and unistd.h I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately. 2003-01-10 13:38:32 +0000 Thomas Vander Stichele * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] PadConnect -> PadLink Original commit message from CVS: PadConnect -> PadLink 2003-01-10 10:22:25 +0000 Thomas Vander Stichele * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t... Original commit message from CVS: another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so that I don't spend double the time resolving conflicts 2002-12-08 14:50:10 +0000 Thomas Vander Stichele * gst/y4m/Makefile.am: [MOVED FROM BAD] parallel install fixes Original commit message from CVS: parallel install fixes 2002-09-18 19:02:52 +0000 Christian Schaller * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] plugins part of license field patch Original commit message from CVS: plugins part of license field patch 2002-06-17 10:29:30 +0000 Thomas Vander Stichele * gst/y4m/Makefile.am: [MOVED FROM BAD] cosmetic change Original commit message from CVS: cosmetic change 2002-05-03 09:59:10 +0000 Thomas Vander Stichele * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] various name fixes and sundry Original commit message from CVS: various name fixes and sundry 2002-04-20 21:42:51 +0000 Andy Wingo * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br... Original commit message from CVS: * a hack to work around intltool's brokenness * a current check for mpeg2dec * details->klass reorganizations * an element browser that uses details->klass * separated cdxa parse out from the avi directory 2002-04-11 20:42:26 +0000 Andy Wingo * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. Original commit message from CVS: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. also, some -Werror fixes. 2002-03-30 17:06:26 +0000 Wim Taymans * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] Changed to the new props API Original commit message from CVS: Changed to the new props API Other small tuff. 2002-03-20 21:45:04 +0000 Andy Wingo * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: [MOVED FROM BAD] s/Gnome-Streamer/GStreamer/ Original commit message from CVS: s/Gnome-Streamer/GStreamer/ 2002-03-19 04:10:06 +0000 Andy Wingo * gst/y4m/Makefile.am: * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: [MOVED FROM BAD] removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with... Original commit message from CVS: * removal of //-style comments * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct, and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory. 2002-03-19 01:39:43 +0000 Andy Wingo * gst/y4m/Makefile.am: [MOVED FROM BAD] s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi... Original commit message from CVS: s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagically, and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag 2002-01-18 11:37:19 +0000 Wrobell * gst/y4m/Makefile.am: [MOVED FROM BAD] - plugins are built without versioning info Original commit message from CVS: - plugins are built without versioning info 2002-01-13 22:27:25 +0000 Wim Taymans * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] Bring the plugins in sync with the new core capsnego system. Original commit message from CVS: Bring the plugins in sync with the new core capsnego system. Added some features, enhancements... 2002-01-12 03:34:27 +0000 David I. Lehn * gst/y4m/Makefile.am: [MOVED FROM BAD] s/filter/plugin/ link plugins to GST_LIBS rearrange rules to a common format Original commit message from CVS: * s/filter/plugin/ * link plugins to GST_LIBS * rearrange rules to a common format 2001-12-23 20:21:20 +0000 Thomas Vander Stichele * gst/y4m/Makefile.am: * gst/y4m/gsty4mencode.c: [MOVED FROM BAD] more fixes Original commit message from CVS: more fixes 2001-12-23 13:17:36 +0000 Thomas Vander Stichele * gst/y4m/Makefile.am: * gst/y4m/gsty4mencode.c: * gst/y4m/gsty4mencode.h: [MOVED FROM BAD] BBB asked me to rename lav to y4m can someone who knows the plugin do this in the source as well ? Original commit message from CVS: BBB asked me to rename lav to y4m can someone who knows the plugin do this in the source as well ? 2009-05-15 18:17:35 +0100 Tim-Philipp Müller * po/Makevars: po: add Makevars magic so we don't get line numbers in *.po files This avoids the number one reason for local modifications in *.po files and and makes things less annoying when working with git (or any other VCS for that matter). 2009-05-15 17:11:27 +0100 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/id3demux.c: * tests/check/elements/souphttpsrc.c: * tests/check/pipelines/flacdec.c: * tests/files/Makefile.am: * tests/files/audiotestsrc.flac: * tests/files/test-cert.pem: * tests/files/test-key.pem: checks: move files required by unit tests into tests/files and make sure they're disted Move unit test data into the directory where it belongs and make in particular the flacdec unit test cd into the directory with the test files instead of making assumptions about the current working directory in that unit test. As a side effect of movng those files, there's only one EXTRA_DIST in tests/check/Makefile.am now, which is likely to work better than having two. Hopefully fixes #582753. 2009-05-14 21:43:14 +0200 Sebastian Dröge * gst/deinterlace/gstdeinterlace.c: deinterlace: If the upstream max latency is unbound return unbound max latency Fixes bug #582661. 2009-05-15 08:44:39 +0200 James Andrewartha * gst/flv/gstflvmux.c: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixertrack.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/v4l2_calls.c: Fix compiler warnings Fixes bug #582715. 2009-05-14 12:32:16 +0200 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lamemp3enc: Improve debugging a bit 2009-05-13 22:46:44 +0200 Josep Torra * configure.ac: Recovered debugutils line accidentally removed in deinterlace2 move. 2009-05-13 10:46:40 +0200 Sebastian Dröge * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-deinterlace.xml: * gst/deinterlace/Makefile.am: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.asm: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/greedyhmacros.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/mmx.h: * gst/deinterlace/tvtime/plugins.h: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/sse.h: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/tomsmocomp/SearchLoop0A.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopBottom.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopEdgeA.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopEdgeA8.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddA.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddA2.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddA6.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddAH.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopOddAH2.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopTop.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopVA.inc: * gst/deinterlace/tvtime/tomsmocomp/SearchLoopVAH.inc: * gst/deinterlace/tvtime/tomsmocomp/StrangeBob.inc: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll2.inc: * gst/deinterlace/tvtime/tomsmocomp/WierdBob.inc: * gst/deinterlace/tvtime/tomsmocomp/tomsmocompmacros.h: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: * gst/deinterlace/tvtime/x86-64_macros.inc: Moved 'deinterlace2' from -bad to -good And rename it to deinterlace. 2009-05-08 15:39:24 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: * gst/deinterlace2/gstdeinterlace2.h: [MOVED FROM BAD 56/56] deinterlace2: Add a disabled mode for passthrough operation Also allow to change the mode in PAUSED and PLAYING by updating the caps if necessary. 2009-04-22 19:43:22 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: * gst/deinterlace2/gstdeinterlace2.h: [MOVED FROM BAD 55/56] deinterlace2: Add documentation and integrate into the build system 2009-04-19 17:18:35 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: [MOVED FROM BAD 54/56] deinterlace2: Make it possible to select interlacing autodetection or to enfore deinterlacing For this add a "mode" property that defaults to "interlaced" for now as most decoders/demuxers don't properly set the "interlaced" field on the caps yet. If this property is set to "auto" the element will work in passthrough mode unless the caps contain the "interlaced" field. 2009-04-17 15:39:59 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: [MOVED FROM BAD 53/56] deinterlace2: Use GST_(DEBUG|WARNING|ERROR)_OBJECT instead of the non-OBJECT ones 2009-04-17 15:39:36 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: [MOVED FROM BAD 52/56] deinterlace2: Reset history if DISCONT is set on the incoming buffer 2009-04-17 15:39:10 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: [MOVED FROM BAD 51/56] deinterlace2: Fix timestamps for buffers with RFF flag set 2009-04-16 17:41:37 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/greedy.c: * gst/deinterlace2/tvtime/greedyh.c: * gst/deinterlace2/tvtime/scalerbob.c: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/weave.c: * gst/deinterlace2/tvtime/weavebff.c: * gst/deinterlace2/tvtime/weavetff.c: [MOVED FROM BAD 50/56] deinterlace2: Rename line_length to row_stride and remove output_stride 2009-04-16 15:52:39 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: [MOVED FROM BAD 49/56] deinterlace2: Implement support for RFF and ONEFIELD buffer flags 2009-04-15 15:46:44 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/greedy.c: * gst/deinterlace2/tvtime/greedyh.c: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: [MOVED FROM BAD 48/56] deinterlace2: Move output buffer from the instance struct to a function parameter 2009-04-15 15:33:17 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: * gst/deinterlace2/gstdeinterlace2.h: [MOVED FROM BAD 47/56] deinterlace2: Add initial support for automatic detection of the field order 2009-04-15 14:47:49 +0200 Sebastian Dröge * gst/deinterlace2/gstdeinterlace2.c: [MOVED FROM BAD 46/56] deinterlace2: Add support for YVYU colorspace This is the same as YUY2 with just Cr and Cb swapped. As we don't make a difference between them when deinterlacing this works. 2008-11-06 14:05:55 +0000 Wim Taymans [MOVED FROM BAD 45/56] gst/deinterlace2/gstdeinterlace2.c: Bring properties into this century. Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_class_init), (gst_deinterlace2_init), (gst_deinterlace2_set_property), (gst_deinterlace2_get_property): Bring properties into this century. 2008-11-04 12:42:30 +0000 Stefan Kost [MOVED FROM BAD 44/56] Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-10-09 19:38:52 +0000 Sebastian Dröge [MOVED FROM BAD 43/56] gst/deinterlace2/tvtime/tomsmocomp.c: Fix unused variable compiler warning when not building Original commit message from CVS: * gst/deinterlace2/tvtime/tomsmocomp.c: (gst_deinterlace_method_tomsmocomp_class_init): Fix unused variable compiler warning when not building X86 assembly. 2008-08-28 17:16:51 +0000 Jan Schmidt [MOVED FROM BAD 42/56] gst/dccp/: Fix compilation on Solaris by including filio.h as needed. Original commit message from CVS: * gst/dccp/gstdccp.c: * gst/dccp/gstdccpclientsrc.c: Fix compilation on Solaris by including filio.h as needed. * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: Fix compilation with Forte - apparently it hates concatenating a macro argument that starts with an underscore?? 2008-08-26 12:33:16 +0000 Sebastian Dröge [MOVED FROM BAD 41/56] gst/deinterlace2/tvtime/tomsmocomp/: Unroll the loop to handle two bytes at once. This should give a small speedup an... Original commit message from CVS: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc: * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc: * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc: Unroll the loop to handle two bytes at once. This should give a small speedup and makes it possible to handle chroma and luma different which is needed later. 2008-08-25 14:37:45 +0000 Sebastian Dröge [MOVED FROM BAD 40/56] gst/deinterlace2/: First part of the C implementation of the tomsmocomp deinterlacing algorithm. This only supports s... Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_method_class_init): * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/tomsmocomp.c: (gst_deinterlace_method_tomsmocomp_class_init): * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc: * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc: * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: First part of the C implementation of the tomsmocomp deinterlacing algorithm. This only supports search-effort=0 currently, is painfully slow and needs some cleanup later when all search-effort settings are implemented in C. 2008-08-02 18:48:17 +0000 Sebastian Dröge [MOVED FROM BAD 39/56] gst/deinterlace2/: Use oil_memcpy() instead of memcpy() as it's faster for the sizes that are usually used here. Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_simple_method_interpolate_scanline), (gst_deinterlace_simple_method_copy_scanline), (gst_deinterlace_simple_method_deinterlace_frame): * gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy): * gst/deinterlace2/tvtime/greedyh.c: (deinterlace_frame_di_greedyh): * gst/deinterlace2/tvtime/scalerbob.c: (deinterlace_scanline_scaler_bob): * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy): * gst/deinterlace2/tvtime/weave.c: (deinterlace_scanline_weave), (copy_scanline): * gst/deinterlace2/tvtime/weavebff.c: (deinterlace_scanline_weave), (copy_scanline): * gst/deinterlace2/tvtime/weavetff.c: (deinterlace_scanline_weave), (copy_scanline): Use oil_memcpy() instead of memcpy() as it's faster for the sizes that are usually used here. 2008-08-02 18:36:11 +0000 Sebastian Dröge [MOVED FROM BAD 38/56] gst/deinterlace2/: Add the remaining tvtime deinterlacing methods and fix the deinterlace_frame() implementation of G... Original commit message from CVS: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_simple_method_deinterlace_frame), (gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method): * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/linear.c: (deinterlace_scanline_linear_c), (deinterlace_scanline_linear_mmx), (deinterlace_scanline_linear_mmxext), (gst_deinterlace_method_linear_class_init), (gst_deinterlace_method_linear_init): * gst/deinterlace2/tvtime/linearblend.c: (deinterlace_scanline_linear_blend_c), (deinterlace_scanline_linear_blend2_c), (deinterlace_scanline_linear_blend_mmx), (deinterlace_scanline_linear_blend2_mmx), (gst_deinterlace_method_linear_blend_class_init), (gst_deinterlace_method_linear_blend_init): * gst/deinterlace2/tvtime/plugins.h: * gst/deinterlace2/tvtime/scalerbob.c: (deinterlace_scanline_scaler_bob), (gst_deinterlace_method_scaler_bob_class_init), (gst_deinterlace_method_scaler_bob_init): * gst/deinterlace2/tvtime/weave.c: (deinterlace_scanline_weave), (copy_scanline), (gst_deinterlace_method_weave_class_init), (gst_deinterlace_method_weave_init): * gst/deinterlace2/tvtime/weavebff.c: (deinterlace_scanline_weave), (copy_scanline), (gst_deinterlace_method_weave_bff_class_init), (gst_deinterlace_method_weave_bff_init): * gst/deinterlace2/tvtime/weavetff.c: (deinterlace_scanline_weave), (copy_scanline), (gst_deinterlace_method_weave_tff_class_init), (gst_deinterlace_method_weave_tff_init): Add the remaining tvtime deinterlacing methods and fix the deinterlace_frame() implementation of GstDeinterlaceSimpleMethod. 2008-08-02 18:30:56 +0000 Sebastian Dröge [MOVED FROM BAD 37/56] gst/deinterlace2/tvtime/vfir.c: Implement the VFIR deinterlacing method as simple method. Original commit message from CVS: * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c), (deinterlace_line_mmx), (gst_deinterlace_method_vfir_class_init): Implement the VFIR deinterlacing method as simple method. 2008-08-02 18:18:54 +0000 Sebastian Dröge [MOVED FROM BAD 36/56] gst/deinterlace2/gstdeinterlace2.*: Add a GstDeinterlaceSimpleMethod subclass of GstDeinterlaceMethod that can be use... Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_simple_method_interpolate_scanline), (gst_deinterlace_simple_method_copy_scanline), (gst_deinterlace_simple_method_deinterlace_frame), (gst_deinterlace_simple_method_class_init), (gst_deinterlace_simple_method_init): * gst/deinterlace2/gstdeinterlace2.h: Add a GstDeinterlaceSimpleMethod subclass of GstDeinterlaceMethod that can be used by simple deinterlacing methods. They only have to provide a function for interpolating a scanline or copying a scanline. 2008-08-02 18:15:49 +0000 Sebastian Dröge [MOVED FROM BAD 35/56] gst/deinterlace2/gstdeinterlace2.c: Respect the latency of the deinterlacing algorithm for the timestamps of every bu... Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_chain): Respect the latency of the deinterlacing algorithm for the timestamps of every buffer. 2008-08-02 18:13:20 +0000 Sebastian Dröge [MOVED FROM BAD 34/56] gst/deinterlace2/tvtime/: Add the MMX registers to the clobbered registers only if __MMX__ is defined. Original commit message from CVS: * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc: Add the MMX registers to the clobbered registers only if __MMX__ is defined. 2008-08-02 18:09:56 +0000 Sebastian Dröge [MOVED FROM BAD 33/56] gst/deinterlace2/: Enable tomsmocomp again as the C port will be ready for the next release. Original commit message from CVS: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method), (gst_deinterlace2_class_init): Enable tomsmocomp again as the C port will be ready for the next release. 2008-08-02 18:02:44 +0000 Sebastian Dröge [MOVED FROM BAD 32/56] gst/deinterlace2/gstdeinterlace2.c: Don't use proxy_getcaps() but implement our own getcaps() function that doubles/h... Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init), (gst_greatest_common_divisor), (gst_fraction_double), (gst_deinterlace2_getcaps), (gst_deinterlace2_setcaps): Don't use proxy_getcaps() but implement our own getcaps() function that doubles/halfs the framerate if all fields should be sent out. 2008-07-18 08:34:06 +0000 Sebastian Dröge [MOVED FROM BAD 31/56] Disable the tomsmocomp algorithm for this release as it's buggy and has no C implementation yet. Original commit message from CVS: * configure.ac: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method), (gst_deinterlace2_class_init), (gst_deinterlace2_init): * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/greedy.c: (gst_deinterlace_method_greedy_l_class_init): * gst/deinterlace2/tvtime/greedyh.c: (gst_deinterlace_method_greedy_h_class_init): * gst/deinterlace2/tvtime/vfir.c: (gst_deinterlace_method_vfir_class_init): Disable the tomsmocomp algorithm for this release as it's buggy and has no C implementation yet. Build the deinterlace2 plugin on all architectures but still mark it as experimental. Build the x86 inline assembly only if GCC inline assembly is supported and only on x86 or amd64. Fixes bug #543286. 2008-07-14 14:13:54 +0000 Edward Hervey [MOVED FROM BAD 30/56] gst/deinterlace2/tvtime/: Fix build on x86_64 Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (gst_deinterlace_method_greedy_l_class_init): * gst/deinterlace2/tvtime/greedyh.c: (gst_deinterlace_method_greedy_h_class_init): * gst/deinterlace2/tvtime/vfir.c: (gst_deinterlace_method_vfir_class_init): Fix build on x86_64 2008-07-13 10:56:45 +0000 Sebastian Dröge [MOVED FROM BAD 29/56] gst/deinterlace2/tvtime/greedyh.asm: Always use the C implementation if width is not a multiple of 4. The assembly op... Original commit message from CVS: * gst/deinterlace2/tvtime/greedyh.asm: Always use the C implementation if width is not a multiple of 4. The assembly optimized version only handle this and calling the C implementation for the remaining part doesn't work because it needs previous calculations. 2008-07-13 10:52:03 +0000 Sebastian Dröge [MOVED FROM BAD 28/56] gst/deinterlace2/tvtime/: Some cleanup, use 3DNOW instead of TDNOW in macros. Original commit message from CVS: * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/greedyh.c: * gst/deinterlace2/tvtime/greedyhmacros.h: Some cleanup, use 3DNOW instead of TDNOW in macros. * gst/deinterlace2/tvtime/tomsmocomp.c: (gst_deinterlace_method_tomsmocomp_class_init): * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: The SSE method in fact only needs MMXEXT, declare it as such. 2008-07-08 13:31:37 +0000 Sebastian Dröge [MOVED FROM BAD 27/56] Don't use declarations after statements in the remaining code. Original commit message from CVS: * ext/spc/gstspc.c: (spc_setup): * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc: Don't use declarations after statements in the remaining code. 2008-07-06 20:43:58 +0000 Sebastian Dröge [MOVED FROM BAD 26/56] gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: Mark internal processing functions as static inline for quite ... Original commit message from CVS: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: Mark internal processing functions as static inline for quite some speedup as they're used only once and need to get many local variables passed as parameter. 2008-07-05 19:20:30 +0000 Sebastian Dröge [MOVED FROM BAD 25/56] gst/deinterlace2/gstdeinterlace2.*: Call the current instance "self" instead of "object". Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_method_deinterlace_frame), (gst_deinterlace2_set_method), (gst_deinterlace2_init), (gst_deinterlace2_reset_history), (gst_deinterlace2_reset), (gst_deinterlace2_set_property), (gst_deinterlace2_get_property), (gst_deinterlace2_pop_history), (gst_deinterlace2_head_history), (gst_deinterlace2_push_history), (gst_deinterlace2_chain), (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event), (gst_deinterlace2_change_state), (gst_deinterlace2_src_event), (gst_deinterlace2_src_query): * gst/deinterlace2/gstdeinterlace2.h: Call the current instance "self" instead of "object". 2008-07-05 19:11:56 +0000 Sebastian Dröge [MOVED FROM BAD 24/56] gst/deinterlace2/gstdeinterlace2.*: Include latency of the method in the returned latency. Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_method_get_latency), (gst_deinterlace2_set_method), (gst_deinterlace2_class_init), (gst_deinterlace2_push_history), (gst_deinterlace2_chain), (gst_deinterlace2_setcaps), (gst_deinterlace2_src_query): * gst/deinterlace2/gstdeinterlace2.h: Include latency of the method in the returned latency. Fix outputting of all fields, i.e. doubling of the framerate. 2008-07-05 16:47:32 +0000 Sebastian Dröge [MOVED FROM BAD 23/56] gst/deinterlace2/: Use a GstObject subtype for the deinterlacing methods and export the different settings for each d... Original commit message from CVS: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace_method_class_init), (gst_deinterlace_method_init), (gst_deinterlace_method_deinterlace_frame), (gst_deinterlace_method_get_fields_required), (gst_deinterlace2_methods_get_type), (_do_init), (gst_deinterlace2_set_method), (gst_deinterlace2_class_init), (gst_deinterlace2_child_proxy_get_child_by_index), (gst_deinterlace2_child_proxy_get_children_count), (gst_deinterlace2_child_proxy_interface_init), (gst_deinterlace2_init), (gst_deinterlace2_finalize), (gst_deinterlace2_chain), (gst_deinterlace2_src_query): * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_c), (deinterlace_greedy_packed422_scanline_mmx), (deinterlace_greedy_packed422_scanline_mmxext), (deinterlace_frame_di_greedy), (gst_deinterlace_method_greedy_l_set_property), (gst_deinterlace_method_greedy_l_get_property), (gst_deinterlace_method_greedy_l_class_init), (gst_deinterlace_method_greedy_l_init): * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C), (deinterlace_frame_di_greedyh), (gst_deinterlace_method_greedy_h_set_property), (gst_deinterlace_method_greedy_h_get_property), (gst_deinterlace_method_greedy_h_class_init), (gst_deinterlace_method_greedy_h_init): * gst/deinterlace2/tvtime/greedyh.h: * gst/deinterlace2/tvtime/plugins.h: * gst/deinterlace2/tvtime/tomsmocomp.c: (gst_deinterlace_method_tomsmocomp_set_property), (gst_deinterlace_method_tomsmocomp_get_property), (gst_deinterlace_method_tomsmocomp_class_init), (gst_deinterlace_method_tomsmocomp_init): * gst/deinterlace2/tvtime/tomsmocomp.h: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir), (gst_deinterlace_method_vfir_class_init), (gst_deinterlace_method_vfir_init): Use a GstObject subtype for the deinterlacing methods and export the different settings for each deinterlacing method via GObject properties. Implement GstChildProxy interface to allow access to the used deinterlacing method and to allow adjusting the different settings. Move global variables of the tomsmocomp deinterlacing method into function local variables to make it possible to use this deinterlacing method from different instances. 2008-07-05 12:22:37 +0000 Sebastian Dröge [MOVED FROM BAD 22/56] gst/deinterlace2/tvtime/greedyh.asm: Support widths that are not a multiply of 4 when using the assembly optimized gr... Original commit message from CVS: * gst/deinterlace2/tvtime/greedyh.asm: Support widths that are not a multiply of 4 when using the assembly optimized greedyh implementations. 2008-07-04 18:54:15 +0000 Sebastian Dröge [MOVED FROM BAD 21/56] gst/deinterlace2/tvtime/greedyh.c: Only build the assembly optimized implementations on x86. Original commit message from CVS: * gst/deinterlace2/tvtime/greedyh.c: (deinterlace_frame_di_greedyh): Only build the assembly optimized implementations on x86. 2008-06-30 07:51:07 +0000 Sebastian Dröge [MOVED FROM BAD 20/56] gst/deinterlace2/: Remove useless file and mark everything possible as static. Original commit message from CVS: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/tvtime/tomsmocomp.c: (tomsmocomp_init), (tomsmocomp_filter_mmx), (tomsmocomp_filter_3dnow), (tomsmocomp_filter_sse), (deinterlace_frame_di_tomsmocomp): * gst/deinterlace2/tvtime/tomsmocomp.h: Remove useless file and mark everything possible as static. * gst/deinterlace2/tvtime/greedy.c: * gst/deinterlace2/tvtime/greedyh.c: Use "_stdint.h" instead of . 2008-06-29 10:56:47 +0000 Sebastian Dröge [MOVED FROM BAD 19/56] gst/deinterlace2/: Get rid of speedy.[ch] as we don't use most of it's code anyway and it doesn't seem to be relicens... Original commit message from CVS: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init): * gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy): * gst/deinterlace2/tvtime/greedyh.c: (deinterlace_frame_di_greedyh): * gst/deinterlace2/tvtime/speedtools.h: * gst/deinterlace2/tvtime/speedy.c: * gst/deinterlace2/tvtime/speedy.h: * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy): * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir): Get rid of speedy.[ch] as we don't use most of it's code anyway and it doesn't seem to be relicensed to LGPL. Use memcpy() instead of the speedy memcpy everywhere instead. * gst/deinterlace2/gstdeinterlace2.h: Remove many unused declarations. 2008-06-28 18:13:08 +0000 Sebastian Dröge [MOVED FROM BAD 18/56] gst/deinterlace2/gstdeinterlace2.c: Divide latency be 2 to convert from fields to frames. Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_src_query): Divide latency be 2 to convert from fields to frames. 2008-06-28 18:10:52 +0000 Sebastian Dröge [MOVED FROM BAD 17/56] gst/deinterlace2/tvtime/greedy.c: Don't use scanlines function from gstdeinterlace2 as it's not appropiate for this m... Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_c), (deinterlace_greedy_packed422_scanline_mmx), (deinterlace_greedy_packed422_scanline_mmxext), (deinterlace_frame_di_greedy): Don't use scanlines function from gstdeinterlace2 as it's not appropiate for this method. Instead implement deinterlace_frame function by taking the one from greedyh. * gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C): Small fix for the C implementation. * gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir): Don't use the scanlines function from gstdeinterlace2 as it's only used for this method and will be removed. Instead implement deinterlace_frame function and make it a bit more efficient. * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_class_init), (gst_deinterlace2_set_method), (gst_deinterlace2_push_history), (gst_deinterlace2_chain), (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event), (gst_deinterlace2_change_state), (gst_deinterlace2_src_event), (gst_deinterlace2_src_query): Fix coding style and remove scanlines function as it's unused now. 2008-06-28 17:25:56 +0000 Sebastian Dröge [MOVED FROM BAD 16/56] gst/deinterlace2/tvtime/: Add a C implementation for the greedyh deinterlacing method, clean up the code a bit and ma... Original commit message from CVS: * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C), (deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method): * gst/deinterlace2/tvtime/greedyhmacros.h: Add a C implementation for the greedyh deinterlacing method, clean up the code a bit and mark the SSE version as MMXEXT as it doesn't require any SSE instructions. 2008-06-27 13:22:34 +0000 Sebastian Dröge [MOVED FROM BAD 15/56] gst/deinterlace2/gstdeinterlace2.c: If we're outputting all fields the framerate has to be doubled. Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_set_property), (gst_deinterlace2_chain), (gst_deinterlace2_setcaps): If we're outputting all fields the framerate has to be doubled. Set duration on the outgoing buffers. 2008-06-25 16:05:08 +0000 Edward Hervey [MOVED FROM BAD 14/56] gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: Remove unneeded macros that break build on macosx. Original commit message from CVS: * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: Remove unneeded macros that break build on macosx. 2008-06-24 12:08:47 +0000 Sebastian Dröge [MOVED FROM BAD 13/56] gst/deinterlace2/tvtime/greedy.c: Optimize MMX/MMXEXT implementations a bit by requiring two less memory accesses and... Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_mmx), (deinterlace_greedy_packed422_scanline_mmxext): Optimize MMX/MMXEXT implementations a bit by requiring two less memory accesses and fix the workaround for the missing right shift on bytes to unset the highest bit of every byte. 2008-06-24 10:15:41 +0000 Sebastian Dröge [MOVED FROM BAD 12/56] gst/deinterlace2/tvtime/greedy.c: Remove sfence instruction as it's not needed and actually is an SSE instruction. Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_mmxext): Remove sfence instruction as it's not needed and actually is an SSE instruction. 2008-06-24 10:12:08 +0000 Sebastian Dröge [MOVED FROM BAD 11/56] gst/deinterlace2/tvtime/greedy.c: Add plain MMX implementation for the greedyl method. Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_mmx), (deinterlace_greedy_packed422_scanline): Add plain MMX implementation for the greedyl method. 2008-06-24 09:40:03 +0000 Sebastian Dröge [MOVED FROM BAD 10/56] gst/deinterlace2/Makefile.am: Move the assembly includes to noinst_HEADERS where they belong. Original commit message from CVS: * gst/deinterlace2/Makefile.am: Move the assembly includes to noinst_HEADERS where they belong. * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c), (deinterlace_line_mmx): Fix C and MMX implementations a bit more. 2008-06-24 09:10:46 +0000 Sebastian Dröge [MOVED FROM BAD 09/56] gst/deinterlace2/tvtime/greedy.c: Fix the C implementation to produce correct results and optimize the Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_c), (deinterlace_greedy_packed422_scanline_mmxext), (deinterlace_greedy_packed422_scanline): Fix the C implementation to produce correct results and optimize the MMXEXT implementation. Handle odd widths and don't read over array boundaries in the MMXEXT implementation. * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c), (deinterlace_line_mmx), (deinterlace_scanline_vfir): Fix a small rounding bug in the MMX implementation, the MMX implementation doesn't actually need MMXEXT instructions so don't mark it as such. Handle odd widths in both implementations. 2008-06-21 09:05:00 +0000 Sebastian Dröge [MOVED FROM BAD 08/56] gst/deinterlace2/tvtime/greedy.c: Implement a C version of the greedy low motion algorithm and mark the assembly opti... Original commit message from CVS: * gst/deinterlace2/tvtime/greedy.c: (deinterlace_greedy_packed422_scanline_sse), (deinterlace_greedy_packed422_scanline_c), (deinterlace_greedy_packed422_scanline): Implement a C version of the greedy low motion algorithm and mark the assembly optimized version as SSE as it uses SSE instructions additional to MMX instructions. 2008-06-20 14:48:40 +0000 Sebastian Dröge [MOVED FROM BAD 07/56] gst/deinterlace2/tvtime/vfir.c: Make it possible to use the vfir method on X86 CPUs without MMXEXT too but use the MM... Original commit message from CVS: * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_mmxext), (deinterlace_line_c), (deinterlace_scanline_vfir): Make it possible to use the vfir method on X86 CPUs without MMXEXT too but use the MMXEXT optimized code whenever possible. 2008-06-20 14:35:25 +0000 Sebastian Dröge [MOVED FROM BAD 06/56] gst/deinterlace2/gstdeinterlace2.*: Reset element state on PAUSED->READY properly, don't leak any buffers when finali... Original commit message from CVS: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_class_init), (gst_deinterlace2_init), (gst_deinterlace2_reset_history), (gst_deinterlace2_reset), (gst_deinterlace2_finalize), (gst_deinterlace2_chain), (gst_deinterlace2_sink_event), (gst_deinterlace2_change_state), (gst_deinterlace2_src_query): * gst/deinterlace2/gstdeinterlace2.h: Reset element state on PAUSED->READY properly, don't leak any buffers when finalizing, allocate buffers with gst_pad_alloc_buffer() and properly return flow returns from gst_pad_push() instead of ignoring them. 2008-06-20 13:45:08 +0000 Sebastian Dröge [MOVED FROM BAD 05/56] gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: Add missing header. Original commit message from CVS: * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h: Add missing header. 2008-06-20 13:24:29 +0000 Sebastian Dröge [MOVED FROM BAD 04/56] Fix compilation on generic x86/amd64 and include deinterlace2 in the build system. Because of several bugs it's still... Original commit message from CVS: * configure.ac: * gst/deinterlace2/Makefile.am: * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc: Fix compilation on generic x86/amd64 and include deinterlace2 in the build system. Because of several bugs it's still enabled only by --enable-experimental. 2008-06-18 06:31:13 +0000 Stefan Kost [MOVED FROM BAD 03/56] Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * examples/app/appsrc-ra.c: * examples/app/appsrc-seekable.c: * examples/app/appsrc-stream.c: * examples/app/appsrc-stream2.c: * ext/directfb/dfbvideosink.h: * ext/metadata/gstbasemetadata.c: * ext/metadata/gstbasemetadata.h: * ext/metadata/metadata.c: * ext/metadata/metadataexif.c: * ext/theora/theoradec.h: * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/speedy.c: * gst/deinterlace2/tvtime/speedy.h: * gst/deinterlace2/tvtime/vfir.c: Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments. 2008-06-11 11:12:49 +0000 Martin Eikermann [MOVED FROM BAD 02/56] gst/deinterlace2/: Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer, which was relicensed to LGPL f... Original commit message from CVS: Based on a patch by: Martin Eikermann * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_method_get_type), (gst_deinterlace2_fields_get_type), (gst_deinterlace2_field_layout_get_type), (gst_deinterlace2_base_init), (gst_deinterlace2_class_init), (gst_deinterlace2_init), (gst_deinterlace2_set_method), (gst_deinterlace2_set_property), (gst_deinterlace2_get_property), (gst_deinterlace2_finalize), (gst_deinterlace2_pop_history), (gst_deinterlace2_head_history), (gst_deinterlace2_push_history), (gst_deinterlace2_deinterlace_scanlines), (gst_deinterlace2_chain), (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event), (gst_deinterlace2_change_state), (gst_deinterlace2_src_event), (gst_deinterlace2_src_query), (gst_deinterlace2_src_query_types), (plugin_init): * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/greedy.c: (copy_scanline), (deinterlace_greedy_packed422_scanline_mmxext), (dscaler_greedyl_get_method): * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/greedyh.c: (deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method), (greedyh_init), (greedyh_filter_mmx), (greedyh_filter_3dnow), (greedyh_filter_sse): * gst/deinterlace2/tvtime/greedyh.h: * gst/deinterlace2/tvtime/greedyhmacros.h: * gst/deinterlace2/tvtime/mmx.h: * gst/deinterlace2/tvtime/plugins.h: * gst/deinterlace2/tvtime/speedtools.h: * gst/deinterlace2/tvtime/speedy.c: (multiply_alpha), (clip255), (comb_factor_packed422_scanline_mmx), (diff_factor_packed422_scanline_c), (diff_factor_packed422_scanline_mmx), (diff_packed422_block8x8_mmx), (diff_packed422_block8x8_c), (packed444_to_packed422_scanline_c), (packed422_to_packed444_scanline_c), (packed422_to_packed444_rec601_scanline_c), (vfilter_chroma_121_packed422_scanline_mmx), (vfilter_chroma_121_packed422_scanline_c), (vfilter_chroma_332_packed422_scanline_mmx), (vfilter_chroma_332_packed422_scanline_c), (kill_chroma_packed422_inplace_scanline_mmx), (kill_chroma_packed422_inplace_scanline_c), (invert_colour_packed422_inplace_scanline_mmx), (invert_colour_packed422_inplace_scanline_c), (mirror_packed422_inplace_scanline_c), (interpolate_packed422_scanline_c), (convert_uyvy_to_yuyv_scanline_mmx), (convert_uyvy_to_yuyv_scanline_c), (interpolate_packed422_scanline_mmx), (interpolate_packed422_scanline_mmxext), (blit_colour_packed422_scanline_c), (blit_colour_packed422_scanline_mmx), (blit_colour_packed422_scanline_mmxext), (blit_colour_packed4444_scanline_c), (blit_colour_packed4444_scanline_mmx), (blit_colour_packed4444_scanline_mmxext), (small_memcpy), (speedy_memcpy_c), (speedy_memcpy_mmx), (speedy_memcpy_mmxext), (blit_packed422_scanline_c), (blit_packed422_scanline_mmx), (blit_packed422_scanline_mmxext), (composite_colour4444_alpha_to_packed422_scanline_c), (composite_colour4444_alpha_to_packed422_scanline_mmxext), (composite_packed4444_alpha_to_packed422_scanline_c), (composite_packed4444_alpha_to_packed422_scanline_mmxext), (composite_packed4444_to_packed422_scanline_c), (composite_packed4444_to_packed422_scanline_mmxext), (composite_alphamask_to_packed4444_scanline_c), (composite_alphamask_to_packed4444_scanline_mmxext), (composite_alphamask_alpha_to_packed4444_scanline_c), (premultiply_packed4444_scanline_c), (premultiply_packed4444_scanline_mmxext), (blend_packed422_scanline_c), (blend_packed422_scanline_mmxext), (quarter_blit_vertical_packed422_scanline_mmxext), (quarter_blit_vertical_packed422_scanline_c), (subpix_blit_vertical_packed422_scanline_c), (a8_subpix_blit_scanline_c), (myround), (init_RGB_to_YCbCr_tables), (init_YCbCr_to_RGB_tables), (rgb24_to_packed444_rec601_scanline_c), (rgba32_to_packed4444_rec601_scanline_c), (packed444_to_rgb24_rec601_scanline_c), (packed444_to_nonpremultiplied_packed4444_scanline_c), (aspect_adjust_packed4444_scanline_c), (setup_speedy_calls), (speedy_get_accel): * gst/deinterlace2/tvtime/speedy.h: * gst/deinterlace2/tvtime/sse.h: * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy), (deinterlace_frame_di_tomsmocomp), (dscaler_tomsmocomp_get_method), (tomsmocomp_init), (tomsmocomp_filter_mmx), (tomsmocomp_filter_3dnow), (tomsmocomp_filter_sse): * gst/deinterlace2/tvtime/tomsmocomp.h: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoop0A.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA8.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA2.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA6.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH2.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVA.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVAH.inc: * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc: * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line), (deinterlace_scanline_vfir), (copy_scanline), (dscaler_vfir_get_method): * gst/deinterlace2/tvtime/x86-64_macros.inc: Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer, which was relicensed to LGPL for GStreamer and in theory provides better and faster results than the simple deinterlace element. Fixes bug #163578. Ported to GStreamer 0.10 but still not enabled or included in the build system by default because of bad artefacts caused by a bug somewhere and as it can be only build on x86/amd64 ATM and requires special CFLAGS. Will be fixed soon. 2008-06-11 11:12:14 +0000 Martin Eikermann [MOVED FROM BAD 01/56] gst/deinterlace2/: Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer, which was relicensed to LGPL f... Original commit message from CVS: Based on a patch by: Martin Eikermann * gst/deinterlace2/Makefile.am: * gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_method_get_type), (gst_deinterlace2_fields_get_type), (gst_deinterlace2_field_layout_get_type), (gst_deinterlace2_base_init), (gst_deinterlace2_class_init), (gst_deinterlace2_init), (gst_deinterlace2_set_method), (gst_deinterlace2_set_property), (gst_deinterlace2_get_property), (gst_deinterlace2_finalize), (gst_deinterlace2_pop_history), (gst_deinterlace2_head_history), (gst_deinterlace2_push_history), (gst_deinterlace2_deinterlace_scanlines), (gst_deinterlace2_chain), (gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event), (gst_deinterlace2_change_state), (gst_deinterlace2_src_event), (gst_deinterlace2_src_query), (gst_deinterlace2_src_query_types), (plugin_init): * gst/deinterlace2/gstdeinterlace2.h: * gst/deinterlace2/tvtime/greedy.c: (copy_scanline), (deinterlace_greedy_packed422_scanline_mmxext), (dscaler_greedyl_get_method): * gst/deinterlace2/tvtime/greedyh.asm: * gst/deinterlace2/tvtime/greedyh.c: (deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method), (greedyh_init), (greedyh_filter_mmx), (greedyh_filter_3dnow), (greedyh_filter_sse): * gst/deinterlace2/tvtime/greedyh.h: * gst/deinterlace2/tvtime/greedyhmacros.h: * gst/deinterlace2/tvtime/mmx.h: * gst/deinterlace2/tvtime/plugins.h: * gst/deinterlace2/tvtime/speedtools.h: * gst/deinterlace2/tvtime/speedy.c: (multiply_alpha), (clip255), (comb_factor_packed422_scanline_mmx), (diff_factor_packed422_scanline_c), (diff_factor_packed422_scanline_mmx), (diff_packed422_block8x8_mmx), (diff_packed422_block8x8_c), (packed444_to_packed422_scanline_c), (packed422_to_packed444_scanline_c), (packed422_to_packed444_rec601_scanline_c), (vfilter_chroma_121_packed422_scanline_mmx), (vfilter_chroma_121_packed422_scanline_c), (vfilter_chroma_332_packed422_scanline_mmx), (vfilter_chroma_332_packed422_scanline_c), (kill_chroma_packed422_inplace_scanline_mmx), (kill_chroma_packed422_inplace_scanline_c), (invert_colour_packed422_inplace_scanline_mmx), (invert_colour_packed422_inplace_scanline_c), (mirror_packed422_inplace_scanline_c), (interpolate_packed422_scanline_c), (convert_uyvy_to_yuyv_scanline_mmx), (convert_uyvy_to_yuyv_scanline_c), (interpolate_packed422_scanline_mmx), (interpolate_packed422_scanline_mmxext), (blit_colour_packed422_scanline_c), (blit_colour_packed422_scanline_mmx), (blit_colour_packed422_scanline_mmxext), (blit_colour_packed4444_scanline_c), (blit_colour_packed4444_scanline_mmx), (blit_colour_packed4444_scanline_mmxext), (small_memcpy), (speedy_memcpy_c), (speedy_memcpy_mmx), (speedy_memcpy_mmxext), (blit_packed422_scanline_c), (blit_packed422_scanline_mmx), (blit_packed422_scanline_mmxext), (composite_colour4444_alpha_to_packed422_scanline_c), (composite_colour4444_alpha_to_packed422_scanline_mmxext), (composite_packed4444_alpha_to_packed422_scanline_c), (composite_packed4444_alpha_to_packed422_scanline_mmxext), (composite_packed4444_to_packed422_scanline_c), (composite_packed4444_to_packed422_scanline_mmxext), (composite_alphamask_to_packed4444_scanline_c), (composite_alphamask_to_packed4444_scanline_mmxext), (composite_alphamask_alpha_to_packed4444_scanline_c), (premultiply_packed4444_scanline_c), (premultiply_packed4444_scanline_mmxext), (blend_packed422_scanline_c), (blend_packed422_scanline_mmxext), (quarter_blit_vertical_packed422_scanline_mmxext), (quarter_blit_vertical_packed422_scanline_c), (subpix_blit_vertical_packed422_scanline_c), (a8_subpix_blit_scanline_c), (myround), (init_RGB_to_YCbCr_tables), (init_YCbCr_to_RGB_tables), (rgb24_to_packed444_rec601_scanline_c), (rgba32_to_packed4444_rec601_scanline_c), (packed444_to_rgb24_rec601_scanline_c), (packed444_to_nonpremultiplied_packed4444_scanline_c), (aspect_adjust_packed4444_scanline_c), (setup_speedy_calls), (speedy_get_accel): * gst/deinterlace2/tvtime/speedy.h: * gst/deinterlace2/tvtime/sse.h: * gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy), (deinterlace_frame_di_tomsmocomp), (dscaler_tomsmocomp_get_method), (tomsmocomp_init), (tomsmocomp_filter_mmx), (tomsmocomp_filter_3dnow), (tomsmocomp_filter_sse): * gst/deinterlace2/tvtime/tomsmocomp.h: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoop0A.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA8.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA2.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA6.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH2.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVA.inc: * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVAH.inc: * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc: * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc: * gst/deinterlace2/tvtime/vfir.c: (deinterlace_line), (deinterlace_scanline_vfir), (copy_scanline), (dscaler_vfir_get_method): * gst/deinterlace2/tvtime/x86-64_macros.inc: Add a deinterlacer plugin based on the tvtime/DScaler deinterlacer, which was relicensed to LGPL for GStreamer and in theory provides better and faster results than the simple deinterlace element. Fixes bug #163578. Ported to GStreamer 0.10 but still not enabled or included in the build system by default because of bad artefacts caused by a bug somewhere and as it can be only build on x86/amd64 ATM and requires special CFLAGS. Will be fixed soon. 2009-05-13 10:30:35 +0200 Sebastian Dröge * configure.ac: flv: Actually add the flv plugin to configure.ac 2009-05-13 09:24:26 +0100 Tim-Philipp Müller * tests/check/pipelines/flacdec.c: checks: fix flacdec unit tests on big-endian machines and under valgrind Flacdec outputs 16-bit samples, so let's check if the value of the first sample is what we expect rather than just the first byte, which may be different from what we expect depending on the host's endianness. Fixes the flacdec unit tests on PPC. Also fix a bunch of leaks in the unit tests to make valgrind happy. Fixes #582420. 2009-05-13 09:18:07 +0100 Tim-Philipp Müller * ext/flac/gstflacdec.c: flacdec: fix buffer leak gst_buffer_replace() will take its own ref, so we still have to unref the buffer if we don't need it any longer. 2009-05-12 21:20:04 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Fix pointer arithmetic This fixes a seeking regression, bug #134522. 2009-05-12 19:22:07 +0100 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: lamemp3enc: add Since tag to gtk-doc chunk 2009-05-12 21:36:31 +0200 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Moved 'flv' from -bad to -good 2009-05-07 17:53:42 +0100 Christian Schaller * gst/flv/gstflvdemux.c: [MOVED FROM BAD 57/57] Add ranks to various muxers and encoders in -bad 2009-04-29 18:52:20 +0100 Tristan Matthews * gst/flv/gstflvmux.c: [MOVED FROM BAD 56/57] flvmux: init variable to NULL to fix compiler warning Fixes #580786. 2009-04-29 13:56:07 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: * gst/flv/gstflvparse.c: [MOVED FROM BAD 55/57] flv: Set/require the framed/parsed fields of the audio/mpeg caps to TRUE 2009-04-29 13:16:25 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: [MOVED FROM BAD 54/57] flv: Always write at least the minimal tags and write the PAR as tags 2009-04-29 13:03:46 +0200 Sebastian Dröge * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: [MOVED FROM BAD 53/57] flv: Add support for muxing some tags 2009-04-29 13:03:27 +0200 Sebastian Dröge * gst/flv/gstflvparse.c: [MOVED FROM BAD 52/57] flv: Add support for title tag 2009-04-29 09:40:41 +0200 Sebastian Dröge * gst/flv/gstflvparse.c: [MOVED FROM BAD 51/57] flv: Fix parsing of tags and add new mappings We shouldn't register a new GstTag for every unknown tag we find as this might lead to conflicts and also those tags are essentially unknown. Add mappings for some known tags and also convert string dates to GDate, as found in many FLV files. 2009-04-22 19:52:05 +0200 Sebastian Dröge * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: * gst/flv/gstflvmux.c: * gst/flv/gstflvmux.h: [MOVED FROM BAD 50/57] flv: Add documentation to flvmux and flvdemux Partially fixes bug #573737. 2009-01-22 13:39:34 +0100 Jan Urbanski * gst/flv/gstflvparse.c: [MOVED FROM BAD 49/57] Add support for ECMA arrays in script tags. Fixes bug #567965. Add support for ECMA arrays in script tags. This fixes seeking on some files that have the seek table stored inside an ECMA array instead of the normal array. 2008-12-03 11:43:00 +0000 Sebastian Dröge [MOVED FROM BAD 48/57] gst/flv/gstflvparse.c: Check if strings are valid UTF8 before using them. Original commit message from CVS: * gst/flv/gstflvparse.c: (FLV_GET_STRING): Check if strings are valid UTF8 before using them. 2008-11-24 11:17:19 +0000 Julien Moutte [MOVED FROM BAD 47/57] gst/flv/gstflvdemux.c: Fix non key unit seeking by always going to the previous keyframe. Mark the discont flag when ... Original commit message from CVS: 2008-11-24 Julien Moutte * gst/flv/gstflvdemux.c: (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull): Fix non key unit seeking by always going to the previous keyframe. Mark the discont flag when we've moved in the file. * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate): MP3 streams are parsed already, makes autoplugged pipelines shorter. 2008-11-04 12:42:30 +0000 Stefan Kost [MOVED FROM BAD 46/57] Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-10-28 18:44:44 +0000 Sebastian Dröge [MOVED FROM BAD 45/57] gst/flv/gstflvdemux.c: Implement position query in time format. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_query): Implement position query in time format. 2008-10-28 18:41:19 +0000 Sebastian Dröge [MOVED FROM BAD 44/57] gst/flv/: Put the GstSegment directly into the instance struct instead of allocating and free'ing it again. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup), (gst_flv_demux_loop), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event), (gst_flv_demux_dispose), (gst_flv_demux_init): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp): Put the GstSegment directly into the instance struct instead of allocating and free'ing it again. Push tags already if only one pad was added, no need to wait for the second one. When generating our index set has_video and has_audio if we find video or audio in case the FLV header has incorrect data. 2008-10-27 09:45:04 +0000 Sebastian Dröge [MOVED FROM BAD 43/57] gst/flv/: Don't memcpy() all data we want to push downstream, instead just create subbuffers and push them downstream. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header), (gst_flv_demux_create_index): * gst/flv/gstflvparse.c: (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type), (gst_flv_parse_header): * gst/flv/gstflvparse.h: Don't memcpy() all data we want to push downstream, instead just create subbuffers and push them downstream. Fix some minor memory leaks. 2008-10-27 09:41:18 +0000 Sebastian Dröge [MOVED FROM BAD 42/57] gst/flv/Makefile.am: Fix (non-critical) syntax error and add all required CFLAGS and LIBS. Original commit message from CVS: * gst/flv/Makefile.am: Fix (non-critical) syntax error and add all required CFLAGS and LIBS. * gst/flv/gstflvparse.c: (FLV_GET_STRING), (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type): Rewrite the script tag parsing to make sure we don't try to read more data than we have. Also use GST_READ_UINT24_BE directly and fix some minor memory leaks. This should make all crashes on fuzzed FLV files disappear. 2008-10-27 09:37:21 +0000 Sebastian Dröge [MOVED FROM BAD 41/57] gst/flv/gstflvparse.c: Properly check everywhere that we have enough data to parse and don't read outside the allocat... Original commit message from CVS: * gst/flv/gstflvparse.c: (FLV_GET_STRING), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_type), (gst_flv_parse_header): Properly check everywhere that we have enough data to parse and don't read outside the allocated memory region. 2008-10-27 09:35:34 +0000 Sebastian Dröge [MOVED FROM BAD 40/57] gst/flv/gstflvparse.c: If the caps change during playback and negotiation fails error out instead of trying to continue. Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): If the caps change during playback and negotiation fails error out instead of trying to continue. 2008-10-27 09:33:40 +0000 Sebastian Dröge [MOVED FROM BAD 39/57] gst/flv/: Add support for Speex audio and allow buffers without valid timestamp in the muxer. Original commit message from CVS: * gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps), (gst_flv_mux_request_new_pad), (gst_flv_mux_write_buffer), (gst_flv_mux_collected): * gst/flv/gstflvmux.h: * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate): Add support for Speex audio and allow buffers without valid timestamp in the muxer. 2008-10-27 09:32:03 +0000 Sebastian Dröge [MOVED FROM BAD 38/57] gst/flv/gstflvdemux.c: Don't post an error message on the bus if sending EOS downstream didn't work. Fixes bug #550454. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_loop), (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull): Don't post an error message on the bus if sending EOS downstream didn't work. Fixes bug #550454. Fix seek event handling to look at the flags of the seek event instead of assuming some random flags, don't send segment-start messages when operating in push mode and push seek events upstream if we couldn't handle them. 2008-10-27 09:27:18 +0000 Sebastian Dröge [MOVED FROM BAD 37/57] gst/flv/gstflvdemux.c: Error out early if pulling a tag failed. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag): Error out early if pulling a tag failed. 2008-10-27 09:25:11 +0000 Sebastian Dröge [MOVED FROM BAD 36/57] gst/flv/: In pull mode we create our own index before doing anything else and don't use the index provided by some fi... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_create_index), (gst_flv_demux_loop): * gst/flv/gstflvparse.c: (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp): * gst/flv/gstflvparse.h: In pull mode we create our own index before doing anything else and don't use the index provided by some files (which are more than often incorrect and cause failed seeks). For push mode we still use the index provided by the file and extend it while doing the playback. 2008-10-27 09:20:01 +0000 Sebastian Dröge [MOVED FROM BAD 35/57] gst/flv/gstflvdemux.c: Instead of using gst_pad_event_default() use a small gst_pad_push_event() wrapper that only do... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_push_src_event), (gst_flv_demux_loop), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event): Instead of using gst_pad_event_default() use a small gst_pad_push_event() wrapper that only does what we want and is much more simple. 2008-10-27 09:14:45 +0000 Sebastian Dröge [MOVED FROM BAD 34/57] gst/flv/gstflvdemux.*: If our index was created by the element and not provided from the outside we should destroy it... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_change_state), (gst_flv_demux_set_index), (gst_flv_demux_init): * gst/flv/gstflvdemux.h: If our index was created by the element and not provided from the outside we should destroy it when starting a new stream to get all old entries removed. 2008-10-27 09:12:33 +0000 Sebastian Dröge [MOVED FROM BAD 33/57] gst/flv/gstflvdemux.c: Improve debugging a bit when pulling a buffer from upstream fails. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range): Improve debugging a bit when pulling a buffer from upstream fails. 2008-10-27 09:10:54 +0000 Sebastian Dröge [MOVED FROM BAD 32/57] gst/flv/: Close the currently playing segment from the streaming thread instead of the thread where the seek event is... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_dispose): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Close the currently playing segment from the streaming thread instead of the thread where the seek event is handled. 2008-10-16 15:21:15 +0000 Sebastian Dröge [MOVED FROM BAD 31/57] gst/flv/gstflvmux.c: Don't set video_codec to the value that actually should go into audio codec, otherwise we create... Original commit message from CVS: * gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps), (gst_flv_mux_write_buffer): Don't set video_codec to the value that actually should go into audio codec, otherwise we create invalid files. Fixes bug #556564. 2008-10-12 17:08:10 +0000 Sebastian Dröge [MOVED FROM BAD 30/57] gst/flv/gstflvdemux.c: Fix regression of handling flow returns in pull mode. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header): Fix regression of handling flow returns in pull mode. Fixes bug #556003. 2008-10-10 16:33:36 +0000 Sebastian Dröge [MOVED FROM BAD 29/57] gst/flv/gstflvparse.c: Use gst_pad_alloc_buffer_and_set_caps() to make sure we get a buffer with caps that we can wor... Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Use gst_pad_alloc_buffer_and_set_caps() to make sure we get a buffer with caps that we can work with (i.e. the pad's caps). Add non-keyframe video frames to the index too but without the keyframe flag. Add audio frames to the index only if we have no video stream. 2008-10-10 16:15:09 +0000 Sebastian Dröge [MOVED FROM BAD 28/57] gst/flv/gstflvparse.c: Create pads from the pad templates, use fixed caps on them and only activate them after the ca... Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Create pads from the pad templates, use fixed caps on them and only activate them after the caps are set. 2008-10-09 16:20:26 +0000 Sebastian Dröge [MOVED FROM BAD 27/57] gst/flv/: Get an approximate duration of the file by looking at the timestamp of the last tag in pull mode. If we get... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_loop): * gst/flv/gstflvparse.c: (gst_flv_parse_tag_timestamp): * gst/flv/gstflvparse.h: Get an approximate duration of the file by looking at the timestamp of the last tag in pull mode. If we get (maybe better) duration from metadata later we'll use that instead. 2008-10-09 15:43:02 +0000 Sebastian Dröge [MOVED FROM BAD 26/57] gst/flv/gstflvdemux.c: Refactor _pull_range() logic with checks into a seperate function to make things a bit more re... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range), (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header): Refactor _pull_range() logic with checks into a seperate function to make things a bit more readable. 2008-10-09 15:26:56 +0000 Sebastian Dröge [MOVED FROM BAD 25/57] gst/flv/gstflvdemux.c: Use gst_element_class_set_details_simple(). Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_chain), (gst_flv_demux_base_init): Use gst_element_class_set_details_simple(). If we get GST_FLOW_NOT_LINKED in the parse loop but at least one of the pads is linked continue the loop. 2008-10-09 10:00:51 +0000 Sebastian Dröge [MOVED FROM BAD 24/57] gst/flv/gstflvparse.c: Correct caps for video codec id 5: It's On2 VP6 with alpha channel which needs a different dec... Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate), (gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate): Correct caps for video codec id 5: It's On2 VP6 with alpha channel which needs a different decoder and has different caps. Add support for audio codec id 14, which is MP3 with 8kHz sampling rate. Fix endianness and signedness for raw audio codec ids. Add support for alaw and mulaw audio. 2008-10-09 09:48:46 +0000 Sebastian Dröge [MOVED FROM BAD 23/57] gst/flv/gstflvdemux.c: Go out of the parse loop as soon as we get an error instead of parsing until the GstAdapter is... Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_chain): Go out of the parse loop as soon as we get an error instead of parsing until the GstAdapter is empty. Add some explanations about the header and tag size. Don't print synchronizing message if everything is fine. 2008-10-09 09:26:58 +0000 Sebastian Dröge [MOVED FROM BAD 22/57] gst/flv/: Add first version of a FLV muxer. The only missing feature is writing of stream metadata. Original commit message from CVS: * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: (plugin_init): * gst/flv/gstflvmux.c: (gst_flv_mux_base_init), (gst_flv_mux_class_init), (gst_flv_mux_init), (gst_flv_mux_finalize), (gst_flv_mux_reset), (gst_flv_mux_handle_src_event), (gst_flv_mux_handle_sink_event), (gst_flv_mux_video_pad_setcaps), (gst_flv_mux_audio_pad_setcaps), (gst_flv_mux_request_new_pad), (gst_flv_mux_release_pad), (gst_flv_mux_write_header), (gst_flv_mux_write_buffer), (gst_flv_mux_collected), (gst_flv_mux_change_state): * gst/flv/gstflvmux.h: Add first version of a FLV muxer. The only missing feature is writing of stream metadata. 2008-06-13 22:46:43 +0000 Julien Moutte [MOVED FROM BAD 21/57] gst/flv/: Introduce demuxing support for AAC and Original commit message from CVS: 2008-06-14 Julien Moutte * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup), (gst_flv_demux_dispose): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate), (gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate), (gst_flv_parse_tag_video): Introduce demuxing support for AAC and H.264/AVC inside FLV. * sys/dshowdecwrapper/gstdshowaudiodec.c: (gst_dshowaudiodec_init), (gst_dshowaudiodec_chain), (gst_dshowaudiodec_push_buffer), (gst_dshowaudiodec_sink_event), (gst_dshowaudiodec_setup_graph): * sys/dshowdecwrapper/gstdshowaudiodec.h: * sys/dshowdecwrapper/gstdshowvideodec.c: (gst_dshowvideodec_init), (gst_dshowvideodec_sink_event), (gst_dshowvideodec_chain), (gst_dshowvideodec_push_buffer), (gst_dshowvideodec_src_getcaps): * sys/dshowdecwrapper/gstdshowvideodec.h: Lot of random fixes to improve stability (ref counting, safety checks...) 2008-04-25 08:07:36 +0000 Wim Taymans [MOVED FROM BAD 20/57] gst/flv/gstflvdemux.c: Forward unknown queries upstream instead of returning FALSE on them. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_query): Forward unknown queries upstream instead of returning FALSE on them. 2008-04-11 23:19:21 +0000 Tim-Philipp Müller [MOVED FROM BAD 19/57] gst/flv/gstflvparse.c: Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes crash caused by a strlen on a... Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script): Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes crash caused by a strlen on a NULL string (#527622). 2007-12-11 11:54:43 +0000 Tim-Philipp Müller [MOVED FROM BAD 18/57] gst/flv/gstflvparse.c: Don't strdup (and thus leak) codec name strings when passing them to gst_tag_list_add(). Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Don't strdup (and thus leak) codec name strings when passing them to gst_tag_list_add(). 2007-12-09 19:37:53 +0000 Edward Hervey [MOVED FROM BAD 17/57] gst/flv/gstflvparse.c: Fix list of supported and known codecs. Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Fix list of supported and known codecs. Emit tag with the codec name so it gets properly reported in totem and other applications. 2007-11-25 10:45:09 +0000 Edward Hervey [MOVED FROM BAD 16/57] gst/flv/gstflvparse.c: Output segment with proper 'stop' value, makes flvdemux 100% compatible with gnonlin. Original commit message from CVS: * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Output segment with proper 'stop' value, makes flvdemux 100% compatible with gnonlin. 2007-11-12 19:22:24 +0000 Edward Hervey [MOVED FROM BAD 15/57] gst/flv/gstflvparse.c: Add mapping for Nellymoser ASAO audio codec. Original commit message from CVS: * gst/flv/gstflvparse.c: Add mapping for Nellymoser ASAO audio codec. (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we actually have data to read at the end of the tag. This avoids trying to allocate negative buffers. 2007-10-22 15:45:49 +0000 Julien Moutte [MOVED FROM BAD 14/57] gst/flv/gstflvparse.c: Don't emit no-more-pads for single pad scenarios as the header is definitely not reliable. We ... Original commit message from CVS: 2007-10-22 Julien MOUTTE * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_type): Don't emit no-more-pads for single pad scenarios as the header is definitely not reliable. We emit them for 2 pads scenarios though to speed up media discovery. 2007-09-27 10:06:23 +0000 Julien Moutte [MOVED FROM BAD 13/57] gst/flv/gstflvparse.c: I got it wrong again, audio rate was not detected correctly in all cases. Original commit message from CVS: 2007-09-27 Julien MOUTTE * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): I got it wrong again, audio rate was not detected correctly in all cases. 2007-09-26 16:30:50 +0000 Julien Moutte [MOVED FROM BAD 12/57] gst/flv/gstflvparse.c: codec_data is needed for every tag not just the first one. (Fix a stupid bug i introduced with... Original commit message from CVS: 2007-09-26 Julien MOUTTE * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): codec_data is needed for every tag not just the first one. (Fix a stupid bug i introduced without testing) 2007-09-26 11:17:08 +0000 Julien Moutte [MOVED FROM BAD 11/57] gst/flv/gstflvparse.c: Fix bit masks operations to be sure we detect the codec_tags and sample rates correctly. Original commit message from CVS: 2007-09-26 Julien MOUTTE * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Fix bit masks operations to be sure we detect the codec_tags and sample rates correctly. Fix raw audio caps generation. 2007-09-12 08:38:22 +0000 Peter Kjellerstedt [MOVED FROM BAD 10/57] gst/: Printf format fixes (#476128). Original commit message from CVS: Patch by: Peter Kjellerstedt * gst-libs/gst/app/gstappsink.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvparse.c: * gst/interleave/deinterleave.c: * gst/switch/gstswitch.c: Printf format fixes (#476128). 2007-08-27 14:56:05 +0000 Julien Moutte [MOVED FROM BAD 09/57] gst/flv/gstflvdemux.c: Make sure we initialize the seek result. Original commit message from CVS: 2007-08-27 Julien MOUTTE * gst/flv/gstflvdemux.c: (gst_flv_demux_handle_seek_pull): Make sure we initialize the seek result. 2007-08-24 17:03:15 +0000 Julien Moutte [MOVED FROM BAD 08/57] gst/flv/gstflvdemux.c: Remove some useless ifdef. Original commit message from CVS: 2007-08-24 Julien MOUTTE * gst/flv/gstflvdemux.c: (gst_flv_demux_flush), (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event), (gst_flv_demux_src_event): Remove some useless ifdef. 2007-08-24 15:31:26 +0000 Julien Moutte [MOVED FROM BAD 07/57] gst/flv/gstflvdemux.c: Implement seeking in push mode. Original commit message from CVS: 2007-08-24 Julien MOUTTE * gst/flv/gstflvdemux.c: (gst_flv_demux_flush), (gst_flv_demux_cleanup), (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push), (gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event), (gst_flv_demux_src_event): Implement seeking in push mode. * gst/flv/gstflvdemux.h: 2007-08-22 14:50:51 +0000 Julien Moutte [MOVED FROM BAD 06/57] gst/flv/: Handle pixel aspect ratio through metadata tags like ASF does. Fluendo muxer supports this and Original commit message from CVS: 2007-08-22 Julien MOUTTE * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup), (gst_flv_demux_pull_tag): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Handle pixel aspect ratio through metadata tags like ASF does. Fluendo muxer supports this and Flash players can support it as well this way. 2007-08-22 14:03:42 +0000 Julien Moutte [MOVED FROM BAD 05/57] gst/flv/: Make sure we don't try filling up the index if no times object was parsed. Fix the way we decide to push ta... Original commit message from CVS: 2007-08-22 Julien MOUTTE * gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag): * gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Make sure we don't try filling up the index if no times object was parsed. Fix the way we decide to push tags and emit no-more-pads. Fix some printf typing in debugging. 2007-08-14 14:56:20 +0000 Wim Taymans [MOVED FROM BAD 04/57] gst/flv/gstflvdemux.c: Fix locking and refcounting on the index. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_set_index), (gst_flv_demux_get_index): Fix locking and refcounting on the index. 2007-08-14 14:22:09 +0000 Julien Moutte [MOVED FROM BAD 03/57] gst/flv/gstflvdemux.c: First method for seeking in pull mode using the index built step by step or coming from metadata. Original commit message from CVS: 2007-08-14 Julien MOUTTE * gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup), (gst_flv_demux_adapter_flush), (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_do_seek), (gst_flv_demux_handle_seek), (gst_flv_demux_sink_event), (gst_flv_demux_src_event), (gst_flv_demux_query), (gst_flv_demux_change_state), (gst_flv_demux_set_index), (gst_flv_demux_get_index), (gst_flv_demux_dispose), (gst_flv_demux_class_init): First method for seeking in pull mode using the index built step by step or coming from metadata. * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (FLV_GET_STRING), (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Parse more metadata types and keyframes index. 2007-07-25 13:29:04 +0000 Julien Moutte [MOVED FROM BAD 02/57] gst/flv/: Handle not linked pads, try to make it reusable, more safety checks. Original commit message from CVS: 2007-07-25 Julien MOUTTE (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_change_state), (gst_flv_demux_dispose), (gst_flv_demux_init): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (FLV_GET_STRING), (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_header): * gst/flv/gstflvparse.h: Handle not linked pads, try to make it reusable, more safety checks. 2007-07-19 15:05:30 +0000 Julien Moutte [MOVED FROM BAD 01/57] Adds a first draft of an FLV demuxer. Original commit message from CVS: 2007-07-19 Julien MOUTTE * configure.ac: * gst/flv/Makefile.am: * gst/flv/gstflvdemux.c: (gst_flv_demux_flush), (gst_flv_demux_cleanup), (gst_flv_demux_chain), (gst_flv_demux_pull_tag), (gst_flv_demux_pull_header), (gst_flv_demux_seek_to_prev_keyframe), (gst_flv_demux_loop), (gst_flv_demux_sink_activate), (gst_flv_demux_sink_activate_push), (gst_flv_demux_sink_activate_pull), (gst_flv_demux_sink_event), (gst_flv_demux_change_state), (gst_flv_demux_dispose), (gst_flv_demux_base_init), (gst_flv_demux_class_init), (gst_flv_demux_init), (plugin_init): * gst/flv/gstflvdemux.h: * gst/flv/gstflvparse.c: (FLV_GET_BEUI24), (FLV_GET_STRING), (gst_flv_demux_query_types), (gst_flv_demux_query), (gst_flv_parse_metadata_item), (gst_flv_parse_tag_script), (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video), (gst_flv_parse_tag_type), (gst_flv_parse_header): * gst/flv/gstflvparse.h: Adds a first draft of an FLV demuxer. It does not do seeking yet, it supports pull and push mode so YES you can use it to play youtube videos directly from an HTTP uri. Not so much testing done yet but it parses metadata, reply to duration queries, etc... 2009-05-12 13:00:46 +0200 Sebastian Dröge * gst/rtp/Makefile.am: rtp: Link to -lm Fixes bug #582281. 2009-05-12 11:16:48 +0200 Sebastian Dröge * tests/check/elements/rganalysis.c: rganalysis: Remove invalid unit test The test creates buffers with non-silence, sets the GAP flag on it and expects rganalysis to ignore the content and assume silence. That's not the way how GAP buffers should be used, if the GAP flag is set elements *can* assume that they only contain silence but they're not *required* to assume that. The GAP flag must only be set on silence buffers. Fixes bug #582252. 2009-05-12 00:48:49 +0100 Jan Schmidt * ChangeLog: * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: * win32/common/config.h: 0.10.14.2 pre-release 2009-05-11 23:13:20 +0100 Jan Schmidt * tests/files/Makefile.am: checks: dist id3-577468-unsynced-tag.tag test file 2009-05-11 21:02:27 +0200 Tristan Matthews * gst/avi/gstavidemux.c: avidemux: initialize variable to 0 Fixes #582218. 2009-05-11 18:21:13 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Only search for the index entry once 2009-05-11 18:18:36 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Use the first index entry if it's after the seek position 2009-05-11 18:15:22 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Use the first entry for a given stream if the first entry is after the seek position 2009-05-11 16:50:48 +0200 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Use binary search for finding the requested index entry when seeking 2009-05-11 15:36:46 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Improve/optimize seeking First of all a keyframe seek should be done to the keyframe right before the requested position and not to the keyframe that is nearest to the requested position. Use per track index arrays and use our new binary search function from core to speed up the search. 2009-05-11 15:36:36 +0200 Sebastian Dröge * configure.ac: Require released versions of core/base 2009-05-11 10:15:00 +0200 Sebastian Dröge * tests/check/Makefile.am: gdkpixbuf: Use the libs and cflags of gdk pixbuf instead of gtk This fixes the build if gdk-pixbuf is found but gtk isn't 2009-05-11 09:58:48 +0200 Sebastian Dröge * configure.ac: Always define the conditional HAVE_GTK to fix configure in some cases 2009-05-10 16:53:07 +0200 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lamemp3enc: Don't write a Xing header 2009-05-10 11:17:23 +0200 Marc-Andre Lureau * autogen.sh: Run libtoolize before aclocal This unbreaks the build in some cases. Fixes bug #582021 2009-05-09 10:50:45 -0700 David Schleef * gst/matroska/matroska-demux.c: matroska: fix printf format to agree with argument 2009-05-08 19:42:10 +0100 Tim-Philipp Müller * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: raw1394: include stdlib.h for strtol() Fixes compiler warning when compiling with xml stuff in core disabled. 2009-05-08 16:40:57 +0200 Edward Hervey * ext/flac/gstflacdec.c: flacdec: Actually output the pending buffer.. and not a blank one. It was previously sending the bogus buffer which was returned from the bufferalloc (required for reverse negotiation apparently) instead of the pending buffer. 2009-05-08 14:24:47 +0100 Christian Schaller * ext/twolame/gsttwolame.c: Switch twolame to primary rank 2009-05-08 12:00:57 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Allow non-string fields in the extra-headers property 2009-05-08 11:35:02 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kdepay.h: rtj2kdepay: add basic JPEG 2000 depayloader 2009-05-08 11:31:02 +0200 Wim Taymans * gst/rtp/gstrtpj2kpay.c: rtpj2kpay: set marker bit correctly 2009-05-08 11:29:04 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Add support for extra-headers appended to the HTTP request This allows to set the Referer header among other things by adding a "extra-headers" property that takes a GstStructure with field=string pairs. Fixes bug #581806. 2009-05-08 10:38:42 +0200 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpj2kpay.h: rtpj2kpay: add a simple JPEG 2000 payloader 2009-05-08 10:31:12 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: we only need to swap bits on LE 2009-05-07 18:10:08 +0100 Christian Schaller * ext/flac/gstflac.c: * ext/jpeg/gstjpeg.c: * ext/libpng/gstpng.c: * ext/speex/gstspeex.c: * gst/avi/gstavi.c: * gst/matroska/matroska-mux.c: Add RANKS for various encoders and muxers 2009-05-07 17:59:52 +0100 Christian Schaller * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: Add ranks to mp3 encoders 2009-05-07 17:59:52 +0100 Christian Schaller * ext/twolame/gsttwolame.c: Add ranks to mp3 encoders 2009-05-07 17:09:44 +0200 Wim Taymans * gst/matroska/matroska-demux.c: matroskademux: add some debugging 2009-05-07 15:58:43 +0200 Wim Taymans * gst/matroska/matroska-demux.c: matroskademux: parse xiph headers length correctly See #580980 2009-05-07 16:25:41 +0200 Gabriel Bouvigne * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrganalysis.h: * gst/replaygain/rganalysis.c: * gst/replaygain/rganalysis.h: rganalysis: Add ability to post level messages Fixes bug #581568. 2009-05-07 10:10:02 +0200 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lamemp3enc: Fixup the bitrate only for CBR Additionally clarify some property descriptions. 2009-05-06 23:56:44 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: refuse some unsupported jpeg formats 2009-05-06 21:47:17 +0200 Alessandro Decina * ext/lame/gstlamemp3enc.c: lame: fix format string in debug statement 2009-05-06 18:06:49 +0200 Wim Taymans * gst/rtp/gstrtptheorapay.c: rtptheorapay: fix description 2009-05-06 16:09:13 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: rewrite quant table handling Rewrite the quant table parsing to also handle multiple tables in one JPEG HDQ segment. Handle more jpeg types by keeping track of the tables used per component and putting the used ones in the quant headers. 2009-04-18 17:23:51 +0100 Jan Schmidt * tests/check/elements/id3v2mux.c: id3v2mux: Make the test failure slightly more informative 2009-04-20 18:33:09 +0100 Jan Schmidt * ext/flac/gstflacdec.c: flac: Make buffers created during seek act like normal buffers. Store the offset and caps when allocating a buffer during seeking, and then allocate a new buffer with buffer_alloc before we push it out. This ensures that in all respects the first buffer decoded during seeking behaves like all other buffers, including allowing downstream re-negotiation. 2009-04-18 18:00:54 +0200 Thomas Vander Stichele * ext/flac/gstflacdec.c: flacdec: don't use pad_alloc when decoding while seeking. Fixes #579422 2009-05-06 13:22:51 +0200 Arnout Vandecappelle * ext/jpeg/gstjpegdec.c: jpegdec: refactored gst_jpeg_dec_parse_image_data Fixes #579808 2009-05-06 13:11:53 +0200 Arnout Vandecappelle * ext/jpeg/gstjpegdec.c: jpegdec: support additional 0xff before end marker. JPEG markers may be preceded by additional 0xff. jpegdec should skip over these, even before the end marker. See #579808 2009-05-06 12:54:22 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: handle input with 1 quant table Also handle input with just one quant table, simply duplicate the quant table. Handle invalid SOF correctly and some small cleanups. Fixes #578257 2009-04-29 15:58:10 +0300 Marco Ballesio * gst/qtdemux/qtdemux.c: qtdemux: fix byte order swapping in 3GPP classification entity tag Fixes #580746. 2009-05-05 16:38:19 +0100 Tim-Philipp Müller * ext/lame/gstlamemp3enc.c: lame: fix compilation with LAME versions < 3.98 lame_set_VBR_quality(), which takes a floating point value for the quality, has been added only in v3.98. Use lame_set_VBR_q(), which takes quality as an integer, for older LAME versions. Fixes #581341. 2009-05-05 17:07:13 +0200 Arnout Vandecappelle * gst/multipart/multipartdemux.c: multipartdemux: avoid reading from inavlid memory Read the timestamp of the incomming buffer before we push it in the adapter and flush it out again as the buffer might be unreffed then and we read from invalid memory. Fixes #581444. 2009-05-05 17:03:29 +0200 Arnout Vandecappelle * gst/multipart/multipartdemux.c: multipartdemux: don't leak dynamic pads Free the dynamic pads data in finalize. Fixes #581432 2009-05-05 16:32:17 +0200 Wim Taymans * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpjpegpay.h: rtpjpegpay: correctly set the type header Don't require width/height on the caps. Use the SOF header to find width/height and fall back to the caps if there is no SOF. Also use the SOF info to find the subsampling and quantization tables used. This allows us to set the right type value in the JPEG rtp header. Deprecate the quality property, it's unused now and it was used wrongly before. Always send full quant tables for now until we have some code to detect default ones. Fixes #580880 2009-05-05 16:28:44 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegdepay.h: rtpjpegdepay: use width/height from payload Use the width and the height from the payload headers and set them on the output caps for added awesomeness. Fix quant parsing, we need to check the type in the lower 6 bits. Add first bits of caching quantization tables. 2009-05-05 16:24:16 +0200 Wim Taymans * ext/jpeg/gstjpegenc.c: jpegenc: set colorspace before _set_defaults() The libjpeg api says that we need to set the colorspace before we call _set_defaults(). Indeed, if we don't do that we end up with some very freaky non-standard quant table and huffman table indexes. 2009-05-05 13:19:19 +0100 Tim-Philipp Müller * tests/Makefile.am: tests: don't build examples if --disable-examples was passed to configure 2009-05-05 12:33:57 +0100 Tim-Philipp Müller * configure.ac: configure: clean up mess around gtk+ checking And don't check for gtk+ when it's not needed (ie. if examples are disabled) 2009-05-05 12:27:21 +0100 Tim-Philipp Müller * configure.ac: * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/pixbufscale.h: configure: make gdk-pixbuf plugin depend only on gdk-pixbuf, not gtk+ 2009-05-04 18:55:12 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix find_stream_by_* functions Fix various version of find_stream_by_* by not trying to convert an int to a pointer and vice versa, for portability reasons. Fixes #581333 2009-05-04 18:32:05 +0200 Chris Winter * gst/rtsp/gstrtspsrc.c: rtspsrc: fix dummy nat packet logic Fix a typo in the dummy NAT packet sending code. Fixes #581329 2009-04-30 10:24:27 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid errors after server eof Server eof (e.g. connection closed) is announced as connection closed, so better record state and act accordingly to prevent (read/write) errors during subsequent teardown/cleanup sequences. #Fixes 580851.(c). 2009-04-30 10:19:27 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: also set base_time on src after flush timestamps following flush/seek should be consistent between UDP and TCP interleaved case. Fixes #580851.(b). 2009-04-30 10:17:23 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: sanity checks on range info A max range that overflows should not be trusted, nor should a max range that equals the min range. Fixes #580851.(a). 2009-05-04 16:16:54 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: use SKIP flag to use SCALE headers We can use the SKIP seek flag to instruct the server to send data faster then normal but with the same bandwidth. Fixes #537609 2009-05-04 14:19:22 +0200 Alessandro Decina * ext/speex/gstspeexdec.c: speexdec: make speex_dec_convert work with same-format values when no data has been decoded. 2009-05-04 12:51:29 +0200 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lamemp3enc: Add a note to the encoding-engine-quality property that says, that this does not affect the bitrate at all. 2009-05-04 12:48:43 +0200 Sebastian Dröge * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame: Implement preset interface 2009-05-04 12:47:03 +0200 Sebastian Dröge * ext/twolame/gsttwolame.c: twolame: Implement preset interface 2009-05-04 12:43:42 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flac: Implement preset interface 2009-05-04 12:41:56 +0200 Sebastian Dröge * ext/speex/gstspeexenc.c: speex: Implement preset interface 2009-05-04 12:40:12 +0200 Sebastian Dröge * ext/wavpack/gstwavpackenc.c: wavpack: Implement preset interface 2009-05-04 12:35:19 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: use binary search for index Use the new binary search method for finding the right index entry faster. 2009-05-04 11:26:56 +0200 Wim Taymans * gst/videobox/gstvideobox.c: videobox: draw the complete U and V planes Round up the scaled U and V width and height so that we always draw the correct amount of pixels to fill the complete image. Fixes #569611 2009-04-30 10:21:40 +0200 Sebastian Dröge * ext/lame/gstlamemp3enc.c: * ext/lame/gstlamemp3enc.h: lamemp3enc: Remove fast-vbr property and rename vbr-quality to quality 2009-04-30 10:16:45 +0200 Sebastian Dröge * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: lame/lamemp3enc: Fix memory leak on FLUSH_STOP 2009-04-30 10:14:54 +0200 Sebastian Dröge * ext/lame/gstlame.c: lame: Deprecate the lame element 2009-04-30 10:10:08 +0200 Sebastian Dröge * ext/lame/gstlamemp3enc.c: lame: Update example pipelines with the new properties 2009-04-29 19:01:44 +0200 Sebastian Dröge * ext/lame/Makefile.am: * ext/lame/gstlame.c: * ext/lame/gstlamemp3enc.c: * ext/lame/gstlamemp3enc.h: * ext/lame/plugin.c: lame: Add lamemp3enc element with much simplified interface This deprecates the lame element and fixes bug #494528. 2009-05-01 19:35:11 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: add some more micro optimisations 2009-04-30 18:41:44 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: * gst/qtdemux/qtdemux_types.c: qtdemux: micro optimize qtdemux a little Sprinkle some G_LIKELY around. Avoid traversing and dumping the tree when debugging is not activated. 2009-04-30 14:22:27 +0200 Wim Taymans * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: add support for subtitle pictures Add support for subtitle pictures. Fixes #568278. 2009-04-30 10:32:39 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: make sure we always signal waiters Always signal the waiters in the async callbacks. Especially for the volume callbacks since this might cause deadlocks. 2009-04-29 18:09:07 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: release state lock before stopping task We need to release the state lock before trying to wait for the task to end because the task might also take the lock. Fixes #577671 2009-04-29 12:19:27 +0200 Hans de Goede * gst/qtdemux/qtdemux.c: qtdemux: handle ac-3 audio fix demuxing of m4v streams with ac-3 audio Fixes #580554 2009-04-29 11:12:36 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Use the tag merge mode that was set on the interface for merging tag events 2009-04-25 09:43:38 +0200 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: fix getaddrinfo error reporting getaddrinfo errors should be reported with gai_strerror instead of errno as spotted by MikeS. 2009-04-27 10:08:39 +0200 Wim Taymans * gst/rtp/gstrtpg726pay.c: g726pay: fix compilation 2009-04-27 10:02:06 +0200 Wim Taymans * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg726pay.h: g726pay: add RFC compliant packetizing Shuffle the input bits according to RFC 3551 for G726 payloads. Add option to force the previous behaviour. Fixes #567140 2009-04-27 09:59:54 +0200 Wim Taymans * gst/rtp/gstrtpg726depay.c: g726depay: add debug category Add a debugging category, add some comments and remove _peek_parent(). 2009-04-26 15:59:50 +0100 Tim-Philipp Müller * configure.ac: id3v2mux: we need taglib 1.5 for ID3v2::RelativeVolumeFrame::setIdentification Bump taglib requirement. 2009-04-24 02:11:28 +0100 Tim-Philipp Müller * tests/check/elements/id3demux.c: * tests/files/id3-577468-unsynced-tag.tag: id3demux: add unit test file for unsynced id3 tags 2009-04-24 01:51:35 +0100 Tim-Philipp Müller * gst/id3demux/id3tags.c: id3demux: parse unsynchronised tags properly We didn't handle unsynchronization at all up to now, which might have caused frames to not be extracted - esp. frames after an APIC picture frame. Fixes #577468. 2009-04-24 01:01:53 +0100 Tim-Philipp Müller * gst/id3demux/id3tags.c: id3demux: pass the right size value for size of all frames to the parser Frame data size is tag size adjusted for size of the tag header and footer, not tag size including header and footer. 2009-04-22 15:24:55 +0200 Patrick Radizi * gst/rtsp/gstrtspsrc.c: rtspsrc: fix some more pad leaks Fix some pad leaks. See #577318. 2009-04-21 22:12:45 +0100 Jan Schmidt * common: Automatic update of common submodule From b3941ea to 6ab11d1 2009-04-21 14:02:01 -0700 Michael Smith * gst/qtdemux/qtdemux.c: qtdemux: override caps based on data from ESDS atoms in mpeg4. If the codec is actually something else (e.g. mjpeg) change the caps to match when parsing the ESDS atom. Also, for AAC, override rate and channels with correct values read from ESDS, since the rate/channels values elsewhere are often wrong. 2009-04-20 19:32:00 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: fix warning for still images by not trying to divide by 0 Don't pass a 0 divisor to gst_util_uint64_scale(), or it will complain in the single image case where fps=0/1 (are we supposed to differentiate between no fps=still image and fps=0/1=variable rate here btw?) 2009-04-20 17:25:34 +0100 Jan Schmidt * gst/udp/gstudpnetutils.c: udp: Fix a simple typo in the previous commit Use #ifdef instead of #if, to fix the build 2009-04-20 15:48:21 +0200 Andy Wingo fix format string in pngdec * ext/libpng/gstpngdec.c: Fix size_t vs unsigned int format in error message. 2009-04-20 15:46:03 +0200 Andy Wingo only use struct ip_mreqn if it is detected * configure.ac: Make an explicit check for struct ip_mreqn. * gst/udp/gstudpnetutils.c: Use HAVE_IP_MREQN instead of the ad-hoc checks. 2009-04-20 13:45:32 +0200 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: Fix push mode buffering sanity check to actually fit the description. 2009-04-19 14:03:38 +0200 Edward Hervey * ext/twolame/gsttwolame.c: twolame: Remove unneeded variable, value assigned was never read. 2009-04-19 14:02:03 +0200 Edward Hervey * ext/lame/gstlame.c: lame: Remove unneeded variable, it's assigned a value never read. 2009-04-18 19:11:06 +0200 Edward Hervey * gst/rtp/gstrtph263pay.c: rtph263pay: And let's not forget to remove the unused variable. 2009-04-18 18:50:32 +0200 Edward Hervey * gst/rtp/gstrtph263pay.c: rtph263pay: Remove dead assignments, the variables are never read after. 2009-04-18 18:49:49 +0200 Edward Hervey * gst/rtp/gstrtpmp4vpay.c: rtpmp4vpay: Remove dead assignment. The value is never read after. 2009-04-18 18:48:55 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove dead assignment. t is being overwritten after, before it's used. 2009-04-18 18:48:06 +0200 Edward Hervey * gst/rtp/gstrtpamrdepay.c: rtpamrdepay: Remove unneeded variable, the value is only read once. 2009-04-18 18:47:05 +0200 Edward Hervey * gst/rtp/gstrtpamrpay.c: rtpamrpay: Remove unneeded variable, the value is only read once. 2009-04-18 18:46:12 +0200 Edward Hervey * gst/goom/filters.c: goom/filters: Remove dead assignment. Value overwritten just after. 2009-04-18 18:45:32 +0200 Edward Hervey * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: Remove dead assignment. Value never read after. 2009-04-18 18:45:07 +0200 Edward Hervey * gst/rtp/gstrtptheorapay.c: rtptheorapay: Remove dead assignment. Value never read after. 2009-04-18 18:43:31 +0200 Edward Hervey * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: Remove unused variable, it's never being read. 2009-04-18 18:42:45 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove dead assignment. 'res' isn't read after. 2009-04-18 18:41:58 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove unused variable. 'res' is never read. 2009-04-18 18:40:48 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Remove dead variable. 'stream' is never read after. 2009-04-18 18:39:48 +0200 Edward Hervey * gst/videobox/gstvideobox.c: videbox: Remove dead assignments. These variables are never read after this point. 2009-04-18 18:38:29 +0200 Edward Hervey * gst/goom/convolve_fx.c: goom: ff and iff are only used in a '#ifdef DRAW_MOTIF' block. 2009-04-18 18:34:11 +0200 Edward Hervey * gst/wavparse/gstwavparse.c: wavparse: Remove dead assignment. res isn't read after this. 2009-04-18 18:32:03 +0200 Edward Hervey * gst/wavparse/gstwavparse.c: wavparse: Remove dead assignments, move variable to where it's needed. The header_read_error label will return GST_FLOW_ERROR 2009-04-18 18:21:22 +0200 Edward Hervey * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: Remove dead assignment. The value of 'str' will never be used in these cases. 2009-04-18 18:19:12 +0200 Edward Hervey * gst/matroska/matroska-demux.c: matroskademux: Remove useless variable. iret was never read outside of that loop, and is always being exited if iret was != GST_FLOW_OK anyway. 2009-04-18 18:17:35 +0200 Edward Hervey * gst/avi/gstavidemux.c: avidemux: Move 'res' to where it's actually being used. res was never used outside of that block except for a dead assignment. 2009-04-18 18:16:33 +0200 Edward Hervey * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: audiofx: Remove unused variable. rz is never used in these methods. 2009-04-18 18:15:39 +0200 Edward Hervey * sys/osxaudio/gstosxringbuffer.c: osxringbuffer: Run gst-indent. 2009-04-18 18:14:49 +0200 Edward Hervey * sys/ximage/gstximagesrc.c: ximage: Remove dead assignments. Those variables are not read after that point. 2009-04-18 18:11:00 +0200 Edward Hervey * ext/dv/gstdvdemux.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/libcaca/gstcacasink.c: * ext/libpng/gstpngdec.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/auparse/gstauparse.c: * gst/effectv/gstquark.c: * gst/flx/gstflxdec.c: * gst/icydemux/gsticydemux.c: * gst/interleave/interleave.c: * gst/matroska/matroska-mux.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/qtdemux/gstrtpxqtdepay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmptealpha.c: * gst/smpte/paint.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/videomixer.c: * gst/wavparse/gstwavparse.c: * sys/ximage/gstximagesrc.c: Remove trivial unused variables detected by CLang static analyzer. 2009-04-18 17:52:00 +0200 Edward Hervey * ext/gconf/gstswitchsink.c: * gst/qtdemux/gstrtpxqtdepay.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtpvrawpay.c: Remove blank {set|get}_property/change_state/finalize methods. 2009-04-18 17:42:55 +0200 Edward Hervey * ext/cairo/gsttimeoverlay.c: * ext/esd/esdsink.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/pulse/pulsesink.c: * gst/alpha/gstalphacolor.c: * gst/cutter/gstcutter.c: * gst/debugutils/efence.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gsttaginject.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstudpsink.c: * gst/videofilter/gstvideobalance.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: Remove unused variables in _class_init Detected by LLVM's CLang static analyzer 2009-04-18 13:54:08 +0100 Jan Schmidt * tests/check/elements/souphttpsrc.c: check: Check whether threads are already initialised before g_thread_init() 2009-04-18 14:32:40 +0200 Josep Torra * gst/rtsp/gstrtspsrc.c: rtspsrc: mark discont on the streams as was said the debug line After a seek mark all streams with discont as it was said in the debug line. Fixes that buffers after a seek are generated without a valid timestamp. 2009-04-18 08:45:18 +0200 Josep Torra * gst/rtsp/gstrtspsrc.c: rtspsrc: map GST_RTSP_EEOF to EOS on server requests Permit properly handle the EOS condition when server report it in a request. 2009-04-18 08:39:57 +0200 Edward Hervey * gst/rtp/gstrtptheoradepay.c: rtptheoradepay: Fix build on macosx. Use G_GSIZE_FORMAT instead of u. 2009-04-16 22:50:59 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: fix sample offset calculation again 2009-04-15 19:32:18 +0100 Tim-Philipp Müller * sys/sunaudio/gstsunaudiomixerctrl.c: sunaudio: fix broken indentation of variable declarations 2009-04-15 19:28:53 +0100 James Andrewartha * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiosink.c: sunaudio: remove some unused variables and goto labels Fixes #579070. 2009-04-15 19:24:49 +0200 James Andrewartha * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: rtph263pay: fix compilation on big-endian Some semicolons were missing from the big-endian structs in gstrtph263pay.h. A GST_DEBUG call was missing a format specifier. Fixes #579069 2009-04-15 20:10:04 +0300 Marco Ballesio * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: * gst/qtdemux/quicktime.c: qtdemux: implement 3GPP (TS 26.244 V8.0.0) Asset metadata handling, Fixes #132193 Implements 3gpp iso metadata tags which are different from mov udta atoms. 2009-04-15 15:51:24 +0200 Peter Kjellerstedt * gst/debugutils/efence.h: debugutils: Use G_BEGIN_DECLS/G_END_DECLS. Use G_BEGIN_DECLS/G_END_DECLS to avoid gst-indent messing up the indentation due to extern "C" { }. 2009-04-15 16:03:27 +0300 Stefan Kost * configure.ac: * docs/plugins/Makefile.am: * gst/debugutils/Makefile.am: * gst/debugutils/breakmydata.c: * gst/debugutils/debug.vcproj: * gst/debugutils/efence.c: * gst/debugutils/efence.h: * gst/debugutils/efence.vcproj: * gst/debugutils/gstdebug.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavigationtest.h: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstnavseek.h: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gstpushfilesrc.h: * gst/debugutils/gsttaginject.c: * gst/debugutils/gsttaginject.h: * gst/debugutils/navigationtest.vcproj: * gst/debugutils/negotiation.c: * gst/debugutils/progressreport.c: * gst/debugutils/progressreport.h: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/debugutils/tests.c: * gst/debugutils/tests.h: debug: rename debug to debugutils to avoid clash with --disable-debug. Fixes #562168 2009-04-15 15:43:04 +0300 Stefan Kost * gst/debug/efence.c: * gst/debug/efence.h: * gst/debug/gstnavigationtest.h: * gst/debug/gstnavseek.h: * gst/debug/gstpushfilesrc.h: * gst/debug/gsttaginject.h: * gst/debug/progressreport.h: * gst/debug/tests.h: debug: indent before renaming 2009-04-15 14:07:57 +0200 Wim Taymans * gst/rtp/gstrtpg726depay.c: g726depay: add property for aal2 force 2009-04-15 13:56:17 +0200 Wim Taymans * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726depay.h: g726depay: implement RFC3551 packing We implemented the AAL2 packing, add the encoding-name for those to the caps and a property to force AAL2 decoding (always TRUE for now). Implement RFC3551 unpacking for regular G726. See #567140. 2009-04-15 00:22:43 +0200 Wim Taymans * gst/rtp/gstrtph263pay.h: rtph263pay: fix build 2009-04-14 18:52:48 +0200 Youness Alaoui * gst/rtp/gstrtph263pay.c: h263pay: various fixes Re-enable mode A support and a property to control it. Fix memory leak of GstRtpH263PayBoundry objects. Fix marker. Fixes #509311 2009-04-14 18:44:51 +0200 Janin Kolenc * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: h263pay: Fix the payloader Fix the H263 payloader to be more RFC 2190 compliant. See #509311 2009-04-14 17:27:05 +0200 Wim Taymans * gst/avi/gstavidemux.c: avidemux: don't push EOS in streaming mode In streaming mode, avidemux is not supposed to send an EOS event downstream but it is supposed to return UNEXPECTED from the chain function instead so that upstream can do the right EOS handling. 2009-04-13 14:03:03 +0200 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: Add initial support for muxing/demuxing Speex audio Note: This is not in the Matroska spec yet Fixes bug #578310. 2009-04-10 21:31:06 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: handle NULL timing info Don't crash when the timing info is not yet available. 2009-04-10 21:42:13 +0300 Stefan Kost * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulse: make it work on 0.9.12 First we ignore request to fill the ringbuffer which are less then a segment. The small request where causing stutter. Then we disable flushing the stream when running against pa 0.9.12 as this triggers an assertiong in the sound server and terminates it. It does not happen with 0.9.10 and 0.9.14. 2009-04-10 14:18:48 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: handle server disconnect in get_time When the server is disconnected or when we are shut down, make our clock return an invalid time instead of erroring out. 2009-04-10 12:01:27 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: bps is signed int to avoid overflow Keep bps as gint instead of guint because we will be doing signed math with it later on and we don't want weird results. 2009-04-10 00:26:44 +0200 LRN * gst/avi/gstavidemux.c: avidemux: add convert query, fix duration query Fix the duration query so that it also works with formats other than TIME, such as DEFAULT to get the number of frames. Add a convert function. Fixes #578052. 2009-04-09 23:43:58 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: check for a stream Don't try to change the stream volume (and other things) when we don't have a stream yet. Just store the values for later. 2009-04-09 18:07:38 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: fix compilation for newer pulseaudio 2009-04-09 17:18:54 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: uncork fixes and use prebuf = 0 We can use prebuf = 0 to instruct pulse to not pause the stream on underflows. This way we can remove the underflow callback. We however have to manually uncork the stream now when we have no available space in the buffer or when we are writing too far away from the current read_index. 2009-04-09 14:38:17 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: handle write errors 2009-04-09 14:16:35 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: write silence on underflow Start filling up the buffer with empty samples when an underflow happens. We need to do this to keep pulseaudio reporting the right time for us. 2009-04-09 13:14:14 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: handle pull-based scheduling Use the default basesink methods for implementing pull based scheduling, it works fine for us. 2009-04-09 12:13:44 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: add beginnings of pull-based scheduling 2009-04-08 18:17:10 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: keep track of clock reset when we switch streams, the clock will reset to 0. Make sure that the provided clock doesn't get stuck when this happens by keeping an initial offset. We also need to make sure that we subtract this offset in samples when writing to the ringbuffer. 2009-04-08 13:52:41 +0200 Wim Taymans * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: rewrite pulsesink Derive from BaseAudioSink and implement our custom ringbuffer that maps to the internal pulseaudio ringbuffer. 2009-04-08 13:52:00 +0200 Wim Taymans * ext/pulse/pulseutil.c: pulse: remove some stray debug lines 2009-04-09 11:30:59 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: use slightly more adaptive formula for QoS Should work at least a tad better if the decoder can't keep up, and should also spread dropped frames a bit more evenly over time. 2009-04-07 22:35:31 +0300 Stefan Kost * gst/wavparse/gstwavparse.c: wavparse: don't leak pad-template gst_element_class_add_pad_template() does not take ownership. 2009-04-04 21:18:55 +0300 Felipe Contreras * common: Automatic update of common submodule From d0ea89e to b3941ea 2009-04-01 01:15:31 +0200 Thomas Vander Stichele * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: add pending_samples so that we only update segment's last stop after really sending the samples 2009-03-15 21:31:49 +0100 Thomas Vander Stichele * tests/check/pipelines/flacdec.c: add debug and an assert 2009-03-15 21:30:32 +0100 Thomas Vander Stichele * ext/flac/gstflacdec.c: add debugging 2009-03-03 10:14:02 +0100 Thomas Vander Stichele * tests/check/Makefile.am: * tests/check/audiotestsrc.flac: * tests/check/pipelines/flacdec.c: add a test to check that we get all decoded bytes from a 10-buffer audiotestsrc flac, in the case of: - a full decode - a decode of a seek for the full file - a decode of a seek for a small part, smaller than the first buffer The test fails because flacdec drops the first outgoing buffer on a seek 2009-03-03 10:06:52 +0100 Thomas Vander Stichele * ext/flac/gstflacdec.c: clipping should also work if it's done on the first buffer starting at 0 2009-04-04 14:54:01 +0200 Edward Hervey * common: Automatic update of common submodule From f8b3d91 to d0ea89e 2009-04-03 09:57:15 +0100 Zaheer Merali * gst/qtdemux/LEGAL: Fix grammar. 2009-04-02 22:41:01 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: allow http:// on the proxy setting Allow and ignore http:// at the start of the proxy setting, like souphttpsrc. Fixes #573173 2009-04-02 21:08:48 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't leak the udpsrc pad Fix memory leak in rtspsrc because we didn't unref the udpsrc pad. See #577318 2009-04-01 17:31:18 -0700 Michael Smith * gst/rtp/gstrtptheorapay.c: rtptheorapay: fix length encoding in packed headers. As for vorbis payloader; this by inspection had the same bug. 2009-04-01 17:23:33 -0700 Michael Smith * gst/rtp/gstrtpvorbispay.c: rtpvorbispay: in packed headers, properly flag multibyte lengths. In the sequence of header lengths, for headers >127 bytes, we use multiple bytes to encode the length. Bytes other than the last must have the top (flag) bit set. 2009-04-02 00:20:02 +0100 Jonathan Matthew * ext/taglib/gstid3v2mux.cc: * tests/check/elements/id3v2mux.c: id3v2mux: write RVA2 frames containing peak/gain volume data 2009-04-02 00:05:14 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: demote some log message from DEBUG to LOG And log decoder object. 2009-04-01 21:15:02 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: implement basic QoS Don't decode frames that are going to be too late anyway. 2009-04-01 12:26:12 +0100 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: don't emit ugly warnings with older rtpjitterbuffer versions The on-npt-stop signals was added only recently to rtpjitterbuffer in -bad, so check if the signal exists before g_signal_connect()ing to it, to avoid warnings. 2009-03-31 19:08:37 +0200 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: add proxy support 2009-03-31 17:16:04 +0300 Stefan Kost * gst/matroska/matroska-mux.c: matroska: don't leak serialized values when writing tags 2009-03-31 17:06:50 +0300 Stefan Kost * gst/matroska/matroska-demux.c: matroska: don't alter passed data and especialy don't leak. If we need different size, Make a copy, work with that and free it. 2009-03-31 16:42:15 +0300 Stefan Kost * gst/goom/plugin_info.c: goom: the structure is not fully initialized, but the copied. Set to fully to 0 to avoid creep of uninitialized values. 2009-03-31 16:25:58 +0300 Stefan Kost * gst/matroska/matroska-mux.c: matroska: init endianess as such and signedness as boolean. 2009-03-31 16:22:42 +0300 Stefan Kost * gst/qtdemux/qtdemux.c: qtdemux: don't use ininitialized var in debug log statement Also make the log statement useful by printing the human readable format name. 2009-03-31 12:01:21 +0300 Stefan Kost * gst/qtdemux/qtdemux.c: qtdemux: don't leak atom data in case of a wrong fourcc 2009-03-31 11:57:36 +0300 Stefan Kost * gst/matroska/matroska-demux.c: matroska: don't leak read data in demuxer 2009-03-31 11:50:41 +0300 Stefan Kost * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: udp: don't use protocol in debug message after freeing 2009-03-30 14:10:15 +0100 Tim-Philipp Müller * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: output should be framed already 2009-03-27 21:17:05 +0000 Tim-Philipp Müller * configure.ac: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flac: require a 'newer' flac and remove support for the legacy flac API 2009-03-27 17:48:13 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: link to the on_npt_stop signal to EOS Connect to the on_npt_stop signal of the session manager to schedule the EOS actions. 2009-03-26 14:39:06 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: some stream synchronization to aid seeking in unbalanced clips Some clips (trailers) may have (length-wise) unbalanced streams, which stalls the pipeline if seeking into that region. Additional stream synchronization can handle this, as well as sparse (subtitle) streams (at some later time ?) 2009-03-26 10:31:18 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: additional safety and sanity checks (push based mode) 2009-03-26 10:18:31 +0100 Wim Taymans * gst/videomixer/videomixer.c: videomixer: some more indent fixes 2009-03-24 16:00:58 +0100 Wim Taymans * gst/videomixer/videomixer.c: videomixer: fix gst-indent screwup 2009-03-25 17:54:35 +0000 Tim-Philipp Müller * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtspsrc.c: * po/POTFILES.in: rtspsrc: better error message when the RTSP extension for Real streams is missing Try to post a decent error message when it looks like we're failing because the Real RTSP extension plugin is missing. Also add i18n bits for rtspsrc so our error messages get translated. 2009-03-25 15:42:15 +0000 Tim-Philipp Müller * gst/avi/gstavi.c: * gst/qtdemux/quicktime.c: i18n: make sure gettext gives us UTF-8 at all times 2009-03-25 01:28:38 +0000 Tim-Philipp Müller * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: rtpmp4apay,rtpmp4depay: fix buffer leaks in AAC payloader and depayloader 2009-03-25 01:22:17 +0000 Tim-Philipp Müller * gst/rtp/gstrtpmp4apay.c: rtpmp4apay: warn if input is unframed 2009-03-22 21:20:57 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: put GstSegment inside the element struct instead of allocating it separately 2009-03-25 10:08:41 +0200 Stefan Kost * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: v4l2src: move duplicated timestamping and buffer metadata code to _create() This will include the latency changes also in the mmap case. 2009-03-25 10:06:48 +0200 Stefan Kost * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: v4l2src: remove win32 ifdefs introduced by commit cff3f46760eac74c9bbd7a36aca44fedf327424b V4l2src is under sys and does not exists/run under windows anyway. 2009-03-24 15:44:42 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: handle FLUSH_STOP event Clean up some state (most notably pad flow returns) to resume proper streaming following flushing seek. 2009-03-24 12:42:13 +0100 Alessandro Decina * gst/avi/gstavidemux.c: avidemux: don't post an error if EOS can't be pushed downstream. This aligns avidemux with other demuxers and fixes a bug using avidemux with a recent gnonlin. 2009-03-23 11:22:08 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: clean up the state change function Make the state change function a bit more readable and only pause after the parent had a change to pause first. 2009-03-09 23:43:55 +0200 Stefan Kost * gst/dtmf/Makefile.am: Makefile.am: no static libs for plugins 2009-03-20 17:22:32 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: support seeking in push based mode 2009-03-20 17:11:39 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: align push based behaviour more with pull based Cater for DELTA_UNIT flag on buffers, keep track of current position, remove and warn about edit lists if any (as those as are de facto discarded anyway), add some debug statements and indent fixes. 2009-03-20 17:03:03 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix mem leaks and prevent excessive buffering in push based mode 2009-03-20 13:27:59 +0000 Jan Schmidt * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Track the corked/uncorked state ourselves Use an instance variable to track whether the stream is corked or not, instead of using PA API that was only introduced in 0.9.11 2009-03-19 18:39:04 +0000 Jan Schmidt * ext/pulse/pulsesink.c: pulse: Make sure the stream is uncorked in the write function If the caps changes, the sink is reset without transitioning through a PAUSED->PLAYING state change, resulting in a corked stream. This avoids the problem by checking that the stream is uncorked when writing samples to it. 2009-03-20 01:02:26 +0000 Tim-Philipp Müller * ext/speex/gstspeexenc.c: speexenc: fix direction of latency query and other upstream queries Don't send queries back to the element they just came from by sending them to the peer of the wrong pad. 2009-03-19 11:10:40 +0000 Tim-Philipp Müller * .gitignore: * tests/check/elements/.gitignore: .gitignore: ignore more 2009-03-18 16:55:27 +0000 Tim-Philipp Müller * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: don't append an extra 0 byte to the codec data The audioMuxVersion structure is packed in such a way that the codec data does not start byte-aligned, which means there's an extra bit of padding at the end. We don't want that bit in the codec data, since some decoders seem get confused when they're fed with an extra codec data byte (also it's just not right of course). 2009-03-19 13:25:57 +0100 Wim Taymans * gst/rtp/gstrtph264depay.c: rtph264depay: fix base64 decoding We can't pass -1 to _decode_step, that functions returns 0 right away instead of decoding up to the string end. 2009-03-19 13:24:02 +0100 David Adam * gst/udp/gstudpnetutils.c: udp: Fix build if on Solaris This patch checks for Solaris and uses ip_mreq instead of ip_mreqn if on this platform. Fixes #575937. 2009-03-18 14:50:17 +0100 Sebastian Dröge * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: rtp: Use GLib functions for encoding/decoding base64 2009-03-16 19:17:24 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add some debug for the timestamps When timestamping in TCP mode, log the first timestamp we put on the buffers. 2009-03-15 23:26:56 +0200 Stefan Kost * sys/v4l2/v4l2src_calls.c: v4l2src: log details if we have them, needed for #575391 2009-03-13 18:32:47 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: convert _ in properties to - -- 2009-03-13 18:28:59 +0100 Edgar E. Iglesias * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: * gst/udp/gstudpsrc.c: * gst/udp/gstudpsrc.h: udpsrc: Add network interface selection Add network interface selection when joining multicast groups. Useful when using the udpsrc on multihomed hosts. Fixes #575234. API: GstUDPSrc::multicast-iface 2009-03-13 15:43:52 +0000 Jan Schmidt * sys/v4l2/v4l2_calls.c: v4l2src: Prepend to lists and reverse them at the end. Gratuitous micro-optimisation - prepend to lists and reverse them, rather than appending to them each time. 2009-03-13 15:40:50 +0000 Jan Schmidt * ext/pulse/pulsesink.c: pulsesink: Wait until there is enough room to write an entire segment When trying to write out a segment, wait until there is enough free space for the entire segment. This helps to reduce ripple in the clock reporting, where the app might query the playback position while only half a segment has been written (and is therefore reported by _delay(), even though the ring buffer has not yet been advanced) 2009-03-12 20:38:42 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: don't send PAUSE when not connected don't send a PAUSE request when we are no longer connected. 2009-03-12 16:10:25 +0100 Laszlo Pandy * ext/flac/gstflacdec.c: Don't call FLAC__ methods before it's initialized. Fixes #516031 In the event handler, gst_flac_dec_sink_event(), two functions are called on the FLAC stream without checking if it has been initialized: FLAC__stream_decoder_flush() FLAC__stream_decoder_process_until_end_of_stream() Both these FLAC__*() functions modify the internal state of the FLAC stream. Later, when the buffers start flowing, gst_flac_dec_chain() tries to initialize the stream. the FLAC__stream_decoder_init_stream() call will fail because the previous calls to FLAC__*() changed the stream state so it is no longer in the initialized state. 2009-03-11 17:59:00 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix timeout check --- 2009-03-11 12:48:03 +0000 Tim-Philipp Müller * win32/MANIFEST: win32: update MANIFEST, fixing 'make dist' config.h.in no longer exists. 2009-03-10 21:14:43 +0200 Stefan Kost * gst/multipart/Makefile.am: makefile: fix typo in no-static plugins rule 2009-03-10 11:01:16 +0100 Wim Taymans * ext/libpng/gstpngdec.c: pngdec: various cleanups. Make some code more readable. Fix a leak when pull range returns a shot buffer. Push EOS after posting the error. 2009-03-10 10:16:27 +0100 Edward Hervey * gst/rtp/gstrtpvorbisdepay.c: gstrtpvorbisdepay: Fix build on macosx 2009-03-01 17:37:56 +0100 Edward Hervey * .gitignore: .gitignore: Ignore m4 directory 2008-11-04 12:42:30 +0000 Stefan Kost [MOVED FROM BAD] Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. 2008-09-02 09:56:44 +0000 Tim-Philipp Müller [MOVED FROM BAD] Enable/fix up translations for these plugins. Original commit message from CVS: * ext/resindvd/plugin.c: (plugin_init): * ext/resindvd/resindvdsrc.c: * ext/twolame/gsttwolame.c: (plugin_init): * gst/aiffparse/aiffparse.c: (plugin_init): Enable/fix up translations for these plugins. * po/LINGUAS: Add 'ca' to LINGUAS. * po/POTFILES.in: * po/POTFILES.skip: Add more files for translation and more files which tools should skip. 2008-08-07 14:34:03 +0000 Sebastian Dröge [MOVED FROM BAD] ext/twolame/gsttwolame.*: Allow raw float samples as input for encoding. Original commit message from CVS: * ext/twolame/gsttwolame.c: (gst_two_lame_sink_setcaps), (gst_two_lame_chain): * ext/twolame/gsttwolame.h: Allow raw float samples as input for encoding. 2008-08-02 17:39:13 +0000 Sebastian Dröge [MOVED FROM BAD] Add TwoLAME MP2 encoding element, based on the LAME element. Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/twolame/Makefile.am: * ext/twolame/gsttwolame.c: (gst_two_lame_mode_get_type), (gst_two_lame_padding_get_type), (gst_two_lame_emphasis_get_type), (gst_two_lame_release_memory), (gst_two_lame_finalize), (gst_two_lame_base_init), (gst_two_lame_class_init), (gst_two_lame_src_setcaps), (gst_two_lame_sink_setcaps), (gst_two_lame_init), (gst_two_lame_set_property), (gst_two_lame_get_property), (gst_two_lame_sink_event), (gst_two_lame_chain), (gst_two_lame_setup), (gst_two_lame_change_state), (gst_two_lame_get_default_settings), (plugin_init): * ext/twolame/gsttwolame.h: Add TwoLAME MP2 encoding element, based on the LAME element. 2009-03-09 23:12:33 +0000 Jan Schmidt * common: Automatic update of common submodule From 7032163 to f8b3d91 2009-03-09 18:07:20 +0100 Wim Taymans * gst/rtp/gstrtpvorbisdepay.c: vorbisdepay: fix some leaks And leak the codebooks. Use glib base64 decoders. Use subbuffers to avoid a memcpy of the headers. 2009-03-09 17:14:12 +0100 Wim Taymans * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: don't lose the first buffer after a seek The flacdec API calls the write callback when performing a seek. We cannot yet push out a buffer at that time so we must keep it and push it out later. Flush out the upstream part of the pipeline when doing a seek. Fixes #574275. 2009-03-09 15:20:05 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: sanitize tag names Sanitize the tag names before turning them into a structure name. We can only add alphanumeric values as the structure name. 2009-03-08 12:04:22 +0100 Sebastian Dröge * common: Automatic update of common submodule From ffa738d to 7032163 2009-03-08 11:19:56 +0100 Sebastian Dröge * common: Automatic update of common submodule From 3f13e4e to ffa738d 2009-03-07 11:45:35 +0100 Sebastian Dröge * common: Automatic update of common submodule From 3c7456b to 3f13e4e 2009-03-07 10:45:40 +0100 Sebastian Dröge * common: Automatic update of common submodule From 57c83f2 to 3c7456b 2009-03-06 21:56:26 +0200 Stefan Kost * sys/v4l2/v4l2src_calls.c: v4l2src: fix pads, so that they are subset of template caps Do not add w=0 | h=0. When we can't get a framerate add fraction range. 2009-03-05 14:08:14 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: fix range parsing Fix parsing of the range headers. 2009-02-10 17:20:57 +0000 Olivier Crête * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirendepay.h: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsirenpay.h: Move siren rtp pay/depay from gst-plugins-farsight 2009-03-04 16:25:34 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix memory leak in close Close the connection even when we fail to send the teardown message. Use the connection url (which is a copy of the src url). 2009-03-04 16:15:05 +0100 Peter Kjellerstedt * tests/check/Makefile.am: check: gst-plugins-good.supp needs to be distributed. 2009-03-04 12:29:50 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix do-rtcp property description --- 2009-03-03 12:20:27 +0100 Edward Hervey * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: Expose the SoupSession 'timeout' property. 2009-03-02 15:07:24 +0100 Edward Hervey * .gitignore: .gitignore: Ignore the m4/ directory 2009-03-02 17:18:55 +0100 Wim Taymans * gst/rtp/gstrtpmp4vpay.c: rtpmp4vpay: Add support for more formats Hack around short header mpeg4 video files and put the short header as the config string. Fixes #572551. 2009-03-02 16:08:23 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: add support for http tunneling Add support for http tunneling and a new rtsph:// uri for it. See #573173. 2009-03-02 09:43:30 +0100 Thomas Vander Stichele Merge branch 'master' of ssh://thomasvs@git.freedesktop.org/git/gstreamer/gst-plugins-good 2009-03-02 08:41:15 +0100 Thomas Vander Stichele * ext/flac/gstflacdec.c: Add/clarify/fix some logging. 2009-03-01 12:47:37 -0800 David Schleef * sys/osxvideo/Makefile.am: Remove hardcoded definition of OBJC 2009-03-01 19:55:26 +0100 Sjoerd Simons * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: Wait for a frame to become available before capturing it Use GstPoll to wait for the fd of the video device to become readable before trying to capture a frame. This speeds up stopping v4l2src a lot as it no longer has to wait for the next frame, especially when capturing with low framerates or when the video device just never generates a frame (which seems a common issue for uvcvideo devices) Fixes bug #563574. 2009-02-14 17:56:05 +0000 Tim-Philipp Müller * gst/law/alaw-decode.c: * gst/law/mulaw-decode.c: alawdec, mulawdec: demote some debug messages from ERROR to WARNING or DEBUG Non-ok flow returns may happen for a variety of perfectly legitimate and expected reasons (temporarily not linked, seeking, pipeline shutdown), so we really shouldn't spew ERROR debug messages to stderr in those cases. Fixes #570781. (Seems like someone already took care of some of these.) 2009-02-28 15:26:00 +0200 René Stadler * gst/replaygain/gstrgvolume.c: rgvolume: Improve log message for peak values >1.0 by clamping explicitly. 2009-02-27 23:25:32 -0800 David Schleef * ext/dv/gstdvdec.c: Fix the field dominance PAL is TFF, NTSC is BFF. Some day I will learn to keep this straight. 2009-02-27 20:40:31 +0100 LRN * sys/directdraw/gstdirectdrawsink.c: directdrawsink: Fix type mismatches Fixes bug #573343. 2009-02-27 20:28:27 +0100 Sebastian Dröge Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good 2009-02-27 20:24:53 +0100 LRN * gst/udp/gstudpnetutils.c: udp: Don't set errno to EAFNOSUPPORT unconditionally Fixes bug #573342. 2009-02-27 11:17:50 -0800 Michael Smith * gst/replaygain/gstrgvolume.c: rgvolume: ignore out-of-range peak values If the peak value is > 1 (and thus nonsensical) ignore it. Prevents rgvolume reducing volume to effectively silent on files with bogus peak values. 2009-02-27 13:29:41 +0100 Mark Nauwelaerts * gst/wavparse/gstwavparse.c: wavparse: Fix SEEK event handling in push mode, and SEEKABLY query handling Standard pull mode loop based SEEK handling fails in push mode, so convert the SEEK event appropriately and dispatch to upstream. Also cater for NEWSEGMENT event handling, and properly inform downstream and application of SEEKABLE capabilities, depending on scheduling mode and upstream. 2009-02-27 11:04:08 +0100 Edward Hervey * gst/matroska/matroska-demux.c: matroskademux: Remove gst_util_dump_mem() calls. 2009-02-26 19:07:35 +0100 Julien Moutte * gst/avi/gstavidemux.c: avidemux: fix SEEK event handling in push mode When in push mode we should not try to handle the SEEK event as there's no code to handle it properly. Propagate upstream. 2009-02-26 19:05:06 +0100 Patrick Radizi * gst/rtsp/gstrtspsrc.h: rtspsrc: add the .h file change too Add the .h file change for the new property. 2009-02-26 19:03:52 +0100 Patrick Radizi * gst/rtsp/gstrtspsrc.c: rtspsrc: add property to disable RTCP Some old servers don't like us doing RTCP and thus we need a property to disable it. See #573173. 2009-02-26 13:19:31 +0100 Jan Smout * gst/udp/gstudpnetutils.c: udp: fix gst_udp_set_loop_ttl() again Fix the gst_udp_set_loop_ttl() function that was commented out in a previous commit. See #573115. 2009-02-26 13:06:17 +0100 Wim Taymans * gst/rtp/gstrtpvrawdepay.c: rtpvrawdepay: fail on interlaced video Fail on interlaced video until we support it. 2009-02-26 13:00:58 +0100 Wim Taymans * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: fail on interlaced video Detect and fail when trying to payload interlaced video. 2009-02-25 20:47:15 -0800 David Schleef * Makefile.am: * configure.ac: * win32/common/config.h.in: Change how win32/common/config.h is updated Generate win32/common/config.h-new directly from config.h.in, using shell variables in configure and some hard-coded information. Change top-level makefile so that 'make win32-update' copies the generated file to win32/common/config.h, which we keep in source control. It's kept in source control so that the git tree is buildable from VS. This change is similar to the one recently applied to GStreamer and gst-plugins-good. The previous config.h file in -good was in pretty bad shape, so unlike core and base, I didn't attempt to leave it strictly the same, but fixed it as necessary. Needs testing I cannot do myself. 2009-02-25 19:58:29 -0800 David Schleef * ext/dv/gstdvdec.c: * ext/dv/gstdvdec.h: dvdec: Add interlacing info to caps and buffers 2009-02-25 14:57:33 +0000 Jan Schmidt * common: * configure.ac: build: Update shave init statement for changes in common. Bump common. 2009-02-25 14:01:26 +0100 Wim Taymans * gst/udp/gstudpsrc.c: udpsrc: fix compilation Fix compilation on systems MSG_ERRQUEUE and IP_RECVERR. 2009-02-19 20:14:10 +0000 Tim-Philipp Müller * ext/jpeg/gstjpegenc.c: jpegenc: error out instead of crashing if no caps have been set Don't crash if we receive a buffer without caps. Fixes #572413. 2009-02-25 11:35:31 +0100 Peter Kjellerstedt * gst/udp/gstudpsrc.c: udpsrc: Make sure the sockaddr length used for recvfrom() is big enough. Previously the sockaddr length used for recvfrom() was calculated as sizeof (struct sockaddr). However, this is too little to hold an IPv6 address, so the full size of the gst_sockaddr union should be used instead. 2009-02-25 11:32:28 +0100 Peter Kjellerstedt * gst/udp/gstudpsrc.c: udpsrc: Unify the use of union gst_sockaddr. 2009-02-25 11:32:07 +0000 Jan Schmidt * common: Automatic update of common submodule From 9cf8c9b to a6ce5c6 2009-02-25 12:05:22 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: avoid crashing on subtitles Avoid a crash in avi with subtitles by only dereferencing the video description when we actually are dealing with video in the _invert function. 2009-02-25 11:45:05 +0200 Stefan Kost * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: docs: various doc fixes No short-desc as we have them in the element details. Also keep things (Makefile.am and sections.txt) sorted. Reword ambigous returns. No text after since please. 2009-02-24 17:58:32 +0000 Jan Schmidt * gst/udp/gstudpsrc.c: udp: Fix strict-aliasing warnings from gcc 4.4.0 Fix strict aliasing warnings by defining a union on the different sockaddr structs that we need. 2009-02-24 17:35:46 +0000 Tim-Philipp Müller * gst/rtp/gstrtph264pay.c: rtp: Fix compiler warning in h264 payloader Fix an undefined behaviour warning from gcc 4.4.0 Patch By: Tim-Philipp Müller Fixes: #570995 Signed-Off-By: Jan Schmidt 2009-02-22 17:23:09 +0000 Jan Schmidt * configure.ac: * docs/plugins/Makefile.am: Use shave for the build output 2009-02-24 14:55:28 +0100 Sebastian Dröge * ext/gconf/Makefile.am: * ext/gconf/gstgconf.c: * ext/gconf/gstgconf.h: * ext/gconf/gstgconfelements.h: gconf: Rename gconf.[ch] to gstgconf.[ch] to prevent name conflicts 2009-02-24 14:41:26 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Also use "(c)inf" to fill the comment tag 2009-01-26 11:06:13 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: perform UDP SETUP according to MS RTSP spec MS RTSP spec states that the UDP port pair used in subsequent SETUP requests for various streams must be identical (since there will actually be only 1 stream of muxed asf packets). Following traditional specs and using different port pairs in the SETUPs for separate streams will result in all but the first one failing and only one stream being streamed. So, in appropriate circumstances, retry UDP SETUP using previously used port pair. Fixes #552650. 2009-02-23 20:49:37 +0100 Aurelien Grimaud * gst/udp/gstudpsrc.c: Read ICMP error messages instead of looping When we are dealing with connected sockets shared between a udpsrc and a udpsink we might receive ICMP connection refused error messages in udpsrc that will cause it to go into a bursty loop because the poll returns right away without a message to read. Instead of looping, read the error message from the error queue in udpsrc. Fixes #567857. 2009-02-23 19:53:58 +0100 Wim Taymans * sys/v4l2/gstv4l2src.c: Conditionally compile code for YVYU Only compile the code for the YVYU format when the format is actually defined. Spotted by tmatth on IRC. 2009-02-17 11:01:47 -0800 Levente Farkas * sys/v4l2/v4l2src_calls.c: v4l2src: Make sort_by_frame_size conditionally compiled sort_by_frame_size is declared static and only used inside an ifdef, so use the same ifdef to define the function. Fixes #572185 Signed-off-by: David Schleef 2009-02-23 17:05:43 +0100 Wim Taymans * sys/v4l2/gstv4l2src.c: Add YVYU format to caps Add YVYU format to the caps. We don't have anything to handle these caps yet, though. 2009-02-23 15:48:41 +0100 Wim Taymans * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: Some cleanups Remove some unused variables. Avoid a useless _resync call. Correctly use a gboolean. 2009-02-23 15:43:51 +0100 Wai-Ming Ho * gst/rtp/gstrtph264pay.c: Always add PPS to the sprop-parameters-set Rework the parsing code that under certain circumstances dropped the PPS from the sprop-parameters-set. Fixes #572854. 2009-02-23 12:14:23 +0100 Arnout Vandecappelle * gst/matroska/matroska-mux.c: Don't do crazy things with 0/1 framerates We use 0/1 framerates to mark variable framerates and matroskamux should not try to calculate a frame duration for it. Fixes #571294. 2009-02-23 11:45:50 +0100 Wim Taymans * configure.ac: Require newer gst-p-b for the RTSP extensions. -- 2009-02-23 11:42:53 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: Call new receive_request method Call the receive_request extension methods so that extensions can handle the server request if they want. 2009-02-23 11:13:30 +0100 Wim Taymans * gst/rtsp/gstrtspext.c: * gst/rtsp/gstrtspext.h: Add method for hadling server requests Add method to handle server requests on the list of RTSP extensions. 2009-02-13 14:39:29 +0100 Wim Taymans * gst/law/alaw-decode.c: * gst/law/mulaw-decode.c: Don't use GST_ERROR for non-error cases. Turn a GST_ERROR line into a GST_DEBUG line so that we don't spam the log with errors. Fixes #570781. 2009-02-22 19:30:32 +0100 Sjoerd Simons * ext/gconf/gstgconfvideosink.c: * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.c: * ext/gconf/gstgconfvideosrc.h: gconfvideo(src|sink): Disconnect GConf notifications Fixes bug #571321. 2009-02-22 19:25:39 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Unref the buffer and not the memory address of the buffer 2009-02-22 18:47:35 +0100 Olivier Crete * gst/law/alaw-decode.c: * gst/law/mulaw-decode.c: alaw/mulaw: Implement _getcaps function for alaw/mulaw decoders Fixes bug #572358. 2009-02-22 18:46:03 +0100 Olivier Crete * gst/law/alaw-encode.c: * gst/law/mulaw-encode.c: alaw/mulaw: Don't require both, rate and channel, to be set in _getcaps Fixes bug #572358. 2009-02-22 18:32:02 +0100 Sebastian Dröge * gst/avi/gstavidemux.c: avidemux: Fix alignment issues by using GST_READ_* Reading integers from random memory addresses will result in SIGBUS on some architectures if the memory address is not correctly aligned. This can happen at two places in avidemux so we should use GST_READ_UINT32_LE and friends here. Fixes bug #572256. 2009-02-22 18:08:59 +0100 Sebastian Dröge * ext/pulse/pulsemixerctrl.c: pulsemixer: Don't use g_atomic_int_(get|set) for accessing the mixer track flags g_atomic_int_(get|set) only work on ints and the flags are an enum (which on most architectures is stored as an int). Also the way the flags were accessed atomically would still leave a possible race condition and we don't do it in any other mixer track implementation, let alone at any other place where an integer could be changed from different threads. Removing the g_atomic_int_(get|set) will only introduce a new race condition on architectures where integers could be half-written while reading them which shouldn't be the case for any modern architecture and if we really care about this we need to use g_atomic_int_(get|set) at many other places too. Apart from that g_atomic_int_(set|get) will result in aliasing warnings if their argument is explicitely casted to an int *. Fixes bug #571153. 2009-02-22 15:52:06 +0000 Jan Schmidt * common: Automatic update of common submodule From 5d7c9cc to 9cf8c9b 2009-02-22 12:41:53 +0100 Sebastian Dröge * ext/raw1394/gsthdv1394src.c: hdv1394src: Don't use void * pointer arithmetic 2009-02-21 11:13:43 -0800 David Schleef * common: Automatic update of common submodule From 80c627d to 5d7c9cc 2009-02-21 18:42:46 +0000 Jan Schmidt * configure.ac: Back to development -> 0.10.14.1 2009-02-20 18:16:02 -0500 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfsrc.c: Document rtpdtmfdepay a bit 2009-02-20 17:41:37 -0500 Olivier Crête * gst/dtmf/gstdtmf.c: Moved dtmf elements from gst-plugins-farsight to -bad 2009-02-20 17:40:57 -0500 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfdepay.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: Fix up documentation blobs SGML 2009-02-20 17:37:43 -0500 Olivier Crête * gst/dtmf/gstdtmf.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfcommon.h: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: Re-indent to Gst style 2009-02-18 13:30:44 -0500 Laurent Glayal * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Missing format directive 2008-12-04 21:21:44 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: [MOVED FROM GST-P-FARSIGHT] Allow setting a maximum duration to a RTP DTMF event 2008-12-04 21:11:17 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfdepay.c: [MOVED FROM GST-P-FARSIGHT] Improve the minimum quanta to make it impossible for the duration to fall down to 0 2008-12-01 18:31:48 -0500 Olivier Crête * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: [MOVED FROM GST-P-FARSIGHT] Allow setting a minimum size of a sound quanta in the dtmf depayloader 2008-12-11 17:54:18 -0500 Olivier Crête * gst/dtmf/.git-darcs-dir: [MOVED FROM GST-P-FARSIGHT] Remove .git-darcs-dir files 2008-12-01 17:37:10 -0500 Håvard Graff * gst/dtmf/gstrtpdtmfdepay.c: [MOVED FROM GST-P-FARSIGHT] Do wierd casting of the volume to make MSVC happy 2008-10-15 16:21:50 -0400 Olivier Crête * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Clarify the documentation of the "event-type" field when specifying dtmf events 2008-07-22 21:39:38 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Remove g_debugs 20080722213938-3e2dc-44a82d017fe66f3112301c410aa0b543de6156ad.gz 2008-06-13 23:57:23 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Take rate from the peers caps if possible 20080613235723-3e2dc-15690ee42708c539e1be12e20e076a5613faea96.gz 2008-06-13 23:41:44 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Put the sample rate in dtmfsrc into a variable 20080613234144-3e2dc-e60070943bec829b703b8821c7aa4351a02deebe.gz 2008-06-13 23:30:06 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Take the clock-rate from the caps in rtpdtmfsrc 20080613233006-3e2dc-a7d4e918643f4f8c1bb2cc2678558c654025920e.gz 2008-04-28 22:22:37 +0000 Olivier Crete * gst/dtmf/Makefile.am: [MOVED FROM GST-P-FARSIGHT] Link modules with libm where required 20080428222237-3e2dc-b1e9120c1e9ca1a510bfd7c27e2d45f0d4a12504.gz 2008-04-12 23:44:18 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfdepay.c: [MOVED FROM GST-P-FARSIGHT] Fix byte ordering issues with dtmfsrc and rtpdtmfdepay.. use of G_STRINGIFY to avoid error on MSVC 20080412234418-4f0f6-4828d1613dfcd564afd236dfc8fb57a299092f83.gz 2008-03-20 19:14:38 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: [MOVED FROM GST-P-FARSIGHT] Fix copyrights again, per smcv's advice.. 20080320191438-4f0f6-671c9db5d996a4601df017ceab4af6d16469c966.gz 2008-03-19 21:17:31 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Make it clear that dtmfsrc also takes named events as input 20080319211731-3e2dc-26c729f6dc8db27e71cf6b22646a81530dbf862f.gz 2008-03-20 18:48:41 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfdepay.c: [MOVED FROM GST-P-FARSIGHT] debug message made into errors because that's what they are... 20080320184841-4f0f6-8a2d283297b02713dade0ae4acaa5f6e0f67eace.gz 2008-03-20 18:39:37 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfdepay.c: [MOVED FROM GST-P-FARSIGHT] Clean unused stuff... 20080320183937-4f0f6-bcb841cdc07f9e9677512f4b50b4b659a58c6783.gz 2008-03-20 18:39:12 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: [MOVED FROM GST-P-FARSIGHT] Fix copyrights 20080320183912-4f0f6-689365d5a406632e3d088fac74e4fb6f8a4eb0ea.gz 2008-03-20 01:13:01 +0000 Youness Alaoui * gst/dtmf/Makefile.am: * gst/dtmf/gstdtmf.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Adding support for rtpdtmfdepay 20080320011301-4f0f6-d36a5d24be20336e36c4796d75476c9b5ee1a7e1.gz 2008-03-19 19:32:51 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] encoding name has to be upper-case 20080319193251-3e2dc-1581b33be9b486e35ec4948009677ccd5ffdc098.gz 2008-03-20 00:51:47 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfcommon.h: * gst/dtmf/gstrtpdtmfdepay.c: * gst/dtmf/gstrtpdtmfdepay.h: [MOVED FROM GST-P-FARSIGHT] Adding necessary files for rtpdtmfdepay 20080320005147-4f0f6-550fe22f70152f3aab3dcd7a6b02cbf81e89232d.gz 2008-03-20 00:50:41 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Fix typos 20080320005041-4f0f6-9d22fa5d155e35b605ea85b1fd9e7197a882a1f0.gz 2008-02-16 13:41:40 +0000 Sjoerd Simons * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] dtmfsrc: Correctly set the endianess in the caps to the machines endianess 20080216134140-93b9a-40a3a9d7ac1679c5e0dfd24a6b91e4aba6cc6496.gz 2007-09-17 17:52:33 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Search&Replace oops 20070917175233-3e2dc-57f579c4b890993f49fa8e9e6470a3eb79d2b922.gz 2007-09-17 17:51:33 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] events dont yet belong in the caps 20070917175133-3e2dc-fd1d83b7826b898110fc571ae7c3440f1887434d.gz 2007-09-17 16:08:20 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Add patch to make it work with maemo dsp sources that payload incorrectly 20070917160820-3e2dc-06b1b1d1b0918b30dabea5a0714cb732b3b8d8dd.gz 2007-09-17 04:26:49 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Oops, set to no preroll when playing->paused too 20070917042649-3e2dc-94adb6aa0617e815a6e233232dabb4bbc48dc82c.gz 2007-09-17 00:36:54 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Complete port to basesrc 20070917003654-3e2dc-db0f84dabd9dd1ac929a0461865b8aaa8ef91a77.gz 2007-09-17 00:24:12 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Add caps negotiation function 20070917002412-3e2dc-ca266816e9629746e9083c5bb8b7f73b94a9b2b0.gz 2007-09-17 00:16:59 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Properly free non-start events 20070917001659-3e2dc-a571777e3ecfb90989f87412f554aa10a31cc2ca.gz 2007-09-17 00:15:52 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Make interval and packet_redundancy into uint 20070917001552-3e2dc-60032e547b3669b87317c981d985c156aab91b40.gz 2007-09-16 19:44:08 +0000 Olivier Crete * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Make the rtp dtmf src use basesrc 20070916194408-3e2dc-734000130dce2434a014acf843d641ff0e60aa5a.gz 2007-09-16 19:41:01 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Make dtmf src code nicer 20070916194101-3e2dc-a8be8c509c65400d1d3962da02e67d15d2054316.gz 2007-09-14 04:20:42 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Implement stopping in a nice thread safe way 20070914042042-3e2dc-1fe257ff4b72aca4b0eb5f285a14650b8df268c3.gz 2007-09-14 04:18:34 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Remove get_times (Wim says its only good for really fake sources) 20070914041834-3e2dc-fff4d5da2a145f19e7b610a1027d2c4d4bc5eae0.gz 2007-09-13 21:21:45 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] using the unlock method of basesrc 20070913212145-4f0f6-0e438a681bf1651c0cc0d8fa3269aed3f1668b6b.gz 2007-09-13 21:12:26 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] more debug 20070913211226-4f0f6-bc32b5828fc8e0323c8a6eee779a38145aacd593.gz 2007-09-13 20:46:14 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] added debugs 20070913204614-4f0f6-68c2a69ae7a1efca6e13c116dbad7f9b686f0242.gz 2007-09-13 19:20:53 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Make sure to unlock the thread when going to ready and to flush the queue when moving to paused or playing 20070913192053-4f0f6-76c3925380d1a30988286170535a65dea64a5583.gz 2007-09-13 17:55:20 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Changed dtmfsrc into a subclass of GstBaseSrc 20070913175520-4f0f6-16ca4bf93690072f3e836d1c8a5b52cf7a421916.gz 2007-09-04 22:57:53 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Add another fix for a possible race condition 20070904225753-4f0f6-5ba8c4260c002bb27eb98e9faba3c15799357b57.gz 2007-09-04 21:52:24 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Add comment to explain push back 20070904215224-3e2dc-d92ac1f403dcf571546a7c53f18809f840eea51d.gz 2007-09-04 20:55:09 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Properly do the locking to avoid race conditions with clock unscheduling 20070904205509-3e2dc-da19900b51af6aedb6547f4f392bef4d1061dec2.gz 2007-09-01 00:03:24 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] oups, I did it again... 20070901000324-4f0f6-3d8b46691ee520537b06c511a5e732f5b812b844.gz 2007-08-31 23:54:28 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] oups, sorry.. DTMF, not RTP_DTMF for this file... 20070831235428-4f0f6-00b606bfb4892e4f217c440b611cc794ab0de55a.gz 2007-08-31 23:44:13 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Fixes the deadlock when pausing the dtmfsrc and rtpdtmfsrc. Had to push something on the async queue to release the blocking async_queue_pop(). Thanks to Olivier for the solution. 20070831234413-4f0f6-793cf35fc43636e7275258cc7063fc068f5efa0a.gz 2007-08-28 22:15:34 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] ClockID when waiting for buffer is now unscheduled when stopping the task. Various fixes to avoid bugs (thanks to -Wall -Werror). Fixes to allow the merge of the branch. 20070828221534-4f0f6-b0d6a4fe48c4e2a16b9ff69cb310087c970ce48e.gz 2007-08-28 17:15:46 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Cleaned up the code a bit, no use of GST_* and return value verification from gst_* 20070828171546-4f0f6-bdeb4b1b7f99f9464aabe5c43bd4a4d2025262b6.gz 2007-08-27 19:56:10 +0000 Olivier Crete * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Fix overly long lines and tabs 20070827195610-3e2dc-396a3fa01e16f184e4109c71fe2deb6e516bdf0d.gz 2007-08-27 19:26:18 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] untabbified dtmfsrc 20070827192618-4f0f6-77d68070464f1b5f9a46cb6eec2d922340143c04.gz 2007-08-27 17:24:24 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Fix RTP timestamps by sending a new_segment event to the payloader 20070827172424-4f0f6-d20907e3d436d50bfe74eb4fc3d2d6d7b6b6dbc5.gz 2007-08-27 17:23:39 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Better handling of packets, we send the same duration for all packets to avoid huge packets when min duration defines are modified. 20070827172339-4f0f6-cc93304437ea376fff6458c74c46c19f6920d329.gz 2007-08-27 17:23:22 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] Changing minimum values to work better on some gateways 20070827172322-4f0f6-5bf2bffa59a8244538dced795fa7d7649452ca91.gz 2007-08-22 20:16:53 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] The DTMF tone generator now respects the volume argument passed in the event 20070822201653-4f0f6-8b7ff874006e11f5a74d0fd91e5a9a43cd082ada.gz 2007-08-22 18:01:33 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] don't know why I did that... 20070822180133-4f0f6-6a7382f6c7d3630f91da384e1904763c7ea6fa1a.gz 2007-08-22 17:55:33 +0000 Youness Alaoui * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Ported the event queue work from dtmfsrc to rtpdtmfsrc Added a queue based system for the rtpdtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each tone, including inter-digit silence. 20070822175533-4f0f6-f27414c406f1f7b00c9a9084a988cf3a7930fe5c.gz 2007-08-22 17:54:44 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: [MOVED FROM GST-P-FARSIGHT] ouch, printing with arguments but without %s.. that made it segfault a few times... 20070822175444-4f0f6-445ea6ce7a9668d04cf999af772a504ec74fb67a.gz 2007-08-22 17:51:26 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Moved the timestamp from the event to dtmfsrc structure since we have only one event at a time, so let's keep it stored in the dtmfsrc struct 20070822175126-4f0f6-53bcda2bd8ae8c56d29e62e69ac19a30e08ad350.gz 2007-08-20 20:38:26 +0000 Youness Alaoui * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Added a queue based system for the dtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each tone, including inter-digit silence. 20070820203826-4f0f6-750a22b612a5e495e767666934465c34fe32074b.gz 2007-08-20 18:48:52 +0000 Youness Alaoui * gst/dtmf/Makefile.am: * gst/dtmf/gstdtmf.c: * gst/dtmf/gstdtmfsrc.c: * gst/dtmf/gstdtmfsrc.h: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Added dtmfsrc, a DTMF Tone Generator, and made it part of the 'dtmf' plugin. 20070820184852-4f0f6-a0d85e67708290aebafa89ab79d3cedd5815b620.gz 2007-08-20 18:48:00 +0000 Youness Alaoui * gst/dtmf/.git-darcs-dir: * gst/dtmf/Makefile.am: * gst/dtmf/gstrtpdtmfsrc.c: * gst/dtmf/gstrtpdtmfsrc.h: [MOVED FROM GST-P-FARSIGHT] Moved rtpdtmf to dtmf directory 20070820184800-4f0f6-fa33ea974510161de8c9951c39087af3613b65a4.gz 2009-02-21 12:47:00 +0100 Thomas Vander Stichele * ext/flac/gstflacdec.c: respect DEFAULT segment by clipping the last buffer to be sent === release 0.10.14 === 2009-02-19 20:09:07 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.14 2009-02-19 20:07:41 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files 2009-02-19 13:16:39 +0000 Jan Schmidt * gst/audiofx/audioecho.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosrc.c: Update Since: tags in autodetect srcs and audioecho 2009-02-19 11:12:58 +0000 Jan Schmidt * ChangeLog: Update ChangeLog for 0.10.13.3 2009-02-19 11:09:03 +0000 Jan Schmidt * configure.ac: * win32/common/config.h: 0.10.13.3 pre-release 2009-02-10 11:25:49 +0100 Mark Nauwelaerts * ext/pulse/pulsemixerctrl.c: pulsemixer: Fix compiler warnings. Cast (enum *) to (int *), not necessarily technically right, but plugs #571153. 2009-02-13 18:03:14 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Issue property change notification in streaming thread, rather than PA thread. pa_threaded_mainloop_lock() (a.o.) and by extension get_property should not be done from a PA thread, but the latter may occur as a result of a property change notification. Fixes #571204 (though current situation not ideal, e.g. post message rather than signal). 2009-02-10 11:27:51 +0100 Edward Hervey * gst/videocrop/gstaspectratiocrop.c: aspectratiocrop: Don't forget to call parent finalize implementation. This fixes a memory leak (leaking the contained elements of the bin). 2009-02-10 08:43:59 +0100 Edward Hervey * sys/osxvideo/osxvideosink.m: osxvideosink: Fix build. Fixes #571038 2009-02-09 12:18:36 +0100 Edward Hervey * common: Bump revision to use for common submodule. 2009-02-07 16:00:49 +0000 Jan Schmidt * ChangeLog: ChangeLog: Update ChangeLog for 0.10.13.2 2009-02-07 15:58:55 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/mt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: po: Update translations for 0.10.13.2 2009-02-07 15:46:07 +0000 Jan Schmidt * configure.ac: * win32/common/config.h: Release 0.10.13.2 2009-02-07 15:40:53 +0000 Jan Schmidt * po/LINGUAS: * po/mt.po: po: Add Maltese translation 2009-02-06 16:16:05 -0800 David Schleef * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: qtdemux: Add handling for stps atoms stps atoms contain "partial sync" information, which means that it's a sync point where pts != dts. This is needed to properly handle MPEG2, H.264, Dirac, etc., in quicktime. 2009-02-05 15:51:42 -0800 Michael Smith * ext/flac/gstflacdec.c: flacdec: if we aborted reading, don't do into an infinite loop. If our read callback ran out of data, so had to abort reading, we return GST_FLOW_ERROR instead of going into an infinite loop. 2009-02-05 10:19:37 -0800 Michael Smith * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: osxvideosink: remove non-embedded mode and fix memory management. Remove non-embedded mode. Embed mode becomes default and only mode. embed property is retained for binary compatibility. Added autorelease pools around all objc functions that might be called from a non-main thread. 2009-02-05 20:02:01 +0100 Thomas Vander Stichele * ext/flac/gstflacdec.c: debug on the object 2009-02-04 16:40:13 -0800 Michael Smith * sys/osxaudio/gstosxringbuffer.c: osxaudio fixes: multichannel and changing caps. Ensure we create the ringbuffer segment size as a multiple of the bytes per sample (fixes 6-channel output). Reset the segoffset when acquiring the ringbuffer, so we don't retain a bogus offset when caps change. 2009-02-04 11:38:30 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Keep track of connected state Keep track of the state of the connection and don't try to send TEARDOWN when the server has closed the connection. 2009-02-04 09:20:28 +0100 Robin Stocker * gst/matroska/matroska-demux.c: Read Matroska Title element for the TITLE tag Not all Matroska files have a Tags element which contains information about the title among other things. Most video Matroska files only contain the Title element so we should parse this too. Fixes bug #570435. 2009-02-03 22:34:38 +0000 Tim-Philipp Müller * configure.ac: configure.ac: bump core/base requirements to released versions 2009-02-03 17:10:30 +0100 Sebastian Dröge * tests/check/elements/audioecho.c: Fix audioecho unit test on 32 bit systems Cast the new value for the "delay" property to GstClockTime. Integers without type are passed to vararg functions with an integer type that can hold a pointer. 2009-02-03 14:09:26 +0200 Stefan Kost * gst/equalizer/gstiirequalizer.c: equalizer: Don't reset frequency bands from user settings. Fixes #570343. Move reallocating the history buffer out of _compute_frequencies() and call the right function as needed. Add some logging and tweak the formatting of existing logging. Simplify setting need_new_coefficients when changing properties. 2009-02-03 11:52:15 +0100 Sebastian Dröge * gst/audiofx/audioecho.c: Use guint64 instead of guint for storing guint64 2009-02-02 18:37:35 +0100 Jonathan Matthew * ext/soup/gstsouphttpsrc.c: Use correct flag for the GNOME proxy configuration Fixes bug #552140. 2009-02-02 13:08:14 +0100 Wim Taymans * tests/icles/v4l2src-test.c: Fix compiler warnings fix compiler warnings due to unused return values of scanf. 2009-01-31 11:08:30 +0100 Sebastian Dröge * tests/icles/v4l2src-test.c: Fix format string compiler warning 2009-01-30 22:24:14 +0200 Stefan Kost * docs/plugins/gst-plugins-good-plugins-docs.sgml: Add releaseinfo with online url. 2009-01-30 18:04:11 +0000 Jan Schmidt * tests/check/Makefile.am: * tests/icles/Makefile.am: Fix up some compile flags 2009-01-30 17:35:49 +0000 Jan Schmidt * gst/videocrop/gstvideocrop.c: Don't use Glib 2.16 function g_strcmp0. 2009-01-30 17:34:45 +0000 Jan Schmidt * gst/qtdemux/qtdemux.c: Don't do void pointer arithmetic 2009-01-30 17:26:19 +0000 Jan Schmidt * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: Fix Forte compiler warnings. Don't do void pointer arithmetic. Don't have an unreachable statement. 2009-01-30 17:29:45 +0000 Jan Schmidt * common: Bump common 2009-01-26 10:33:55 +0100 Edward Hervey * gst/avi/gstavidemux.c: Remove useless processing for non-raw formats 2009-01-30 15:34:31 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: Add support for the 'Requirement' and 'Encoder' tags 2009-01-30 15:33:19 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: Modify private-tag name formatter so that it doesn't go mad at fourcc starting with '(c)'. 2009-01-30 14:40:51 +0100 Brijesh Singh * sys/v4l2/gstv4l2tuner.c: Fix comparison of the tuner norms The V4L2 tuner norms that a device supports could be a subset of some norm (e.g. NTSC instead of NTSC_M). The comparison should be done by & instead of ==. See http://www.linuxtv.org/downloads/video4linux/API/V4L2_API/spec-single/v4l2.html#STANDARD Fixes bug #569820. 2009-01-30 08:53:06 +0100 Edward Hervey * autogen.sh: * common: Use a symbolic link for the pre-commit client-side hook 2009-01-29 14:08:56 +0100 Thijs Vermeir * gst/videocrop/gstaspectratiocrop.c: Only unref the peer when there is one. 2009-01-29 11:07:59 +0200 Stefan Kost * gst/avi/gstavimux.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: * sys/osxvideo/osxvideosink.m: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: Remove version numbers from a few gst-launch examples. The majority of the examples doe not use -0.10 and this will also help us to maintain the docs. 2009-01-29 10:10:08 +0200 Stefan Kost * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: Update and add documentation for platform specific plugins (sys). Link to properties. Correct titles for examples. Fix examples. 2009-01-29 09:45:25 +0200 Stefan Kost * gst/multipart/multipartmux.c: Add ' to framerate argument and remove the word 'simple' as all our pipelines are apparently simple. 2009-01-29 09:42:56 +0200 Stefan Kost * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: Add examples for the jpeg elements. 2009-01-28 21:40:11 +0000 Jan Schmidt * ext/pulse/pulsesink.c: Fix compile error in the last commit 2009-01-28 20:34:40 +0000 Jan Schmidt * configure.ac: * ext/pulse/pulseprobe.c: * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: Rewrite the pulse plugin, conditionally enabling new behaviour with newer pulseaudio. Fixes: #567794 * Hook pulsesink's volume property up with the stream volume -- not the sink volume in PA. * Read the device description directly from the sink instead of going via the mixer. * Properly implement _reset() methods for both sink and source to avoid deadlocks when shutting down a pipeline. * Replace all simple pa_threaded_mainloop_wait() by proper loops to guarantee that we wait for the right event in case multiple events are fired. While this is not strictly necessary in many cases it certainly is more correct and makes me sleep better at night. * Replace CHECK_DEAD_GOTO macros with proper functions * Extend the number of supported channels to 32 since that is the actual limit in PA. * Get rid of _dispose() methods since we don't need them. * Increase the volume property upper limit of the sink to 1000. * Reset function pointers after we disconnect a stream/context. Better fix for bug 556986. * Reset the state of the element properly if open/prepare fails * Cork the PA stream when the pipeline is paused. This allows the PA * daemon to close audio device on pause and thus save a bit of power. * Set PA stream properties based on GST tags such as GST_TAG_TITLE, GST_TAG_ARTIST, and so on. Signed-off-by: Lennart Poettering 2009-01-28 17:46:06 +0200 Stefan Kost * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfaudiosrc.c: * ext/gconf/gstgconfvideosink.c: * ext/gconf/gstgconfvideosrc.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/hal/gsthalaudiosink.c: * ext/hal/gsthalaudiosrc.c: * ext/hal/hal.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libcaca/gstcacasink.h: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/pulse/pulsemixer.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackparse.c: * gst/matroska/matroska-mux.h: * gst/udp/gstudpsrc.c: Update and add documentation for plugins with deps (ext). Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. Fix warnings that gtk-doc points out. 2009-01-28 15:57:20 +0100 Sebastian Dröge * gst/audiofx/audioecho.c: * gst/audiofx/audioecho.h: Limit the delay by a new max-delay property Introduce a new max-delay property that can only be set before going to PLAYING or PAUSED. This is used to limit the maximum delay and is set to the current delay by default. Using this will make sure that we have enough data in our internal ringbuffer for the echo. With dynamic reallocation of the ringbuffer as used before silence could've been used as the echo directly after setting a new delay. 2009-01-28 11:58:42 +0100 Edward Hervey * win32/common/config.h: Revert previous bogus commit 2009-01-28 12:29:42 +0200 Stefan Kost * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioecho.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/auparse/gstauparse.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/cutter/gstcutter.c: * gst/debug/gstpushfilesrc.c: * gst/debug/gsttaginject.c: * gst/debug/progressreport.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/law/mulaw.c: * gst/level/gstlevel.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/qtdemux/qtdemux.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/spectrum/gstspectrum.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/videomixer.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * win32/common/config.h: Update and add documentation for plugins with no deps (gst). Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. 2009-01-27 23:09:05 +0200 Stefan Kost * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: Fix example apps by drawing in the main-loop. 2009-01-27 20:33:02 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: fix build of aspectratio crop unit test in uninstalled environment. 2009-01-27 20:30:02 +0000 Tim-Philipp Müller * .gitignore: Make git ignore backup files 2009-01-26 16:14:47 +0100 Peter Kjellerstedt * gst/multipart/multipartdemux.c: Plug a memory leak in a debug message. 2009-01-22 15:59:40 +0100 Peter Kjellerstedt * gst/udp/gstudpnetutils.c: Correct return value from gst_udp_get_addr() when no known family is found. 2009-01-26 09:51:36 +0100 Jonathan Matthew * configure.ac: * ext/soup/gstsouphttpsrc.c: Use libsoup-gnome for proxy configuration if available If libsoup-gnome is found use this as it will give us the GNOME proxy configuration. Otherwise use normal libsoup. The GNOME proxy configuration will only be used if the proxy properties are not set on souphttpsrc and if the http_proxy environment variable is not set. Fixes bug #552140. 2009-01-25 19:26:46 -0800 David Schleef * gst/qtdemux/qtdemux.c: Add a few more video fourcc's 2009-01-24 14:48:00 +0100 Thijs Vermeir * gst/videocrop/gstaspectratiocrop.c: * tests/check/Makefile.am: * tests/check/elements/aspectratiocrop.c: Add unit test for aspectratiocrop Fixes bug #527951 Add unit test for aspectratiocrop and refactor this element. Added finalize function to cleanup leaking mutex. 2009-01-25 14:34:09 +0000 Jan Schmidt * tests/check/elements/.gitignore: Ignore check binaries 2009-01-24 18:28:06 +0100 Sebastian Dröge * gst/audiofx/audioecho.c: Save some allocations if the echo delay is increased often Save some allocations if the echo delay is increased often during playback by always allocating enough memory to hold data up to the next complete second, i.e. in the worst case allocate memory for one additional second. 2009-01-24 14:25:08 +0100 Thijs Vermeir * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Update plugin version in documentation 2009-01-23 21:47:40 +0100 Thijs Vermeir * gst/videocrop/gstvideocrop.c: Fix link in documentation of videocrop element 2009-01-23 21:46:13 +0100 Thijs Vermeir * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-videocrop.xml: * gst/videocrop/gstaspectratiocrop.c: Add documentation for aspectratiocrop 2009-01-24 13:21:39 +0100 Sebastian Dröge * win32/common/config.h: Update win32/common/config.h for the new development cycle 2009-01-24 11:53:40 +0100 Sebastian Dröge * gst/audiofx/audioecho.c: Add note that audioecho's reverb sounds metallic Add a note to the docs that audioecho's reverb will sound metallic. This happens because for a real reverb filter additional filtering is necessary. Also note which values should be used for the delay property to get an echo effect. 2009-01-23 23:38:10 +0000 Jan Schmidt * .gitignore: * docs/plugins/.gitignore: * po/.gitignore: * tests/examples/audiofx/.gitignore: More entries for the gitignores 2009-01-23 20:36:27 +0100 Thijs Vermeir * tests/check/elements/videocrop.c: skip video/x-raw-gray in videocrop unit test A recent commit added video/x-raw-gray support to videocrop. However this lets the videocrop unit test fail. Because videotestsrc can't generate this format. 2009-01-23 15:39:46 +0100 Thijs Vermeir * gst/videocrop/Makefile.am: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstaspectratiocrop.h: * gst/videocrop/gstvideocrop.c: Add aspectratiocrop element. Fixes bug #527951 Add new aspectratiocrop element that crops the video to a specified aspect ratio using videocrop. 2009-01-23 10:49:28 +0100 Thijs Vermeir * gst/videocrop/gstvideocrop.c: Fix navigation event forwarding while cropping. Fixes bug #567992. Fix the navigation event forwarding while cropping by adjusting the mouse position by the amount of cropped pixels. 2009-01-23 10:04:39 +0100 Brian Cameron * configure.ac: Fix linking on Solaris. Fixes bug #568809. Check for the socket library which is needed for socket() on Solaris. 2009-01-22 22:41:43 +0000 Jan Schmidt * configure.ac: Bump version number again -> 0.10.13.1 2009-01-22 22:41:01 +0000 Jan Schmidt * gst-plugins-good.doap: Add releases 0.10.12 and 0.10.13 to the doap file 2009-01-22 18:08:50 +0200 Stefan Kost * common: Update common snapshot. 2009-01-22 14:25:07 +0000 Jan Schmidt * configure.ac: * win32/common/config.h: Back to devel -> 0.10.12.1 2009-01-22 01:29:40 +0000 Jan Schmidt * configure.ac: Release 0.10.12 2009-01-21 17:22:39 -0800 David Schleef * gst/qtdemux/qtdemux.c: Fix for security advisory TKADV2009-0xx Fix potential buffer overflows while reading quicktime headers. Security issue noticed by Tobias Klein. 2009-01-21 12:56:55 +0000 Jan Schmidt * ext/flac/gstflacdec.c: Fix typo and small flaw in flac decoder 2009-01-22 13:49:35 +0100 Sebastian Dröge * common: Fix pre-commit hook 2009-01-22 10:40:34 +0100 Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-videocrop.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audioecho.c: * gst/audiofx/audioecho.h: * gst/audiofx/audiofx.c: * tests/check/Makefile.am: * tests/check/elements/audioecho.c: Rename audioreverb to audioecho. Fixes bug #568395. The element can add an echo and a simple reverb effect to an audio stream but for a real reverb filter it would need some additional filtering to prevent a metallic-sounding result. 2009-01-22 12:21:29 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: Free leftover udp ports (if any) when a setup request fails. 2009-01-22 06:05:26 +0100 Edward Hervey * autogen.sh: * common: Install and use pre-commit indentation hook from common 2009-01-21 13:25:06 +0100 Wim Taymans * ext/flac/gstflacdec.c: Whitespace fixes and some improved debug lines. 2009-01-21 04:31:58 +0100 Edward Hervey * autogen.sh: autogen.sh : Use git submodule 2009-01-20 15:33:05 +0000 Tim-Philipp Müller sys/v4l2/gstv4l2src.c: Fix error code (the message string also needs love, but not today). Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read): Fix error code (the message string also needs love, but not today). 2009-01-19 11:44:36 +0000 Luotao Fu gst/videocrop/gstvideocrop.c: Add 8bit grayscale support to videocrop plugin. Fixes #567952. Original commit message from CVS: Patch by: Luotao Fu * gst/videocrop/gstvideocrop.c: (gst_video_crop_get_image_details_from_caps): Add 8bit grayscale support to videocrop plugin. Fixes #567952. 2009-01-19 11:22:06 +0000 Sebastian Dröge gst/audiofx/audioreverb.c: Set the default value in the instance init function. Original commit message from CVS: * gst/audiofx/audioreverb.c: (gst_audio_reverb_init): Set the default value in the instance init function. 2009-01-19 11:19:08 +0000 Sebastian Dröge Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-spectrum.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init), (gst_audio_reverb_class_init), (gst_audio_reverb_init), (gst_audio_reverb_finalize), (gst_audio_reverb_set_property), (gst_audio_reverb_get_property), (gst_audio_reverb_setup), (gst_audio_reverb_stop), (gst_audio_reverb_transform_ip): * gst/audiofx/audioreverb.h: * tests/check/Makefile.am: * tests/check/elements/audioreverb.c: (setup_reverb), (cleanup_reverb), (GST_START_TEST), (audioreverb_suite): Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567874. 2009-01-19 10:13:53 +0000 Sebastian Dröge gst/spectrum/gstspectrum.*: Implement a simple compensation algorithm for rounding errors. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: Implement a simple compensation algorithm for rounding errors. This makes sure that a spectrum message is posted on the bus every interval nanoseconds. Fixes bug #567955. 2009-01-15 21:16:45 +0000 Michael Smith sys/osxaudio/Makefile.am: Link against CoreServices (needed for osx 10.4) and fix up the linker flags. Fixes #567853. Original commit message from CVS: * sys/osxaudio/Makefile.am: Link against CoreServices (needed for osx 10.4) and fix up the linker flags. Fixes #567853. 2009-01-15 14:53:18 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Catch invalid and commonly wrong playback rates in the elst atoms. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_segments): Catch invalid and commonly wrong playback rates in the elst atoms. Fixes #567800. 2009-01-15 11:40:23 +0000 Sebastian Dröge gst/spectrum/gstspectrum.c: Don't call gst_fft_f32_free() with NULL to prevent a crash. Fixes bug #567642. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state): Don't call gst_fft_f32_free() with NULL to prevent a crash. Fixes bug #567642. 2009-01-14 15:44:18 +0000 Sebastian Dröge gst/spectrum/gstspectrum.*: Use correct types for frame/fft counters and some minor cleanup. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: Use correct types for frame/fft counters and some minor cleanup. 2009-01-14 15:37:07 +0000 Lennart Poettering ext/pulse/pulseprobe.c: Fix refcount loop, resulting in a thread leak. Fixes bug #567746. Original commit message from CVS: Patch by: Lennart Poettering * ext/pulse/pulseprobe.c: (gst_pulseprobe_new), (gst_pulseprobe_free): Fix refcount loop, resulting in a thread leak. Fixes bug #567746. 2009-01-14 10:46:54 +0000 Sebastian Dröge gst/spectrum/: Post a spectrum message on the bus for every interval, even if the interval is small than the length o... Original commit message from CVS: * gst/spectrum/Makefile.am: * gst/spectrum/README: * gst/spectrum/gstspectrum.c: (gst_spectrum_base_init), (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_reset_state), (gst_spectrum_finalize), (gst_spectrum_set_property), (gst_spectrum_start), (gst_spectrum_stop), (gst_spectrum_setup), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: Post a spectrum message on the bus for every interval, even if the interval is small than the length of the FFT. Fixes bug #567642. Major cleanup of the spectrum element. 2009-01-13 19:23:57 +0000 Sebastian Dröge Add audioiirfilter and audiofirfilter elements which allow generic IIR/FIR filters to be implemented by providing the... Original commit message from CVS: * configure.ac: * gst/audiofx/Makefile.am: * gst/audiofx/audiofirfilter.c: (gst_audio_fir_filter_base_init), (gst_audio_fir_filter_class_init), (gst_audio_fir_filter_update_kernel), (gst_audio_fir_filter_init), (gst_audio_fir_filter_setup), (gst_audio_fir_filter_finalize), (gst_audio_fir_filter_set_property), (gst_audio_fir_filter_get_property): * gst/audiofx/audiofirfilter.h: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audioiirfilter.c: (gst_audio_iir_filter_base_init), (gst_audio_iir_filter_class_init), (gst_audio_iir_filter_update_coefficients), (gst_audio_iir_filter_init), (gst_audio_iir_filter_setup), (gst_audio_iir_filter_finalize), (gst_audio_iir_filter_set_property), (gst_audio_iir_filter_get_property): * gst/audiofx/audioiirfilter.h: Add audioiirfilter and audiofirfilter elements which allow generic IIR/FIR filters to be implemented by providing the filter coefficients. Fixes bug #567577. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-wavparse.xml: Add documentation for the audioiirfilter and audiofirfilter elements. * tests/check/Makefile.am: * tests/check/elements/audiofirfilter.c: (on_message), (on_rate_changed), (on_handoff), (GST_START_TEST), (audiofirfilter_suite): * tests/check/elements/audioiirfilter.c: (on_message), (on_rate_changed), (on_handoff), (GST_START_TEST), (audioiirfilter_suite): * tests/examples/Makefile.am: * tests/examples/audiofx/Makefile.am: * tests/examples/audiofx/firfilter-example.c: (on_message), (on_rate_changed), (main): * tests/examples/audiofx/iirfilter-example.c: (on_message), (on_rate_changed), (main): Add unit tests and example applications for the two filter elements. 2009-01-13 19:09:19 +0000 Thiago Sousa Santos gst/qtdemux/qtdemux.c: Fix format string for guint64. Original commit message from CVS: Patch by: Thiago Sousa Santos * gst/qtdemux/qtdemux.c: Fix format string for guint64. 2009-01-13 19:04:09 +0000 Michael Smith sys/osxaudio/Makefile.am: osxaudio plugin now requires AudioUnit framework, so link against that. Original commit message from CVS: * sys/osxaudio/Makefile.am: osxaudio plugin now requires AudioUnit framework, so link against that. Clean up tabs v spaces while I'm there. 2009-01-13 17:49:07 +0000 Wim Taymans tests/examples/rtp/server-alsasrc-PCMA.c: Add some example code for printing the RTP manager stats. Original commit message from CVS: * tests/examples/rtp/server-alsasrc-PCMA.c: (print_source_stats), (print_stats), (main): Add some example code for printing the RTP manager stats. 2009-01-13 08:24:25 +0000 Sebastian Dröge gst/audiofx/: Use a custom mutex for protecting the instance fields instead of the GstObject lock. Using the latter c... Original commit message from CVS: * gst/audiofx/audiochebband.c: (gst_audio_cheb_band_class_init), (gst_audio_cheb_band_init), (gst_audio_cheb_band_finalize), (gst_audio_cheb_band_set_property): * gst/audiofx/audiochebband.h: * gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_class_init), (gst_audio_cheb_limit_init), (gst_audio_cheb_limit_finalize), (gst_audio_cheb_limit_set_property): * gst/audiofx/audiocheblimit.h: * gst/audiofx/audiowsincband.c: (gst_audio_wsincband_class_init), (gst_audio_wsincband_init), (gst_audio_wsincband_finalize), (gst_audio_wsincband_set_property): * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init), (gst_audio_wsinclimit_finalize), (gst_audio_wsinclimit_set_property): * gst/audiofx/audiowsinclimit.h: Use a custom mutex for protecting the instance fields instead of the GstObject lock. Using the latter can lead to deadlocks, especially with the FIR filters when updating the latency. 2009-01-11 19:03:38 +0000 Sebastian Dröge gst/audiofx/: Implement a base class for generic audio FIR filters. Original commit message from CVS: * gst/audiofx/Makefile.am: * gst/audiofx/audiofxbasefirfilter.c: (gst_audio_fx_base_fir_filter_dispose), (gst_audio_fx_base_fir_filter_base_init), (gst_audio_fx_base_fir_filter_class_init), (gst_audio_fx_base_fir_filter_init), (gst_audio_fx_base_fir_filter_push_residue), (gst_audio_fx_base_fir_filter_setup), (gst_audio_fx_base_fir_filter_transform), (gst_audio_fx_base_fir_filter_start), (gst_audio_fx_base_fir_filter_stop), (gst_audio_fx_base_fir_filter_query), (gst_audio_fx_base_fir_filter_query_type), (gst_audio_fx_base_fir_filter_event), (gst_audio_fx_base_fir_filter_set_kernel): * gst/audiofx/audiofxbasefirfilter.h: * gst/audiofx/audiofxbaseiirfilter.c: Implement a base class for generic audio FIR filters. * gst/audiofx/audiowsincband.c: (gst_gst_audio_wsincband_mode_get_type), (gst_gst_audio_wsincband_window_get_type), (gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init), (gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel), (gst_audio_wsincband_setup), (gst_audio_wsincband_set_property), (gst_audio_wsincband_get_property): * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_mode_get_type), (gst_audio_wsinclimit_window_get_type), (gst_audio_wsinclimit_base_init), (gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init), (gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup), (gst_audio_wsinclimit_set_property), (gst_audio_wsinclimit_get_property): * gst/audiofx/audiowsinclimit.h: * tests/check/elements/audiowsincband.c: (GST_START_TEST): * tests/check/elements/audiowsinclimit.c: (GST_START_TEST): Use this new base class for audiowsincband and audiowsinclimit. Also cleanup both elements. 2009-01-08 18:17:13 +0000 Michael Smith gst/qtdemux/qtdemux.c: In push mode, error out if we get EOS before we've created any srcpads. Original commit message from CVS: * gst/qtdemux/qtdemux.c: In push mode, error out if we get EOS before we've created any srcpads. Handle (in pull mode) some files that have a truncated moov atom where the final sub-atom is a 'free' atom and the contents of that are not present in the file. 2009-01-08 15:56:46 +0000 Mark Nauwelaerts gst/matroska/: Some cleanups, refactoring and minor enhancements in caps handling. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps): * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): Some cleanups, refactoring and minor enhancements in caps handling. * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init), (gst_matroska_mux_init), (gst_matroska_pad_reset), (gst_matroska_pad_free), (gst_matroska_mux_reset), (gst_matroska_mux_video_pad_setcaps), (gst_matroska_mux_request_new_pad): * tests/check/elements/matroskamux.c: (teardown_src_pad): Only remove, release or reset what is appropriate upon state change. 2009-01-07 20:38:50 +0000 Jan Schmidt ext/pulse/pulsesink.*: Use a mutex to protect the current stream pointer, and ignore callbacks for stream objects tha... Original commit message from CVS: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: Use a mutex to protect the current stream pointer, and ignore callbacks for stream objects that have been destroyed already. Fixes problems with unprepare/prepare cycles caused by the input caps changing, without reintroducing bug #556986. 2009-01-07 16:09:47 +0000 Jan Schmidt sys/v4l2/gstv4l2src.c: Remove () from translateable string, so that it makes more sense. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: Remove () from translateable string, so that it makes more sense. 2009-01-07 09:43:13 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Minor fix/cleanup in header field calculation. Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps): Minor fix/cleanup in header field calculation. 2009-01-06 17:48:10 +0000 Mark Nauwelaerts gst/matroska/matroska-mux.*: Remove internal taglist and fully use tagsetter interface. Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event), (gst_matroska_mux_finish): * gst/matroska/matroska-mux.h: Remove internal taglist and fully use tagsetter interface. 2009-01-06 14:50:29 +0000 Mark Nauwelaerts gst/avi/gstavimux.*: Ensure header size invariance during subsequent rewrite by using tags snapshot. Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_reset), (gst_avi_mux_riff_get_avi_header): * gst/avi/gstavimux.h: Ensure header size invariance during subsequent rewrite by using tags snapshot. 2009-01-05 17:31:13 +0000 Sebastian Dröge ext/pulse/pulsesink.c: Don't wait for the pulse mainloop when destroying the stream. Original commit message from CVS: * ext/pulse/pulsesink.c: (gst_pulsesink_destroy_stream): Don't wait for the pulse mainloop when destroying the stream. Fixes a deadlock when the pulsedaemon goes away while pulsesink is PLAYING. Fixes bug #556986. 2009-01-05 12:30:40 +0000 Sascha Hauer sys/v4l2/gstv4l2src.c: Add support for grayscale v4l2 devices. Fixes bug #566616. Original commit message from CVS: Patch by: Sascha Hauer Luotao Fu * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure), (gst_v4l2_get_caps_info): Add support for grayscale v4l2 devices. Fixes bug #566616. 2009-01-05 11:42:09 +0000 Mark Nauwelaerts gst/qtdemux/: Streamline tag handling and pass unparsed tags as binary blob in private tag. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_tag_add_str), (qtdemux_tag_add_tmpo), (qtdemux_tag_add_covr), (qtdemux_tag_add_date), (qtdemux_tag_add_gnre), (qtdemux_tag_add_blob), (qtdemux_parse_udta): * gst/qtdemux/qtdemux.h: * gst/qtdemux/quicktime.c: (plugin_init): Streamline tag handling and pass unparsed tags as binary blob in private tag. 2009-01-05 10:13:29 +0000 Sebastian Dröge gst/audiofx/: Implement a base class for IIR filters. Original commit message from CVS: * gst/audiofx/Makefile.am: * gst/audiofx/audiofxbaseiirfilter.c: (gst_audio_fx_base_iir_filter_base_init), (gst_audio_fx_base_iir_filter_dispose), (gst_audio_fx_base_iir_filter_class_init), (gst_audio_fx_base_iir_filter_init), (gst_audio_fx_base_iir_filter_calculate_gain), (gst_audio_fx_base_iir_filter_set_coefficients), (gst_audio_fx_base_iir_filter_setup), (process), (gst_audio_fx_base_iir_filter_transform_ip), (gst_audio_fx_base_iir_filter_stop): * gst/audiofx/audiofxbaseiirfilter.h: Implement a base class for IIR filters. * gst/audiofx/audiochebband.c: (gst_audio_cheb_band_base_init), (gst_audio_cheb_band_class_init), (gst_audio_cheb_band_init), (generate_coefficients), (gst_audio_cheb_band_set_property), (gst_audio_cheb_band_setup): * gst/audiofx/audiochebband.h: * gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_base_init), (gst_audio_cheb_limit_class_init), (gst_audio_cheb_limit_init), (generate_coefficients), (gst_audio_cheb_limit_set_property), (gst_audio_cheb_limit_setup): * gst/audiofx/audiocheblimit.h: Use the IIR filter base class for the chebyshev filters. 2009-01-02 20:39:34 +0000 Justin Karnegas sys/osxaudio/: Rewrite osxaudio to work more flexibly and more reliably, using a different abstraction layer of corea... Original commit message from CVS: Patch by: Justin Karnegas and Michael Smith * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudioelement.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: Rewrite osxaudio to work more flexibly and more reliably, using a different abstraction layer of coreaudio that is the recommended way of doing low-level audio I/O on OSX. Fixes byg #564948. 2009-01-02 16:31:13 +0000 Wim Taymans tests/examples/rtp/server-decodebin-H263p-AMR.sh: Add example RTP transcoding pipeline from any file decodedable with... Original commit message from CVS: * tests/examples/rtp/server-decodebin-H263p-AMR.sh: Add example RTP transcoding pipeline from any file decodedable with uridecodebin. 2009-01-02 15:20:48 +0000 Wim Taymans tests/examples/rtp/: Add two C examples of using gstrtpbin as a sender and a receiver. Original commit message from CVS: * tests/examples/rtp/.cvsignore: * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main): * tests/examples/rtp/server-alsasrc-PCMA.c: (main): Add two C examples of using gstrtpbin as a sender and a receiver. 2008-12-31 11:20:55 +0000 Jan Schmidt ChangeLog: Remove conflict marker from ChangeLog Original commit message from CVS: * ChangeLog: Remove conflict marker from ChangeLog 2008-12-28 09:50:31 +0000 j^ gst/qtdemux/qtdemux.c: Add codec mapping for xvid, fmp4 and ac3 tracks. Original commit message from CVS: Patch by: j^ * gst/qtdemux/qtdemux.c: (qtdemux_video_caps), (qtdemux_audio_caps): Add codec mapping for xvid, fmp4 and ac3 tracks. Fixes #565850 2008-12-23 12:10:41 +0000 Wim Taymans ext/jpeg/gstsmokeenc.*: Implement getcaps function. Original commit message from CVS: * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init), (gst_smokeenc_getcaps), (gst_smokeenc_setcaps), (gst_smokeenc_chain), (gst_smokeenc_change_state): * ext/jpeg/gstsmokeenc.h: Implement getcaps function. Set caps on the pad and on all outgoing buffers. Fixes #565441. 2008-12-19 09:36:45 +0000 Stefan Kost ext/pulse/pulsemixerctrl.c: And remove temporary comment pointing to the bug ticket. Original commit message from CVS: * ext/pulse/pulsemixerctrl.c: And remove temporary comment pointing to the bug ticket. * gst/avi/gstavimux.c: Move reoccuring logging to LOG and log instance too. 2008-12-17 17:28:39 +0000 Stefan Kost ext/pulse/pulsemixerctrl.c: Don't leak the pa_operation. Original commit message from CVS: * ext/pulse/pulsemixerctrl.c: Don't leak the pa_operation. 2008-12-16 16:19:26 +0000 Stefan Kost configure.ac: Require core cvs. Original commit message from CVS: * configure.ac: Require core cvs. 2008-12-16 16:07:48 +0000 Stefan Kost gst/avi/gstavimux.c: Rename api from _flush to _reset_tags. Original commit message from CVS: * gst/avi/gstavimux.c: Rename api from _flush to _reset_tags. 2008-12-16 14:22:51 +0000 Stefan Kost gst/avi/gstavimux.c: Use new tagsetter api to flush tags. Original commit message from CVS: * gst/avi/gstavimux.c: Use new tagsetter api to flush tags. 2008-12-16 13:14:39 +0000 Sebastian Dröge tests/check/elements/deinterleave.c: Increase timeout to 3 minutes to prevent timeouts. Original commit message from CVS: * tests/check/elements/deinterleave.c: (deinterleave_suite): Increase timeout to 3 minutes to prevent timeouts. 2008-12-16 12:52:24 +0000 Sebastian Dröge tests/check/elements/interleave.c: Increase timeout to 3 minutes to prevent timeouts. Original commit message from CVS: * tests/check/elements/interleave.c: (interleave_suite): Increase timeout to 3 minutes to prevent timeouts. 2008-12-16 11:57:01 +0000 Stefan Kost gst/avi/gstavimux.*: Totally remove the internal taglists and fully use tagsetter. Original commit message from CVS: * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: Totally remove the internal taglists and fully use tagsetter. 2008-12-15 15:59:53 +0000 Stefan Kost gst/avi/gstavimux.c: Instead of filtering wrongly just use the mergemode. Applications is use KEEP_ALL if they want t... Original commit message from CVS: * gst/avi/gstavimux.c: Instead of filtering wrongly just use the mergemode. Applications is use KEEP_ALL if they want to supress tag-events. Fixes #563221 for avi for real (I hope). Everyone chime in, before I fix the others. 2008-12-15 12:45:35 +0000 Stefan Kost ext/pulse/pulsemixerctrl.c: Add note about memleak. Original commit message from CVS: * ext/pulse/pulsemixerctrl.c: Add note about memleak. 2008-12-13 16:23:09 +0000 Edward Hervey m4/Makefile.am: A couple more .m4 that aren't shipped anymore with gettext 0.17. Original commit message from CVS: * m4/Makefile.am: A couple more .m4 that aren't shipped anymore with gettext 0.17. 2008-12-13 15:34:01 +0000 Edward Hervey Switch to using GstStaticPadTemplate. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_base_init), (gst_flac_dec_init): * gst/law/alaw-decode.c: (gst_alaw_dec_base_init), (gst_alaw_dec_init): * gst/law/alaw-encode.c: (gst_alaw_enc_base_init), (gst_alaw_enc_init): * gst/law/alaw.c: (plugin_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init), (gst_mulawdec_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init), (gst_mulawenc_init): * gst/law/mulaw.c: (plugin_init): Switch to using GstStaticPadTemplate. * gst/udp/gstudpnetutils.c: (gst_udp_get_addr): Don't forget to free the addrinfo structure. * gst/wavparse/gstwavparse.c: (gst_wavparse_reset), (gst_wavparse_sink_activate): Don't forget to unref the GstAdapter. 2008-12-13 12:58:24 +0000 Edward Hervey m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we... Original commit message from CVS: * m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we can remove it from the list of files to dist. 2008-12-10 15:03:23 +0000 Christian Schaller * gst-plugins-good.spec.in: smaller spec file updates Original commit message from CVS: smaller spec file updates 2008-12-09 17:55:22 +0000 Stefan Kost gst/avi/gstavidemux.c: More logging. Original commit message from CVS: * gst/avi/gstavidemux.c: More logging. * gst/avi/gstavimux.c: Handle more metadata fields. Better estimate of metadata size. Don't merge received tags, if application has specified tags using GST_TAG_MERGE_REPLACE_ALL. Fixes #563221 for avi. 2008-12-09 14:30:03 +0000 Sebastian Dröge tests/check/Makefile.am: Also ignore pulsemixer for the states unit test. Original commit message from CVS: * tests/check/Makefile.am: Also ignore pulsemixer for the states unit test. 2008-12-09 14:19:16 +0000 Wim Taymans gst/rtp/gstrtpjpegdepay.c: Add an EOI marker at the end of the jpeg frame when it's missing. Original commit message from CVS: * gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_process): Add an EOI marker at the end of the jpeg frame when it's missing. Fixes #563056. 2008-12-09 10:47:14 +0000 Sebastian Dröge tests/check/elements/videocrop.c: Update the unit test for the new color values for BT.601 red. Original commit message from CVS: * tests/check/elements/videocrop.c: (check_1x1_buffer): Update the unit test for the new color values for BT.601 red. Fixes bug #563510. 2008-12-09 10:28:11 +0000 Tim-Philipp Müller ext/dv/gstdvdemux.c: Restore previous behaviour of not passing QoS and navigation events upstream, which presumably w... Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_src_event): Restore previous behaviour of not passing QoS and navigation events upstream, which presumably wasn't meant to be changed. 2008-12-09 09:39:53 +0000 Sebastian Dröge ext/dv/gstdvdemux.c: Add srcpads only when needed and remove them again when going back to READY. This prevents stall... Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_add_video_pad), (gst_dvdemux_add_audio_pad), (gst_dvdemux_remove_pads), (gst_dvdemux_demux_audio), (gst_dvdemux_demux_video), (gst_dvdemux_chain), (gst_dvdemux_loop), (gst_dvdemux_change_state): Add srcpads only when needed and remove them again when going back to READY. This prevents stalled pipelines if there's no audio inside the DV stream, which happens for many MXF files. 2008-12-09 09:09:25 +0000 Sebastian Dröge tests/check/elements/souphttpsrc.c: The ports in libsoup are unsigned integers and not signed integers. Original commit message from CVS: * tests/check/elements/souphttpsrc.c: (GST_START_TEST), (run_server): The ports in libsoup are unsigned integers and not signed integers. 2008-12-08 18:31:00 +0000 Sebastian Dröge ext/dv/gstdvdemux.c: Forward all events upstream unless it's something we really don't handle. This fixes latency con... Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_src_event): Forward all events upstream unless it's something we really don't handle. This fixes latency configuration of pipelines. 2008-12-08 18:24:21 +0000 Sebastian Dröge ext/dv/: Really call dv_init() exactly one time, not one time for the demuxer and one time for the decoder. Original commit message from CVS: * ext/dv/gstdv.c: (plugin_init): * ext/dv/gstdvdec.c: (gst_dvdec_class_init): * ext/dv/gstdvdemux.c: (gst_dvdemux_class_init): Really call dv_init() exactly one time, not one time for the demuxer and one time for the decoder. 2008-12-08 12:37:45 +0000 Wim Taymans gst/rtp/gstrtpmp4apay.c: Copy incomming timestamp to outgoing packets. Original commit message from CVS: * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_handle_buffer): Copy incomming timestamp to outgoing packets. 2008-12-08 12:36:21 +0000 Wim Taymans gst/rtp/gstrtpmp4vpay.c: Don't try to push packets before we could find a valid config startcode. Fixes #563509. Original commit message from CVS: * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush), (gst_rtp_mp4v_pay_event): Don't try to push packets before we could find a valid config startcode. Fixes #563509. 2008-12-07 19:22:48 +0000 Brian Cameron sys/sunaudio/gstsunaudiomixerctrl.c: Set the mixer fd before calling ioctl() on it. Fixes bug #563414. Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_open): Set the mixer fd before calling ioctl() on it. Fixes bug #563414. 2008-12-07 19:01:35 +0000 Alexandre Rostovtsev configure.ac: Make usage of libv4l optional by a configure parameter. Original commit message from CVS: Patch by: Alexandre Rostovtsev * configure.ac: Make usage of libv4l optional by a configure parameter. Fixes bug #563504. 2008-12-05 09:24:18 +0000 Sebastian Dröge Add documentation for matroskamux and matroskademux and update the inspection xml files. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: Add documentation for matroskamux and matroskademux and update the inspection xml files. 2008-12-04 20:10:58 +0000 Sebastian Dröge configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change. Original commit message from CVS: * configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change. 2008-12-04 19:47:21 +0000 Sebastian Dröge configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros. Original commit message from CVS: * configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros. 2008-11-30 16:24:45 +0000 Sebastian Dröge gst/udp/gstmultiudpsink.c: Provide the parameters that are required for the format string to fix a compiler warning. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render): Provide the parameters that are required for the format string to fix a compiler warning. 2008-11-29 20:05:41 +0000 Stefan Kost gst/autodetect/gstautoaudiosrc.c: Fix classification. Original commit message from CVS: * gst/autodetect/gstautoaudiosrc.c: Fix classification. 2008-11-29 13:31:55 +0000 Sebastian Dröge Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s... Original commit message from CVS: Patch by: Cygwin Ports maintainer * autogen.sh: * configure.ac: Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will still work. Fixes bug #556091. 2008-11-28 15:10:50 +0000 Peter Kjellerstedt gst/udp/gstmultiudpsink.c: Make gst_multiudpsink_render() ignore errors from sendto() instead of breaking streaming. ... Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render): Make gst_multiudpsink_render() ignore errors from sendto() instead of breaking streaming. Emit a warning instead. Fixes #562572. 2008-11-27 16:43:24 +0000 Ron McOuat Add support for basic and digest authentication in souphttpsrc. Original commit message from CVS: Patch by: Ron McOuat * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init), (gst_soup_http_src_init), (gst_soup_http_src_dispose), (gst_soup_http_src_set_property), (gst_soup_http_src_get_property), (gst_soup_http_src_authenticate_cb), (gst_soup_http_src_start): * ext/soup/gstsouphttpsrc.h: * tests/check/elements/souphttpsrc.c: (basic_auth_cb), (digest_auth_cb), (run_test), (GST_START_TEST), (souphttpsrc_suite), (run_server): Add support for basic and digest authentication in souphttpsrc. Fixes bug #561775. 2008-11-27 12:13:39 +0000 Sebastian Dröge gst/wavenc/: Add support for a-law and mu-law encoded wav files. Fixes bug #562434. Original commit message from CVS: Patch by: Pepijn Van Eeckhoudt * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf), (gst_wavenc_sink_setcaps), (gst_wavenc_change_state): * gst/wavenc/gstwavenc.h: * gst/wavenc/riff.h: Add support for a-law and mu-law encoded wav files. Fixes bug #562434. 2008-11-27 11:22:56 +0000 이문형 gst/rtsp/gstrtspsrc.c: Prevent further read/write actions taken to the connect-failed socket by erroring out quickly.... Original commit message from CVS: Patch by: 이문형 * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp): Prevent further read/write actions taken to the connect-failed socket by erroring out quickly. See #562258. 2008-11-26 21:19:47 +0000 Stefan Kost tests/examples/level/level-example.c: Set fakesink to sync. Otherwise people might question the message interval. Nev... Original commit message from CVS: * tests/examples/level/level-example.c: Set fakesink to sync. Otherwise people might question the message interval. Nevertheless the timestamp in the message is what matters. 2008-11-25 18:13:25 +0000 Wim Taymans tests/icles/.cvsignore: cvsignore newly generated file. Original commit message from CVS: * tests/icles/.cvsignore: cvsignore newly generated file. 2008-11-25 18:03:02 +0000 Wim Taymans gst/rtp/: Fix the descriptions and fix some email addresses. Original commit message from CVS: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps): * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps): * gst/rtp/gstrtpac3depay.h: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps): * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps): * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps): * gst/rtp/gstrtph263depay.h: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps): * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpjpegdepay.h: * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps): * gst/rtp/gstrtpmp1sdepay.h: * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps): * gst/rtp/gstrtpmp2tdepay.h: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp2tpay.h: * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps): * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4apay.h: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps): * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps): * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event): * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvdepay.h: * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process): * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtpsv3vdepay.h: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtptheorapay.h: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers): * gst/rtp/gstrtpvorbispay.h: * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps): * gst/rtp/gstrtpvrawpay.c: Fix the descriptions and fix some email addresses. 2008-11-25 17:47:24 +0000 Julien Moutte gst/qtdemux/qtdemux.c: Add MPG1 and MPG2 fourcc to supported qtdemux video codecs as I found some video clips using t... Original commit message from CVS: 2008-11-25 Julien Moutte * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add MPG1 and MPG2 fourcc to supported qtdemux video codecs as I found some video clips using those. 2008-11-25 16:26:16 +0000 Wim Taymans gst/autodetect/: Post an error when we can't set the internal ghostpad target. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect): * gst/autodetect/gstautoaudiosrc.c: (gst_auto_audio_src_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset), (gst_auto_video_sink_detect): * gst/autodetect/gstautovideosrc.c: (gst_auto_video_src_detect): Post an error when we can't set the internal ghostpad target. 2008-11-25 16:06:22 +0000 Wim Taymans gst/videocrop/gstvideocrop.*: Fix renegotiation when changing properties using the new basetransform features. Fixes ... Original commit message from CVS: * gst/videocrop/gstvideocrop.c: (gst_video_crop_init), (gst_video_crop_transform), (gst_video_crop_transform_caps), (gst_video_crop_set_caps), (gst_video_crop_set_property): * gst/videocrop/gstvideocrop.h: Fix renegotiation when changing properties using the new basetransform features. Fixes #561502. * tests/icles/Makefile.am: * tests/icles/videocrop2-test.c: (make_pipeline), (main): Add crazy interactive test unit for dynamically changing properties. 2008-11-24 12:20:29 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Add some more debugging. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (new_session_pad), (gst_rtspsrc_parse_range): Add some more debugging. Use the reanges received from the server unconditionally. Fixes #561625. 2008-11-23 15:08:45 +0000 Stefan Kost ext/pulse/pulsesink.c: Change #if 0 to something more expresive and add pointer to related bug ticket. Original commit message from CVS: * ext/pulse/pulsesink.c: Change #if 0 to something more expresive and add pointer to related bug ticket. 2008-11-23 11:17:01 +0000 Sebastian Dröge * ChangeLog: ChangeLog surgery Original commit message from CVS: ChangeLog surgery 2008-11-23 11:14:42 +0000 Tal Shalif gst/qtdemux/qtdemux.c: Use G_{BIG,LITTLE}_ENDIAN instead of the non-GLib variants as the latter don't exist on some s... Original commit message from CVS: Patch by: Tal Shalif * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps): Use G_{BIG,LITTLE}_ENDIAN instead of the non-GLib variants as the latter don't exist on some systems (mingw). Fixes bug #561992. 2008-11-21 13:43:29 +0000 Zeeshan Ali ext/soup/gstsouphttpsrc.c: Add transferMode.dnla.org header to HTTP requests as this is required by the DLNA specs an... Original commit message from CVS: Patch by: Zeeshan Ali * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_build_message): Add transferMode.dnla.org header to HTTP requests as this is required by the DLNA specs and doesn't hurt in other situations. Fixes bug #561802. 2008-11-20 23:59:07 +0000 Michael Smith sys/osxvideo/osxvideosink.*: Handle video window resizing more correctly, avoiding crashes when embedding the window ... Original commit message from CVS: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Handle video window resizing more correctly, avoiding crashes when embedding the window and resizing it. 2008-11-20 22:56:58 +0000 Michael Smith gst/udp/: Fix multiudpsink on OSX by passing the specific length of the socket, refactor that into a function shared ... Original commit message from CVS: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpnetutils.c: * gst/udp/gstudpnetutils.h: * gst/udp/gstudpsrc.c: Fix multiudpsink on OSX by passing the specific length of the socket, refactor that into a function shared with the same thing in udpsrc. 2008-11-20 20:07:26 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Fix the scaling code. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (uint64_ceiling_scale), (gst_wavparse_calculate_duration), (gst_wavparse_stream_headers): Fix the scaling code. Fix parsing of the INFO chunks, we were reading the wrong number of bytes. Fixes #561580. 2008-11-20 14:30:40 +0000 Jan Schmidt gst/matroska/matroska-mux.c: Fix NULL pointer dereference of an unset codec_id in the recently added Dirac paths Original commit message from CVS: * gst/matroska/matroska-mux.c: Fix NULL pointer dereference of an unset codec_id in the recently added Dirac paths 2008-11-20 13:58:43 +0000 Jan Schmidt tests/check/Makefile.am: Just keep disabling elements that hang the states test until it works. Original commit message from CVS: * tests/check/Makefile.am: Just keep disabling elements that hang the states test until it works. 2008-11-20 13:46:47 +0000 Jan Schmidt ext/libpng/gstpngenc.c: Don't flush downstream after every buffer - that's not what this libpng callback is for at all! Original commit message from CVS: * ext/libpng/gstpngenc.c: Don't flush downstream after every buffer - that's not what this libpng callback is for at all! 2008-11-17 14:04:20 +0000 Tim-Philipp Müller sys/v4l2/v4l2src_calls.c: Turns out we don't always get the frame sizes in a predefined order from lowest to highest ... Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format_and_size), (sort_by_frame_size), (gst_v4l2src_probe_caps_for_format): Turns out we don't always get the frame sizes in a predefined order from lowest to highest resolution, so let's just sort the list by frame size once we've queried the possible resolutions rather than assume any particular order. Fixes probed caps for the camera in my HP2133 mini notebook and makes v4l2src default to a decent size. 2008-11-16 14:41:32 +0000 Edward Hervey gst/matroska/: Make mkvdemux aware of E-AC3. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps): * gst/matroska/matroska-ids.h: Make mkvdemux aware of E-AC3. 2008-11-14 18:41:29 +0000 Wim Taymans gst/rtp/: Add a jpeg depayloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_base_init), (gst_rtp_jpeg_depay_class_init), (gst_rtp_jpeg_depay_init), (gst_rtp_jpeg_depay_finalize), (MakeTables), (MakeQuantHeader), (MakeHuffmanHeader), (MakeDRIHeader), (MakeHeaders), (gst_rtp_jpeg_depay_setcaps), (gst_rtp_jpeg_depay_process), (gst_rtp_jpeg_depay_change_state), (gst_rtp_jpeg_depay_plugin_init): * gst/rtp/gstrtpjpegdepay.h: Add a jpeg depayloader. * gst/rtp/gstrtpjpegpay.c: Set the default properties on the payloader to better defaults. 2008-11-14 15:42:32 +0000 Stefan Kost sys/v4l2/gstv4l2.c: Give it a primary rank for autovideosrc. Original commit message from CVS: * sys/v4l2/gstv4l2.c: Give it a primary rank for autovideosrc. 2008-11-14 11:41:55 +0000 Bjorn Ostby gst/rtp/: Add JPEG payloader. Fixes #560756. Original commit message from CVS: Patch by: Bjorn Ostby * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpjpegpay.c: (gst_rtp_jpeg_pay_base_init), (gst_rtp_jpeg_pay_class_init), (gst_rtp_jpeg_pay_init), (gst_rtp_jpeg_pay_setcaps), (gst_rtp_jpeg_pay_header_size), (gst_rtp_jpeg_pay_read_quant_table), (gst_rtp_jpeg_pay_scan_marker), (gst_rtp_jpeg_pay_handle_buffer), (gst_rtp_jpeg_pay_set_property), (gst_rtp_jpeg_pay_get_property), (gst_rtp_jpeg_pay_plugin_init): * gst/rtp/gstrtpjpegpay.h: Add JPEG payloader. Fixes #560756. 2008-11-13 17:45:59 +0000 Fabricio Godoy sys/: Fix some spelling mistakes. Fixes #556802. Original commit message from CVS: Patch by: Fabricio Godoy * sys/oss/gstosssink.c: (gst_oss_sink_open): * sys/oss/gstosssrc.c: (gst_oss_src_open): * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_mmap): * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): Fix some spelling mistakes. Fixes #556802. 2008-11-13 16:24:59 +0000 Stefan Kost gst/equalizer/: Add presets for equalizer. Fixes #522183. Original commit message from CVS: * gst/equalizer/GstIirEqualizer10Bands.prs: * gst/equalizer/GstIirEqualizer3Bands.prs: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: Add presets for equalizer. Fixes #522183. 2008-11-13 16:17:38 +0000 Wim Taymans gst/rtsp/: Remove google extension again, it's not needed anymore because we never send multiple transports anymore. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspgoogle.c: * gst/rtsp/gstrtspgoogle.h: Remove google extension again, it's not needed anymore because we never send multiple transports anymore. 2008-11-13 16:11:16 +0000 Eric Zhang gst/rtsp/gstrtspsrc.*: Add property to configure NAT traversal method. Original commit message from CVS: Based on patch by: Eric Zhang * gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp_sinks), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send_dummy_packets), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add property to configure NAT traversal method. Ignore EOS from the internal sinks. Implement sending dummy packets as a (simple) method to open up some firewalls. Send PLAY request to the server after we started the udp sources. Fixes #559545. 2008-11-13 14:04:40 +0000 Yotam gst/rtp/gstrtpmp4vpay.c: Flush the remaining frames on EOS. Fixes #560641. Original commit message from CVS: Patch by: Yotam * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event): Flush the remaining frames on EOS. Fixes #560641. 2008-11-12 16:37:06 +0000 Jan Schmidt gst/rtp/gstrtpg729pay.c: Fix compiler warning about printf formatting. Original commit message from CVS: * gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_handle_buffer): Fix compiler warning about printf formatting. 2008-11-12 11:55:14 +0000 Andy Wingo gst/qtdemux/qtdemux.*: Queue up new segment events instead of sending them from the seeking thread. Original commit message from CVS: * gst/qtdemux/qtdemux.h (struct _GstQTDemux): * gst/qtdemux/qtdemux.c (gst_qtdemux_do_seek): Queue up new segment events instead of sending them from the seeking thread. Fixes #559288. (gst_qtdemux_push_pending_newsegment): New helper, sends out queued newsegment events. (gst_qtdemux_loop_state_movie): Voilà, call it here. Only need to call it here, as we only seek when looping, and only push in the movie state. 2008-11-11 19:52:05 +0000 Mark Nauwelaerts gst/qtdemux/: Add cover and alternative copyright tag, and enhance some existing ones by marking them as container at... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_tag_add_tmpo), (qtdemux_tag_add_covr), (qtdemux_parse_udta): * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: Add cover and alternative copyright tag, and enhance some existing ones by marking them as container atoms. 2008-11-11 17:33:00 +0000 Wim Taymans gst/rtp/gstrtpg729pay.c: Don't ignore the return value of setcaps. Original commit message from CVS: * gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_set_caps): Don't ignore the return value of setcaps. 2008-11-11 17:29:03 +0000 Olivier Crete gst/rtp/gstrtpg729pay.*: Replace G729 payloader with an improved version. Fixes #532409. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtpg729pay.c: (gst_rtp_g729_pay_base_init), (gst_rtp_g729_pay_class_init), (gst_rtp_g729_pay_init), (gst_rtp_g729_pay_set_caps), (gst_rtp_g729_pay_handle_buffer): * gst/rtp/gstrtpg729pay.h: Replace G729 payloader with an improved version. Fixes #532409. 2008-11-11 16:00:48 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Only send one transport at a time for improved compatibility with some broken servers. See #53... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string), (gst_rtspsrc_change_state): Only send one transport at a time for improved compatibility with some broken servers. See #537832. 2008-11-11 15:16:31 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Only pause/play in the seek handler when the source was playing. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek): Only pause/play in the seek handler when the source was playing. Fixes #529379. 2008-11-11 12:18:23 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Fix muxing of Dirac streams if the input already has the format we need, i.e. is the out... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_handle_dirac_packet): Fix muxing of Dirac streams if the input already has the format we need, i.e. is the output of matroskademux. 2008-11-11 10:06:01 +0000 Stefan Kost gst/avi/gstavimux.c: Don't segfault on string typed tags being NULL. Fixes #560155. Original commit message from CVS: * gst/avi/gstavimux.c: Don't segfault on string typed tags being NULL. Fixes #560155. 2008-11-10 16:44:45 +0000 Mark Nauwelaerts gst/matroska/matroska-mux.c: Fix mapping AAC profile to Matroska codec id. Original commit message from CVS: * gst/matroska/matroska-mux.c: (aac_codec_data_to_codec_id), (gst_matroska_mux_audio_pad_setcaps): Fix mapping AAC profile to Matroska codec id. 2008-11-10 16:36:09 +0000 Mark Nauwelaerts gst/qtdemux/qtdemux.c: Refactor some raw audio caps building, and handle >16-bit cases. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps): Refactor some raw audio caps building, and handle >16-bit cases. Fix/replace building caps from a string description. 2008-11-10 13:59:27 +0000 Thomas Vander Stichele gst/: Make author name consistent with others. Original commit message from CVS: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/cutter/gstcutter.c: Make author name consistent with others. 2008-11-10 12:13:21 +0000 Eric Zhang gst/rtsp/gstrtspsrc.c: Pause the RTSP stream before doing a new play request. Original commit message from CVS: Based on patch by: Eric Zhang * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek), (gst_rtspsrc_stream_configure_udp_sink): Pause the RTSP stream before doing a new play request. Make sure that adding the udpsinks does not cause the rtspsrc to become a sink. Fixes #559547. 2008-11-05 14:42:35 +0000 Sebastian Dröge gst/matroska/: Implement Dirac muxing into Matroska comforming to the spec, i.e. put all Dirac packages up to a pictu... Original commit message from CVS: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: (gst_matroska_pad_free), (gst_matroska_mux_handle_dirac_packet), (gst_matroska_mux_write_data): Implement Dirac muxing into Matroska comforming to the spec, i.e. put all Dirac packages up to a picture into a Matroska block. TODO: Implement writing of the ReferenceBlock Matroska elements, currently the Dirac muxing is only 100% correct if Matroska version 2 is selected for muxing. 2008-11-04 12:32:48 +0000 Bastien Nocera Optionally use libv4l to access v4l2 devices. Fixes bug #545033. Original commit message from CVS: Patch by: Bastien Nocera , Hans de Goede * configure.ac: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read): * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities), (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_close), (gst_v4l2_get_norm), (gst_v4l2_set_norm), (gst_v4l2_get_frequency), (gst_v4l2_set_frequency), (gst_v4l2_signal_strength), (gst_v4l2_get_attribute), (gst_v4l2_set_attribute), (gst_v4l2_get_input), (gst_v4l2_set_input): * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize), (gst_v4l2_buffer_new), (gst_v4l2_buffer_pool_finalize), (gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate), (gst_v4l2src_fill_format_list), (gst_v4l2src_probe_caps_for_format_and_size), (gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init), (gst_v4l2src_capture_start), (gst_v4l2src_capture_stop), (gst_v4l2src_get_nearest_size): Optionally use libv4l to access v4l2 devices. Fixes bug #545033. 2008-11-04 12:28:34 +0000 Stefan Kost Don't install static libs for plugins. Fixes #550851 for -good. Original commit message from CVS: * ext/aalib/Makefile.am: * ext/annodex/Makefile.am: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/esd/Makefile.am: * ext/flac/Makefile.am: * ext/gconf/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/hal/Makefile.am: * ext/jpeg/Makefile.am: * ext/ladspa/Makefile.am: * ext/libcaca/Makefile.am: * ext/libmng/Makefile.am: * ext/libpng/Makefile.am: * ext/mikmod/Makefile.am: * ext/pulse/Makefile.am: * ext/raw1394/Makefile.am: * ext/shout2/Makefile.am: * ext/soup/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/Makefile.am: * ext/wavpack/Makefile.am: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debug/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/interleave/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/median/Makefile.am: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/oldcore/Makefile.am: * gst/qtdemux/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtsp/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * sys/directdraw/Makefile.am: * sys/directsound/Makefile.am: * sys/oss/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxvideo/Makefile.am: * sys/sunaudio/Makefile.am: * sys/v4l2/Makefile.am: * sys/waveform/Makefile.am: * sys/ximage/Makefile.am: Don't install static libs for plugins. Fixes #550851 for -good. 2008-10-31 18:17:50 +0000 Sebastian Dröge ext/flac/Makefile.am: Include $(FLAC_CFLAGS) in CFLAGS to make sure to find the FLAC headers. Original commit message from CVS: * ext/flac/Makefile.am: Include $(FLAC_CFLAGS) in CFLAGS to make sure to find the FLAC headers. This fixes compilation if FLAC is installed in an uncommon location that is not already handled by other CFLAGS. Fixes bug #558711. 2008-10-31 10:08:50 +0000 Wim Taymans sys/v4l2/v4l2src_calls.c: Guard more uncommon formats with ifdefs so that we can compile on older versions. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank): Guard more uncommon formats with ifdefs so that we can compile on older versions. 2008-10-31 10:00:18 +0000 Nick Haddad gst/avi/gstavidemux.c: Invert other uncompressed RGB formats. Fixes #558554. Original commit message from CVS: Patch by: Nick Haddad * gst/avi/gstavidemux.c: (gst_avi_demux_is_uncompressed), (gst_avi_demux_invert), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data): Invert other uncompressed RGB formats. Fixes #558554. 2008-10-30 15:08:49 +0000 Sebastian Dröge gst/wavenc/gstwavenc.*: Add support for float/double as input and remove the (nowadays) useless parsing of the depth ... Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf), (gst_wavenc_sink_setcaps), (gst_wavenc_change_state): * gst/wavenc/gstwavenc.h: Add support for float/double as input and remove the (nowadays) useless parsing of the depth as we require width==depth. 2008-10-30 10:31:35 +0000 Wim Taymans gst/rtp/: Narrow down the caps of the mpeg audio pay/depayloaders to only accept mpeg version 1. Fixes #558427. Original commit message from CVS: * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps): * gst/rtp/gstrtpmpapay.c: Narrow down the caps of the mpeg audio pay/depayloaders to only accept mpeg version 1. Fixes #558427. 2008-10-29 18:28:25 +0000 Wim Taymans gst/rtp/gstrtpL16pay.c: Only put an integral amount of samples in the RTP packet. Original commit message from CVS: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_getcaps): Only put an integral amount of samples in the RTP packet. Fixes #556641. 2008-10-28 17:42:02 +0000 Wim Taymans gst/rtp/gstrtpchannels.*: Add method to get possible channel positions. Original commit message from CVS: * gst/rtp/gstrtpchannels.c: (gst_rtp_channels_get_by_index): * gst/rtp/gstrtpchannels.h: Add method to get possible channel positions. 2008-10-28 17:39:48 +0000 Wim Taymans gst/rtp/Makefile.am: Also commit updated makefile Original commit message from CVS: * gst/rtp/Makefile.am: Also commit updated makefile 2008-10-28 14:56:08 +0000 Sebastian Dröge gst/wavenc/gstwavenc.c: Don't allow width=32,depth=24 as input. WAV requires that the width is the next integer multi... Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Don't allow width=32,depth=24 as input. WAV requires that the width is the next integer multiply of 8 from the depth. 2008-10-28 10:01:49 +0000 Wim Taymans gst/rtp/: Add mappings for multichannel support. Does not completely just work because the getcaps function does not ... Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps): * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_getcaps): * gst/rtp/gstrtpchannels.c: (check_channels), (gst_rtp_channels_get_by_pos), (gst_rtp_channels_get_by_order), (gst_rtp_channels_create_default): * gst/rtp/gstrtpchannels.h: Add mappings for multichannel support. Does not completely just work because the getcaps function does not yet return the allowed channel mappings. See #556641. 2008-10-28 06:50:57 +0000 Stefan Kost gst/goom/: Add license headers in all source files. Remove filter.c from Original commit message from CVS: * gst/goom/Makefile.am: * gst/goom/README: * gst/goom/config_param.c: * gst/goom/convolve_fx.c: * gst/goom/drawmethods.c: * gst/goom/drawmethods.h: * gst/goom/filters.c: * gst/goom/filters_mmx.s: * gst/goom/flying_stars_fx.c: * gst/goom/goom.h: * gst/goom/goom_config.h: * gst/goom/goom_config_param.h: * gst/goom/goom_core.c: * gst/goom/goom_filters.h: * gst/goom/goom_fx.h: * gst/goom/goom_graphic.h: * gst/goom/goom_plugin_info.h: * gst/goom/goom_tools.c: * gst/goom/goom_tools.h: * gst/goom/goom_typedefs.h: * gst/goom/goom_visual_fx.h: * gst/goom/graphic.c: * gst/goom/ifs.c: * gst/goom/ifs.h: * gst/goom/lines.c: * gst/goom/lines.h: * gst/goom/mathtools.c: * gst/goom/mathtools.h: * gst/goom/mmx.c: * gst/goom/motif_goom1.h: * gst/goom/motif_goom2.h: * gst/goom/plugin_info.c: * gst/goom/ppc_drawings.h: * gst/goom/ppc_zoom_ultimate.h: * gst/goom/sound_tester.c: * gst/goom/sound_tester.h: * gst/goom/surf3d.c: * gst/goom/surf3d.h: * gst/goom/tentacle3d.c: * gst/goom/tentacle3d.h: * gst/goom/v3d.c: * gst/goom/v3d.h: * gst/goom/xmmx.c: Add license headers in all source files. Remove filter.c from EXTRA_DIST, as its in SOURCES already. Mention the files in the REDME which are not used right now. Fixes #557709. 2008-10-27 11:28:30 +0000 Olivier Crete gst/rtp/gstrtpL16pay.c: Implement getcaps in rtpL16pay. Fixes #556484. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_class_init), (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_getcaps): Implement getcaps in rtpL16pay. Fixes #556484. 2008-10-27 11:03:53 +0000 Wim Taymans gst/rtp/gstrtpL16depay.c: Check if clock-rate and channels are valid. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps), (gst_rtp_L16_depay_process): Check if clock-rate and channels are valid. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process): Don't ignore the return value of set_caps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpamrdepay.h: Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set output caps on the buffers, the base class does that for us. The subclass will make sure we are negotiated. * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps), (gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset): * gst/rtp/gstrtpdvdepay.h: Clean up caps negotiation. The subclass will make sure we are negotiated. * gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init), (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process): * gst/rtp/gstrtpg729depay.h: The subclass will make sure we are negotiated. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps), (gst_rtp_gsm_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps): Clean up caps negotiation. Don't ignore the return value of set_outcaps. * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps), (gst_rtp_h263_depay_process): Clean up caps negotiation. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps), (gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer): * gst/rtp/gstrtph263pay.h: Don't ignore the return value of set_outcaps. Do some more timestamps. * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps), (gst_rtp_h263p_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush), (gst_rtp_h263p_pay_handle_buffer): * gst/rtp/gstrtph263ppay.h: Don't ignore the return value of set_outcaps. Do some more timestamps. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. Fix possible caps leak. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Add some more debug info. * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps), (gst_rtp_ilbc_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps): Clean up caps negotiation. * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps), (gst_rtp_mp4a_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. No need to set caps on buffers, subclass does that for us. * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps), (gst_rtp_mp4v_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): Clean up caps negotiation. Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process): Clean up caps negotiation. Actually set output caps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps), (gst_rtp_pcma_depay_process): Clean up caps negotiation. Set output buffer duration because we can. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps), (gst_rtp_pcmu_depay_process): Clean up caps negotiation. Use the marker bit to set the DISCONT flag on outgoing buffers. * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init), (gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process): Clean up caps negotiation. Set output caps on the pad and header buffers. Set duration on output buffers because we can. * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps), (gst_rtp_sv3v_depay_process): Clean up caps negotiation. No need to validate the buffer, the base class does that for us. No need to set caps out output buffers, subclass does that. * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps), (gst_rtp_theora_depay_process): Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_flush_packet), (encode_base64), (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id), (gst_rtp_theora_pay_handle_buffer): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps), (gst_rtp_vorbis_depay_process): Don't ignore the return value of setcaps. No need to validate the buffer, the base class does that for us. * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers): Don't ignore the return value of set_outcaps. * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps): Clean up caps negotiation, don't ignore setcaps return. * gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps): Don't ignore the return value of set_outcaps. 2008-10-27 10:35:07 +0000 Wim Taymans gst/matroska/matroska-demux.c: Forward unknown events upstream. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_src_event): Forward unknown events upstream. 2008-10-27 10:33:20 +0000 Wim Taymans tests/check/elements/icydemux.c: Add some refcount check Original commit message from CVS: * tests/check/elements/icydemux.c: (icydemux_found_pad): Add some refcount check * tests/check/elements/rtp-payloading.c: (rtp_pipeline_run): Don't ignore the result of write(), fixes a compiler warning for me. * tests/icles/videobox-test.c: (main): Make the output a little more pretty. 2008-10-27 09:26:19 +0000 Stefan Kost ext/esd/esdmon.c: Add doc blob. Original commit message from CVS: * ext/esd/esdmon.c: Add doc blob. 2008-10-27 09:21:44 +0000 Stefan Kost docs/plugins/: Add the docs of the new elements. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-autodetect.xml: Add the docs of the new elements. 2008-10-27 09:04:37 +0000 Sebastian Dröge gst/autodetect/: Fix "Since" tags in the documentation. Original commit message from CVS: * gst/autodetect/gstautoaudiosrc.c: (gst_auto_audio_src_class_init): * gst/autodetect/gstautovideosrc.c: (gst_auto_video_src_class_init): Fix "Since" tags in the documentation. 2008-10-27 09:00:29 +0000 Sjoerd Simons ext/soup/gstsouphttpsrc.c: Add support for souphttpsrc to act as a live source. This makes it possible to get timesta... Original commit message from CVS: Patch by: Sjoerd Simons * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init), (gst_soup_http_src_set_property), (gst_soup_http_src_get_property): Add support for souphttpsrc to act as a live source. This makes it possible to get timestamped buffers in combination with the "do-timestamp" property. Fixes bug #556019. 2008-10-27 08:54:30 +0000 Stefan Kost gst/autodetect/: Implement src plugins. Little code/string cleanup in the sinks. Original commit message from CVS: * gst/autodetect/Makefile.am: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautoaudiosrc.h: * gst/autodetect/gstautodetect.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/autodetect/gstautovideosrc.h: Implement src plugins. Little code/string cleanup in the sinks. Fixes #523813. 2008-10-27 08:45:11 +0000 Peter Kjellerstedt gst/matroska/matroska-mux.c: Fix a memory leak when pads are requested but the pipeline never goes into PLAYING. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset), (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad): Fix a memory leak when pads are requested but the pipeline never goes into PLAYING. Correctly remove request pads, no matter if they have collected data or not. Fixes bug #557710. 2008-10-27 08:40:02 +0000 Sebastian Dröge gst/udp/gstudpnetutils.h: Define the correct WINVER so getaddinfo() can be used when using mingw32. Fixes bug #557294. Original commit message from CVS: Patch by: * gst/udp/gstudpnetutils.h: Define the correct WINVER so getaddinfo() can be used when using mingw32. Fixes bug #557294. 2008-10-27 08:36:43 +0000 Sebastian Dröge gst/udp/: Fix "argument type mismatch" compiler warnings on Windows. Original commit message from CVS: Patch by: * gst/udp/gstdynudpsink.c: (gst_dynudpsink_render): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render): * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Fix "argument type mismatch" compiler warnings on Windows. Fixes bug #557293. 2008-10-27 08:30:51 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Don't calculate the filter coefficients for every single buffer but only when it's n... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (update_coefficients): Don't calculate the filter coefficients for every single buffer but only when it's needed. Fixes bug #557260. 2008-10-26 20:05:43 +0000 Jan Schmidt configure.ac: Back to development -> 0.10.11.1 Original commit message from CVS: * configure.ac: Back to development -> 0.10.11.1 2008-10-26 20:04:21 +0000 Jan Schmidt gst-plugins-good.doap: Fix version number of 0.10.11 release in doap file Original commit message from CVS: * gst-plugins-good.doap: Fix version number of 0.10.11 release in doap file === release 0.10.11 === 2008-10-24 22:41:18 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.11 Original commit message from CVS: Release 0.10.11 2008-10-24 22:20:47 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files Original commit message from CVS: Update .po files 2008-10-24 16:30:53 +0000 Jan Schmidt configure.ac: Commit 0.10.10.4 pre-release Original commit message from CVS: * configure.ac: Commit 0.10.10.4 pre-release 2008-10-21 12:42:45 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Fix VPRP chunk setup in avimux. Original commit message from CVS: * gst/avi/gstavimux.c: Fix VPRP chunk setup in avimux. Fixes: #556010 Patch By: Mark Nauwelaerts 2008-10-21 12:38:35 +0000 Wim Taymans gst/videobox/gstvideobox.c: support dynamically changing properties in videobox Original commit message from CVS: * gst/videobox/gstvideobox.c: support dynamically changing properties in videobox Fixed: #557085 Patch By: Wim Taymans 2008-10-16 17:10:42 +0000 Jan Schmidt configure.ac: 0.10.10.3 pre-release Original commit message from CVS: * configure.ac: 0.10.10.3 pre-release 2008-10-16 15:30:22 +0000 Jan Schmidt tests/check/Makefile.am: Don't run the states test on pulsesrc and pulsesink Original commit message from CVS: * tests/check/Makefile.am: Don't run the states test on pulsesrc and pulsesink 2008-10-16 11:52:44 +0000 Jan Schmidt configure.ac: Commit 0.10.10.2 pre-release bump that actually went out on 2008-10-11 Original commit message from CVS: * configure.ac: Commit 0.10.10.2 pre-release bump that actually went out on 2008-10-11 2008-10-15 15:42:29 +0000 Edward Hervey gst/avi/gstavidemux.c: Skip entries for streams that don't have a output pad yet, thereby avoiding calling pad functi... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan): Skip entries for streams that don't have a output pad yet, thereby avoiding calling pad functions with a NULL pad. Fixes #556424 2008-10-15 09:39:27 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Remove previous wrong commit Original commit message from CVS: * gst/qtdemux/qtdemux.c: Remove previous wrong commit * tests/check/elements/icydemux.c: (icydemux_found_pad): Remove problematic and useless refcount check. Fixes #556381 2008-10-15 09:27:27 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Remove problematic and useless refcount check. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_node): Remove problematic and useless refcount check. Fixes #556381 2008-10-13 18:10:25 +0000 Stefan Kost Don't install static libs for plugins. Fixes #550851 for ugly. Original commit message from CVS: * ext/a52dec/Makefile.am: * ext/amrnb/Makefile.am: * ext/cdio/Makefile.am: * ext/dvdnav/Makefile.am: * ext/dvdread/Makefile.am: * ext/lame/Makefile.am: * ext/mad/Makefile.am: * ext/mpeg2dec/Makefile.am: * ext/sidplay/Makefile.am: * gst/ac3parse/Makefile.am: * gst/asfdemux/Makefile.am: * gst/dvdlpcmdec/Makefile.am: * gst/dvdsub/Makefile.am: * gst/iec958/Makefile.am: * gst/mpegaudioparse/Makefile.am: * gst/mpegstream/Makefile.am: * gst/realmedia/Makefile.am: * gst/synaesthesia/Makefile.am: Don't install static libs for plugins. Fixes #550851 for ugly. 2008-10-10 12:28:34 +0000 Jan Schmidt ext/flac/: Cast some size_t arguments to guint to avoid compiler warnings on 64-bit systems. Original commit message from CVS: * ext/flac/gstflacdec.c (gst_flac_dec_read_stream): * ext/flac/gstflacenc.c (gst_flac_enc_write_callback): Cast some size_t arguments to guint to avoid compiler warnings on 64-bit systems. 2008-10-09 14:27:12 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Return TRUE instead of FALSE from the event handler when we swallowed the event. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event): Return TRUE instead of FALSE from the event handler when we swallowed the event. 2008-10-08 15:59:56 +0000 Christian Schaller * gst-plugins-good.spec.in: remove old CDIO plugin now in ugly Original commit message from CVS: remove old CDIO plugin now in ugly 2008-10-08 14:47:14 +0000 Wim Taymans gst/avi/gstavidemux.c: Reset header state. Fixes #555321. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index): Reset header state. Fixes #555321. 2008-10-08 13:31:44 +0000 Wim Taymans gst/avi/gstavidemux.*: For timestamping audio packets we need to take into account the amount of blocks in one entry ... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index): * gst/avi/gstavidemux.h: For timestamping audio packets we need to take into account the amount of blocks in one entry using the blockalign. Fixes some sync issues with zero-padded audio blocks in the beginning of avi files. 2008-10-08 10:42:26 +0000 Wim Taymans gst/multifile/gstmultifilesrc.c: Implement DEFAULT and BUFFER position queries. See #555260. Original commit message from CVS: * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init), (gst_multi_file_src_query): Implement DEFAULT and BUFFER position queries. See #555260. 2008-10-08 09:29:00 +0000 Edward Hervey sys/ximage/gstximagesrc.c: Fix build for systems that don't have XDamage. Original commit message from CVS: * sys/ximage/gstximagesrc.c: (gst_ximage_src_stop): Fix build for systems that don't have XDamage. 2008-10-07 09:58:13 +0000 Wim Taymans tests/examples/rtp/: Add some more H263p server and client examples. Original commit message from CVS: * tests/examples/rtp/client-H263p.sdp: * tests/examples/rtp/client-H263p.sh: * tests/examples/rtp/server-VTS-H263p.sh: Add some more H263p server and client examples. 2008-10-03 17:03:07 +0000 Tim-Philipp Müller configure.ac: Depend on released versions of core and base. Original commit message from CVS: * configure.ac:: Depend on released versions of core and base. 2008-10-03 16:13:32 +0000 Wim Taymans ext/pulse/: Return -1 instead of 0 in error cases. Fixes #554771. Original commit message from CVS: * ext/pulse/pulsesink.c: (gst_pulsesink_write): * ext/pulse/pulsesrc.c: (gst_pulsesrc_read): Return -1 instead of 0 in error cases. Fixes #554771. 2008-10-03 15:54:07 +0000 Wim Taymans sys/ximage/gstximagesrc.c: Stop leaking the cursor image. Original commit message from CVS: * sys/ximage/gstximagesrc.c: (gst_ximage_src_start), (gst_ximage_src_stop), (gst_ximage_src_ximage_get): Stop leaking the cursor image. Unref the last_ximage and the cached cursor image on shutdown. Fixes #551570. 2008-10-03 11:32:47 +0000 Wim Taymans sys/v4l2/gstv4l2object.h: Getting the Class from an instance is not just a matter of casting it to the class struct b... Original commit message from CVS: * sys/v4l2/gstv4l2object.h: Getting the Class from an instance is not just a matter of casting it to the class struct but it involves calling G_OBJECT_GET_CLASS on the instance. Fixes #549784. 2008-10-01 21:22:26 +0000 Michael Smith configure.ac: Fix libs for linking directsound. Original commit message from CVS: * configure.ac: Fix libs for linking directsound. * sys/directsound/gstdirectsoundsink.c: Fix buffer sizing to prevent racing the ringbuffer at startup. Add volume property. 2008-09-27 00:43:07 +0000 Jan Schmidt ext/pulse/pulsesink.c: Fix problems with pulsesink randomly erroring with code 'OK' after a format change on the stre... Original commit message from CVS: * ext/pulse/pulsesink.c: Fix problems with pulsesink randomly erroring with code 'OK' after a format change on the stream by waiting when disconnecting the stream. 2008-09-26 14:44:49 +0000 Wim Taymans gst/rtp/gstrtpamrdepay.c: Mark DISCONT on output buffers when the marker bit signals a new talk spurt. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init), (gst_rtp_amr_depay_process): Mark DISCONT on output buffers when the marker bit signals a new talk spurt. * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer): Set the marker bit for buffers with a DISCONT flag to signal a talk spurt. 2008-09-26 13:55:48 +0000 Wim Taymans gst/rtp/: Added MP4A-LATM payloader to match the depayloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type), (gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init), (gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize), (gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps), (gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer), (gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init): * gst/rtp/gstrtpmp4apay.h: Added MP4A-LATM payloader to match the depayloader. 2008-09-25 15:11:16 +0000 Wim Taymans gst/videomixer/videomixer.c: Handle segments a little better. Fixes #537361. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues), (gst_videomixer_sink_event): Handle segments a little better. Fixes #537361. 2008-09-25 12:07:46 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Don't assume the server supports PAUSE by default. Fixes #551048. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods): Don't assume the server supports PAUSE by default. Fixes #551048. 2008-09-25 11:30:35 +0000 Wim Taymans gst/udp/gstudpsrc.c: Switch on the socket family to get the addrlen size right. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_set_uri), (gst_udpsrc_start): Switch on the socket family to get the addrlen size right. 2008-09-25 10:34:39 +0000 Daniel Franke gst/udp/gstudpsrc.c: OS X's bind() implementation is picky about its addrlen parameter and fails with EINVAL if it is... Original commit message from CVS: Patch by: Daniel Franke * gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start): OS X's bind() implementation is picky about its addrlen parameter and fails with EINVAL if it is larger than expected for the socket's address family. Set the length to the expected length instead. Fixes #553191. 2008-09-23 18:08:56 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Handle the case where we cannot do desribe or when the describe result does not contain a vali... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Handle the case where we cannot do desribe or when the describe result does not contain a valid SDP message. 2008-09-23 17:31:22 +0000 Wim Taymans gst/udp/gstmultiudpsink.c: Fix setting the qos. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_set_property): Fix setting the qos. 2008-09-17 14:50:42 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Some 'broken' files out there have atom lengths of zero... which basically results in qtdemux ... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header), (gst_qtdemux_chain): Some 'broken' files out there have atom lengths of zero... which basically results in qtdemux consuming that atom again and again until the *end of night* ! Detect that and emits an adequate element error message. 2008-09-17 13:49:04 +0000 Jan Schmidt gst/: Fix build flags order. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/matroska/Makefile.am: Fix build flags order. * tests/check/elements/audioamplify.c: (GST_START_TEST): * tests/check/elements/audiodynamic.c: (GST_START_TEST): * tests/check/elements/audioinvert.c: (GST_START_TEST): * tests/check/elements/audiopanorama.c: (GST_START_TEST): Format fixes. * tests/check/elements/multifile.c: Pull in unistd.h 2008-09-15 21:10:23 +0000 Wim Taymans gst/rtp/gstrtpmp4gdepay.*: Handle interleaved streams by reordering AU in a queue. Original commit message from CVS: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_clear_queue), (gst_rtp_mp4g_depay_flush_queue), (gst_rtp_mp4g_depay_queue), (gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state): * gst/rtp/gstrtpmp4gdepay.h: Handle interleaved streams by reordering AU in a queue. 2008-09-15 16:04:26 +0000 Wim Taymans gst/rtp/gstrtpmp4gdepay.c: Change some of the ranges in the caps, mostly for the amount of bits we can use. Original commit message from CVS: * gst/rtp/gstrtpmp4gdepay.c: (gst_bs_parse_init), (gst_bs_parse_read), (gst_rtp_mp4g_depay_process): Change some of the ranges in the caps, mostly for the amount of bits we can use. Added a little bitstream parse and use it to parse the AU header fields. Check for malformed and wrongly sized packets better. Implement more header field parsing. Handle the size of fragmented packets correctly. 2008-09-14 11:32:15 +0000 Jonathan Matthew gst/qtdemux/qtdemux.c: Add mapping for 'tiff' => image/tiff Original commit message from CVS: Patch by: Jonathan Matthew * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add mapping for 'tiff' => image/tiff Fixes #552213 2008-09-11 11:26:06 +0000 Tim-Philipp Müller ext/raw1394/: Pretend to care about the result of write() which works around compiler warnings. Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (SEND_COMMAND): * ext/raw1394/gsthdv1394src.c: (SEND_COMMAND): Pretend to care about the result of write() which works around compiler warnings. 2008-09-04 09:25:59 +0000 Tim-Philipp Müller ext/flac/gstflacenc.c: Make sure the desired default values are actually set, not only registered as defaults (actual... Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_class_init): Make sure the desired default values are actually set, not only registered as defaults (actual problem is that the stereo-specific values are only updated if channels==2, which is not the case yet when the object is created, so the default values for the mid-side-stereo and loose-mid-side-stereo settings are never set in _update_quality()). Makes flacenc create smaller files by default (for stereo input), and fixes #550791. 2008-09-03 12:39:35 +0000 Mark Nauwelaerts gst/qtdemux/: Add support for video/mj2 mime-type and its additional atoms/boxes. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state), (gst_qtdemux_loop_state_header), (qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps): * gst/qtdemux/qtdemux.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: Add support for video/mj2 mime-type and its additional atoms/boxes. Fixes #550646. 2008-09-03 11:10:25 +0000 Stefan Kost gst/debug/gsttaginject.c: Add warning when tags parameter is unparsable and give example for quoting in the docs. Original commit message from CVS: * gst/debug/gsttaginject.c: Add warning when tags parameter is unparsable and give example for quoting in the docs. 2008-09-02 15:27:49 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Add mapping for IMA Loki SDL MJPEG ADPCM codec. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps): Add mapping for IMA Loki SDL MJPEG ADPCM codec. Add some alternative byteswapped mappings that seem to pop up sometimes. Fixes #550288. 2008-09-02 09:40:38 +0000 Tim-Philipp Müller po/: Add 'ca' to LINGUAS; add some more files with translations and some files which should be ignored by translation... Original commit message from CVS: * po/LINGUAS: * po/POTFILES.in: * po/POTFILES.skip: Add 'ca' to LINGUAS; add some more files with translations and some files which should be ignored by translation tools. 2008-09-02 08:51:04 +0000 Sebastian Dröge ext/speex/: Use integer encoding and decoding functions instead of converting the integer input to float in the eleme... Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data): * ext/speex/gstspeexdec.h: * ext/speex/gstspeexenc.c: (gst_speex_enc_encode): * ext/speex/gstspeexenc.h: Use integer encoding and decoding functions instead of converting the integer input to float in the element. The libspeex integer functions are doing this for us already or, if libspeex was compiled in integer mode, they're doing everything using integer arithmetics. Also saves some copying around. 2008-09-01 13:29:29 +0000 Tim-Philipp Müller configure.ac: Fix --disable-external Original commit message from CVS: * configure.ac: Fix --disable-external 2008-08-31 17:09:18 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.*: Handle non-zero start timestamps and stream discontinuities correctly. This only has an ... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset), (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain): * ext/wavpack/gstwavpackenc.h: Handle non-zero start timestamps and stream discontinuities correctly. This only has an effect if we're muxing into a container format as the raw WavPack stream must contain continous sample numbers. 2008-08-31 15:02:09 +0000 Sebastian Dröge ext/speex/gstspeexenc.c: Correct the timestamp and granulepos calculation by one Speex frame. Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speex_enc_encode): Correct the timestamp and granulepos calculation by one Speex frame. 2008-08-31 14:39:57 +0000 Sebastian Dröge ext/speex/gstspeexdec.c: Correctly take the granulepos from upstream if possible and correctly handle the granulepos ... Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data): Correctly take the granulepos from upstream if possible and correctly handle the granulepos in various calculations: the granulepos is the sample number of the _last_ sample in a frame, not the first. * ext/speex/gstspeexenc.c: (gst_speex_enc_sinkevent), (gst_speex_enc_encode), (gst_speex_enc_chain), (gst_speex_enc_change_state): * ext/speex/gstspeexenc.h: Handle non-zero start timestamps in the encoder and detect/handle stream discontinuities. Fixes bug #547075. 2008-08-31 08:32:45 +0000 Craig Keogh ext/annodex/gstcmmlparser.c: Fix compiler warnings caused by passing a string as format string instead of "%s" and th... Original commit message from CVS: Patch by: Craig Keogh * ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk): Fix compiler warnings caused by passing a string as format string instead of "%s" and then the string. This is only exposed by -Wformat=2 as used by default on Ubuntu. Fixes bug #550015. 2008-08-30 14:15:03 +0000 Tim-Philipp Müller Make stuff compile with GST_DISABLE_GST_DEBUG. Original commit message from CVS: * ext/raw1394/gsthdv1394src.c: (gst_hdv1394src_create): * gst/alpha/gstalpha.c: (gst_alpha_get_unit_size): * gst/audiofx/audiocheblimit.c: (generate_coefficients): * gst/avi/gstavidemux.c: (gst_avi_demux_src_convert): * gst/matroska/ebml-read.c: (gst_ebml_read_element_id), (gst_ebml_read_element_length): * gst/matroska/matroska-demux.c: (gst_matroska_demux_check_subtitle_buffer): Make stuff compile with GST_DISABLE_GST_DEBUG. 2008-08-29 00:28:55 +0000 Michael Smith gst/law/: Ref caps before passing to gst_pad_template_new(), since that takes ownership. Original commit message from CVS: * gst/law/alaw.c: * gst/law/mulaw.c: Ref caps before passing to gst_pad_template_new(), since that takes ownership. 2008-08-28 10:09:16 +0000 Mersad Jelacic gst/multipart/: Convert audio/x-adpcm to and from the audio/G726-X in the muxer and demuxer. Fixes #549551. Original commit message from CVS: Patch by: Mersad Jelacic * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: (gst_multipart_mux_get_mime): Convert audio/x-adpcm to and from the audio/G726-X in the muxer and demuxer. Fixes #549551. 2008-08-27 16:12:39 +0000 Edward Hervey sys/osxaudio/: Fix the build on macosx. Original commit message from CVS: * sys/osxaudio/gstosxaudiosink.c: (gst_osx_audio_sink_select_device): * sys/osxaudio/gstosxaudiosrc.c: (gst_osx_audio_src_create_ringbuffer), (gst_osx_audio_src_select_device): * sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_acquire): Fix the build on macosx. 2008-08-27 15:42:11 +0000 Tim-Philipp Müller gst/icydemux/gsticydemux.c: Small docs fix: in the example pipeline, we need to pass iradio-mode=true to the source, ... Original commit message from CVS: * gst/icydemux/gsticydemux.c: Small docs fix: in the example pipeline, we need to pass iradio-mode=true to the source, so the server actually sends an ICY stream. 2008-08-27 00:08:20 +0000 Michael Smith sys/osxaudio/gstosxaudio.c: Oops. Revert more completely. Original commit message from CVS: * sys/osxaudio/gstosxaudio.c: Oops. Revert more completely. 2008-08-26 23:57:05 +0000 Michael Smith sys/osxaudio/gstosxaudio.c: Revert accidental element rename from testing. Original commit message from CVS: * sys/osxaudio/gstosxaudio.c: Revert accidental element rename from testing. 2008-08-26 23:53:40 +0000 Jan Schmidt gst-plugins-good.doap: Pull in 0.10.10 doap entry from release branch Original commit message from CVS: * gst-plugins-good.doap: Pull in 0.10.10 doap entry from release branch 2008-08-26 23:05:57 +0000 Jan Schmidt configure.ac: Update version number to reflect 0.10.10 release from branch. Original commit message from CVS: * configure.ac: Update version number to reflect 0.10.10 release from branch. 2008-08-26 21:13:08 +0000 Michael Smith sys/osxaudio/: Rewrite caps setting and ring buffer initialisation. Original commit message from CVS: * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxaudio/gstosxaudiosrc.h: * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: Rewrite caps setting and ring buffer initialisation. Previously we never told CoreAudio what format we were going to send it, so it only worked due to luck, and not at all on some hardware. Now we explicitly advertise what formats the hardware supports, and then configure the selected one correctly. 2008-08-26 12:27:11 +0000 Stefan Kost sys/v4l2/: Fix memory leaks. Small code cleanups : No need for empty _init(). No need to memset instance structures. ... Original commit message from CVS: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: Fix memory leaks. Small code cleanups : No need for empty _init(). No need to memset instance structures. Some more FIXME's. 2008-08-26 08:11:26 +0000 Stefan Kost tests/icles/.cvsignore: Ignore more. Original commit message from CVS: * tests/icles/.cvsignore: Ignore more. 2008-08-26 08:00:57 +0000 Stefan Kost gst/: Ignore files. Original commit message from CVS: * gst/goom/.cvsignore: * gst/goom2k1/.cvsignore: Ignore files. 2008-08-26 07:51:42 +0000 Stefan Kost ext/cairo/gsttextoverlay.c: Fix compiler warning. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: Fix compiler warning. 2008-08-26 05:42:15 +0000 David Schleef ext/cairo/gsttextoverlay.c: Fix obvious memleak. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: Fix obvious memleak. 2008-08-25 14:15:43 +0000 Edward Hervey gst/matroska/: Add Real[Audio|Video] support to Matroska containers. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_send_event), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps): * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps), (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_finish): Add Real[Audio|Video] support to Matroska containers. It works fine for: * decoding real audio/video streams contained in mkv * 'transmuxing' real (.rm) files into .mkv files It will not work though for encoding real[audio/video] streams that don't contain the 'mdpr_data' extra data on the caps. The reason why this will not work is because I never intended to duplicate virtually all the 'mdpr' block creation into mkvmux. Fixes #536067 2008-08-25 09:48:06 +0000 Wim Taymans gst/law/: The encoder can't really renegotiate at the time they perform a pad-alloc so make the srcpads use fixed caps. Original commit message from CVS: * gst/law/alaw-encode.c: (gst_alaw_enc_init), (gst_alaw_enc_chain): * gst/law/mulaw-conversion.c: * gst/law/mulaw-encode.c: (gst_mulawenc_init), (gst_mulawenc_chain): The encoder can't really renegotiate at the time they perform a pad-alloc so make the srcpads use fixed caps. Check the buffer size after a pad-alloc because the returned size might not be right when the downstream element does not know the size of the new buffer (capsfilter). Fixes #549073. 2008-08-23 15:43:49 +0000 Filippo Argiolas sys/v4l2/gstv4l2tuner.c: v4l2src doesn't have a property named "norm" so don't try to notify about changes to that pr... Original commit message from CVS: Patch by: Filippo Argiolas * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_set_norm_and_notify): v4l2src doesn't have a property named "norm" so don't try to notify about changes to that property. The "norm" property and related code are commented out currently. Fixes bug #549090. 2008-08-23 15:33:49 +0000 Mike Ruprecht sys/v4l2/gstv4l2object.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged ... Original commit message from CVS: Patch by: Mike Ruprecht * sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices): Reprobe devices again instead of taking a cached list as new devices could've been plugged in. Fixes bug #549062. 2008-08-22 16:04:02 +0000 Sebastian Dröge gst/autodetect/Makefile.am: Don't link the autodetect plugin with GConf as it doesn't use GConf. Fixes bug #545463. Original commit message from CVS: * gst/autodetect/Makefile.am: Don't link the autodetect plugin with GConf as it doesn't use GConf. Fixes bug #545463. 2008-08-22 12:24:23 +0000 Sebastian Dröge gst/matroska/ebml-read.c: Change some GST_ELEMENT_ERRORs to GST_ERROR_OBJECT to make it possible to ignore errors and... Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_element_id), (gst_ebml_read_element_length), (gst_ebml_read_uint), (gst_ebml_read_sint), (gst_ebml_read_float), (gst_ebml_read_header): Change some GST_ELEMENT_ERRORs to GST_ERROR_OBJECT to make it possible to ignore errors and not post any ERROR messages on the bus. * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_contents): Ignore any errors and not just EOS when parsing the contents of a SeekHead. Errors here are usually caused by truncated files and playback of the file works fine. Fixes playback of the audio_only_chapter_seekbroken.mka file from the MPlayer samples archive. 2008-08-22 11:29:26 +0000 Zaheer Abbas Merali gst/multipart/: Conform to RFC2046. audio/basic is mulaw 8000Hz mono. Original commit message from CVS: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: Conform to RFC2046. audio/basic is mulaw 8000Hz mono. 2008-08-21 21:56:19 +0000 Ole André Vadla Ravnås * ChangeLog: * sys/directdraw/gstdirectdrawsink.c: sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc, gst_directdraw_sink_bufferpool_clear): Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc, gst_directdraw_sink_bufferpool_clear): Fix two more buffer ref leaks. 2008-08-21 15:28:09 +0000 Ole André Vadla Ravnås sys/directdraw/gstdirectdrawsink.c: Fix buffer ref leak. Original commit message from CVS: Patch by: Ole André Vadla Ravnås * sys/directdraw/gstdirectdrawsink.c: (gst_directdraw_sink_show_frame): Fix buffer ref leak. 2008-08-21 13:27:12 +0000 Sebastian Dröge gst/wavenc/gstwavenc.c: Revert the last commit. wavenc still supports width!=depth for 32 bit width. Thanks Tim. Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Revert the last commit. wavenc still supports width!=depth for 32 bit width. Thanks Tim. 2008-08-21 13:22:06 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: If the duration of a block is unknown only use the timestamp for the first lace and us... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_blockgroup_or_simpleblock): If the duration of a block is unknown only use the timestamp for the first lace and use GST_CLOCK_TIME_NONE as duration for the following laces. Otherwise every lace has the same timestamp which leads to various problems. Really fixes bug #548831. 2008-08-21 12:56:01 +0000 Sebastian Dröge gst/wavenc/gstwavenc.c: If we're not allowing width!=depth in wavenc we should also disable the code that was added t... Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): If we're not allowing width!=depth in wavenc we should also disable the code that was added to support width!=depth. 2008-08-21 12:52:47 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Don't calculate the default duration of a frame from the audio sampling rate. This onl... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream): Don't calculate the default duration of a frame from the audio sampling rate. This only works for raw audio if every frame contains a single sample and results in broken buffer durations for other formats if no specified default duration is given or the blocks have no duration. Fixes bug #548831. 2008-08-21 12:34:33 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks are used for tex... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_blockgroup_or_simpleblock): Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks are used for text/plain subtitles as a gap-filler in some files. 2008-08-21 12:12:00 +0000 Wim Taymans sys/v4l2/gstv4l2src.c: Add S910 and PWC formats with a low priority. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure), (gst_v4l2_get_caps_info): Add S910 and PWC formats with a low priority. * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank), (gst_v4l2src_probe_caps_for_format): Add more debugging. 2008-08-20 21:54:35 +0000 Tim-Philipp Müller ext/flac/gstflacenc.c: Fix compilation against older libflac versions. Original commit message from CVS: * ext/flac/gstflacenc.c: Fix compilation against older libflac versions. 2008-08-20 17:46:48 +0000 Sebastian Dröge ext/pulse/: Use GST_BOILERPLATE everywhere and fix coding style at some places. Original commit message from CVS: * ext/pulse/pulsemixer.c: (gst_pulsemixer_class_init), (gst_pulsemixer_set_property), (gst_pulsemixer_get_property): * ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb), (gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_timeout_event), (gst_pulsemixer_ctrl_set_volume): * ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_new): * ext/pulse/pulseprobe.c: (gst_pulseprobe_open): * ext/pulse/pulsesink.c: (gst_pulsesink_class_init), (gst_pulsesink_init), (gst_pulsesink_open), (gst_pulsesink_prepare), (gst_pulsesink_write), (gst_pulsesink_delay), (gst_pulsesink_reset): * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init), (gst_pulsesrc_init): Use GST_BOILERPLATE everywhere and fix coding style at some places. Fix a locking issue in pulsesink's prepare function. * ext/pulse/pulseutil.c: (gst_pulse_channel_map_to_gst): Check if the created channel layout is valid for GStreamer. 2008-08-20 17:42:21 +0000 Wim Taymans gst/rtsp/gstrtspgoogle.c: Things that can happen when your brain is in google mode trying to deal with their google r... Original commit message from CVS: * gst/rtsp/gstrtspgoogle.c: Things that can happen when your brain is in google mode trying to deal with their google rtsp server extensions and trying to type your google mail account. 2008-08-20 17:30:19 +0000 Wim Taymans gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi... Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send), (gst_rtsp_google_after_send), (gst_rtsp_google_get_transports), (_do_init), (gst_rtsp_google_base_init), (gst_rtsp_google_class_init), (gst_rtsp_google_init), (gst_rtsp_google_finalize), (gst_rtsp_google_change_state), (gst_rtsp_google_extension_init): * gst/rtsp/gstrtspgoogle.h: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fixes #546465. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send), (gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause): Make transport setup code a bit better using GString. Add some more debug. Check for closed connections before doing anything on them. 2008-08-20 17:17:55 +0000 Sebastian Dröge ext/pulse/: If downstream provides no channel layout and >2 channels should be used use the default layout that pulse... Original commit message from CVS: * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init), (gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate), (gst_pulsesrc_prepare): * ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map), (gst_pulse_channel_map_to_gst): * ext/pulse/pulseutil.h: If downstream provides no channel layout and >2 channels should be used use the default layout that pulseaudio chooses and also add this layout to the caps. Fixes bug #547258. 2008-08-20 11:51:38 +0000 Peter Kjellerstedt gst/udp/: Avoid leaking internally allocated file descriptors when setting custom file descriptors. Fixes #543101. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/udp/gstdynudpsink.c: (gst_dynudpsink_init), (gst_dynudpsink_finalize), (gst_dynudpsink_set_property), (gst_dynudpsink_init_send), (gst_dynudpsink_close): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init), (gst_multiudpsink_finalize), (gst_multiudpsink_set_property): * gst/udp/gstudpsrc.c: (gst_udpsrc_finalize), (gst_udpsrc_set_property): Avoid leaking internally allocated file descriptors when setting custom file descriptors. Fixes #543101. 2008-08-20 11:48:46 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Don't try to configure RTCP back to the server when the server did not give us a valid port nu... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink): Don't try to configure RTCP back to the server when the server did not give us a valid port number. 2008-08-20 10:59:52 +0000 Wim Taymans gst/videobox/gstvideobox.c: Use new basetransform method to renegotiate. Fixes #544956. Original commit message from CVS: * gst/videobox/gstvideobox.c: (gst_video_box_set_property): Use new basetransform method to renegotiate. Fixes #544956. * tests/icles/Makefile.am: * tests/icles/videobox-test.c: (make_pipeline), (main): Add videobox renegotiation example. 2008-08-19 21:03:22 +0000 David Schleef gst/wavenc/gstwavenc.c: Remove depth ranges and replace with sane values. Fixes #548530. Original commit message from CVS: * gst/wavenc/gstwavenc.c: Remove depth ranges and replace with sane values. Fixes #548530. 2008-08-18 15:05:32 +0000 Sebastian Dröge ext/pulse/: The bytes_per_sample and silence_sample fields of the GstRingBufferSpec are already filled with the corre... Original commit message from CVS: * ext/pulse/pulsesink.c: (gst_pulsesink_prepare): * ext/pulse/pulsesrc.c: (gst_pulsesrc_prepare): The bytes_per_sample and silence_sample fields of the GstRingBufferSpec are already filled with the correct values by gst_ring_buffer_parse_caps() so there's no need to set them again with wrong values. 2008-08-16 14:54:56 +0000 Edward Hervey gst/avi/gstavidemux.c: Some AVI 2.0 (ODML) files don't respect the 'specifications' completely and instead of using t... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_read_subindexes_push): Some AVI 2.0 (ODML) files don't respect the 'specifications' completely and instead of using the 'ix##' nomenclature, use '##ix'. They're still valid though, this fixes the duration and indexes for virtually all the ODML files I have. 2008-08-15 17:26:18 +0000 Olivier Crete gst/rtp/: Update the vorbis RTP pay/depay to RFC 5215. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps), (gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers): Update the vorbis RTP pay/depay to RFC 5215. Fixes #547842. 2008-08-14 22:07:02 +0000 David Schleef gst/qtdemux/qtdemux.c: Add 'hdv6' as a HDV format for 1080i/60 with 3:2 pulldown, i.e., 24p. Original commit message from CVS: * gst/qtdemux/qtdemux.c: Add 'hdv6' as a HDV format for 1080i/60 with 3:2 pulldown, i.e., 24p. 2008-08-14 12:47:09 +0000 Wim Taymans tests/check/elements/level.c: Fix compilation some more. Original commit message from CVS: * tests/check/elements/level.c: (GST_START_TEST): Fix compilation some more. 2008-08-14 11:44:59 +0000 Tim-Philipp Müller configure.ac: Require -base CVS for wavparse acid chunk parsing. Original commit message from CVS: * configure.ac:: Require -base CVS for wavparse acid chunk parsing. 2008-08-13 13:57:01 +0000 Sebastian Dröge ext/pulse/pulsesink.*: Add "device-name" property to pulsesink too and currently commented out and not working suppor... Original commit message from CVS: * ext/pulse/pulsesink.c: (gst_pulsesink_class_init), (gst_pulsesink_init), (gst_pulsesink_finalize), (gst_pulsesink_set_volume), (gst_pulsesink_get_volume), (gst_pulsesink_set_property), (gst_pulsesink_get_property), (gst_pulsesink_prepare), (gst_pulsesink_change_state): * ext/pulse/pulsesink.h: Add "device-name" property to pulsesink too and currently commented out and not working support for a "volume" property. 2008-08-13 13:17:15 +0000 Thijs Vermeir configure.ac: Remove more cdio stuff (moved to ugly) Original commit message from CVS: * configure.ac: Remove more cdio stuff (moved to ugly) 2008-08-13 12:37:26 +0000 Laszlo Pandy ext/pulse/pulsesrc.c: Add "device-name" property, which provides a human readable string for the audio device, to mak... Original commit message from CVS: Patch by: Laszlo Pandy * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init), (gst_pulsesrc_get_property): Add "device-name" property, which provides a human readable string for the audio device, to make it more consisten with other audio sources. Fixes bug #547519. 2008-08-13 12:34:13 +0000 Sebastian Dröge ext/pulse/: Improve debugging a bit by including the parent object in pulsemixerctrl and pulseprobe objects and using... Original commit message from CVS: * ext/pulse/pulsemixer.c: (gst_pulsemixer_change_state): * ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb), (gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_new), (gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_timeout_event): * ext/pulse/pulsemixerctrl.h: * ext/pulse/pulseprobe.c: (gst_pulseprobe_open), (gst_pulseprobe_enumerate), (gst_pulseprobe_new), (gst_pulseprobe_free), (gst_pulseprobe_needs_probe), (gst_pulseprobe_probe_property), (gst_pulseprobe_get_values): * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: (gst_pulsesink_init): * ext/pulse/pulsesrc.c: (gst_pulsesrc_init), (gst_pulsesrc_delay), (gst_pulsesrc_change_state): Improve debugging a bit by including the parent object in pulsemixerctrl and pulseprobe objects and using GST_WARNING_OBJECT instead of GST_WARNING. Use the parent GObject subclass instead of a random struct as GObject parameter for G_OBJECT_WARN_INVALID_PROPERTY_ID. This fixes a crash when probing for another property than "device". 2008-08-13 12:21:22 +0000 Laszlo Pandy ext/pulse/pulsemixer.c: Fix property probing after the device property is set by calling set_server when the server p... Original commit message from CVS: Patch by: Laszlo Pandy * ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property): Fix property probing after the device property is set by calling set_server when the server property changes. Fixes bug #547518. 2008-08-13 12:11:34 +0000 Laszlo Pandy ext/pulse/pulsemixer.c: Fix property probing after the device property is set by calling set_server when the server p... Original commit message from CVS: Patch by: Laszlo Pandy * ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property): Fix property probing after the device property is set by calling set_server when the server property changes. Fixes bug #547518. 2008-08-13 12:01:01 +0000 Laszlo Pandy ext/pulse/: Implement GstPropertyProbe interface on pulsesink for detecting sink devices and on pulsesrc for detectin... Original commit message from CVS: Patch by: Laszlo Pandy * ext/pulse/pulsesink.c: (gst_pulsesink_interface_supported), (gst_pulsesink_implements_interface_init), (gst_pulsesink_init_interfaces), (gst_pulsesink_init), (gst_pulsesink_finalize), (gst_pulsesink_set_property), (gst_pulsesink_get_type): * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported), (gst_pulsesrc_init_interfaces), (gst_pulsesrc_init), (gst_pulsesrc_finalize), (gst_pulsesrc_set_property): * ext/pulse/pulsesrc.h: Implement GstPropertyProbe interface on pulsesink for detecting sink devices and on pulsesrc for detecting source devices. Fixes bugs #547227 and #547217. 2008-08-13 09:17:20 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Don't terminate on fabs(in)>1.0. Init doubles as doubles. Original commit message from CVS: * gst/spectrum/gstspectrum.c: Don't terminate on fabs(in)>1.0. Init doubles as doubles. 2008-08-13 08:33:57 +0000 Edward Hervey sys/v4l2/gstv4l2src.c: Properly set the maximum latency value, in the same way it is done in v4lsrc. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_query): Properly set the maximum latency value, in the same way it is done in v4lsrc. * sys/v4l2/v4l2src_calls.c: Simplify fraction equality check, no need to use GValues for this. 2008-08-12 12:04:24 +0000 Edward Hervey sys/v4l2/gstv4l2src.c: Add warning messages stating exactly why the latency query failed. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_query): Add warning messages stating exactly why the latency query failed. * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture): In some cases, the negotiated framerate might be the default one which is already set internally. But we still need to mark it down in fps_n and fps_d so that the latency query can happen properly. 2008-08-12 11:28:47 +0000 Edward Hervey docs/plugins/inspect/plugin-1394.xml: Whoops, forgot one doc file for people who can't/don't build the raw1394 plugin. Original commit message from CVS: * docs/plugins/inspect/plugin-1394.xml: Whoops, forgot one doc file for people who can't/don't build the raw1394 plugin. 2008-08-12 09:22:29 +0000 Jan Schmidt Pull changes from 0.10.9.2 pre-release branch moving the libcdio Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-cdio.xml: * ext/Makefile.am: * ext/cdio/Makefile.am: * ext/cdio/gstcdio.c: * ext/cdio/gstcdio.h: * ext/cdio/gstcdiocddasrc.c: * ext/cdio/gstcdiocddasrc.h: Pull changes from 0.10.9.2 pre-release branch moving the libcdio CDDA source to -ugly. * po/LINGUAS: * po/POTFILES.in: * po/id.po: Pull in new translation from 0.10.9.2 release branch. 2008-08-11 15:05:13 +0000 Edward Hervey docs/plugins/: Integrate documentation for new hdv1394src element. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: Integrate documentation for new hdv1394src element. 2008-08-11 14:36:13 +0000 Edward Hervey ext/raw1394/: mpeg2-ts (HDV) variant of firewire capture element. Original commit message from CVS: * ext/raw1394/Makefile.am: * ext/raw1394/gst1394.c: (plugin_init): * ext/raw1394/gsthdv1394src.c: (_do_init), (gst_hdv1394src_base_init), (gst_hdv1394src_class_init), (gst_hdv1394src_init), (gst_hdv1394src_dispose), (gst_hdv1394src_set_property), (gst_hdv1394src_get_property), (gst_hdv1394src_from_raw1394handle), (gst_hdv1394src_iec61883_receive), (gst_hdv1394src_bus_reset), (gst_hdv1394src_create), (gst_hdv1394src_discover_avc_node), (gst_hdv1394src_start), (gst_hdv1394src_stop), (gst_hdv1394src_unlock), (gst_hdv1394src_update_device_name), (gst_hdv1394src_uri_get_type), (gst_hdv1394src_uri_get_protocols), (gst_hdv1394src_uri_get_uri), (gst_hdv1394src_uri_set_uri), (gst_hdv1394src_uri_handler_init): * ext/raw1394/gsthdv1394src.h: mpeg2-ts (HDV) variant of firewire capture element. Fixes #350830 2008-08-11 10:53:06 +0000 Edward Hervey gst/level/gstlevel.c: Fix compilation (also known as the classic 'fix code that someone committed without compiling i... Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_message_new): Fix compilation (also known as the classic 'fix code that someone committed without compiling it first'). 2008-08-10 19:40:27 +0000 Stefan Kost tests/check/elements/level.c: Add a test for level in stereo mode. Original commit message from CVS: * tests/check/elements/level.c: Add a test for level in stereo mode. 2008-08-10 19:35:05 +0000 Stefan Kost tests/examples/spectrum/: Demo how to draw analyzer results synced to the clock. Original commit message from CVS: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: Demo how to draw analyzer results synced to the clock. 2008-08-10 15:52:42 +0000 Stefan Kost gst/level/gstlevel.c: Little renaming (l -> level). Original commit message from CVS: * gst/level/gstlevel.c: Little renaming (l -> level). * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: Also send full timestamp/duration details here. 2008-08-10 11:32:03 +0000 Stefan Kost gst/level/gstlevel.*: Send same timestamp/duration details as videoanalysis. This gives applications better chance to... Original commit message from CVS: * gst/level/gstlevel.c: * gst/level/gstlevel.h: Send same timestamp/duration details as videoanalysis. This gives applications better chance to sync analysis results with playback. 2008-08-09 14:02:27 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: We need to drop one additional buffer for FLAC as the fLaC marker and STREAMINFO block a... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_handle_sink_event), (flac_streamheader_to_codecdata): We need to drop one additional buffer for FLAC as the fLaC marker and STREAMINFO block are merged into one buffer in the caps. Also don't pretend to support NEWSEGMENT events, otherwise we will most probably write some invalid data. 2008-08-09 13:48:22 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Add support for muxing FLAC into Matroska containers. Original commit message from CVS: * gst/matroska/matroska-mux.c: (flac_streamheader_to_codecdata), (gst_matroska_mux_audio_pad_setcaps): Add support for muxing FLAC into Matroska containers. Fixes bug #311586. 2008-08-09 08:58:26 +0000 Sebastian Dröge ext/flac/gstflacenc.c: Actually provide the variables required for the format string. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_check_discont): Actually provide the variables required for the format string. 2008-08-08 16:20:26 +0000 Sebastian Dröge gst/matroska/matroska-demux.*: Close the current segment if we're doing a non-flushing seek and send the close-segmen... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_element_send_event), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop): * gst/matroska/matroska-demux.h: Close the current segment if we're doing a non-flushing seek and send the close-segment and the new segment of the seek from the streaming thread. 2008-08-08 15:20:24 +0000 Sebastian Dröge ext/flac/gstflacenc.*: Handle non-zero start timestamps correctly, mark header packets as Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_write_callback), (gst_flac_enc_check_discont), (gst_flac_enc_chain), (gst_flac_enc_change_state): * ext/flac/gstflacenc.h: Handle non-zero start timestamps correctly, mark header packets as IN_CAPS and print a warning and suggest using audiorate if stream discontinuities are detected. When FLAC supports flushing the encoder somehow this should be done for discontinuities instead. Remove some unused variables from the instance struct. 2008-08-07 17:14:39 +0000 Christian Schaller * gst-plugins-good.spec.in: add pulseaudio to plugins list in spec file Original commit message from CVS: add pulseaudio to plugins list in spec file 2008-08-07 16:14:42 +0000 Frederic Crozat Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822). Original commit message from CVS: Patch by: Frederic Crozat * ext/dvdread/dvdreadsrc.c: (plugin_init): * ext/lame/gstlame.c: (plugin_init): * gst/asfdemux/gstasf.c: (plugin_init): Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822). 2008-08-07 16:13:41 +0000 Sebastian Dröge ext/flac/gstflacenc.c: If seeking failed return the appropiate return value to FLAC. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback): If seeking failed return the appropiate return value to FLAC. Otherwise it thinks seeking was successfull and tries to rewrite parts of the headers which then get appended to the output. 2008-08-07 16:11:00 +0000 Frederic Crozat Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822). Original commit message from CVS: Patch by: Frederic Crozat * ext/esd/gstesd.c: (plugin_init): * ext/flac/gstflac.c: (plugin_init): * ext/shout2/gstshout2.c: (plugin_init): * ext/wavpack/gstwavpack.c: (plugin_init): * sys/oss/gstossaudio.c: (plugin_init): * sys/v4l2/gstv4l2.c: (plugin_init): Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822). 2008-08-07 14:40:13 +0000 Sebastian Dröge ext/flac/gstflacdec.c: Add FIXME for 0.11 to simply output everything with width=32 as given by FLAC and let audiocon... Original commit message from CVS: * ext/flac/gstflacdec.c: Add FIXME for 0.11 to simply output everything with width=32 as given by FLAC and let audioconvert handle the conversions instead of doing them in flacdec. 2008-08-07 10:22:32 +0000 Jan Schmidt sys/v4l2/v4l2src_calls.c: When outputting a pad template range for the size, include a framerate range too, to avoid ... Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format): When outputting a pad template range for the size, include a framerate range too, to avoid 'not a real subset of template caps' errors. 2008-08-06 15:34:55 +0000 Jonathan Matthew ext/flac/: Port flactag to 0.10, add documentation for it and clean it up a bit. Original commit message from CVS: Based on a patch by: Jonathan Matthew * ext/flac/Makefile.am: * ext/flac/gstflac.c: (plugin_init): * ext/flac/gstflactag.c: (gst_flac_tag_setup_interfaces), (gst_flac_tag_base_init), (gst_flac_tag_class_init), (gst_flac_tag_dispose), (gst_flac_tag_init), (gst_flac_tag_sink_setcaps), (gst_flac_tag_chain), (gst_flac_tag_change_state): * ext/flac/gstflactag.h: Port flactag to 0.10, add documentation for it and clean it up a bit. Fixes bug #413841. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-flac.xml: * ext/flac/gstflacdec.c: (gst_flac_dec_base_init): * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.c: (gst_flac_enc_base_init): * ext/flac/gstflacenc.h: Add flactag and flacenc to the documentation and mark the private parts of the flacdec instance structure as private. Also use gst_element_class_set_details_simple() in flacdec and flacenc. 2008-08-06 13:12:07 +0000 Stefan Kost gst/qtdemux/qtdemux.c: Use audio/x-qdm for caps. Collect some info - mplayer has a decoder for it but ffmpeg does not. Original commit message from CVS: * gst/qtdemux/qtdemux.c: Use audio/x-qdm for caps. Collect some info - mplayer has a decoder for it but ffmpeg does not. 2008-08-05 15:05:44 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Handle the list chunk and use gst_riff_parse_info() to parse the info sub-chunk. Original commit message from CVS: * gst/wavparse/gstwavparse.c: Handle the list chunk and use gst_riff_parse_info() to parse the info sub-chunk. 2008-08-05 14:22:12 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Handle the acid chunk and send tempo as part of tags. Other fields are interesting too, b... Original commit message from CVS: * gst/wavparse/gstwavparse.c: Handle the acid chunk and send tempo as part of tags. Other fields are interesting too, but need more tag-definitions. Fixes #545433. 2008-08-05 14:16:32 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Refactor wavparse. Call _reset() from dispose() and move old code from dispose into reset... Original commit message from CVS: * gst/wavparse/gstwavparse.c: Refactor wavparse. Call _reset() from dispose() and move old code from dispose into reset. This way we don't leak taglists when we abort parsing. Fix some comments. Move code for skipping a chunk into extra function. Replace chunk sizes with a const to ease readability. 2008-08-05 13:57:57 +0000 Aurelien Grimaud gst/rtsp/gstrtspsrc.c: Improve udp port setup. Fixes #545710. Original commit message from CVS: Patch by: Aurelien Grimaud * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_alloc_udp_ports): Improve udp port setup. Fixes #545710. 2008-08-05 13:54:18 +0000 Wim Taymans gst/rtp/: Add MP1S depayloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_base_init), (gst_rtp_mp1s_depay_class_init), (gst_rtp_mp1s_depay_init), (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process), (gst_rtp_mp1s_depay_set_property), (gst_rtp_mp1s_depay_get_property), (gst_rtp_mp1s_depay_change_state), (gst_rtp_mp1s_depay_plugin_init): * gst/rtp/gstrtpmp1sdepay.h: Add MP1S depayloader. * gst/rtsp/URLS: Some more sample rtsp streams. 2008-08-05 08:43:45 +0000 Wim Taymans gst/rtsp/URLS: Add another URL. Original commit message from CVS: * gst/rtsp/URLS: Add another URL. * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags): * tests/check/elements/rglimiter.c: (GST_START_TEST): Add some more debug info. 2008-08-04 09:16:40 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Provide cbSize field for audio extra_data size, and take care to pad extra_data. Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header): Provide cbSize field for audio extra_data size, and take care to pad extra_data. 2008-08-04 07:23:07 +0000 Stefan Kost gst/qtdemux/qtdemux.c: Return the result of gst_pad_{start,stop}_task instead of hard-coded Original commit message from CVS: * gst/qtdemux/qtdemux.c: Return the result of gst_pad_{start,stop}_task instead of hard-coded TRUE. 2008-08-04 07:17:38 +0000 Stefan Kost gst/qtdemux/: Add keyword tag support. Fixes #520694 for qtdemux. Original commit message from CVS: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: Add keyword tag support. Fixes #520694 for qtdemux. 2008-08-04 07:05:33 +0000 Stefan Kost gst/qtdemux/qtdemux.c: Add support for tmpo tag (BPM). Original commit message from CVS: * gst/qtdemux/qtdemux.c: Add support for tmpo tag (BPM). 2008-08-03 12:23:49 +0000 Sebastian Dröge ext/flac/gstflacenc.c: Set an estimate for the total number of samples that will be encoded if possible to help decod... Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_query_peer_total_samples), (gst_flac_enc_sink_setcaps), (gst_flac_enc_write_callback): Set an estimate for the total number of samples that will be encoded if possible to help decoders if the streaminfo can't be rewritten later (like when muxing into Ogg containers). Add a warning if we get header packets after data packets as those will get lost when muxing into Ogg, i.e. rewriting the headers doesn't work. 2008-08-03 11:38:22 +0000 Sebastian Dröge ext/flac/gstflacdec.c: Support decoding of all depths between 4 and 32 bits and read the depth from the streaminfo he... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback), (gst_flac_dec_write): Support decoding of all depths between 4 and 32 bits and read the depth from the streaminfo header if needed. Also support all sampling rates between 1 and 655350 Hz. * ext/flac/gstflacenc.c: (gst_flac_enc_caps_append_structure_with_widths), (gst_flac_enc_sink_getcaps), (gst_flac_enc_sink_setcaps), (gst_flac_enc_chain): * ext/flac/gstflacenc.h: Support encoding in all bit depths supported by the streamable subformat (i.e. 8, 12, 16, 20 and 24 bits) and all sampling rates between 1 Hz and 655350 Hz. 2008-08-03 09:23:14 +0000 Sebastian Dröge ext/flac/gstflacenc.c: Support encoding of up to 8 channels. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_init), (gst_flac_enc_sink_getcaps): Support encoding of up to 8 channels. 2008-08-02 21:39:01 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.*: Fix seeking race condition in #540300 Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: Fix seeking race condition in #540300 Patch By: Wouter Cloetens 2008-08-02 18:35:21 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: When receiving a SEEK event on a specific pad first search for a seek table entry for ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek), (gst_matroska_demux_element_send_event), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_handle_src_event): When receiving a SEEK event on a specific pad first search for a seek table entry for the stream of the pad and then fall back to an entry for a different stream. 2008-08-02 18:20:44 +0000 Sebastian Dröge Build depend on core CVS for the attachment tag. Original commit message from CVS: * configure.ac: * gst/matroska/matroska-ids.c: (gst_matroska_register_tags): * gst/matroska/matroska-ids.h: Build depend on core CVS for the attachment tag. 2008-08-02 18:18:05 +0000 Sebastian Dröge Decode the codec private data and following ContentEncoding if necessary. Original commit message from CVS: * configure.ac: * gst/matroska/Makefile.am: * gst/matroska/lzo.c: (get_byte), (get_len), (copy), (copy_backptr), (lzo1x_decode), (main): * gst/matroska/lzo.h: * gst/matroska/matroska-demux.c: (gst_matroska_demux_read_track_encoding), (gst_matroska_decompress_data), (gst_matroska_decode_data), (gst_matroska_decode_buffer), (gst_matroska_decode_content_encodings), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_add_stream), (gst_matroska_demux_parse_blockgroup_or_simpleblock): * gst/matroska/matroska-ids.h: Decode the codec private data and following ContentEncoding if necessary. Support bzip2, lzo and header stripped compression. For lzo use the ffmpeg lzo implementation as liblzo is GPL licensed. Fix zlib decompression. 2008-08-02 18:11:32 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Fix muxing of MP3/MP2 with different MPEG versions by calculating the duration of a fram... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_setcaps): Fix muxing of MP3/MP2 with different MPEG versions by calculating the duration of a frame with the new mpegaudioversion caps field. 2008-08-02 18:06:20 +0000 Sebastian Dröge gst/matroska/matroska-demux.*: Allow an infinite number of stream inside Matroska containers and use a GPtrArray for ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_finalize), (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_demux_combine_flows), (gst_matroska_demux_reset), (gst_matroska_demux_stream_from_num), (gst_matroska_demux_tracknumber_unique), (gst_matroska_demux_add_stream), (gst_matroska_demux_send_event), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_sync_streams), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_loop): * gst/matroska/matroska-demux.h: Allow an infinite number of stream inside Matroska containers and use a GPtrArray for storing them instead of allowing "only" 127 streams. 2008-08-02 18:01:36 +0000 Sebastian Dröge gst/matroska/: Fix indention everywhere. A broken indent version has added newlines after every single declaration so... Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_class_init), (gst_ebml_read_change_state), (gst_ebml_read_element_level_up), (gst_ebml_read_peek_bytes), (gst_ebml_read_element_id), (gst_ebml_read_element_length), (gst_ebml_peek_id), (gst_ebml_read_get_length), (gst_ebml_read_skip), (gst_ebml_read_buffer), (gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint), (_ext2dbl), (gst_ebml_read_float), (gst_ebml_read_ascii), (gst_ebml_read_date), (gst_ebml_read_master), (gst_ebml_read_binary), (gst_ebml_read_header): * gst/matroska/ebml-write.c: (gst_ebml_write_element_id), (gst_ebml_write_element_size), (gst_ebml_write_uint), (gst_ebml_write_sint), (gst_ebml_write_ascii), (gst_ebml_write_master_start), (gst_ebml_write_master_finish), (gst_ebml_replace_uint): * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_read_track_encoding), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_add_stream), (gst_matroskademux_do_index_seek), (gst_matroska_demux_send_event), (gst_matroska_demux_element_send_event), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_handle_src_event), (gst_matroska_demux_init_stream), (gst_matroska_demux_parse_tracks), (gst_matroska_demux_parse_index_cuetrack), (gst_matroska_demux_parse_index_pointentry), (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_metadata_id_simple_tag), (gst_matroska_demux_parse_metadata_id_tag), (gst_matroska_demux_parse_metadata), (gst_matroska_demux_parse_attached_file), (gst_matroska_demux_parse_attachments), (gst_matroska_demux_parse_chapters), (gst_matroska_ebmlnum_uint), (gst_matroska_ebmlnum_sint), (gst_matroska_demux_push_hdr_buf), (gst_matroska_demux_push_flac_codec_priv_data), (gst_matroska_demux_push_xiph_codec_priv_data), (gst_matroska_demux_push_dvd_clut_change_event), (gst_matroska_demux_add_mpeg_seq_header), (gst_matroska_demux_add_wvpk_header), (gst_matroska_demux_check_subtitle_buffer), (gst_matroska_decode_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_parse_cluster), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_parse_contents), (gst_matroska_demux_loop_stream_parse_id), (gst_matroska_demux_loop_stream), (gst_matroska_demux_loop), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps), (gst_matroska_demux_subtitle_caps), (gst_matroska_demux_change_state): * gst/matroska/matroska-ids.c: * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init), (gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event), (gst_matroska_mux_video_pad_setcaps), (xiph3_streamheader_to_codecdata), (vorbis_streamheader_to_codecdata), (theora_streamheader_to_codecdata), (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad), (gst_matroska_mux_track_header), (gst_matroska_mux_start), (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish), (gst_matroska_mux_best_pad), (gst_matroska_mux_write_data), (gst_matroska_mux_collected), (gst_matroska_mux_change_state): Fix indention everywhere. A broken indent version has added newlines after every single declaration some time ago. 2008-08-02 17:59:05 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: If no Tracks are found error out instead of trying it again until the end of time. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_loop_stream_parse_id): If no Tracks are found error out instead of trying it again until the end of time. 2008-08-02 17:57:31 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Fix demuxing of raw integer audio. The samples are unsigned only for 8 bit and signed ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps): Fix demuxing of raw integer audio. The samples are unsigned only for 8 bit and signed otherwise, not the other way around. 2008-08-02 17:54:04 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Add more raw YUV formats to the list of supported formats. Original commit message from CVS: * gst/matroska/matroska-mux.c: Add more raw YUV formats to the list of supported formats. 2008-08-02 17:52:16 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Add support for muxing raw float audio now that the spec defines the endianness and add ... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_setcaps): Add support for muxing raw float audio now that the spec defines the endianness and add support for muxing raw integer audio with 24 and 32 bits. Allow muxing of more than 8 audio channels. 2008-08-02 17:47:32 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Add locking to the global array of used track UIDs to prevent random crashes if more tha... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_create_uid), (gst_matroska_mux_reset), (gst_matroska_mux_start): Add locking to the global array of used track UIDs to prevent random crashes if more than a single matrosmux instance is used. Use 64 bit values for the track UIDs. Use the global GRandom of GLib instead of creating our own one for the few random numbers we need every single time. 2008-08-02 17:18:47 +0000 Sebastian Dröge ext/flac/gstflacdec.c: Always post the audio-codec tag, not only if other tags are present. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_setup_seekable_decoder), (gst_flac_dec_setup_stream_decoder), (gst_flac_dec_update_metadata): Always post the audio-codec tag, not only if other tags are present. 2008-08-01 23:26:50 +0000 Jan Schmidt configure.ac: Back to development -> 0.10.9.1 Original commit message from CVS: * configure.ac: Back to development -> 0.10.9.1 2008-08-01 15:58:47 +0000 Christian Schaller * gst-plugins-good.spec.in: add missing gstreamer plugins to spec file Original commit message from CVS: add missing gstreamer plugins to spec file === release 0.10.9 === 2008-07-31 22:10:17 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst-plugins-good.doap: * win32/common/config.h: Release 0.10.9 Original commit message from CVS: Release 0.10.9 2008-07-31 21:50:44 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/it.po: * po/ja.po: * po/lt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files Original commit message from CVS: Update .po files 2008-07-31 21:26:48 +0000 Jan Schmidt ext/soup/gstsouphttpsrc.c: Don't throw an error when soup completes a msg with status 'cancelled', as that indicates ... Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: Don't throw an error when soup completes a msg with status 'cancelled', as that indicates we cancelled a request while shutting down or seeking, and it's not an error. Fixes: #540300 again. 2008-07-31 14:24:27 +0000 Sebastian Dröge ext/lame/gstlame.c: Use the default for the strict-iso property too. Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_get_default_settings): Use the default for the strict-iso property too. Allow a bitrate setting of 0, which lets lame choose the default value and which makes it possible to set the compression-ratio property. 2008-07-29 16:57:16 +0000 Sebastian Dröge ext/lame/gstlame.*: Get the defaults settings of LAME in the plugin initialization function and return FALSE here if ... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init), (gst_lame_chain), (gst_lame_get_default_settings), (plugin_init): * ext/lame/gstlame.h: Get the defaults settings of LAME in the plugin initialization function and return FALSE here if something goes wrong. This removes the hacky failing instance init function. Use LAMEs default value for all settings instead of overwriting some of them. Overwriting some of them gives unexpected results if one only sets a preset. Fixes bug #498004. 2008-07-28 20:17:46 +0000 Jan Schmidt configure.ac: 0.10.8.4 pre-release Original commit message from CVS: * configure.ac: 0.10.8.4 pre-release 2008-07-27 15:56:27 +0000 Sebastian Dröge ext/lame/gstlame.c: Use LAME's default for the min/max/mean VBR bitrate. Setting our own defaults will restrict the b... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_init): Use LAME's default for the min/max/mean VBR bitrate. Setting our own defaults will restrict the bitrate when using the presets in a bad way. Fixes bug #498004. 2008-07-27 11:01:12 +0000 Sebastian Dröge Put the MPEG audio version into the caps as "mpegaudioversion". Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_setcaps): * gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_type_frame_length_from_header), (mp3_caps_create), (gst_mp3parse_chain): Put the MPEG audio version into the caps as "mpegaudioversion". This is different from "mpegversion". 2008-07-25 14:50:03 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Fix segment-stop regression. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment): Fix segment-stop regression. Add documentation regarding segments in quicktime files by Wim Taymans. Fixes #544509 2008-07-24 23:55:58 +0000 Jan Schmidt configure.ac: 0.10.8.3 pre-release Original commit message from CVS: * configure.ac: 0.10.8.3 pre-release * po/LINGUAS: * po/pt_BR.po: Add pt_BR translation 2008-07-23 22:01:20 +0000 Michael Smith gst/goom/: Fix build with MSVC: include glib.h to define inline appropriately, use header guards where needed. Original commit message from CVS: * gst/goom/convolve_fx.c: * gst/goom/filters.c: * gst/goom/goom_config.h: * gst/goom/goom_core.c: * gst/goom/goom_tools.h: Fix build with MSVC: include glib.h to define inline appropriately, use header guards where needed. * gst/udp/gstudpnetutils.c: * gst/udp/gstudpsrc.c: Fix build with MSVC: use WSA* constants/functions where appropriate, use g_snprintf rather than snprintf. Fixes #544433. 2008-07-22 18:25:08 +0000 Sebastian Dröge ext/lame/gstlame.*: Fix build with lame >= 3.97. The padding type and cwlimit settings are deprecated now and the fun... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init), (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup): * ext/lame/gstlame.h: Fix build with lame >= 3.97. The padding type and cwlimit settings are deprecated now and the function declarations are hidden in the headers so deprecate the GObject properties for them and remove them in 0.11. Fixes bug #544039. 2008-07-22 06:32:03 +0000 Stefan Kost gst/debug/gsttaginject.*: Sent tags in _transform_ip() instead of _start(). Fixes #543404 partially. Original commit message from CVS: * gst/debug/gsttaginject.c: * gst/debug/gsttaginject.h: Sent tags in _transform_ip() instead of _start(). Fixes #543404 partially. 2008-07-19 14:12:39 +0000 Jan Schmidt configure.ac: 0.10.8.2 pre-release Original commit message from CVS: * configure.ac: 0.10.8.2 pre-release 2008-07-19 13:50:53 +0000 Jan Schmidt ext/Makefile.am: Finish hooking up pulseaudio plugin to the build. Original commit message from CVS: * ext/Makefile.am: Finish hooking up pulseaudio plugin to the build. * ext/pulse/pulsemixerctrl.c: Fix compilation error. 2008-07-19 13:23:29 +0000 Jan Schmidt po/: Add new lithunian translation, and add french to the LINGUAS file. Original commit message from CVS: * po/LINGUAS: * po/lt.po: Add new lithunian translation, and add french to the LINGUAS file. 2008-07-19 13:08:42 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.c: Fix Soup HTTP source seeking. Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: Fix Soup HTTP source seeking. Patch By: Wouter Cloetens Fixes: #540300 * tests/check/elements/.cvsignore: Ignore new check programs. 2008-07-19 01:01:13 +0000 Jan Schmidt Move replaygain and interleave plugins from -bad. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-replaygain.xml: * tests/check/Makefile.am: Move replaygain and interleave plugins from -bad. Fixes: #543406 Fixes: #536228 2008-07-18 20:03:07 +0000 Mark Nauwelaerts gst/qtdemux/qtdemux.c: Revert ISO base media spec based pixel-aspect-ratio calculation. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse_trak): Revert ISO base media spec based pixel-aspect-ratio calculation. Fixes #543300. 2008-07-17 16:42:53 +0000 Edward Hervey sys/osxvideo/osxvideosink.m: Fix minor build issues on macosx. Original commit message from CVS: * sys/osxvideo/osxvideosink.m: Fix minor build issues on macosx. Fixes #543054 2008-07-17 14:40:51 +0000 Tim-Philipp Müller Only use -Wno-attributes (which is there to work around a bug in the taglib 1.5 headers) if the c++ compiler actually... Original commit message from CVS: * configure.ac:: * ext/taglib/Makefile.am:: Only use -Wno-attributes (which is there to work around a bug in the taglib 1.5 headers) if the c++ compiler actually supports it (#543255). 2008-07-17 13:54:38 +0000 Benoit Fouet sys/v4l2/gstv4l2src.c: Avoid compiler warning by initialising variable to NULL (#543259). Original commit message from CVS: Patch by: Benoit Fouet * sys/v4l2/gstv4l2src.c: (gst_v4l2src_negotiate): Avoid compiler warning by initialising variable to NULL (#543259). 2008-07-14 17:17:47 +0000 Sebastian Dröge gst/debug/gsttaginject.c: Don't pass NULL taglists to gst_tag_list_is_empty(). Original commit message from CVS: * gst/debug/gsttaginject.c: (gst_tag_inject_start): Don't pass NULL taglists to gst_tag_list_is_empty(). 2008-07-14 17:15:42 +0000 Sebastian Dröge tests/check/elements/: Don't use declarations after statements. Original commit message from CVS: * tests/check/elements/cmmldec.c: (GST_START_TEST): * tests/check/elements/rtp-payloading.c: (rtp_pipeline_create), (rtp_pipeline_run): * tests/check/elements/souphttpsrc.c: (souphttpsrc_suite): Don't use declarations after statements. 2008-07-14 16:28:25 +0000 Mark Nauwelaerts ext/jpeg/gstjpegdec.c: Align documentation with reality. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: Align documentation with reality. 2008-07-14 13:11:14 +0000 Sebastian Dröge gst/udp/gstudpnetutils.c: EAI_ADDRFAMILY was obsoleted in BSD at some point. Define it to the old value (1) if it's n... Original commit message from CVS: * gst/udp/gstudpnetutils.c: EAI_ADDRFAMILY was obsoleted in BSD at some point. Define it to the old value (1) if it's not defined which should not cause any problems as we're using it internal only anyway. 2008-07-14 13:02:48 +0000 Alessandro Decina gst/avi/gstavidemux.c: Fix build of avidemux on big endian architectures. Original commit message from CVS: Patch by: Alessandro Decina * gst/avi/gstavidemux.c: (gst_avi_demux_riff_parse_vprp): Fix build of avidemux on big endian architectures. 2008-07-10 20:47:56 +0000 Thiago Sousa Santos gst/qtdemux/qtdemux.c: Correctly distinguish 8bit vs 16bit raw audio. Fixes #542410. Original commit message from CVS: Patch by: Thiago Sousa Santos * gst/qtdemux/qtdemux.c: (qtdemux_audio_caps): Correctly distinguish 8bit vs 16bit raw audio. Fixes #542410. 2008-07-10 18:51:11 +0000 Stefan Kost Document one more. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-stereo.xml: * gst/stereo/gststereo.c: Document one more. 2008-07-08 21:05:18 +0000 Mark Nauwelaerts gst/qtdemux/qtdemux.c: Set pixel-aspect-ratio in caps using display width and height provided in track. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse_trak): Set pixel-aspect-ratio in caps using display width and height provided in track. 2008-07-08 13:59:51 +0000 Sebastian Dröge configure.ac: Don't include ERROR_CFLAGS in GST_CXXFLAGS as it might include flags that are invalid for C++. Fixes bu... Original commit message from CVS: * configure.ac: Don't include ERROR_CFLAGS in GST_CXXFLAGS as it might include flags that are invalid for C++. Fixes bug #516509. 2008-07-08 12:51:34 +0000 Sebastian Dröge Don't use declarations after statements and variable length arrays. Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri): * ext/speex/gstspeexenc.c: (gst_speex_enc_sink_getcaps): * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config): * sys/v4l2/gstv4l2src.c: (gst_v4l2src_fixate): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format): * tests/examples/equalizer/demo.c: (message_handler): * tests/examples/spectrum/demo-audiotest.c: (message_handler): * tests/examples/spectrum/demo-osssrc.c: (message_handler): Don't use declarations after statements and variable length arrays. 2008-07-07 21:28:58 +0000 Daniel Drake sys/v4l2/v4l2src_calls.c: Try progressive video if interlaced fails. Fixes bug #541956 and the usage of v4l2src on OLPC. Original commit message from CVS: Patch by: Daniel Drake * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture), (gst_v4l2src_get_nearest_size): Try progressive video if interlaced fails. Fixes bug #541956 and the usage of v4l2src on OLPC. 2008-07-07 15:34:12 +0000 Sebastian Dröge gst/rtp/gstrtpspeexdepay.*: Revert last change: Only the jitterbuffer is able to convert RTP to Original commit message from CVS: * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init), (gst_rtp_speex_depay_process): * gst/rtp/gstrtpspeexdepay.h: Revert last change: Only the jitterbuffer is able to convert RTP to Gstreamer timestamps and normal (de)payloaders should simply copy it. Reopens bug #541787. 2008-07-07 10:30:51 +0000 Stefan Kost gst/rtp/gstrtpvrawdepay.c: Include stdlib.h for atoi(). Original commit message from CVS: * gst/rtp/gstrtpvrawdepay.c: Include stdlib.h for atoi(). * gst/rtsp/gstrtspsrc.c: Use floating point math for latencies < 0 sec in log output. 2008-07-07 10:16:07 +0000 Tomasz Grobelny gst/rtp/gstrtpspeexdepay.*: Take timestamp from the RTP packet as a first step to fix problems with transmission over... Original commit message from CVS: Patch by: Tomasz Grobelny * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init), (gst_rtp_speex_depay_process): * gst/rtp/gstrtpspeexdepay.h: Take timestamp from the RTP packet as a first step to fix problems with transmission over RTP when the network is not reliable. Fixes bug #541787. 2008-07-05 19:01:28 +0000 Tero Saarni gst/udp/gstudpsrc.c: Fix parsing of udp:// URIs containing IPv6 addresses. Original commit message from CVS: Patch by: Tero Saarni * gst/udp/gstudpsrc.c: (gst_udpsrc_set_uri): Fix parsing of udp:// URIs containing IPv6 addresses. Fixes bug #541650. 2008-07-04 20:43:07 +0000 Mark Nauwelaerts ext/gdk_pixbuf/gstgdkpixbuf.c: Do not leak incoming buffers. Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain): Do not leak incoming buffers. 2008-07-03 19:27:53 +0000 Damien Lespiau configure.ac: Fix build of the RTP plugin with mingw32 by linking to ws2_32 for htons() and htonl(). Fixes bug #541412. Original commit message from CVS: Patch by: Damien Lespiau * configure.ac: Fix build of the RTP plugin with mingw32 by linking to ws2_32 for htons() and htonl(). Fixes bug #541412. 2008-07-02 09:51:16 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Handle position and duration query in DEFAULT format if the pad's track has a default ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init), (gst_matroska_demux_add_stream), (gst_matroska_demux_query), (gst_matroska_demux_element_query), (gst_matroska_demux_handle_src_query), (gst_matroska_demux_handle_seek_event): Handle position and duration query in DEFAULT format if the pad's track has a default frame duration set. Fix seeking now that the segment's duration doesn't contain the (possibly wrong or inaccurate) duration of the Matroska file. 2008-07-02 09:04:50 +0000 Sebastian Dröge gst/matroska/ebml-read.c: Use NAN constant instead of 0.0/0.0 if possible. NAN is defined in math.h except on MSVC wh... Original commit message from CVS: * gst/matroska/ebml-read.c: (_ext2dbl): Use NAN constant instead of 0.0/0.0 if possible. NAN is defined in math.h except on MSVC where it is defined in xmath.h. Fixes compilation with MSVC. 2008-07-02 08:57:04 +0000 Sebastian Dröge gst/matroska/matroska-demux.*: Don't set the segment duration to the duration from the Matroska header as this value ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_handle_src_query), (gst_matroska_demux_parse_info), (gst_matroska_demux_loop_stream_parse_id): * gst/matroska/matroska-demux.h: Don't set the segment duration to the duration from the Matroska header as this value could be wrong and is just informational. 2008-07-02 08:47:00 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: If no Tracks element is found until the first Cluster is found search it and error out... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_loop_stream_parse_id): If no Tracks element is found until the first Cluster is found search it and error out if none is found in the complete file. 2008-07-02 08:14:35 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Resync non-subtitle tracks too if a too large gap compared to other tracks is detected. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams): Resync non-subtitle tracks too if a too large gap compared to other tracks is detected. 2008-07-01 13:28:02 +0000 Wim Taymans gst/rtp/: Add raw video pay and depayloaders, see RFC4175. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_base_init), (gst_rtp_vraw_depay_class_init), (gst_rtp_vraw_depay_init), (gst_rtp_vraw_depay_setcaps), (gst_rtp_vraw_depay_process), (gst_rtp_vraw_depay_change_state), (gst_rtp_vraw_depay_plugin_init): * gst/rtp/gstrtpvrawdepay.h: * gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_get_type), (gst_rtp_vraw_pay_base_init), (gst_rtp_vraw_pay_class_init), (gst_rtp_vraw_pay_init), (gst_rtp_vraw_pay_finalize), (gst_rtp_vraw_pay_setcaps), (gst_rtp_vraw_pay_handle_buffer), (gst_rtp_vraw_pay_plugin_init): * gst/rtp/gstrtpvrawpay.h: Add raw video pay and depayloaders, see RFC4175. 2008-06-30 22:53:39 +0000 Jan Schmidt ext/libpng/gstpngdec.c: Don't return GST_FLOW_ERROR when buffer_alloc fails - return whatever it returned. Original commit message from CVS: * ext/libpng/gstpngdec.c: Don't return GST_FLOW_ERROR when buffer_alloc fails - return whatever it returned. 2008-06-29 19:52:51 +0000 Mark Nauwelaerts gst/avi/avi-ids.h: Add vprp chunk related structures. Original commit message from CVS: * gst/avi/avi-ids.h: Add vprp chunk related structures. * gst/avi/gstavidemux.c: (gst_avi_demux_riff_parse_vprp), (gst_avi_demux_parse_stream): Parse optional vprp chunk and add calculated pixel-aspect-ratio to caps. Fixes #539482. * gst/avi/gstavimux.h: * gst/avi/gstavimux.c: (gst_avi_mux_pad_reset), (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_riff_get_avi_header): Add a vprp chunk if non-trival pixel-aspect-ratio provided in caps. 2008-06-28 19:31:46 +0000 Mark Nauwelaerts tests/check/elements/avimux.c: Adjust avimux unit test according to increased streamheader size. Original commit message from CVS: * tests/check/elements/avimux.c: (check_avimux_pad): Adjust avimux unit test according to increased streamheader size. 2008-06-27 18:11:01 +0000 David Schleef gst/qtdemux/qtdemux.c: Add Dirac stream type Original commit message from CVS: * gst/qtdemux/qtdemux.c: Add Dirac stream type 2008-06-27 15:25:00 +0000 Mark Nauwelaerts gst/avi/gstavimux.*: Add 8 bytes to current streamheader to make for a complete one and to make more players happy. ... Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header): * gst/avi/gstavimux.h: Add 8 bytes to current streamheader to make for a complete one and to make more players happy. Fixes #519460. 2008-06-26 16:36:47 +0000 Tim-Philipp Müller sys/v4l2/v4l2_calls.c: Don't include unused gstv4l2xoverlay.h. Fixes build in case where X11 headers are not installed. Original commit message from CVS: * sys/v4l2/v4l2_calls.c:: Don't include unused gstv4l2xoverlay.h. Fixes build in case where X11 headers are not installed. 2008-06-26 10:07:46 +0000 Wim Taymans ext/dv/gstdv.c: Fix compilation. Original commit message from CVS: * ext/dv/gstdv.c: (plugin_init): Fix compilation. 2008-06-26 09:37:23 +0000 Edward Hervey ext/dv/gstdv.c: Marking rank of dvdec as GST_RANK_MARGINAL since it's the slowest Original commit message from CVS: * ext/dv/gstdv.c: (plugin_init): Marking rank of dvdec as GST_RANK_MARGINAL since it's the slowest DV decoder available. Fixes #532393 2008-06-25 08:12:18 +0000 Sebastian Dröge gst/udp/gstudpsrc.c: Call getsockname() after the call to bind() to get updated values for the port, etc. This fixes ... Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_start): Call getsockname() after the call to bind() to get updated values for the port, etc. This fixes the usage of udpsrc on anonymous binding and it's usage by rtspsrc. Fixes bugs #539372, #539548. Thanks to Aurelien Grimaud for pointing out the obvious fix. 2008-06-25 07:57:26 +0000 Sebastian Dröge tests/check/pipelines/wavpack.c: Remove workaround for a bug in identity that is fixed in 0.10.20. Original commit message from CVS: * tests/check/pipelines/wavpack.c: (bus_handler): Remove workaround for a bug in identity that is fixed in 0.10.20. 2008-06-25 06:36:58 +0000 Jason Donenfeld ext/soup/gstsouphttpsrc.c: Fix HTTP auth support with user/password passed via the URI. Original commit message from CVS: Patch by: Jason Donenfeld * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb): Fix HTTP auth support with user/password passed via the URI. Fixes bug #540067. 2008-06-24 15:42:33 +0000 Tim-Philipp Müller configure.ac: Depend on released versions of core and -base. Original commit message from CVS: * configure.ac: Depend on released versions of core and -base. 2008-06-23 16:13:40 +0000 Julien Moutte gst/matroska/matroska-demux.c: Fix buggy format strings in macros. (makes it build on OS X again...) Original commit message from CVS: 2008-06-23 Julien Moutte * gst/matroska/matroska-demux.c: (gst_matroska_demux_read_track_encoding), (gst_matroska_demux_parse_blockgroup_or_simpleblock): Fix buggy format strings in macros. (makes it build on OS X again...) 2008-06-20 16:24:11 +0000 Thomas Vander Stichele gst/: Added debug. Original commit message from CVS: * gst/rtp/gstrtptheorapay.c: * gst/udp/gstmultiudpsink.c: Added debug. 2008-06-20 15:21:59 +0000 Christian Schaller * ChangeLog: * common: * configure.ac: switch v4l2src from experimental to normal build. Fixes #536831 Original commit message from CVS: switch v4l2src from experimental to normal build. Fixes #536831 2008-06-19 11:24:54 +0000 Wim Taymans gst/rtp/gstrtpg726pay.c: Remove unused variable so that we can compile again. Original commit message from CVS: * gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_setcaps): Remove unused variable so that we can compile again. 2008-06-19 11:06:29 +0000 Peter Kjellerstedt gst/rtp/gstrtpg726pay.c: No need to check for audio/G723 and audio/32KADPCM here as they are no longer supported. Original commit message from CVS: * gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_setcaps): No need to check for audio/G723 and audio/32KADPCM here as they are no longer supported. 2008-06-19 10:58:57 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Use G_GINT64_CONSTANT, this fixes the duration query on files without known length. Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset), (gst_wavpack_parse_src_query), (gst_wavpack_parse_create_src_pad): Use G_GINT64_CONSTANT, this fixes the duration query on files without known length. 2008-06-19 10:48:57 +0000 Sebastian Dröge gst/matroska/: Fix demuxing of WavPack files. Muxing is still broken. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_wvpk_header), (gst_matroska_demux_audio_caps): * gst/matroska/matroska-ids.h: Fix demuxing of WavPack files. Muxing is still broken. 2008-06-19 09:12:55 +0000 Sebastian Dröge gst/matroska/: Add a "vfunc" to the track context for postprocessing frames and convert the wavpack and subtitle post... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_track_free), (gst_matroska_demux_add_mpeg_seq_header), (gst_matroska_demux_add_wvpk_header), (gst_matroska_demux_check_subtitle_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps), (gst_matroska_demux_subtitle_caps): * gst/matroska/matroska-ids.h: Add a "vfunc" to the track context for postprocessing frames and convert the wavpack and subtitle postprocessing to this vfunc. Copy buffer flags in those functions to the new buffers too. Parse CodecState elements of Blocks. Add a postprocessing function for MPEG video that adds the sequence header from the codec private data or codec state to the frames if it's not already there. 2008-06-19 08:22:16 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: If a gap of more than 1/2 second is found in one stream send a Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_blockgroup_or_simpleblock): If a gap of more than 1/2 second is found in one stream send a NEWSEGMENT event to not stall the pipeline if the gap is too large. This also fixes Matroska files where the first buffer doesn't start at timestamp 0. Fixes bug #429322. The duration of a block is the default duration multiplied with the number of laces. Every lace is one frame and the default duration is the duration of one frame. This fixes playback of files that use lacing for some tracks. 2008-06-18 20:09:28 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Update FIXME/TODOs and only ignore EOS at the central, important place instead of seve... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_contents_seekentry): Update FIXME/TODOs and only ignore EOS at the central, important place instead of several places. 2008-06-18 16:55:05 +0000 Wim Taymans gst/rtp/gstrtpg726pay.c: Fix caps, See #538891. Original commit message from CVS: * gst/rtp/gstrtpg726pay.c: Fix caps, See #538891. 2008-06-18 10:28:20 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: Improve debug output everywhere and fix the EOS logic. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_stream_from_num), (gst_matroska_demux_encoding_cmp), (gst_matroska_demux_encoding_order_unique), (gst_matroska_demux_read_track_encoding), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_tracknumber_unique), (gst_matroska_demux_add_stream), (gst_matroska_demux_init_stream), (gst_matroska_demux_parse_tracks), (gst_matroska_demux_parse_index_cuetrack), (gst_matroska_demux_parse_index_pointentry), (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_metadata_id_simple_tag), (gst_matroska_demux_parse_metadata_id_tag), (gst_matroska_demux_parse_metadata), (gst_matroska_demux_parse_attached_file), (gst_matroska_demux_parse_attachments), (gst_matroska_demux_parse_chapters), (gst_matroska_demux_sync_streams), (gst_matroska_decode_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_parse_cluster), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_parse_contents), (gst_matroska_demux_loop_stream_parse_id), (gst_matroska_demux_loop): Improve debug output everywhere and fix the EOS logic. Check the values of the ContentEncoding elements more strictly and don't use tracks for which it's invalid. Check that the track number is unique for this stream. Check that seek positions are below G_MAXINT64 as our seeks are int64-based and overflows will fail badly. After seeks also don't push SimpleBlocks until the first one containing a keyframe is found. Before this was done only for normal Blocks. Update some FIXME/TODOs. * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes), (gst_ebml_read_utf8), (gst_ebml_read_header): Improve debug output. * gst/matroska/matroska-ids.c: (gst_matroska_track_init_video_context): * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): Remove eye mode and don't parse it anymore. We can't use that information in GStreamer yet so it's useless. 2008-06-18 10:12:57 +0000 mersad gst/rtp/: Added G726 pay/depayloaders. Fixes #538891. Original commit message from CVS: Patch by: mersad * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_base_init), (gst_rtp_g726_depay_class_init), (gst_rtp_g726_depay_init), (gst_rtp_g726_depay_setcaps), (gst_rtp_g726_depay_process), (gst_rtp_g726_depay_plugin_init): * gst/rtp/gstrtpg726depay.h: * gst/rtp/gstrtpg726pay.c: (gst_rtp_g726_pay_base_init), (gst_rtp_g726_pay_class_init), (gst_rtp_g726_pay_init), (gst_rtp_g726_pay_setcaps), (gst_rtp_g726_pay_plugin_init): * gst/rtp/gstrtpg726pay.h: Added G726 pay/depayloaders. Fixes #538891. 2008-06-17 10:14:47 +0000 Wim Taymans gst/rtsp/URLS: Some more urls. Original commit message from CVS: * gst/rtsp/URLS: Some more urls. * gst/smpte/barboxwipes.c: Add a comment * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Fix typo, add audioresample to the pipeline. 2008-06-17 10:05:55 +0000 Wim Taymans ext/libmng/: Somewhat port mngenc and mngdec to 0.10. Does not work yet and has many bits ifdeffed out still. Original commit message from CVS: * ext/libmng/Makefile.am: * ext/libmng/gstmng.c: (plugin_init): * ext/libmng/gstmngdec.c: (gst_mng_dec_base_init), (gst_mng_dec_class_init), (gst_mng_dec_sink_setcaps), (gst_mng_dec_init), (gst_mng_dec_src_getcaps), (gst_mng_dec_loop), (gst_mng_dec_get_property), (gst_mng_dec_set_property), (mngdec_error), (mngdec_openstream), (mngdec_closestream), (gst_mng_dec_sink_event), (mngdec_readdata), (mngdec_settimer), (mngdec_processheader), (mngdec_getcanvasline), (mngdec_refresh), (gst_mng_dec_change_state): * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.c: (gst_mng_enc_base_init), (gst_mng_enc_class_init), (gst_mng_enc_sink_setcaps), (gst_mng_enc_init), (gst_mng_enc_chain), (gst_mng_enc_get_property), (gst_mng_enc_set_property): * ext/libmng/gstmngenc.h: Somewhat port mngenc and mngdec to 0.10. Does not work yet and has many bits ifdeffed out still. 2008-06-16 11:34:54 +0000 Sebastian Dröge gst/matroska/matroska-demux.c: When comparing index elements with the same time compare their block number. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_index_compare): When comparing index elements with the same time compare their block number. 2008-06-16 11:31:06 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_attached_file) Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_attached_file) Init variable to NULL to avoid compiler warning. 2008-06-16 10:59:39 +0000 Sebastian Dröge gst/matroska/: Parse Attachments and post them as GST_TAG_IMAGE if we detect it as image and otherwise as GST_TAG_ATT... Original commit message from CVS: * gst/matroska/Makefile.am: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_parse_attached_file), (gst_matroska_demux_parse_attachments), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_loop_stream_parse_id): * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.c: (gst_matroska_register_tags): * gst/matroska/matroska-ids.h: * gst/matroska/matroska.c: (plugin_init): Parse Attachments and post them as GST_TAG_IMAGE if we detect it as image and otherwise as GST_TAG_ATTACHMENT. Include filename and description of the attachments in the caps. Fixes bug #537622. 2008-06-16 10:09:03 +0000 Wim Taymans ext/speex/gstspeexenc.c: Add mode property. Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speex_enc_mode_get_type), (gst_speex_enc_class_init), (gst_speex_enc_sink_getcaps), (gst_speex_enc_get_latency), (gst_speex_enc_get_query_types), (gst_speex_enc_src_query), (gst_speex_enc_init), (gst_speex_enc_setup), (gst_speex_enc_push_buffer), (gst_speex_enc_chain), (gst_speex_enc_get_property), (gst_speex_enc_set_property): Add mode property. Some cleanups, add more debug info. Add latency query. 2008-06-16 09:54:27 +0000 Sebastian Dröge gst/matroska/ebml-read.c: Return GST_FLOW_UNEXPECTED instead of GST_FLOW_ERROR on short reads. Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes): Return GST_FLOW_UNEXPECTED instead of GST_FLOW_ERROR on short reads. If we get less bytes than requested we can't do anything except doing our EOS logic. 2008-06-15 19:09:54 +0000 Sebastian Dröge gst/matroska/: Use a GArray for storing the Cue (i.e. seek) information, store the CueTrackPositions for every track,... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroskademux_do_index_seek), (gst_matroska_demux_parse_index_cuetrack), (gst_matroska_demux_parse_index_pointentry), (gst_matroska_index_compare), (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_metadata): * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: Use a GArray for storing the Cue (i.e. seek) information, store the CueTrackPositions for every track, store the block number and optimize searching in the array by sorting it after the last element was added. Fix a small memory leak when trying to parse a tags element that was already parsed. 2008-06-15 15:29:29 +0000 Sebastian Dröge gst/matroska/matroska-mux.*: Don't write another SeekHead which indexes all Clusters to the end of the file. This isn... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_reset), (gst_matroska_mux_start), (gst_matroska_mux_finish), (gst_matroska_mux_write_data): * gst/matroska/matroska-mux.h: Don't write another SeekHead which indexes all Clusters to the end of the file. This isn't useful for anything and just increases filesize. 2008-06-15 15:01:30 +0000 Sebastian Dröge gst/matroska/ebml-read.c: Prevent unaligned memory access when reading floats. Original commit message from CVS: * gst/matroska/ebml-read.c: (_ext2dbl), (gst_ebml_read_float): Prevent unaligned memory access when reading floats. 2008-06-15 14:08:41 +0000 Sebastian Dröge gst/matroska/: Make sure that every Tags element is only parsed once and it's containing tags are only posted once. Original commit message from CVS: * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_parse_metadata): * gst/matroska/matroska-demux.h: Make sure that every Tags element is only parsed once and it's containing tags are only posted once. 2008-06-15 09:43:25 +0000 Sebastian Dröge gst/matroska/: Handle EBML elements like Void or CRC32 in the EbmlRead base class already. They're not useful in the ... Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_peek_id), (gst_ebml_read_header): * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_parse_tracks), (gst_matroska_demux_parse_index_cuetrack), (gst_matroska_demux_parse_index_pointentry), (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_metadata_id_simple_tag), (gst_matroska_demux_parse_metadata_id_tag), (gst_matroska_demux_parse_metadata), (gst_matroska_demux_parse_attachments), (gst_matroska_demux_parse_chapters), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_parse_cluster), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_parse_contents), (gst_matroska_demux_loop_stream_parse_id): Handle EBML elements like Void or CRC32 in the EbmlRead base class already. They're not useful in the matroska parser and only cause additional code. 2008-06-14 15:51:25 +0000 Sebastian Dröge gst/matroska/: Reverse the level list as we usually are only interested in the first element or want to add a new fir... Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_level_free), (gst_ebml_finalize), (gst_ebml_read_change_state), (gst_ebml_read_element_level_up), (gst_ebml_read_master): * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_contents_seekentry): Reverse the level list as we usually are only interested in the first element or want to add a new first element. Having the first element stored at the end and calling g_list_last() and g_list_append() is more expensive. Also use GSlice for allocating the GstEbmlLevel structs. 2008-06-13 21:13:46 +0000 Tim-Philipp Müller gst/debug/gsttaginject.c: Don't unref NULL taglist in finalize. Don't use c++ style comments. Original commit message from CVS: * gst/debug/gsttaginject.c: (gst_tag_inject_finalize), (gst_tag_inject_class_init), (gst_tag_inject_init): Don't unref NULL taglist in finalize. Don't use c++ style comments. 2008-06-13 19:14:41 +0000 Sebastian Dröge gst/matroska/: Use gst_value_serialize() and gst_value_deserialize() for transforming tags from some GType to a strin... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_metadata_id_simple_tag): * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_write_data): Use gst_value_serialize() and gst_value_deserialize() for transforming tags from some GType to a string and the other way around. The default transformations in GLib don't include transformations from string to number types. 2008-06-13 19:07:03 +0000 Sebastian Dröge gst/matroska/matroska-demux.*: Only parse Tracks, SeekHead and SegmentInfo elements once but allow Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset), (gst_matroska_demux_parse_tracks), (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_attachments), (gst_matroska_demux_parse_chapters), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_loop_stream_parse_id): * gst/matroska/matroska-demux.h: Only parse Tracks, SeekHead and SegmentInfo elements once but allow Tags multiple times. The first ones can appear more than once but must contain the same content as the first for backup purposes so we ignore all but the first one. Tags can appear multiple times with different content. Jump to all elements except Clusters that are available from a SeekHead to make it more likely to have all required informations before getting to the first Clusters. Add dummy functions for parsing Attachments and Chapters. 2008-06-13 14:33:52 +0000 Stefan Kost gst/replaygain/: More doc updates. Original commit message from CVS: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: More doc updates. 2008-06-13 11:59:23 +0000 Stefan Kost docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-app.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdaudio.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstinterlace.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-mythtv.xml * docs/plugins/inspect/plugin-nas.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-oss4.xml * docs/plugins/inspect/plugin-rawparse.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rfbsrc.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-selector.xml: * docs/plugins/inspect/plugin-sndfile.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-subenc.xml * docs/plugins/inspect/plugin-timidity.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-vcdsrc.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/dc1394/gstdc1394.c: * ext/directfb/dfbvideosink.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/mplex/gstmplex.cc: * ext/musicbrainz/gsttrm.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst-libs/gst/app/gstappsink.c: * gst/deinterlace/gstdeinterlace.c: * gst/dvdspu/gstdvdspu.c: * gst/festival/gstfestival.c: * gst/freeze/gstfreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/modplug/gstmodplug.cc: * gst/nuvdemux/gstnuvdemux.c: Add missing elements to docs. Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types. 2008-06-13 11:54:05 +0000 Wim Taymans gst/udp/gstudpsrc.*: Add property to control automatic join/leave of multicast groups. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop): * gst/udp/gstudpsrc.h: Add property to control automatic join/leave of multicast groups. Add G_LIKELY. Remove setting caps on buffers explicitly, basesrc does that for us now. Improve debug info. Convert some non-fatal error into warnings. Use g_ntohs for better portability. Leave multicast groups when stopping. When using external sockets, use getsockname() on them to fill up the addr structure before calling methods that use the structure. Should all fix #536903. API: GstUDPSrc::auto-multicast property 2008-06-13 11:47:28 +0000 Wim Taymans gst/udp/gstudpnetutils.c: Use g_ntohl for better portability. Original commit message from CVS: * gst/udp/gstudpnetutils.c: (gst_udp_is_multicast): Use g_ntohl for better portability. 2008-06-13 11:45:54 +0000 Wim Taymans gst/udp/gstmultiudpsink.c: Fix a typo and do some small cleanups. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send), (gst_multiudpsink_remove): Fix a typo and do some small cleanups. 2008-06-13 09:39:41 +0000 Olivier Crete gst/rtp/gstrtptheoradepay.c: Make the delivery-method mandatory on the caps and only accept inline for now. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps): Make the delivery-method mandatory on the caps and only accept inline for now. Reverse strcmp checks for delivery-method. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps): Make delivery method optional when parsing caps and note this in the caps. Reverse strcmp checks for delivery-method. * gst/rtp/gstrtpvorbispay.c: Update a comment to note that the delivery-method is optional, Fixes #537675. 2008-06-13 06:57:21 +0000 Stefan Kost Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs alrea... Original commit message from CVS: * docs/plugins/gst-plugins-ugly-plugins-docs.sgml: * docs/plugins/gst-plugins-ugly-plugins-sections.txt: * ext/a52dec/gsta52dec.c: * ext/amrnb/amrnbdec.c: * ext/amrnb/amrnbenc.c: * ext/amrnb/amrnbparse.c: * ext/lame/gstlame.c: * ext/mad/gstmad.c: * ext/sidplay/gstsiddec.cc: * gst/asfdemux/gstrtspwms.c: * gst/mpegaudioparse/gstxingmux.c: * gst/realmedia/rademux.c: * gst/realmedia/rdtmanager.c: * gst/realmedia/rtspreal.c: * gst/synaesthesia/gstsynaesthesia.c: Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs already). Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types. 2008-06-13 05:52:17 +0000 Stefan Kost Do not use short_description in section docs for elements. We extract them from element details and there will be war... Original commit message from CVS: * ext/lame/gstlame.c: * ext/sidplay/gstsiddec.cc: * gst/mpegaudioparse/gstxingmux.c: Do not use short_description in section docs for elements. We extract them from element details and there will be warnings if they differ. 2008-06-12 17:30:06 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Set udpsrc for receiving data from multicast groups to PAUSED instead of leaving them in READY... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_mcast): Set udpsrc for receiving data from multicast groups to PAUSED instead of leaving them in READY. Fixes #537832. 2008-06-12 12:14:38 +0000 Stefan Kost gst/avi/gstavimux.c: Simplify code. gst_tag_list_merge() does the NULL checks. Add a FIXME for a random constant in t... Original commit message from CVS: * gst/avi/gstavimux.c: Simplify code. gst_tag_list_merge() does the NULL checks. Add a FIXME for a random constant in tagmuxing code. 2008-06-11 14:28:44 +0000 Stefan Kost gst/debug/gsttaginject.*: Now actually adding the new element. Original commit message from CVS: * gst/debug/gsttaginject.c: * gst/debug/gsttaginject.h: Now actually adding the new element. 2008-06-11 14:11:16 +0000 Stefan Kost Remove dummy plugin_init. Remove some undefined entries from doc- section file. Add taginject element and rebuild doc... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst/debug/Makefile.am: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstdebug.c: * gst/debug/gstnavseek.c: * gst/debug/gstpushfilesrc.c: * gst/debug/gstpushfilesrc.h: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/progressreport.h: * gst/debug/rndbuffersize.c: * gst/debug/testplugin.c: Remove dummy plugin_init. Remove some undefined entries from doc- section file. Add taginject element and rebuild docs for it. 2008-06-11 11:27:46 +0000 Sebastian Dröge gst/matroska/matroska-mux.c: Update the counter for the number of streams when pads are added or removed. This will m... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad), (gst_matroska_mux_write_data): Update the counter for the number of streams when pads are added or removed. This will make sure that a seek table is generated for files with just one audio stream. 2008-06-11 11:18:23 +0000 Sebastian Dröge gst/matroska/: Add some more tags, improve debugging a bit and make sure that Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_metadata_id_simple_tag): * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_simple_tag): Add some more tags, improve debugging a bit and make sure that GValue transformation has succeeded before using the result as a tag. 2008-06-11 08:56:16 +0000 Olivier Crete gst/rtp/gstrtptheorapay.c: The Theora RTP payloader only supports the "inline" delievery method so let's declare this... Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtptheorapay.c: The Theora RTP payloader only supports the "inline" delievery method so let's declare this on the caps of the static pad template. Fixes bug #537675. 2008-06-10 17:20:45 +0000 Wim Taymans gst/videomixer/videomixer.c: Remove bogus check. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues), (gst_videomixer_blend_buffers), (gst_videomixer_update_queues): Remove bogus check. 2008-06-10 16:25:24 +0000 Wim Taymans gst/videomixer/videomixer.c: Use stream_time to synchronize the object properties. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues), (gst_videomixer_blend_buffers): Use stream_time to synchronize the object properties. Use running_time of the master pad to timestamp outgoing buffers. Fix the initial segment event to extend an unknown amount of time. Fixes #537361. 2008-06-10 11:05:30 +0000 Wim Taymans gst/avi/gstavidemux.c: Try to ignore unparsable/unknown streams and give a warning instead of erroring out. Fixes #53... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull): Try to ignore unparsable/unknown streams and give a warning instead of erroring out. Fixes #537377. 2008-06-10 10:44:53 +0000 Sebastian Dröge gst/matroska/ebml-write.c: Use GDOUBLE_TO_BE() instead of (probably slower) custom code. Original commit message from CVS: * gst/matroska/ebml-write.c: (gst_ebml_write_float): Use GDOUBLE_TO_BE() instead of (probably slower) custom code. * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init), (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_track_free), (gst_matroska_demux_encoding_cmp), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_add_stream), (gst_matroska_demux_handle_src_query), (gst_matroska_demux_init_stream), (gst_matroska_demux_parse_index_cuetrack), (gst_matroska_demux_parse_index_pointentry), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_metadata_id_simple_tag), (gst_matroska_demux_parse_metadata), (gst_matroska_demux_add_wvpk_header), (gst_matroska_decode_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_parse_cluster), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_loop_stream_parse_id), (gst_matroska_demux_loop), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps), (gst_matroska_demux_subtitle_caps): * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.c: (gst_matroska_track_init_subtitle_context): * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init), (gst_matroska_mux_class_init), (gst_matroska_mux_init), (gst_matroska_mux_create_uid), (gst_matroska_mux_reset), (gst_matroska_mux_video_pad_setcaps), (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_subtitle_pad_setcaps), (gst_matroska_mux_request_new_pad), (gst_matroska_mux_track_header), (gst_matroska_mux_start), (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish), (gst_matroska_mux_write_data), (gst_matroska_mux_collected), (gst_matroska_mux_set_property): Add many FIXMEs/TODOs all over the matroska muxer and demuxer elements, do some checks for valid values in the demuxer, handle tracktimecodescale in the demuxer, set correct default values for all settings in the demuxer, review and add all missing matroska IDs and some more raw YUV formats, and some trivial cleanup. 2008-06-10 08:59:17 +0000 Sebastian Dröge ext/pulse/: Some smaller cleanup. Use G_PARAM_STATIC_STRINGS, gst_element_class_set_details_simple() and fix coding s... Original commit message from CVS: * ext/pulse/pulsemixer.c: (gst_pulsemixer_base_init), (gst_pulsemixer_class_init): * ext/pulse/pulsesink.c: (gst_pulsesink_base_init), (gst_pulsesink_class_init), (gst_pulsesink_prepare): * ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported), (gst_pulsesrc_base_init), (gst_pulsesrc_class_init), (gst_pulsesrc_prepare): Some smaller cleanup. Use G_PARAM_STATIC_STRINGS, gst_element_class_set_details_simple() and fix coding style a bit more. 2008-06-10 08:22:17 +0000 Sebastian Dröge Add documentation to the pulseaudio plugin and run make update in docs/plugins. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * ext/pulse/plugin.c: * ext/pulse/pulsemixer.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: Add documentation to the pulseaudio plugin and run make update in docs/plugins. 2008-06-10 06:52:44 +0000 Brian Cameron sys/sunaudio/gstsunaudiomixerctrl.c: Improvements for the SunAudio mixer by handling mute as no gain for tracks that ... Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_get_volume), (gst_sunaudiomixer_ctrl_set_volume): Improvements for the SunAudio mixer by handling mute as no gain for tracks that have a gain property but no mute property. Fixes bug #536067. 2008-06-10 06:45:33 +0000 Sebastian Dröge Add pulseaudio GStreamer element from gst-pulse. Development will continue here instead of pulseaudio SVN. Fixes bug ... Original commit message from CVS: * configure.ac: * ext/pulse/Makefile.am: * ext/pulse/plugin.c: (plugin_init): * ext/pulse/pulsemixer.c: (gst_pulsemixer_interface_supported), (gst_pulsemixer_implements_interface_init), (gst_pulsemixer_init_interfaces), (gst_pulsemixer_base_init), (gst_pulsemixer_class_init), (gst_pulsemixer_init), (gst_pulsemixer_finalize), (gst_pulsemixer_set_property), (gst_pulsemixer_get_property), (gst_pulsemixer_change_state): * ext/pulse/pulsemixer.h: * ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_context_state_cb), (gst_pulsemixer_ctrl_sink_info_cb), (gst_pulsemixer_ctrl_source_info_cb), (gst_pulsemixer_ctrl_subscribe_cb), (gst_pulsemixer_ctrl_success_cb), (gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_close), (gst_pulsemixer_ctrl_new), (gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_list_tracks), (gst_pulsemixer_ctrl_timeout_event), (restart_time_event), (gst_pulsemixer_ctrl_set_volume), (gst_pulsemixer_ctrl_get_volume), (gst_pulsemixer_ctrl_set_record), (gst_pulsemixer_ctrl_set_mute): * ext/pulse/pulsemixerctrl.h: * ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_class_init), (gst_pulsemixer_track_init), (gst_pulsemixer_track_new): * ext/pulse/pulsemixertrack.h: * ext/pulse/pulseprobe.c: (gst_pulseprobe_context_state_cb), (gst_pulseprobe_sink_info_cb), (gst_pulseprobe_source_info_cb), (gst_pulseprobe_invalidate), (gst_pulseprobe_open), (gst_pulseprobe_enumerate), (gst_pulseprobe_close), (gst_pulseprobe_new), (gst_pulseprobe_free), (gst_pulseprobe_get_properties), (gst_pulseprobe_needs_probe), (gst_pulseprobe_probe_property), (gst_pulseprobe_get_values), (gst_pulseprobe_set_server): * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: (gst_pulsesink_base_init), (gst_pulsesink_class_init), (gst_pulsesink_init), (gst_pulsesink_destroy_stream), (gst_pulsesink_destroy_context), (gst_pulsesink_finalize), (gst_pulsesink_dispose), (gst_pulsesink_set_property), (gst_pulsesink_get_property), (gst_pulsesink_context_state_cb), (gst_pulsesink_stream_state_cb), (gst_pulsesink_stream_request_cb), (gst_pulsesink_stream_latency_update_cb), (gst_pulsesink_open), (gst_pulsesink_close), (gst_pulsesink_prepare), (gst_pulsesink_unprepare), (gst_pulsesink_write), (gst_pulsesink_delay), (gst_pulsesink_success_cb), (gst_pulsesink_reset), (gst_pulsesink_change_title), (gst_pulsesink_event), (gst_pulsesink_get_type): * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported), (gst_pulsesrc_implements_interface_init), (gst_pulsesrc_init_interfaces), (gst_pulsesrc_base_init), (gst_pulsesrc_class_init), (gst_pulsesrc_init), (gst_pulsesrc_destroy_stream), (gst_pulsesrc_destroy_context), (gst_pulsesrc_finalize), (gst_pulsesrc_dispose), (gst_pulsesrc_set_property), (gst_pulsesrc_get_property), (gst_pulsesrc_context_state_cb), (gst_pulsesrc_stream_state_cb), (gst_pulsesrc_stream_request_cb), (gst_pulsesrc_open), (gst_pulsesrc_close), (gst_pulsesrc_prepare), (gst_pulsesrc_unprepare), (gst_pulsesrc_read), (gst_pulsesrc_delay), (gst_pulsesrc_change_state), (gst_pulsesrc_get_type): * ext/pulse/pulsesrc.h: * ext/pulse/pulseutil.c: (gst_pulse_fill_sample_spec), (gst_pulse_client_name), (gst_pulse_gst_to_channel_map): * ext/pulse/pulseutil.h: Add pulseaudio GStreamer element from gst-pulse. Development will continue here instead of pulseaudio SVN. Fixes bug #400679. Only changes over gst-pulse SVN are added copyright to the top of files and coding style changes. 2008-06-09 20:02:05 +0000 Benjamin Kampmann ext/cdio/: Also extract album title and album genre from CD-TEXT if available (#537021). Original commit message from CVS: Patch by: Benjamin Kampmann * ext/cdio/gstcdio.c: (gst_cdio_get_cdtext), (gst_cdio_add_cdtext_album_tags): * ext/cdio/gstcdio.h: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open): Also extract album title and album genre from CD-TEXT if available (#537021). 2008-06-09 08:52:04 +0000 Sjoerd Simons sys/v4l2/gstv4l2src.c: Improve negotiation a bit more by picking the smallest possible resolution that is larger than... Original commit message from CVS: Patch by: Sjoerd Simons * sys/v4l2/gstv4l2src.c: (gst_v4l2src_negotiate): Improve negotiation a bit more by picking the smallest possible resolution that is larger than the resolution specified in the first caps entry of the peer caps. Fixes bug #536994. 2008-06-09 08:42:49 +0000 Bastien Nocera sys/v4l2/: Fix compilation with newer GIT kernels that deprecated Original commit message from CVS: Patch by: Bastien Nocera * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): Fix compilation with newer GIT kernels that deprecated V4L2_CID_HCENTER and V4L2_CID_VCENTER. Fixes bug #536317. 2008-06-07 18:48:54 +0000 Tim-Philipp Müller Require libcdio >= 0.76. Original commit message from CVS: * configure.ac: * ext/cdio/gstcdio.c: * ext/cdio/gstcdio.h: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open): Require libcdio >= 0.76. 2008-06-05 11:07:17 +0000 Sebastian Dröge gst/interleave/: Properly implement duration and position queries in bytes format. We have to take the upstream reply... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads), (gst_deinterleave_src_query): * gst/interleave/interleave.c: (gst_interleave_src_query_duration), (gst_interleave_src_query): Properly implement duration and position queries in bytes format. We have to take the upstream reply and divide/multiply it by the number of channels to get the correct result. 2008-06-05 09:45:00 +0000 Thijs Vermeir gst/avi/gstavidemux.c: Catch UNEXPECTED when downstream has reached end of segment in reverse mode. Original commit message from CVS: * gst/avi/gstavidemux.c: Catch UNEXPECTED when downstream has reached end of segment in reverse mode. 2008-06-04 18:08:35 +0000 Thijs Vermeir gst/avi/gstavidemux.c: Fix typo in comment Original commit message from CVS: * gst/avi/gstavidemux.c: Fix typo in comment 2008-06-04 18:03:24 +0000 Thijs Vermeir gst/avi/gstavidemux.c: Because we don't know the frame order we need to push till the next keyframe Original commit message from CVS: * gst/avi/gstavidemux.c: Because we don't know the frame order we need to push till the next keyframe 2008-06-04 17:39:31 +0000 Sjoerd Simons sys/v4l2/gstv4l2src.c: Provide a custom negotiation function to make sure to pick the highest possible framerate and ... Original commit message from CVS: Patch by: Sjoerd Simons * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_fixate), (gst_v4l2src_negotiate): Provide a custom negotiation function to make sure to pick the highest possible framerate and resolution. Fixes bug #536646. 2008-06-04 16:49:26 +0000 Thijs Vermeir gst/avi/gstavidemux.c: Set EOS when going out of the segment in reverse playback Original commit message from CVS: * gst/avi/gstavidemux.c: Set EOS when going out of the segment in reverse playback 2008-06-04 15:19:46 +0000 Tim-Philipp Müller ext/taglib/Makefile.am: Add -Wno-attributes to CXXFLAGS to suppress warning caused by taglib headers (with gcc 4.3.1). Original commit message from CVS: * ext/taglib/Makefile.am:: Add -Wno-attributes to CXXFLAGS to suppress warning caused by taglib headers (with gcc 4.3.1). 2008-06-04 11:59:18 +0000 Peter Kjellerstedt gst/rtsp/gstrtspsrc.c: Use the new gst_rtsp_connection_get_ip() to access the IP address of a GstRTSPConnection since... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink): Use the new gst_rtsp_connection_get_ip() to access the IP address of a GstRTSPConnection since it is a private member. 2008-06-04 10:42:46 +0000 Tim-Philipp Müller Use new utility functions in libgsttag to process coverart (#512333). Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer): * gst/id3demux/id3v2frames.c: (parse_picture_frame): Use new utility functions in libgsttag to process coverart (#512333). 2008-06-04 08:54:09 +0000 Sebastian Dröge ext/flac/gstflacdec.c: We actually support left/side, right/side and mid/side files. The conversion to normal, interl... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_write): We actually support left/side, right/side and mid/side files. The conversion to normal, interleaved stereo is done by libflac. 2008-06-04 07:36:07 +0000 Sebastian Dröge gst/matroska/ebml-write.c: Unref the write cache in finalize if it was set and add add "FIXME" to a comment that need... Original commit message from CVS: * gst/matroska/ebml-write.c: (gst_ebml_write_finalize), (gst_ebml_write_set_cache): Unref the write cache in finalize if it was set and add add "FIXME" to a comment that needs it. 2008-06-04 06:48:46 +0000 Sebastian Dröge gst/interleave/interleave.*: Use an always increasing integer for the number in the name of the requested sink pads t... Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_pad_get_type), (gst_interleave_pad_get_property), (gst_interleave_pad_class_init), (gst_interleave_request_new_pad), (gst_interleave_release_pad): * gst/interleave/interleave.h: Use an always increasing integer for the number in the name of the requested sink pads to guarantuee a unique name. Add a "channel" property to GstInterleavePad to make it possible for applications to retrieve the channel number in the output for every pad. Use g_type_register_static_simple() instead of g_type_register_static() to save some relocations. 2008-06-03 14:35:59 +0000 Sebastian Dröge gst/interleave/interleave.c: Stop GstCollectPads before calling the parent's state change function when going from PA... Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_pad_get_type), (gst_interleave_change_state): Stop GstCollectPads before calling the parent's state change function when going from PAUSED to READY as we otherwise deadlock. Fixes bug #536258. 2008-06-03 09:03:19 +0000 Sebastian Dröge gst/interleave/interleave.c: Use new gst_audio_check_channel_positions() function and register the GstInterleavePad t... Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_check_channel_positions), (gst_interleave_set_channel_positions), (gst_interleave_class_init): Use new gst_audio_check_channel_positions() function and register the GstInterleavePad type from a threadsafe context. 2008-06-02 16:10:00 +0000 Thijs Vermeir gst/avi/gstavidemux.*: Implement reverse playback. Fixes #535300. Original commit message from CVS: Patch by: Thijs Vermeir * gst/avi/gstavidemux.c: (gst_avi_demux_index_next), (gst_avi_demux_index_prev), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_do_seek), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry): * gst/avi/gstavidemux.h: Implement reverse playback. Fixes #535300. Small cleanups. 2008-06-02 12:42:14 +0000 Sebastian Dröge gst/interleave/interleave.*: Allow setting channel positions via a property and allow using the channel positions on ... Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_pad_get_type), (gst_interleave_finalize), (gst_audio_check_channel_positions), (gst_interleave_set_channel_positions), (gst_interleave_class_init), (gst_interleave_init), (gst_interleave_set_property), (gst_interleave_get_property), (gst_interleave_request_new_pad), (gst_interleave_release_pad), (gst_interleave_sink_setcaps), (gst_interleave_src_query_duration), (gst_interleave_src_query_latency), (gst_interleave_collected): * gst/interleave/interleave.h: Allow setting channel positions via a property and allow using the channel positions on the input as the channel positions of the output. Fix some broken logic and memory leaks. * tests/check/Makefile.am: * tests/check/elements/interleave.c: (src_handoff_float32), (sink_handoff_float32), (GST_START_TEST), (interleave_suite): Add unit tests for checking correct handling of channel positions. 2008-06-02 12:22:56 +0000 Sebastian Dröge gst/videomixer/videomixer.c: When using gst_element_iterate_pads() one has to unref every pad after usage. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_query_duration), (gst_videomixer_query_latency): When using gst_element_iterate_pads() one has to unref every pad after usage. 2008-05-31 16:53:23 +0000 Bastien Nocera gst/qtdemux/: Improve meta-data handling, add 'comment', 'description' and 'copyright' tag handling. Original commit message from CVS: Patch by: Bastien Nocera * gst/qtdemux/qtdemux.c: (qtdemux_tag_add_str), (qtdemux_parse_udta): * gst/qtdemux/qtdemux_fourcc.h: Improve meta-data handling, add 'comment', 'description' and 'copyright' tag handling. Fixes #535935 2008-05-31 15:30:41 +0000 Julien Moutte gst/qtdemux/qtdemux.c: Make sure we we don't clip the segment's stop using the main segment duration as that could cr... Original commit message from CVS: 2008-05-31 Julien Moutte * gst/qtdemux/qtdemux.c: (gst_qtdemux_find_keyframe), (gst_qtdemux_find_segment), (gst_qtdemux_perform_seek), (gst_qtdemux_seek_to_previous_keyframe), (gst_qtdemux_activate_segment), (gst_qtdemux_loop): Make sure we we don't clip the segment's stop using the main segment duration as that could crop quite some video frames. Make reverse playback support more robust and support edit lists. Support seeking to the last frame, and fix reverse looping playback. Add some debugging. * win32/common/config.h: Updated. 2008-05-31 08:37:00 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Don't clip float/double samples, correctly unset passthrough mode and use better rou... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_transform_ip): Don't clip float/double samples, correctly unset passthrough mode and use better rounding for integer samples. 2008-05-30 11:03:57 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.*: Update the filter coefficients only when needed in the transform_ip function and cor... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_init), (setup_filter), (set_passthrough), (update_coefficients), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_transform_ip): * gst/equalizer/gstiirequalizer.h: Update the filter coefficients only when needed in the transform_ip function and correctly set the element into passthrough mode if the gain of all bands is 0. 2008-05-29 11:30:16 +0000 Sebastian Keller gst/alpha/gstalpha.c: Try to skip pixels or areas that are too dark or too bright for us to do meaningfull color dete... Original commit message from CVS: Based on patch by: Sebastian Keller * gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init), (gst_alpha_set_property), (gst_alpha_get_property), (gst_alpha_chroma_key_ayuv), (gst_alpha_chromakey_row_i420): Try to skip pixels or areas that are too dark or too bright for us to do meaningfull color detection. Added properties to control the sensitivity to light and darkness. Added some small cleanups. Fixes #512345. 2008-05-28 20:01:32 +0000 Jan Schmidt Ignore some more generated things Original commit message from CVS: * docs/plugins/.cvsignore: * tests/check/elements/.cvsignore: Ignore some more generated things * tests/check/Makefile.am: Ignore OSS elements in the state changes test too. 2008-05-28 16:22:36 +0000 Wim Taymans docs/plugins/: Add SMPTE effect elements to docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: Add SMPTE effect elements to docs. 2008-05-28 14:31:05 +0000 Stefan Kost Document whats first shown on the fdo plugin docs page :) Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/raw1394/gstdv1394src.c: Document whats first shown on the fdo plugin docs page :) 2008-05-28 14:07:21 +0000 Stefan Kost Rename audiovoice to audiokaraoke and add it to the docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiokaraoke.h: * gst/audiofx/audiovoice.c: * gst/audiofx/audiovoice.h: Rename audiovoice to audiokaraoke and add it to the docs. 2008-05-28 13:28:20 +0000 Stefan Kost Document aasink and cacasink. Original commit message from CVS: * REQUIREMENTS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * ext/aalib/gstaasink.c: * ext/libcaca/gstcacasink.c: Document aasink and cacasink. 2008-05-28 08:36:44 +0000 Sebastian Dröge gst/videomixer/videomixer.*: duration and latency queries. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_reset), (gst_videomixer_init), (gst_videomixer_query_duration), (gst_videomixer_query_latency), (gst_videomixer_query), (gst_videomixer_blend_buffers): * gst/videomixer/videomixer.h: Implement position (in time), duration and latency queries. 2008-05-28 08:14:16 +0000 Sebastian Dröge gst/interleave/interleave.c: Implement latency query. Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_src_query_duration), (gst_interleave_src_query_latency), (gst_interleave_src_query): Implement latency query. 2008-05-27 17:55:30 +0000 Edward Hervey gst/videomixer/videomixer.*: Implement proper seek/newsegment handling. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_reset), (gst_videomixer_init), (gst_videomixer_request_new_pad), (gst_videomixer_fill_queues), (forward_event_func), (forward_event), (gst_videomixer_src_event), (gst_videomixer_sink_event): * gst/videomixer/videomixer.h: Implement proper seek/newsegment handling. Based on adder's implementation. Fixes #535121 2008-05-26 16:25:15 +0000 j^ gst/qtdemux/qtdemux.c: Add caps for DVCPRO50 and DVCPRO HD PAL/NTSC. See #526481. Original commit message from CVS: Patch by: j^ * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add caps for DVCPRO50 and DVCPRO HD PAL/NTSC. See #526481. 2008-05-26 15:51:41 +0000 Wim Taymans gst/audiofx/: Add simple voice removal element. Yay karaoke. Original commit message from CVS: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audiovoice.c: (gst_audio_voice_base_init), (gst_audio_voice_class_init), (gst_audio_voice_init), (update_filter), (gst_audio_voice_set_property), (gst_audio_voice_get_property), (gst_audio_voice_setup), (gst_audio_voice_transform_int), (gst_audio_voice_transform_float), (gst_audio_voice_transform_ip): * gst/audiofx/audiovoice.h: Add simple voice removal element. Yay karaoke. 2008-05-26 15:39:26 +0000 William M. Brack sys/v4l2/v4l2src_calls.c: Fix potential caps leak. Original commit message from CVS: Patch by: William M. Brack * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format): Fix potential caps leak. If we can't get the framerate with an ioctl, try to get it with the current norm. Fixes #520092. 2008-05-26 15:14:55 +0000 William M. Brack sys/v4l2/v4l2src_calls.c: If we fail to get the frame intervals, simply don't touch the framerates on the template ca... Original commit message from CVS: Patch by: William M. Brack * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format_and_size): If we fail to get the frame intervals, simply don't touch the framerates on the template caps instead of discarding the format. See #520092. 2008-05-26 14:52:51 +0000 William M. Brack sys/v4l2/gstv4l2src.c: Add NV12, NV21 and bayer support. See #520092. Original commit message from CVS: Patch by: William M. Brack * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure), (gst_v4l2_get_caps_info): Add NV12, NV21 and bayer support. See #520092. 2008-05-26 13:51:38 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Unbreak segment activation again. Fixes #531672. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_find_segment), (gst_qtdemux_activate_segment): Unbreak segment activation again. Fixes #531672. 2008-05-26 10:28:47 +0000 Sebastian Dröge gst/interleave/deinterleave.c: Add another example launch line. Original commit message from CVS: * gst/interleave/deinterleave.c: Add another example launch line. * gst/interleave/interleave.c: (interleave_24), (gst_interleave_finalize), (gst_interleave_base_init), (gst_interleave_class_init), (gst_interleave_init), (gst_interleave_request_new_pad), (gst_interleave_release_pad), (gst_interleave_change_state), (__remove_channels), (__set_channels), (gst_interleave_sink_getcaps), (gst_interleave_set_process_function), (gst_interleave_sink_setcaps), (gst_interleave_sink_event), (gst_interleave_src_query_duration), (gst_interleave_src_query), (forward_event_func), (forward_event), (gst_interleave_src_event), (gst_interleave_collected): * gst/interleave/interleave.h: Major rewrite of interleave using GstCollectpads. This new version also supports almost all raw audio formats and has better caps negotiation. Fixes bug #506594. Also update docs and add some more examples. * tests/check/elements/interleave.c: (interleave_chain_func), (GST_START_TEST), (src_handoff_float32), (sink_handoff_float32), (interleave_suite): Add some more extensive unit tests for interleave. 2008-05-26 09:57:40 +0000 Wim Taymans Don't use _gst_pad(). Original commit message from CVS: * examples/switch/switcher.c: (switch_timer): * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init): * gst/rtpmanager/gstrtpclient.c: (create_stream): * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp), (gst_sdp_demux_stream_configure_udp_sink): * tests/check/elements/deinterleave.c: (GST_START_TEST), (pad_added_setup_data_check_float32_8ch_cb): * tests/check/elements/rganalysis.c: (send_eos_event), (send_tag_event): Don't use _gst_pad(). 2008-05-25 16:09:39 +0000 Sebastian Dröge ext/flac/: Set the channel layout when decoding FLAC files with more than 2 channels as defined by the FLAC spec. Fix... Original commit message from CVS: * ext/flac/Makefile.am: * ext/flac/gstflacdec.c: (gst_flac_dec_write): Set the channel layout when decoding FLAC files with more than 2 channels as defined by the FLAC spec. Fixes bug #534570. Also don't try to decode left/side, right/side and mid/side files as we don't support this at all. 2008-05-24 12:55:39 +0000 Tim-Philipp Müller configure.ac: We need -base CVS (rtsp). Original commit message from CVS: * configure.ac: We need -base CVS (rtsp). 2008-05-22 19:47:53 +0000 Sebastian Dröge docs/plugins/: Add interleave/deinterleave to the docs and while at that run make update in docs/plugins. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-app.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdaudio.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-nas.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-rawparse.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rfbsrc.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-selector.xml: * docs/plugins/inspect/plugin-sndfile.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-vcdsrc.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: Add interleave/deinterleave to the docs and while at that run make update in docs/plugins. * gst/interleave/deinterleave.c: Add a parapraph about using a queue and audioconvert after the source pads to the docs. 2008-05-22 18:55:09 +0000 Sebastian Dröge gst/interleave/deinterleave.*: Don't set a getcaps() function on the src pads as it's not required and the default ge... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps): * gst/interleave/deinterleave.h: Don't set a getcaps() function on the src pads as it's not required and the default getcaps() function returns the correct results for our src pads. Complete documentation and add myself to the authors of the element. 2008-05-22 14:49:08 +0000 Tim-Philipp Müller gst/udp/Makefile.am: Add -D_GNU_SOURCE to CFLAGS so we get things like EAI_ADDRFAMILY when including netdb.h when bui... Original commit message from CVS: * gst/udp/Makefile.am: Add -D_GNU_SOURCE to CFLAGS so we get things like EAI_ADDRFAMILY when including netdb.h when building against glibc >= 2.8. 2008-05-22 11:19:03 +0000 Julien Moutte gst/smpte/gstsmptealpha.c: Fix debug statement arguments. Original commit message from CVS: 2008-05-22 Julien Moutte * gst/smpte/gstsmptealpha.c: (gst_smpte_alpha_setcaps): Fix debug statement arguments. * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_setup_qos_dscp): * gst/udp/gstudpnetutils.c: (gst_udp_join_group), (gst_udp_leave_group): Fix IP and IPV6 options to make it work on more platforms. 2008-05-21 17:51:09 +0000 Wim Taymans tests/check/elements/: Don't use gst_element_get_pad(), it's a bad, bad method. Original commit message from CVS: * tests/check/elements/avimux.c: (setup_src_pad), (teardown_src_pad): * tests/check/elements/icydemux.c: (icydemux_found_pad), (GST_START_TEST): * tests/check/elements/matroskamux.c: (setup_src_pad), (teardown_src_pad), (setup_sink_pad), (teardown_sink_pad): * tests/check/elements/videocrop.c: (video_crop_get_test_caps), (GST_START_TEST): * tests/check/elements/wavpackparse.c: (wavpackparse_found_pad), (setup_wavpackparse), (cleanup_wavpackparse): Don't use gst_element_get_pad(), it's a bad, bad method. 2008-05-21 17:39:38 +0000 Wim Taymans Don't use gst_element_get_pad(), it's a bad method. Original commit message from CVS: * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset), (do_toggle_element): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset), (do_toggle_element): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset), (do_toggle_element): * ext/gconf/gstswitchsink.c: (gst_switch_commit_new_kid): * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_reset), (do_toggle_element): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_reset), (do_toggle_element): * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset), (gst_auto_audio_sink_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset), (gst_auto_video_sink_detect): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string), (gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr): * tests/icles/videocrop-test.c: (test_with_caps), (video_crop_get_test_caps): Don't use gst_element_get_pad(), it's a bad method. 2008-05-21 17:35:50 +0000 Wim Taymans gst/udp/: Joining a multicast group and setting the loop/ttl properties are totally unrelated tasks are must be separ... Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal): * gst/udp/gstudpnetutils.c: (gst_udp_set_loop_ttl), (gst_udp_join_group): * gst/udp/gstudpnetutils.h: * gst/udp/gstudpsrc.c: (gst_udpsrc_start): Joining a multicast group and setting the loop/ttl properties are totally unrelated tasks are must be separated. 2008-05-21 14:09:41 +0000 Stefan Kost gst/avi/gstavimux.c: Also support alaw/mulaw. Original commit message from CVS: * gst/avi/gstavimux.c: Also support alaw/mulaw. 2008-05-21 13:47:43 +0000 Wim Taymans gst/udp/gstmultiudpsink.*: Add a fixme for the auto-multicast property. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_setup_qos_dscp), (gst_multiudpsink_add_internal): * gst/udp/gstmultiudpsink.h: Add a fixme for the auto-multicast property. Fix some confusing debug messages. Disable setting a qos value by default. 2008-05-21 11:38:17 +0000 Gustaf Räntilä gst/udp/gstmultiudpsink.c: Ignore EPERM errors from sendto. Fixes #533619. Original commit message from CVS: Patch by: Gustaf Räntilä * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render): Ignore EPERM errors from sendto. Fixes #533619. 2008-05-21 10:51:52 +0000 Henrik Eriksson gst/udp/gstmultiudpsink.*: Add qos-dscp property to manage the Quality of service. Original commit message from CVS: Patch by: Henrik Eriksson * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_init), (gst_multiudpsink_setup_qos_dscp), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal): * gst/udp/gstmultiudpsink.h: Add qos-dscp property to manage the Quality of service. 2008-05-21 10:09:23 +0000 Wim Taymans gst/rtp/gstrtptheoradepay.c: Improve debugging of the ident. Original commit message from CVS: * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_process): Improve debugging of the ident. 2008-05-21 09:56:02 +0000 Bruno Santos gst/udp/gstudpnetutils.*: Provide a bunch of helper methods to deal with IPv4 and IPv6 transparently. Original commit message from CVS: Patch by: Bruno Santos * gst/udp/gstudpnetutils.c: (gst_udp_get_addr), (gst_udp_join_group), (gst_udp_leave_group), (gst_udp_is_multicast): * gst/udp/gstudpnetutils.h: Provide a bunch of helper methods to deal with IPv4 and IPv6 transparently. * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_init), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (join_multicast), (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal), (gst_multiudpsink_remove): * gst/udp/gstmultiudpsink.h: Add multicast TTL and loopback properties. Use the helper methods to implement ip4 and ip6. * gst/udp/gstudpsrc.c: (gst_udpsrc_create), (gst_udpsrc_start): * gst/udp/gstudpsrc.h: Use the helper methods to implement ip4 and ip6. Fixes #515962. 2008-05-21 09:38:48 +0000 Patrick Radizi gst/multipart/multipartdemux.*: Don't blindly copy the mime-type as the caps name because they not always map directl... Original commit message from CVS: Patch by: Patrick Radizi * gst/multipart/multipartdemux.c: (gst_multipart_demux_class_init), (gst_multipart_demux_get_gstname), (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain): * gst/multipart/multipartdemux.h: Don't blindly copy the mime-type as the caps name because they not always map directly. Instead use a hashtable with common mappings. Fixes #533287. 2008-05-20 17:27:35 +0000 Michael Meeks ext/esd/esdsink.c: When we post an error, we must return -1 to let the parent know that we cannot write the segment e... Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_write): When we post an error, we must return -1 to let the parent know that we cannot write the segment else it will loop and continue to call us again forever. Patch by Michael Meeks. 2008-05-20 14:24:21 +0000 Stefan Kost gst/videomixer/videomixer.c: Add missing incudes. Original commit message from CVS: * gst/videomixer/videomixer.c: Add missing incudes. 2008-05-20 13:57:44 +0000 Peter Kjellerstedt gst/rtp/gstrtph264pay.*: Correct a typo (sinle -> single). Original commit message from CVS: * gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type), (gst_rtp_h264_pay_handle_buffer): * gst/rtp/gstrtph264pay.h: Correct a typo (sinle -> single). 2008-05-20 11:33:05 +0000 Wim Taymans gst/rtp/gstrtph264depay.*: Add experimental support for outputting quicktime-like AVC output in addition to the exist... Original commit message from CVS: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init), (gst_rtp_h264_depay_set_property), (gst_rtp_h264_depay_get_property), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): * gst/rtp/gstrtph264depay.h: Add experimental support for outputting quicktime-like AVC output in addition to the existing bytestream output. * gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type), (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init), (gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property): * gst/rtp/gstrtph264pay.h: Make the parsing mode configurable, for some inputs we don't need to scan every byte for start codes. Only set the marker bit on ACCESS units. 2008-05-20 10:47:10 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Use a bigger type in integer mode for the intermediate results to prevent overflows.... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: Use a bigger type in integer mode for the intermediate results to prevent overflows. This fixes the crippled sound when using the equalizer in integer mode. Fixes bug #510865. 2008-05-20 10:42:33 +0000 Jan Schmidt gst/videomixer/videomixer.*: Instead of a random number for the request pad id's, use a counter. Original commit message from CVS: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: Instead of a random number for the request pad id's, use a counter. Register the videomixerpad class from the element's class_init where it's safer, and allows the docs generator to scan it. 2008-05-20 09:29:28 +0000 Wim Taymans gst/smpte/: Add new plugin that adds the SMPTE transition in the alpha channel of Original commit message from CVS: * gst/smpte/Makefile.am: * gst/smpte/gstsmpte.c: (gst_smpte_plugin_init): * gst/smpte/gstsmpte.h: * gst/smpte/gstsmptealpha.c: (gst_smpte_alpha_transition_type_get_type), (gst_smpte_alpha_get_type), (gst_smpte_alpha_base_init), (gst_smpte_alpha_class_init), (gst_smpte_alpha_update_mask), (gst_smpte_alpha_setcaps), (gst_smpte_alpha_get_unit_size), (gst_smpte_alpha_init), (gst_smpte_alpha_finalize), (gst_smpte_alpha_do_ayuv), (gst_smpte_alpha_do_i420), (gst_smpte_alpha_transform), (gst_smpte_alpha_set_property), (gst_smpte_alpha_get_property), (gst_smpte_alpha_plugin_init): * gst/smpte/gstsmptealpha.h: * gst/smpte/plugin.c: (plugin_init): Add new plugin that adds the SMPTE transition in the alpha channel of I420 and AYUV frames so that they can be blended with videomixer later on. Uses all niceties such as using base transform for efficient alloc and negotiation. It currently requires GstController to control the position in the transition effect. 2008-05-19 21:05:03 +0000 Stefan Kost Try using thaytans new mechanism to get extra classes into plugin docs. Aparently works for the Eq. For VideoMixer th... Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.types: * gst/videomixer/videomixer.c: Try using thaytans new mechanism to get extra classes into plugin docs. Aparently works for the Eq. For VideoMixer the GObject stuff is missing still. 2008-05-19 12:32:06 +0000 Sebastian Dröge tests/check/elements/deinterleave.c: Set keep-positions property to TRUE for the 8 channel test to ensure that the or... Original commit message from CVS: * tests/check/elements/deinterleave.c: (GST_START_TEST): Set keep-positions property to TRUE for the 8 channel test to ensure that the original channel position is set on the output. 2008-05-19 07:46:05 +0000 Sebastian Dröge gst/interleave/deinterleave.*: Add a property to select whether channel positions should be kept on the mono output b... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property), (gst_deinterleave_get_property): * gst/interleave/deinterleave.h: Add a property to select whether channel positions should be kept on the mono output buffers or should be dropped. 2008-05-18 19:27:59 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Set proper rate in avi stream header for PCM audio, and also do some more sanity checks on caps ... Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps): Set proper rate in avi stream header for PCM audio, and also do some more sanity checks on caps in this case. Fixes #511489. 2008-05-17 19:39:53 +0000 Sebastian Dröge gst/interleave/deinterleave.*: Queue events until src pads were added and they can be sent. Otherwise downstream will... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_finalize), (gst_deinterleave_init), (gst_deinterleave_sink_event), (gst_deinterleave_process), (gst_deinterleave_sink_activate_push): * gst/interleave/deinterleave.h: Queue events until src pads were added and they can be sent. Otherwise downstream will never get the first newsegment event. 2008-05-17 14:05:03 +0000 Sebastian Dröge gst/interleave/deinterleave.c: Always set the channel positions when gst_audio_get_channel_positions() returns someth... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps), (gst_deinterleave_getcaps): Always set the channel positions when gst_audio_get_channel_positions() returns something, even if they're not set in the caps. This makes sure that the output channels can be interleaved again correctly in the mono/stereo cases too. Don't ask for the peercaps of the current pad in getcaps() as this might call getcaps() again and deadlock. 2008-05-17 10:38:18 +0000 Sebastian Dröge sys/v4l2/gstv4l2src.c: Don't include the gstv4l2xoverlay.h header as the XOverlay support isn't implemented at all ye... Original commit message from CVS: * sys/v4l2/gstv4l2src.c: Don't include the gstv4l2xoverlay.h header as the XOverlay support isn't implemented at all yet and this requires X headers to be installed. Fixes bug #533264. 2008-05-16 21:56:24 +0000 Sebastian Dröge gst/interleave/: Add support for all raw audio formats and provide better negotiation if the caps are changing. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.c: (deinterleave_24), (gst_deinterleave_finalize), (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps), (gst_deinterleave_set_process_function), (gst_deinterleave_sink_setcaps), (__remove_channels), (__set_channels), (gst_deinterleave_getcaps), (gst_deinterleave_process), (gst_deinterleave_chain), (gst_deinterleave_sink_activate_push): * gst/interleave/deinterleave.h: Add support for all raw audio formats and provide better negotiation if the caps are changing. Don't allow changes of the channel positions and set the position of the corresponding channel on the src pad caps. General cleanup and smaller bugfixes. * tests/check/elements/deinterleave.c: (float_buffer_check_probe): Check the channel positions on the output buffer caps. 2008-05-16 17:50:20 +0000 Jan Schmidt Fix some compiler warnings. Original commit message from CVS: * ext/wavpack/gstwavpackstreamreader.c: * tests/examples/spectrum/demo-audiotest.c: * tests/examples/spectrum/demo-osssrc.c: Fix some compiler warnings. 2008-05-14 18:28:46 +0000 Wim Taymans gst/rtp/gstrtph264depay.c: Small comment added. Original commit message from CVS: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process): Small comment added. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_decode_nal), (gst_rtp_h264_pay_parse_sps_pps), (gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer): Debug string cleanups (remove trailing \n) Refactor and clean up the payloader a bit and make sure that we only put one NAL unit in an RTP packet even if the input buffer contains multiple NAL units. Add suport for AVC format input. 2008-05-14 17:58:50 +0000 Peter Kjellerstedt gst/rtp/gstrtph264pay.*: Make it possible to specify profile-level-id and sprop-parameter-sets using properties in ca... Original commit message from CVS: * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property): * gst/rtp/gstrtph264pay.h: Make it possible to specify profile-level-id and sprop-parameter-sets using properties in case they are not available in-stream. 2008-05-14 14:19:47 +0000 Tim-Philipp Müller tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, since it causes weird invalid free errors in ... Original commit message from CVS: * tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, since it causes weird invalid free errors in valgrind/libc after _exit for some reason. * tests/check/elements/deinterleave.c: (pads_created), (set_channel_positions), (src_handoff_float32_8ch), (float_buffer_check_probe), (pad_added_setup_data_check_float32_8ch_cb), (make_fake_src_8chans_float32), (GST_START_TEST), (deinterleave_suite): Add some more deinterleave unit test bits I had locally. 2008-05-14 12:52:15 +0000 Stefan Kost docs/plugins/: Remove ladspa fro plugin-docs, its in gst-plugins-bad. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-ladspa.xml: Remove ladspa fro plugin-docs, its in gst-plugins-bad. 2008-05-14 07:32:44 +0000 Sebastian Dröge gst/interleave/: Split definitions into separate header files for better documentation generation. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.h: * gst/interleave/interleave.h: * gst/interleave/plugin.h: Split definitions into separate header files for better documentation generation. * gst/interleave/deinterleave.c: (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps), (gst_deinterleave_process): Don't use alloca, allow caps changes as long as the number of channels does not change, don't use g_warning, return NOT_NEGOTIATED as early as possible and some other cleanup. * gst/interleave/interleave.c: (gst_interleave_base_init), (gst_interleave_class_init): Do some random cleanup. * tests/check/Makefile.am: * tests/check/elements/deinterleave.c: (GST_START_TEST), (deinterleave_chain_func), (deinterleave_pad_added), (deinterleave_suite): Add unit tests for the deinterleave element. 2008-05-13 20:25:20 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Send an initial BYTE segment to inform downstream of later seeking, and to forego sync attempts. Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_start_file): Send an initial BYTE segment to inform downstream of later seeking, and to forego sync attempts. 2008-05-13 08:59:41 +0000 Wim Taymans gst/rtp/gstrtpg729depay.c: Fix wrong caps string. Original commit message from CVS: * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_setcaps): Fix wrong caps string. 2008-05-13 08:35:55 +0000 Olivier Crete gst/rtp/: Added G729 pay and depayloaders. Fixes #532409. Original commit message from CVS: Based on patch by: Olivier Crete * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_base_init), (gst_rtp_g729_depay_class_init), (gst_rtp_g729_depay_init), (gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process), (gst_rtp_g729_depay_plugin_init): * gst/rtp/gstrtpg729depay.h: * gst/rtp/gstrtpg729pay.c: (gst_rtpg729pay_base_init), (gst_rtpg729pay_class_init), (gst_rtpg729pay_init), (gst_rtpg729pay_setcaps), (gst_rtp_g729_pay_plugin_init): * gst/rtp/gstrtpg729pay.h: Added G729 pay and depayloaders. Fixes #532409. 2008-05-13 08:21:26 +0000 Wim Taymans ext/speex/gstspeexdec.c: Fix the calculation of the duration of the concealment packets. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_sink_event): Fix the calculation of the duration of the concealment packets. 2008-05-12 18:27:24 +0000 Olivier Crete gst/rtp/: Add DV pay and depayloaders. Fixes #532423. Original commit message from CVS: Based on patch by: Olivier Crete * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_base_init), (gst_rtp_dv_depay_class_init), (gst_rtp_dv_depay_init), (parse_encode), (gst_rtp_dv_depay_setcaps), (calculate_difblock_location), (gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset), (gst_rtp_dv_depay_change_state), (gst_rtp_dv_depay_plugin_init): * gst/rtp/gstrtpdvdepay.h: * gst/rtp/gstrtpdvpay.c: (gst_dv_pay_mode_get_type), (gst_rtp_dv_pay_base_init), (gst_rtp_dv_pay_class_init), (gst_rtp_dv_pay_init), (gst_dv_pay_set_property), (gst_dv_pay_get_property), (gst_rtp_dv_pay_setcaps), (gst_dv_pay_negotiate), (include_dif), (gst_rtp_dv_pay_handle_buffer), (gst_rtp_dv_pay_plugin_init): * gst/rtp/gstrtpdvpay.h: Add DV pay and depayloaders. Fixes #532423. 2008-05-12 16:35:39 +0000 Mark Nauwelaerts gst/matroska/matroska-demux.c: Convert subtitle palette info in VobSub private data from VobSub's (buggy) RGB to YUV. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_dvd_clut_change_event): Convert subtitle palette info in VobSub private data from VobSub's (buggy) RGB to YUV. 2008-05-12 15:26:01 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Do not leave fourcc stream header field empty upon reset. Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_pad_reset): Do not leave fourcc stream header field empty upon reset. Fixes #519301. 2008-05-11 14:43:26 +0000 Jan Schmidt Add goom2k1 into the docs. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: Add goom2k1 into the docs. 2008-05-08 16:58:02 +0000 Wouter Cloetens gst/rtsp/gstrtspsrc.c: Support Digest authentication. Fixes #532065. Original commit message from CVS: Based on patch by: Wouter Cloetens * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string), (gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth): Support Digest authentication. Fixes #532065. 2008-05-08 10:20:52 +0000 Stefan Kost gst/level/gstlevel.c: Also support 32bit (e.g. whe having it after 'mad'). Add more notes about whats needed for libo... Original commit message from CVS: * gst/level/gstlevel.c: Also support 32bit (e.g. whe having it after 'mad'). Add more notes about whats needed for liboil acceleration. Simplify docs a bit. 2008-05-08 08:15:34 +0000 Sjoerd Simons gst/matroska/matroska-mux.c: Update the track duration if the old one was invalid. Original commit message from CVS: Patch by: Sjoerd Simons * gst/matroska/matroska-mux.c: (gst_matroska_mux_collected): Update the track duration if the old one was invalid. Fixes bug #532117. 2008-05-07 16:36:04 +0000 Ole André Vadla Ravnås gst/rtp/gstrtph264pay.c (gst_rtp_h264_pay_parse_sps_pps): Use GST_STR_NULL when trying to print sps and pps strings t... Original commit message from CVS: * gst/rtp/gstrtph264pay.c (gst_rtp_h264_pay_parse_sps_pps): Use GST_STR_NULL when trying to print sps and pps strings that could be NULL, as this might crash on some platforms. 2008-05-07 15:33:52 +0000 Haakon Sporsheim sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_setup_ddraw): Do IDirectDrawClipper_SetHWnd() if the window I... Original commit message from CVS: patch by: Haakon Sporsheim * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_setup_ddraw): Do IDirectDrawClipper_SetHWnd() if the window ID has already been set after creating the clipper. 2008-05-07 15:28:06 +0000 Haakon Sporsheim sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame): Added checking of surface lost case after an uns... Original commit message from CVS: patch by: Haakon Sporsheim * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame): Added checking of surface lost case after an unsuccessful IDirectDrawSurface7_Lock() call. If surface is lost, return GST_FLOW_OK. 2008-05-07 15:19:47 +0000 Haakon Sporsheim * ChangeLog: * sys/directdraw/gstdirectdrawsink.c: sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame, Original commit message from CVS: patch by: Haakon Sporsheim * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame, WndProc, gst_directdraw_sink_window_thread): Improved Windows message loop and fixed window destruction issue. When the window which DirectDraw is rendering to is destroyed, the render/show_frame function will return GST_FLOW_ERROR. Partially fixes #520885. 2008-05-07 15:09:10 +0000 Haakon Sporsheim sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_set_caps): Fixed mid stream resolution change bug, the offscr... Original commit message from CVS: patch by: Haakon Sporsheim * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_set_caps): Fixed mid stream resolution change bug, the offscreen surface is now released when set_caps is called. Partially fixes #520885. 2008-05-07 14:56:22 +0000 Ole André Vadla Ravnås * ChangeLog: * sys/directdraw/gstdirectdrawsink.c: sys/directdraw/gstdirectdrawsink.c Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc): Make it so that gst_directdraw_sink_buffer_alloc uses the right width/height. Especially when looking through the pool of buffers, make sure that the width/height of caps is used instead of the already negotiated dimensions. For example if a buffer with different caps is requested, i.e. higher resolution, the caller would get a buffer with the old dimensions and thus corrupt the heap. 2008-05-07 14:43:39 +0000 Ole André Vadla Ravnås * sys/directdraw/gstdirectdrawsink.c: sys/directdraw/gstdirectdrawsink.c Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc): Clear the flags on recycled buffers from buffer_alloc. Partially fixes #520885. The right fix this time. 2008-05-07 14:39:45 +0000 Ole André Vadla Ravnås * sys/directdraw/gstdirectdrawsink.c: sys/directdraw/gstdirectdrawsink.c Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc): Reverting previous commit, it had it all mixed up, was for a different patch (major automation screw-up). Sorry! 2008-05-07 13:48:28 +0000 Ole André Vadla Ravnås * ChangeLog: * sys/directdraw/gstdirectdrawsink.c: sys/directdraw/gstdirectdrawsink.c Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_buffer_alloc): Clear the flags on recycled buffers from buffer_alloc. Partially fixes #520885. 2008-05-07 11:22:51 +0000 Ole André Vadla Ravnås gst/rtp/gstrtpilbcpay.c: Added missing stdlib.h include for strtol(), and made include ordering and style consistent ... Original commit message from CVS: * gst/rtp/gstrtpilbcpay.c: Added missing stdlib.h include for strtol(), and made include ordering and style consistent with the corresponding depayloader. 2008-05-07 09:52:34 +0000 Ole André Vadla Ravnås gst/rtp/gstrtpilbcpay.c: Added missing stdlib.h include for strtol(), and made include ordering and style consistent ... Original commit message from CVS: * gst/rtp/gstrtpilbcpay.c: Added missing stdlib.h include for strtol(), and made include ordering and style consistent with the corresponding depayloader. 2008-05-07 08:03:51 +0000 Tim-Philipp Müller configure.ac: Error out if we don't have the required core/base versions. Original commit message from CVS: * configure.ac: Error out if we don't have the required core/base versions. 2008-05-06 09:33:46 +0000 Thijs Vermeir sys/osxvideo/cocoawindow.m: Fix compiler warnings on PPC64. Fixes bug #499318. Original commit message from CVS: Patch by: Thijs Vermeir * sys/osxvideo/cocoawindow.m: Fix compiler warnings on PPC64. Fixes bug #499318. 2008-05-05 11:19:13 +0000 Sjoerd Simons gst/rtsp/gstrtspsrc.c: Don't leak file descriptors on error. Fixes #531532. Original commit message from CVS: Patch by: Sjoerd Simons * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_open): Don't leak file descriptors on error. Fixes #531532. 2008-05-03 09:18:22 +0000 Sebastian Dröge ext/gconf/: When we can't create a fakesink/fakesrc complain instead of unreffing Original commit message from CVS: * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset), (gst_gconf_audio_src_change_state): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset), (gst_gconf_video_sink_change_state): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset), (gst_gconf_video_src_change_state): * ext/gconf/gstswitchsink.c: (gst_switch_sink_reset), (gst_switch_commit_new_kid), (gst_switch_sink_change_state): When we can't create a fakesink/fakesrc complain instead of unreffing NULL pointers and crashing later. See bug #530535. 2008-05-02 12:44:18 +0000 Wim Taymans gst/rtp/gstrtph263pdepay.c: Add some more debug info and guard against small payloads. Original commit message from CVS: * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process): Add some more debug info and guard against small payloads. * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process): Set duration on outgoing buffers because we can. 2008-05-02 12:39:03 +0000 Olivier Crete ext/speex/gstspeexenc.c: Add negotiation for the speex channels and rate. Fixes #465146. Original commit message from CVS: Patch by: Olivier Crete * ext/speex/gstspeexenc.c: (gst_speex_enc_sink_getcaps), (gst_speex_enc_init), (gst_speex_enc_chain): Add negotiation for the speex channels and rate. Fixes #465146. 2008-05-02 12:34:22 +0000 Olivier Crete gst/rtp/gstrtpspeexpay.c: Add negotiation for the speec channels and rate. See #465146. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_getcaps): Add negotiation for the speec channels and rate. See #465146. 2008-05-02 12:24:55 +0000 Olivier Crete gst/rtp/gstrtpilbcpay.c: Add negotiation for the ILBC mode. See #465146. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_class_init), (gst_rtpilbcpay_sink_setcaps), (gst_rtpilbcpay_sink_getcaps): Add negotiation for the ILBC mode. See #465146. 2008-05-02 11:32:31 +0000 Stefan Kost ext/soup/gstsouphttpsrc.c: Include stdlib to fix the build. Use g_free instead of free, libsoup uses glib. Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: Include stdlib to fix the build. Use g_free instead of free, libsoup uses glib. 2008-05-02 09:09:58 +0000 j^ gst/qtdemux/qtdemux.c: Add more mpeg2 variants. Fixes #530886. Original commit message from CVS: Patch by: j^ * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add more mpeg2 variants. Fixes #530886. 2008-05-01 10:52:11 +0000 Youness Alaoui gst/udp/gstudpsrc.c: Don't error out if we get an ICMP destination-unreachable message when trying to read packets on... Original commit message from CVS: Patch by: Youness Alaoui * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Don't error out if we get an ICMP destination-unreachable message when trying to read packets on win32 (#529454). 2008-04-30 12:18:41 +0000 Tim-Philipp Müller Use new error code for encrypted streams (which requires core CVS). Original commit message from CVS: * configure.ac: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Use new error code for encrypted streams (which requires core CVS). 2008-04-30 12:10:02 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix swapped pad template names, spotted by Thiago Sousa Santos. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_videosrc_template), (gst_qtdemux_audiosrc_template): Fix swapped pad template names, spotted by Thiago Sousa Santos. 2008-04-30 09:48:11 +0000 Wim Taymans ext/speex/gstspeexdec.c: Produce concealment data when time progresses in a segment update. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_sink_event), (speex_dec_chain_parse_data): Produce concealment data when time progresses in a segment update. 2008-04-29 14:11:45 +0000 Wim Taymans ext/speex/gstspeexdec.c: Try to preserve input timestamps when we can. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data), (speex_dec_chain): Try to preserve input timestamps when we can. Do beginnings of error concealment. 2008-04-28 22:38:11 +0000 Michael Smith gst/debug/gstnavigationtest.c: MSVC doesn't provide rint(), define an adequate replacement locally as elsewhere. Original commit message from CVS: * gst/debug/gstnavigationtest.c: MSVC doesn't provide rint(), define an adequate replacement locally as elsewhere. 2008-04-28 11:16:32 +0000 Julien Moutte gst/debug/rndbuffersize.c: Fix printf format to pacify Mac OSX's gcc. Original commit message from CVS: 2008-04-28 Julien Moutte * gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop): Fix printf format to pacify Mac OSX's gcc. 2008-04-25 19:34:31 +0000 Tim-Philipp Müller gst/debug/rndbuffersize.c: Bring rndbuffersize element into a state that doesn't require us to move it to -bad immedi... Original commit message from CVS: * gst/debug/rndbuffersize.c: (DEFAULT_SEED), (DEFAULT_MIN), (DEFAULT_MAX), (src_template), (sink_template), (gst_rnd_buffer_size_base_init), (gst_rnd_buffer_size_class_init), (gst_rnd_buffer_size_init), (gst_rnd_buffer_size_activate), (gst_rnd_buffer_size_loop), (gst_rnd_buffer_size_plugin_init): Bring rndbuffersize element into a state that doesn't require us to move it to -bad immediately. For one, fix up default min/max values so that the element actuall works using the default values. Also, don't ignore flow return values and do some kind of minimal eos logic. Allow min=max to pull fixed-sized buffers. Bunch of other gratuitious clean-ups. 2008-04-25 19:24:00 +0000 Tim-Philipp Müller docs/plugins/: Add docs for gdkpixbufsink; update docs to CVS version. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Add docs for gdkpixbufsink; update docs to CVS version. 2008-04-25 18:45:33 +0000 Wim Taymans tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Remove test sync-offset by default. Original commit message from CVS: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Remove test sync-offset by default. 2008-04-25 13:31:48 +0000 Tim-Philipp Müller gst/: Use GLib versions of htonl, htons, ntohl and ntohs in order to avoid problems on win32 (#529707). Original commit message from CVS: * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_chain): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add_internal): * gst/udp/gstudpsrc.c: (gst_udpsrc_start): Use GLib versions of htonl, htons, ntohl and ntohs in order to avoid problems on win32 (#529707). 2008-04-25 12:52:44 +0000 Jesús Corrius gst/goom/: Fix build with mingw32: use rand() instead of random() and replace bzero() with memset(). Fixes #529692. Original commit message from CVS: Patch by: Jesús Corrius * gst/goom/filters.c: (zoomVector): * gst/goom/goom_core.c: (init_buffers): Fix build with mingw32: use rand() instead of random() and replace bzero() with memset(). Fixes #529692. 2008-04-25 07:56:12 +0000 Wim Taymans gst/avi/gstavidemux.c: Fix typo in comments. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows): Fix typo in comments. * tests/examples/rtp/client-H263p-PCMA.sdp: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H264-PCMA.sdp: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-H264.sdp: * tests/examples/rtp/client-H264.sh: * tests/examples/rtp/client-PCMA.sdp: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Add some more docs and fix examples. 2008-04-24 22:04:57 +0000 Sebastian Dröge tests/check/elements/multifile.c: Include stdlib.h and unistd.h for mkdtemp. Some platforms have it declared in the f... Original commit message from CVS: * tests/check/elements/multifile.c: Include stdlib.h and unistd.h for mkdtemp. Some platforms have it declared in the former, some have it declared in the latter. 2008-04-24 22:01:52 +0000 Sebastian Dröge Stop using deprecated GLib functions. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_set_property): * gst/debug/tests.c: (md5_get_value): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps): * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps): * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps): * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps): Stop using deprecated GLib functions. 2008-04-24 21:17:42 +0000 Jan Schmidt configure.ac: Back to development -> 0.10.8.1 Original commit message from CVS: * configure.ac: Back to development -> 0.10.8.1 === release 0.10.8 === === release 0.10.8 === 2008-04-23 23:40:48 +0000 Jan Schmidt * NEWS: * RELEASE: Release 0.10.8 a little harder (edited the release notes) Original commit message from CVS: Release 0.10.8 a little harder (edited the release notes) 2008-04-23 23:26:24 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst-plugins-good.doap: * po/LINGUAS: * win32/common/config.h: Release 0.10.8 Original commit message from CVS: Release 0.10.8 2008-04-23 23:18:44 +0000 Jan Schmidt * common: * po/af.po: * po/az.po: * po/bg.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/it.po: * po/ja.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files Original commit message from CVS: Update .po files 2008-04-22 00:29:00 +0000 Jan Schmidt configure.ac: 0.10.7.4 pre-release Original commit message from CVS: * configure.ac: 0.10.7.4 pre-release 2008-04-22 00:18:52 +0000 Jan Schmidt gst/goom/: Free a bunch of stuff, and initialise things to fix leaks and valgrind warnings in the testsuite. Original commit message from CVS: * gst/goom/config_param.c: (goom_plugin_parameters_free): * gst/goom/convolve_fx.c: (convolve_init), (convolve_free): * gst/goom/filters.c: (zoomFilterVisualFXWrapper_free): * gst/goom/flying_stars_fx.c: (fs_free): * gst/goom/goom_config_param.h: * gst/goom/goom_core.c: (goom_init), (goom_close): * gst/goom/goom_plugin_info.h: * gst/goom/gstgoom.c: (gst_goom_finalize): * gst/goom/lines.c: (goom_lines_free): * gst/goom/plugin_info.c: (plugin_info_init), (plugin_info_free): * gst/goom/surf3d.c: (grid3d_free): * gst/goom/surf3d.h: * gst/goom/tentacle3d.c: (tentacle_free): Free a bunch of stuff, and initialise things to fix leaks and valgrind warnings in the testsuite. Fixes: #529268 2008-04-21 21:54:11 +0000 Sebastian Dröge tests/check/elements/rganalysis.c: Don't leak a tag list. Fixes bug #529285. Original commit message from CVS: * tests/check/elements/rganalysis.c: (GST_START_TEST): Don't leak a tag list. Fixes bug #529285. 2008-04-21 08:21:14 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Ref caps as the return value for the request_pt_map signal. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (request_pt_map), (gst_rtspsrc_configure_caps): Ref caps as the return value for the request_pt_map signal. Remove some caps weirdness when configuring a stream. See #528245. 2008-04-18 18:47:43 +0000 Tim-Philipp Müller tests/icles/gdkpixbufsink-test.c: Add cast to placate gcc 4.1.2. Original commit message from CVS: * tests/icles/gdkpixbufsink-test.c: Add cast to placate gcc 4.1.2. 2008-04-17 23:00:29 +0000 Jan Schmidt configure.ac: 0.10.7.3 pre-release Original commit message from CVS: * configure.ac: 0.10.7.3 pre-release 2008-04-17 22:32:16 +0000 Jan Schmidt tests/check/Makefile.am: Disable some more elements in the state test. Original commit message from CVS: * tests/check/Makefile.am: Disable some more elements in the state test. Add a define so the soup test can find the test files it needs at runtime. * tests/check/elements/souphttpsrc.c: (run_server): Add a define so the soup test can find the test files it needs at runtime. 2008-04-17 18:08:53 +0000 Jan Schmidt gst/goom/convolve_fx.c: Don't ever draw the GOOM logo. Original commit message from CVS: * gst/goom/convolve_fx.c: (convolve_apply): Don't ever draw the GOOM logo. Fixes: #528615 2008-04-17 10:24:32 +0000 Edward Hervey ext/: gst_atomic_int_set ==> g_atomic_int_set Original commit message from CVS: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdemux.c: gst_atomic_int_set ==> g_atomic_int_set 2008-04-16 10:31:17 +0000 Tim-Philipp Müller Strip out the config/script parsing stuff, we don't need it. Original commit message from CVS: * configure.ac: * gst/goom/Makefile.am: * gst/goom/convolve_fx.c: * gst/goom/default_scripts.h: * gst/goom/goom.h: * gst/goom/goom_core.c: (choose_a_goom_line): * gst/goom/goom_plugin_info.h: * gst/goom/goomsl.c: * gst/goom/goomsl.h: * gst/goom/goomsl_hash.c: * gst/goom/goomsl_hash.h: * gst/goom/goomsl_heap.c: * gst/goom/goomsl_heap.h: * gst/goom/goomsl_private.h: * gst/goom/plugin_info.c: Strip out the config/script parsing stuff, we don't need it. Fixes #527999. 2008-04-15 16:58:36 +0000 Tim-Philipp Müller gst/goom/plugin_info.c: Disable altivec optimisations for 32-bit PPC as well to make things build properly on all PPC... Original commit message from CVS: * gst/goom/plugin_info.c: (setOptimizedMethods): Disable altivec optimisations for 32-bit PPC as well to make things build properly on all PPC systems. Fixes #528143 2008-04-14 20:01:44 +0000 Tim-Philipp Müller gst-plugins-good.spec.in: Update for souphttpsrc plugin which has moved to -good. Original commit message from CVS: * gst-plugins-good.spec.in: Update for souphttpsrc plugin which has moved to -good. 2008-04-14 13:38:32 +0000 Mark Nauwelaerts gst/matroska/matroska-demux.c: Fix open-ended seeks in matroskademux Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_seek_event): Fix open-ended seeks in matroskademux Patch by: Mark Nauwelaerts Fixes: #526557 2008-04-13 23:13:32 +0000 Jan Schmidt tests/check/Makefile.am: Add soup test certificates to the dist. Original commit message from CVS: * tests/check/Makefile.am: Add soup test certificates to the dist. 2008-04-13 17:43:52 +0000 Jan Schmidt ext/Makefile.am: Remove LADSPA reference I missed. Original commit message from CVS: * ext/Makefile.am: Remove LADSPA reference I missed. 2008-04-13 13:06:39 +0000 Sebastian Dröge ext/soup/gstsouphttpsrc.c: Give souphttpsrc GST_RANK_PRIMARY to make it the default HTTP source over gnome-vfs and ev... Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: (plugin_init): Give souphttpsrc GST_RANK_PRIMARY to make it the default HTTP source over gnome-vfs and everything else. Fixes bug #527848. 2008-04-12 23:47:23 +0000 Jan Schmidt Remove LADSPA plugin. Fixes: #515978 Original commit message from CVS: * configure.ac: * ext/Makefile.am: Remove LADSPA plugin. Fixes: #515978 2008-04-12 23:30:54 +0000 Jan Schmidt Move soup plugin from -bad (Fixes: #523124) Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-soup.xml: * ext/Makefile.am: * tests/check/Makefile.am: Move soup plugin from -bad (Fixes: #523124) 2008-04-11 11:08:35 +0000 Jan Schmidt * ChangeLog: Fix the Changelog - actually speex <= 1.1.12 are vulnerable. Original commit message from CVS: Fix the Changelog - actually speex <= 1.1.12 are vulnerable. 2008-04-11 10:32:20 +0000 Jan Schmidt ext/speex/gstspeexdec.c: Fix bounds checking of mode in Speex header, which may produce negative numbers in speex < 1... Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_header): Fix bounds checking of mode in Speex header, which may produce negative numbers in speex < 1.1.12 2008-04-10 07:11:51 +0000 Sebastian Dröge tests/check/elements/souphttpsrc.c: Increase the timeout for the internet tests to 250 seconds and check for NULL cap... Original commit message from CVS: * tests/check/elements/souphttpsrc.c: (got_buffer), (souphttpsrc_suite): Increase the timeout for the internet tests to 250 seconds and check for NULL caps instead of just crashing. The real fix would be to implement an shoutcast server for the unit test instead of relying on a working internet connection. Fixes bug #521749. 2008-04-09 16:11:40 +0000 Tim-Philipp Müller gst/goom/: Remove a bunch of font/text related code that we don't need. Original commit message from CVS: * gst/goom/Makefile.am: * gst/goom/gfontlib.c: * gst/goom/gfontlib.h: * gst/goom/gfontrle.c: * gst/goom/gfontrle.h: * gst/goom/goom.h: * gst/goom/goom_core.c: (goom_update): * gst/goom/goom_plugin_info.h: * gst/goom/gstgoom.c: (gst_goom_chain): * gst/goom/plugin_info.c: Remove a bunch of font/text related code that we don't need. 2008-04-09 14:02:37 +0000 Tim-Philipp Müller gst/goom/: Change license of these files to LGPL, as permitted by the author, Guillaume Borios. See #515073. Original commit message from CVS: * gst/goom/ppc_drawings.s: * gst/goom/ppc_zoom_ultimate.s: Change license of these files to LGPL, as permitted by the author, Guillaume Borios. See #515073. 2008-04-09 13:31:22 +0000 Stefan Kost gst/goom/: As hinted in Bug #518213, revert one change and fix warnings properly. Original commit message from CVS: * gst/goom/convolve_fx.c: * gst/goom/motif_goom1.h: * gst/goom/motif_goom2.h: As hinted in Bug #518213, revert one change and fix warnings properly. This fixes both #518213 and #520073 for me. 2008-04-09 12:02:55 +0000 Jan Schmidt gst/matroska/: Fix the Forte build by making function declaration signatures match the implementations. Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_seek): * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_loop): Fix the Forte build by making function declaration signatures match the implementations. 2008-04-08 19:49:34 +0000 Tim-Philipp Müller sys/oss/: More logging when probing (see #518474), some comments in _reset(). Original commit message from CVS: * sys/oss/gstosshelper.c: (gst_oss_helper_rate_check_rate): * sys/oss/gstosssink.c: (gst_oss_sink_reset): * sys/oss/gstosssrc.c: (gst_oss_src_reset): More logging when probing (see #518474), some comments in _reset(). 2008-04-07 17:18:48 +0000 Julien Moutte gst/rtp/gstrtph264pay.c: Fix build because of a bad argument number. Original commit message from CVS: 2008-04-07 Julien Moutte * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Fix build because of a bad argument number. 2008-04-06 18:28:09 +0000 Tim-Philipp Müller tests/icles/: Interactive test app for gdkpixbufsink. Original commit message from CVS: * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/gdkpixbufsink-test.c: Interactive test app for gdkpixbufsink. 2008-04-06 09:01:42 +0000 Sjoerd Simons ext/soup/gstsouphttpsrc.c: Only ignore actual redirects not all responses when in state Original commit message from CVS: Patch by: Sjoerd Simons * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_response_cb): Only ignore actual redirects not all responses when in state GST_SOUP_HTTP_SRC_SESSION_IO_STATUS_RUNNING. Fixes bug #526337. 2008-04-06 08:57:59 +0000 Damien Lespiau configure.ac: Actually build dlls when cross-compiling with mingw32. Original commit message from CVS: Patch by: Damien Lespiau * configure.ac: Actually build dlls when cross-compiling with mingw32. Fixes bug #526247. 2008-04-05 12:00:46 +0000 Tim-Philipp Müller ext/hal/hal.c: Don't munge device string to 'default:x' for capture devices. Original commit message from CVS: * ext/hal/hal.c: (gst_hal_get_alsa_element): Don't munge device string to 'default:x' for capture devices. Fixes #525833. 2008-04-04 19:00:19 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Always use GSlice as we actually depend on GLib 2.12 already. Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_index_entry_free): Always use GSlice as we actually depend on GLib 2.12 already. 2008-04-04 11:26:40 +0000 Tim-Philipp Müller configure.ac: Require core/base 0.10.18 for ARGB caps parsing fixes in libgstvideo. Original commit message from CVS: * configure.ac: Require core/base 0.10.18 for ARGB caps parsing fixes in libgstvideo. Also bump the GLib requirement to the current de-facto requirement (ie. 2.12). 2008-04-04 10:32:21 +0000 Wim Taymans gst/rtp/gstrtph264pay.*: Parse codec_data for future AVC compatibility. Original commit message from CVS: * gst/rtp/gstrtph264pay.c: (encode_base64), (gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_handle_buffer): * gst/rtp/gstrtph264pay.h: Parse codec_data for future AVC compatibility. Fail when we encounter AVC data for now. 2008-04-04 09:50:10 +0000 Tim-Philipp Müller gst/spectrum/gstspectrum.c: Rename property enums and default defines for the properties to match the property names ... Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_set_property), (gst_spectrum_get_property), (gst_spectrum_message_new): Rename property enums and default defines for the properties to match the property names and rephrase property descriptions to make them a bit clearer (hopefully). See #518188. 2008-04-03 22:59:44 +0000 Tim-Philipp Müller tests/check/: Add unit test for gdkpixbufsink element. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/gdkpixbufsink.c: Add unit test for gdkpixbufsink element. 2008-04-03 22:50:48 +0000 Tim-Philipp Müller ext/gdk_pixbuf/: Add gdkpixbufsink element for easy snapshotting (#525946). Original commit message from CVS: * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init): * ext/gdk_pixbuf/gstgdkpixbufsink.c: (gst_gdk_pixbuf_sink_base_init), (gst_gdk_pixbuf_sink_class_init), (gst_gdk_pixbuf_sink_init), (gst_gdk_pixbuf_sink_start), (gst_gdk_pixbuf_sink_stop), (gst_gdk_pixbuf_sink_set_caps), (gst_gdk_pixbuf_sink_pixbuf_destroy_notify), (gst_gdk_pixbuf_sink_get_pixbuf_from_buffer), (gst_gdk_pixbuf_sink_handle_buffer), (gst_gdk_pixbuf_sink_preroll), (gst_gdk_pixbuf_sink_render), (gst_gdk_pixbuf_sink_set_property), (gst_gdk_pixbuf_sink_get_property): * ext/gdk_pixbuf/gstgdkpixbufsink.h: Add gdkpixbufsink element for easy snapshotting (#525946). 2008-04-03 20:25:34 +0000 Sebastian Dröge tests/check/pipelines/wavpack.c: Bump timeout from 3 to 60 seconds. Original commit message from CVS: * tests/check/pipelines/wavpack.c: (wavpack_suite): Bump timeout from 3 to 60 seconds. 2008-04-03 20:21:15 +0000 Sebastian Dröge tests/check/pipelines/.cvignore: Remove useless file. Original commit message from CVS: * tests/check/pipelines/.cvignore: Remove useless file. * tests/check/pipelines/.cvsignore: Add new test to .cvsignore. 2008-04-03 20:05:31 +0000 Sebastian Dröge tests/check/: Add unit test that encodes and decodes some data, checks that it is still the same and that all timesta... Original commit message from CVS: * tests/check/Makefile.am: * tests/check/pipelines/wavpack.c: (bus_handler), (identity_handoff), (fakesink_handoff), (GST_START_TEST), (wavpack_suite), (main): Add unit test that encodes and decodes some data, checks that it is still the same and that all timestamps/offsets are perfect. 2008-04-03 18:28:28 +0000 Sebastian Dröge ext/wavpack/: Use GSlice for allocating index entries and use gst_element_class_set_details_simple(). Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_index_entry_new), (gst_wavpack_parse_index_entry_free), (gst_wavpack_parse_base_init), (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset): Use GSlice for allocating index entries and use gst_element_class_set_details_simple(). 2008-04-02 22:37:29 +0000 Brian Cameron sys/sunaudio/: Fix up copyrights (#525860). Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudio.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixertrack.c: * sys/sunaudio/gstsunaudiomixertrack.h: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: * sys/sunaudio/gstsunaudiosrc.c: * sys/sunaudio/gstsunaudiosrc.h: Fix up copyrights (#525860). 2008-04-02 16:10:33 +0000 Christian Schaller * gst-plugins-good.spec.in: add new goom plugin to spec file Original commit message from CVS: add new goom plugin to spec file 2008-04-02 15:42:27 +0000 Tim-Philipp Müller gst/goom/goomsl.c: Check return value of fread() to avoid compiler warnings. Original commit message from CVS: * gst/goom/goomsl.c: (gsl_read_file): Check return value of fread() to avoid compiler warnings. 2008-04-01 11:00:43 +0000 mersad gst/law/: Make negotiation a bit modern. Original commit message from CVS: Based on patch by: mersad * gst/law/alaw-decode.c: (gst_alaw_dec_sink_setcaps), (gst_alaw_dec_chain), (gst_alaw_dec_change_state): * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: (gst_alaw_enc_chain): * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps), (gst_mulawdec_chain), (gst_mulawdec_change_state): * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.c: (gst_mulawenc_chain): Make negotiation a bit modern. Use pad_alloc. Fixes #525359. 2008-03-31 22:06:14 +0000 David Schleef gst/goom/xmmx.c: Fix constraints on asm code so that it compiles consistently. Fixes #522278. Original commit message from CVS: * gst/goom/xmmx.c: Fix constraints on asm code so that it compiles consistently. Fixes #522278. 2008-03-27 09:36:58 +0000 Brian Cameron sys/sunaudio/: Fix up the mixer tracks to use a volume range of 0-255, which is what the sun audio API uses. This sim... Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_get_volume), (gst_sunaudiomixer_ctrl_set_volume): * sys/sunaudio/gstsunaudiomixertrack.c: (gst_sunaudiomixer_track_new): Fix up the mixer tracks to use a volume range of 0-255, which is what the sun audio API uses. This simplifies the code and avoids rounding errors. Fixes #524593. 2008-03-26 15:10:08 +0000 Edgard Lima * ChangeLog: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: Add device-fd property to make it possible to apps to call ioctl's. Original commit message from CVS: Add device-fd property to make it possible to apps to call ioctl's. 2008-03-25 16:44:20 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Unbreak streaming mode again. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (next_entry_size): Unbreak streaming mode again. 2008-03-25 12:39:22 +0000 Tim-Philipp Müller sys/v4l2/v4l2src_calls.c: Remove superfluous DEBUG macro. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture): Remove superfluous DEBUG macro. 2008-03-25 12:33:09 +0000 William M. Brack sys/v4l2/v4l2src_calls.c: Check whether the device supports setting the framerate before trying to set it and then po... Original commit message from CVS: Based on patch by: William M. Brack * sys/v4l2/v4l2src_calls.c: (fractions_are_equal), (gst_v4l2src_set_capture): Check whether the device supports setting the framerate before trying to set it and then posting a warning or error if it doesn't work (#516649, #520092). Also compare fractions more correctly. 2008-03-24 12:32:59 +0000 Rene Stadler Make rganalysis and rglimiter elements GAP-flag aware. Original commit message from CVS: * gst/replaygain/gstrganalysis.c (gst_rg_analysis_init), (gst_rg_analysis_transform_ip): * gst/replaygain/gstrglimiter.c (gst_rg_limiter_init), (gst_rg_limiter_transform_ip): Make rganalysis and rglimiter elements GAP-flag aware. * tests/check/elements/rganalysis.c: (test_gap_buffers), (rganalysis_suite): * tests/check/elements/rglimiter.c (test_gap), (rglimiter_suite): Add tests to verify gap-awareness. 2008-03-23 13:31:15 +0000 Tim-Philipp Müller gst/goom/Makefile.am: Remove ppc assembler optimisations from the build until they actually build (they also seem to ... Original commit message from CVS: * gst/goom/Makefile.am: Remove ppc assembler optimisations from the build until they actually build (they also seem to have GPL headers). 2008-03-23 12:48:44 +0000 Tim-Philipp Müller m4/Makefile.am: Better not dist files that don't exist any longer (lrint*m4). Original commit message from CVS: * m4/Makefile.am: Better not dist files that don't exist any longer (lrint*m4). 2008-03-22 19:26:04 +0000 Sebastian Dröge ext/soup/gstsouphttpsrc.c: Don't autoplug souphttpsrc for dav/davs. This is better handled by Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb), (gst_soup_http_src_chunk_allocator), (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_uri_get_protocols): Don't autoplug souphttpsrc for dav/davs. This is better handled by GIO and GnomeVFS as they provide authentication. Don't leak the icy caps if we already set them and get a new icy-metaint header. Try harder to set the icy caps on the output buffer to have correct caps for the first buffer already. * tests/check/elements/souphttpsrc.c: (got_buffer), (GST_START_TEST): Check that we get a buffer with application/x-icy caps if iradio-mode is enabled and we have an icecast URL. 2008-03-22 18:18:46 +0000 Sebastian Dröge ext/soup/gstsouphttpsrc.c: Actually set the icy caps on our src pad if we have icecast data. Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_chunk_allocator): Actually set the icy caps on our src pad if we have icecast data. Fixes bug #523854. 2008-03-21 13:36:27 +0000 Sebastian Dröge Remove lrint/lrintf checks. We don't use it anywhere. Original commit message from CVS: * configure.ac: * m4/lrint.m4: * m4/lrintf.m4: Remove lrint/lrintf checks. We don't use it anywhere. 2008-03-19 19:56:59 +0000 Stefan Kost gst/freeze/: Add example to source code documentation blob and remove the 3 line Original commit message from CVS: * gst/freeze/FAQ: * gst/freeze/Makefile.am: * gst/freeze/gstfreeze.c: Add example to source code documentation blob and remove the 3 line FAQ. * gst/interleave/interleave.c: Add a source code documentation blob. 2008-03-18 15:03:06 +0000 Andy Wingo * ChangeLog: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy) Original commit message from CVS: 2008-03-18 Andy Wingo * sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy) (gst_osx_video_sink_osxwindow_new): Actually set a lock on the task, whoopdee. (cocoa_event_loop): Pacify the taymans by upping the usleepage to 2 ms. 2008-03-18 11:50:08 +0000 Andy Wingo sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy) Original commit message from CVS: 2008-03-18 Andy Wingo * sys/osxvideo/osxvideosink.m (gst_osx_video_sink_osxwindow_destroy) (gst_osx_video_sink_osxwindow_new, cocoa_event_loop): * sys/osxvideo/osxvideosink.h (struct _GstOSXVideoSink): If we need to run an event loop, do so in a task instead of assuming that there will be a GMainLoop. Fixes #523134. 2008-03-17 19:50:58 +0000 William M. Brack sys/v4l2/v4l2src_calls.c: Make sure the probed frame sizes are reversed in the resulting caps also when using V4L2_FR... Original commit message from CVS: Patch by: William M. Brack * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format_and_size), (gst_v4l2src_probe_caps_for_format): Make sure the probed frame sizes are reversed in the resulting caps also when using V4L2_FRMSIZE_STEPWISE (so they end up highest resolution first); also remove unused variable. (Partly fixes #520092) 2008-03-17 15:56:01 +0000 Ole André Vadla Ravnås gst/rtsp/gstrtspsrc.c: Call WSAStartup() and WSACleanup before using the Winsock API. Original commit message from CVS: Patch by: Ole André Vadla Ravnås * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize): Call WSAStartup() and WSACleanup before using the Winsock API. See #520808. 2008-03-16 15:01:07 +0000 Stefan Kost gst/avi/gstavidemux.c: Erm, the buffer-size is just guint, no need for the special format specifier. Original commit message from CVS: * gst/avi/gstavidemux.c: Erm, the buffer-size is just guint, no need for the special format specifier. 2008-03-16 14:34:45 +0000 Tim-Philipp Müller gst/goom/: Small fixes to build more on PPC: ifdef out code that uses unknown define; add newline at end of header fi... Original commit message from CVS: * gst/goom/plugin_info.c: * gst/goom/ppc_zoom_ultimate.h: Small fixes to build more on PPC: ifdef out code that uses unknown define; add newline at end of header file to avoid compiler warning. Assembler code still doesn't build though. 2008-03-16 14:04:16 +0000 Stefan Kost gst/avi/gstavidemux.c: Fix up my last commit. Use G_GUINT32_FORMAT for the guint32 debug log. Original commit message from CVS: * gst/avi/gstavidemux.c: Fix up my last commit. Use G_GUINT32_FORMAT for the guint32 debug log. Also downgrade a GST_WARNING to GST_DEBUG and add a comment. 2008-03-15 22:10:38 +0000 Stefan Kost gst/avi/gstavidemux.c: Chunksize is uint32. Fix format specifier. Original commit message from CVS: * gst/avi/gstavidemux.c: Chunksize is uint32. Fix format specifier. 2008-03-14 15:53:01 +0000 Christian Schaller * ChangeLog: * gst/rtsp/COPYING.MIT: fix license file, remove extra line copied over by mistake Original commit message from CVS: fix license file, remove extra line copied over by mistake 2008-03-13 14:30:45 +0000 Sebastian Dröge gst/audiofx/audiofx.c: Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values. Original commit message from CVS: * gst/audiofx/audiofx.c: Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values. 2008-03-13 09:45:09 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.*: Try to resume on server disconnect. Fixes bug #522134. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_init), (gst_soup_http_src_finished_cb), (gst_soup_http_src_response_cb), (gst_soup_http_src_build_message), (gst_soup_http_src_create): * ext/soup/gstsouphttpsrc.h: Try to resume on server disconnect. Fixes bug #522134. 2008-03-11 23:12:04 +0000 Mark Nauwelaerts sys/oss/gstosssrc.*: Cache probed caps, so _get_caps() during recording doesn't cause ioctl calls which may disrupt t... Original commit message from CVS: Patch by: Mark Nauwelaerts * sys/oss/gstosssrc.c: (gst_oss_src_init), (gst_oss_src_getcaps), (gst_oss_src_close): * sys/oss/gstosssrc.h: Cache probed caps, so _get_caps() during recording doesn't cause ioctl calls which may disrupt the recording (fixes #521875). 2008-03-11 16:23:04 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Make sure we always send a DISCONT after a seek by setting the sample index to an undefined va... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek), (gst_qtdemux_activate_segment), (gst_qtdemux_prepare_current_sample), (gst_qtdemux_loop_state_movie), (qtdemux_parse_trak): Make sure we always send a DISCONT after a seek by setting the sample index to an undefined value after a seek. 2008-03-11 15:18:43 +0000 Tim-Philipp Müller gst/avi/gstavisubtitle.h: Fix up IS_FOO macros, which makes gtk-doc much happier. Original commit message from CVS: * gst/avi/gstavisubtitle.h: (GST_IS_AVI_SUBTITLE), (GST_IS_AVI_SUBTITLE_CLASS): Fix up IS_FOO macros, which makes gtk-doc much happier. 2008-03-08 19:29:20 +0000 Tim-Philipp Müller tests/icles/Makefile.am: Move the -lgstfoo where it belongs. Original commit message from CVS: * tests/icles/Makefile.am: Move the -lgstfoo where it belongs. 2008-03-08 19:14:22 +0000 Tim-Philipp Müller * ChangeLog: ChangeLog surgery Original commit message from CVS: ChangeLog surgery 2008-03-08 04:40:32 +0000 Sebastian Dröge gst/matroska/ebml-ids.h: Add ID for EBML CRC32 elements. Original commit message from CVS: * gst/matroska/ebml-ids.h: Add ID for EBML CRC32 elements. * gst/matroska/Makefile.am: * gst/matroska/ebml-read.c: (gst_ebml_finalize), (gst_ebml_read_class_init), (gst_ebml_read_peek_bytes), (gst_ebml_read_get_length), (_ext2dbl), (gst_ebml_read_float), (gst_ebml_read_header): Support reading 80bit floats, add finalize method to clean up in any case, support reading length/id elements with any length as long as it's smaller than our supported maximum, don't leak buffers if reading as much data as we wanted failed and some smaller cleanup. 2008-03-08 04:21:34 +0000 Olivier Crete gst/rtp/gstrtph263pdepay.c: Check that a buffer is large enough before reading from it. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process): Check that a buffer is large enough before reading from it. Fixes bug #521102. 2008-03-07 15:54:09 +0000 Wim Taymans gst/udp/gstudpsrc.c: Fix compilation after removing the GstPollMode from the constructor. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_start): Fix compilation after removing the GstPollMode from the constructor. 2008-03-07 13:08:42 +0000 Sebastian Dröge Check for sinh(), cosh() and asinh() and define our own implementations if they're not available. Fixes bug #520880. Original commit message from CVS: * configure.ac: * gst/audiofx/Makefile.am: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/math_compat.h: Check for sinh(), cosh() and asinh() and define our own implementations if they're not available. Fixes bug #520880. 2008-03-07 12:40:18 +0000 Olivier Crete ext/speex/gstspeexenc.c: Unref the buffers only once when handling not-negotiated errors. Original commit message from CVS: Patch by: Olivier Crete * ext/speex/gstspeexenc.c: (gst_speex_enc_chain): Unref the buffers only once when handling not-negotiated errors. Fixes bug #520764. 2008-03-07 10:01:40 +0000 Ole André Vadla Ravnås gst/udp/gstudpsrc.c: Properly balance WSA_Cleanup with WSA_Startup. Original commit message from CVS: Patch by: Ole André Vadla Ravnås * gst/udp/gstudpsrc.c: (gst_udpsrc_finalize), (gst_udpsrc_start), (gst_udpsrc_stop): Properly balance WSA_Cleanup with WSA_Startup. Also make the poll controllable on windows. Fixes #520888. 2008-03-06 19:47:48 +0000 Wim Taymans gst/matroska/: Handle return values from pull_range in a more granular way to properly shut down on seeks. Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes), (gst_ebml_read_pull_bytes), (gst_ebml_read_element_id), (gst_ebml_read_element_length), (gst_ebml_peek_id), (gst_ebml_read_skip), (gst_ebml_read_buffer), (gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint), (gst_ebml_read_float), (gst_ebml_read_ascii), (gst_ebml_read_utf8), (gst_ebml_read_date), (gst_ebml_read_master), (gst_ebml_read_binary), (gst_ebml_read_header): * gst/matroska/ebml-read.h: * gst/matroska/matroska-demux.c: (gst_matroska_demux_combine_flows), (gst_matroska_demux_reset), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_add_stream), (gst_matroska_demux_handle_src_query), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_init_stream), (gst_matroska_demux_parse_tracks), (gst_matroska_demux_parse_index_cuetrack), (gst_matroska_demux_parse_index_pointentry), (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_metadata_id_simple_tag), (gst_matroska_demux_parse_metadata_id_tag), (gst_matroska_demux_parse_metadata), (gst_matroska_demux_sync_streams), (gst_matroska_demux_push_hdr_buf), (gst_matroska_demux_push_flac_codec_priv_data), (gst_matroska_demux_push_xiph_codec_priv_data), (gst_matroska_demux_add_wvpk_header), (gst_matroska_demux_check_subtitle_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_parse_cluster), (gst_matroska_demux_parse_contents_seekentry), (gst_matroska_demux_parse_contents), (gst_matroska_demux_loop_stream_parse_id), (gst_matroska_demux_loop_stream), (gst_matroska_demux_loop): * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: Handle return values from pull_range in a more granular way to properly shut down on seeks. Combine return values from push. Implement proper error handling. Prepare for handling seeking correctly. 2008-03-03 22:01:56 +0000 Jan Schmidt gst/matroska/ebml-read.c: Use GINT64 formatting constants from GLIB. Original commit message from CVS: * gst/matroska/ebml-read.c: Use GINT64 formatting constants from GLIB. * gst/matroska/matroska-demux.c: Add some guards to avoid a possible division by 0 and crashing with NULL events on some systems. Use gst_gdouble_to_guint64 somewhere instead of an implicit conversion. * gst/matroska/matroska-mux.c: Check for invalid timestamps in a bunch of places to avoid writing bogus durations into the output file. Fix some double<->gint64 conversions that weren't using gst_guint64_to_gdouble 2008-03-03 13:03:43 +0000 Peter Kjellerstedt configure.ac: Move the checks for bison, flex and as to the program section and the check for gcc inline asm to the c... Original commit message from CVS: * configure.ac: Move the checks for bison, flex and as to the program section and the check for gcc inline asm to the compiler characteristics section. 2008-03-03 12:10:55 +0000 Peter Kjellerstedt configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4... Original commit message from CVS: * configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#498222) 2008-02-29 12:35:24 +0000 Michael Smith gst/videomixer/videomixer.c: Don't call gst_object_sync_values() unless we have a valid timestamp. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_blend_buffers): Don't call gst_object_sync_values() unless we have a valid timestamp. 2008-02-29 06:18:55 +0000 David Schleef gst/matroska/: Fix Dirac mapping. I had previously added a VfW-type mapping, but it looks like Dirac will get a nati... Original commit message from CVS: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: Fix Dirac mapping. I had previously added a VfW-type mapping, but it looks like Dirac will get a native Matroska mapping, and this is the most likely method. 2008-02-28 23:56:30 +0000 David Schleef gst/avi/gstavimux.c: Add Dirac encoding Original commit message from CVS: * gst/avi/gstavimux.c: Add Dirac encoding 2008-02-28 11:51:24 +0000 Peter Kjellerstedt gst/udp/gstudpsrc.*: Port to GstPoll. See #505417. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop), (gst_udpsrc_stop): * gst/udp/gstudpsrc.h: Port to GstPoll. See #505417. 2008-02-28 08:37:44 +0000 Sebastian Dröge gst/law/mulaw-decode.c: Return GST_FLOW_NOT_NEGOTIATED when the caps are not set yet on the srcpad. We need rate and ... Original commit message from CVS: * gst/law/mulaw-decode.c: (gst_mulawdec_chain): Return GST_FLOW_NOT_NEGOTIATED when the caps are not set yet on the srcpad. We need rate and channels before we can do any processing. Fixes bug #519088. 2008-02-26 10:09:38 +0000 Jan Schmidt configure.ac: Detect and indicate if GCC inline assembly syntax is available. Original commit message from CVS: * configure.ac: Detect and indicate if GCC inline assembly syntax is available. * gst/goom/Makefile.am: * gst/goom/convolve_fx.c: * gst/goom/flying_stars_fx.c: * gst/goom/goom_config.h: * gst/goom/goom_core.c: * gst/goom/goomsl.c: * gst/goom/ifs.c: * gst/goom/mmx.c: * gst/goom/plugin_info.c: * gst/goom/xmmx.c: Fix various GCC-isms, and only build the inline assembly with compilers that support GCC inline assembly. Fix a couple of other warnings shown with Forte. 2008-02-26 05:36:17 +0000 Wouter Cloetens Add support for specifying a list of cookies to be passed in the HTTP request. Fixes bug #518722. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init), (gst_soup_http_src_init), (gst_soup_http_src_dispose), (gst_soup_http_src_set_property), (gst_soup_http_src_get_property), (gst_soup_http_src_create): * ext/soup/gstsouphttpsrc.h: * tests/check/elements/souphttpsrc.c: (run_test), (GST_START_TEST), (souphttpsrc_suite): Add support for specifying a list of cookies to be passed in the HTTP request. Fixes bug #518722. 2008-02-25 12:03:46 +0000 Stefan Kost gst/goom/xmmx.c: Use 'emms' instead of 'femms' to not crash on cpus that do not implement this 3dnow specific instruc... Original commit message from CVS: * gst/goom/xmmx.c: Use 'emms' instead of 'femms' to not crash on cpus that do not implement this 3dnow specific instruction. 2008-02-25 10:32:35 +0000 Sebastian Dröge gst/goom/plugin_info.c: Use extended MMX for draw_line() too if available, not only normal MMX. Original commit message from CVS: * gst/goom/plugin_info.c: (setOptimizedMethods): Use extended MMX for draw_line() too if available, not only normal MMX. 2008-02-25 06:50:31 +0000 Sebastian Dröge ext/jpeg/gstjpeg.c: Remove (commented out) smoke typefinder. This is in base now. Original commit message from CVS: * ext/jpeg/gstjpeg.c: (plugin_init): Remove (commented out) smoke typefinder. This is in base now. 2008-02-23 15:02:15 +0000 Jan Schmidt gst/goom2k1/: Rename the installed library, and don't register the same Original commit message from CVS: * gst/goom2k1/Makefile.am: * gst/goom2k1/gstgoom.c: Rename the installed library, and don't register the same GType name as the new goom. 2008-02-23 12:23:38 +0000 Tim-Philipp Müller Check for and define ERROR_CXXFLAGS and use them when building Original commit message from CVS: * configure.ac: * ext/taglib/Makefile.am: Check for and define ERROR_CXXFLAGS and use them when building C++ code (#516509). 2008-02-23 12:10:16 +0000 Tim-Philipp Müller gst/goom/: Call oil_init(), otherwise oil_get_cpu_flags() won't return anything useful. Export goom debug category so... Original commit message from CVS: * gst/goom/gstgoom.c: (goom_debug), (plugin_init): * gst/goom/plugin_info.c: (goom_debug), (GST_CAT_DEFAULT), (setOptimizedMethods): Call oil_init(), otherwise oil_get_cpu_flags() won't return anything useful. Export goom debug category so we can get rid of the VERBOSE define and the printfs. 2008-02-23 11:53:27 +0000 Tim-Philipp Müller gst/goom/: Compile fixes for x86-64. Original commit message from CVS: * gst/goom/goomsl_heap.c: (align_it): * gst/goom/plugin_info.c: (setOptimizedMethods): Compile fixes for x86-64. 2008-02-23 03:10:55 +0000 Bastien Nocera gst/goom/Makefile.am: Don't compile lex or yacc outputs with warnings, but add other CFLAGS Original commit message from CVS: * gst/goom/Makefile.am: Don't compile lex or yacc outputs with warnings, but add other CFLAGS * gst/goom/goomsl.c (gsl_instr_set_namespace), (gsl_instr_add_param), (iflow_execute), (gsl_enternamespace), (calculate_labels), (gsl_read_file): * gst/goom/goomsl_lex.l: * gst/goom/goomsl_yacc.y: * gst/goom/plugin_info.c: Remove a few live printf, and fprintf, replace exit() calls with g_assert_not_reached() if it not optimal for a library 2008-02-23 02:38:03 +0000 Bastien Nocera gst/goom/Makefile.am: Remove the warnings being disabled, fix linkage on x86, spotted by Sebastian Dröge Original commit message from CVS: * gst/goom/Makefile.am: Remove the warnings being disabled, fix linkage on x86, spotted by Sebastian Dröge * gst/goom/convolve_fx.c (convolve_init), (create_output_with_brightness), (convolve_apply): * gst/goom/filters.c (zoomFilterVisualFXWrapper_create): * gst/goom/goomsl.c: * gst/goom/ifs.c (ifs_update), (ifs_visualfx_create): * gst/goom/plugin_info.c: * gst/goom/tentacle3d.c (tentacle_fx_create): Fix warnings, and disable the motifs in the convolve_fx plugin (they were causing warnings, and they were just "Goom" in funny letterring) 2008-02-23 01:51:37 +0000 Bastien Nocera configure.ac: Add checks for Flex/Yacc/Bison and other furry animals, for the new goom 2k4 based plugin Original commit message from CVS: 2008-02-23 Bastien Nocera * configure.ac: Add checks for Flex/Yacc/Bison and other furry animals, for the new goom 2k4 based plugin * gst/goom/*: Update to use goom 2k4, uses liboil to detect CPU optimisations (not working yet), move the old plugin to... * gst/goom2k1/*: ... here, in case somebody is sick enough Fixes #515073 2008-02-22 14:55:57 +0000 Tim-Philipp Müller ext/lame/gstlame.c: Fix broken GST_ELEMENT_ERROR macro, fixes compile with the Sun Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_setcaps): Fix broken GST_ELEMENT_ERROR macro, fixes compile with the Sun Workshop 12 compiler, but probably also crashes (#517985). 2008-02-22 09:56:03 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Post the server response code in an error message instead of a generic 'error' message. Fixes ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Post the server response code in an error message instead of a generic 'error' message. Fixes #517237. 2008-02-22 07:20:03 +0000 Wouter Cloetens Implement zero-copy and make the buffer size configurable. Original commit message from CVS: Patch by: Wouter Cloetens * configure.ac: * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_cancel_message), (gst_soup_http_src_finished_cb), (gst_soup_http_src_chunk_free), (gst_soup_http_src_chunk_allocator), (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_create), (gst_soup_http_src_start), (gst_soup_http_src_set_proxy): * ext/soup/gstsouphttpsrc.h: Implement zero-copy and make the buffer size configurable. Prefix proxy URIs with "http://" if they don't start with it already and catch errors earlier, fixes hanging in some situations. Fixes bug #514948. 2008-02-22 06:22:39 +0000 Sebastian Dröge tests/check/Makefile.am: Ignore gconfaudiosrc for the states unit test too. It will fallback to alsasrc if the gconf ... Original commit message from CVS: * tests/check/Makefile.am: Ignore gconfaudiosrc for the states unit test too. It will fallback to alsasrc if the gconf settings can't be read and not everybody has alsa. 2008-02-22 06:06:06 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.*: Always report the duration if we know it in push mode and don't return 0 just to make ... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query), (gst_wavpack_parse_create_src_pad): * ext/wavpack/gstwavpackparse.h: Always report the duration if we know it in push mode and don't return 0 just to make totem believe we can't seek in push mode. Newer totem version use the SEEKING query which properly reports if we can seek or not. 2008-02-22 05:39:01 +0000 Jens Granseuer tests/examples/equalizer/demo.c: C89 fix, moving variable declarations to the beginning of the block. Fixes bug #517933. Original commit message from CVS: Patch by: Jens Granseuer * tests/examples/equalizer/demo.c: (main): C89 fix, moving variable declarations to the beginning of the block. Fixes bug #517933. 2008-02-21 23:47:37 +0000 Jan Schmidt configure.ac: Back to development... Original commit message from CVS: * configure.ac: Back to development... === release 0.10.7 === 2008-02-21 00:09:07 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.prerequisites: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * gst-plugins-good.doap: * po/LINGUAS: * win32/common/config.h: Release 0.10.7 - Red Door Black Original commit message from CVS: Release 0.10.7 - Red Door Black 2008-02-20 22:51:08 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/en_GB.po: * po/es.po: * po/eu.po: * po/fi.po: * po/hu.po: * po/it.po: * po/ja.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * po/zh_HK.po: * po/zh_TW.po: Update .po files Original commit message from CVS: Update .po files 2008-02-19 10:47:20 +0000 Sebastian Dröge gst/alpha/Makefile.am: Link alpha plugin with libgstbase. Fixes bug #517386. Original commit message from CVS: * gst/alpha/Makefile.am: Link alpha plugin with libgstbase. Fixes bug #517386. 2008-02-18 11:13:35 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Init values to -1 instead of the default 0 value. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream): Init values to -1 instead of the default 0 value. Fixes #516524. 2008-02-14 14:50:30 +0000 Stefan Kost tests/examples/spectrum/spectrum-example.c: Add missing include to fix compilation when libxml usage is disabled. Original commit message from CVS: * tests/examples/spectrum/spectrum-example.c: Add missing include to fix compilation when libxml usage is disabled. Fixes: #516371 2008-02-12 23:38:19 +0000 Wim Taymans fixes: #514889 Original commit message from CVS: patch by: Wim Taymans fixes: #514889 * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: Fix various leaks shown up in valgrind - free sprops and buffer in error cases in H264 payloader - fix leak in mp4g depayloader when construction the caps - don't leak config string in the mp4g payloader - don't leak buffers and headers in theora and vorbis payloaders * tests/check/elements/rtp-payloading.c: Fix the RTP data test - Actually send valid amr data to the payloader instead of 20 zero-bytes - The mp4g payloader expects codec_data on the caps 2008-02-12 21:36:40 +0000 Sébastien Moutte win32/MANIFEST: Add libgstpng.dsp to MANIFEST. Original commit message from CVS: * win32/MANIFEST: Add libgstpng.dsp to MANIFEST. * win32/vs6/libgstaudiofx.dsp: Add new source files to VS project file. 2008-02-12 13:34:52 +0000 Jan Schmidt sys/ximage/gstximagesrc.c: Initialise variables when opening the X display rather than in _start(), as the display ca... Original commit message from CVS: * sys/ximage/gstximagesrc.c: Initialise variables when opening the X display rather than in _start(), as the display can be opened before that. Fixes: #515985 2008-02-12 12:22:48 +0000 Sebastian Dröge sys/directdraw/gstdirectdrawsink.c: Properly chain up finalize functions. Fixes bug #515980. Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_class_init), (gst_ddrawsurface_finalize), (gst_directdraw_sink_finalize): Properly chain up finalize functions. Fixes bug #515980. 2008-02-12 11:38:54 +0000 Sebastian Dröge sys/v4l2/v4l2src_calls.c: Chain up the finalize functions. Fixes bug #515984. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize), (gst_v4l2_buffer_class_init), (gst_v4l2_buffer_pool_finalize), (gst_v4l2_buffer_pool_class_init): Chain up the finalize functions. Fixes bug #515984. 2008-02-12 11:14:36 +0000 Sebastian Dröge sys/ximage/ximageutil.c: Chain up in the finalize function for our custom buffer sub-class. Original commit message from CVS: * sys/ximage/ximageutil.c: Chain up in the finalize function for our custom buffer sub-class. Patch by: Sebastian Dröge Fixes: #515706 2008-02-12 11:12:43 +0000 Sebastian Dröge gst/debug/efence.c: Properly chain up finalize method. Fixes bug #515979. Original commit message from CVS: * gst/debug/efence.c: (gst_fenced_buffer_finalize), (gst_fenced_buffer_class_init): Properly chain up finalize method. Fixes bug #515979. 2008-02-12 11:09:08 +0000 Jan Schmidt sys/ximage/gstximagesrc.c: Free allocated Damage memory before closing our connection to the Original commit message from CVS: * sys/ximage/gstximagesrc.c: Free allocated Damage memory before closing our connection to the X server. Fixes: #515706 2008-02-12 05:21:46 +0000 Sebastian Dröge tests/check/elements/souphttpsrc.c: Include glib/gprintf.h for g_vasprintf(). Fixes bug #515564. Original commit message from CVS: * tests/check/elements/souphttpsrc.c: Include glib/gprintf.h for g_vasprintf(). Fixes bug #515564. 2008-02-12 05:14:16 +0000 Sebastian Dröge Add a few libjpeg suppressions and initialize a variable to make smokeenc valgrind clean. Fixes bug #515701. Original commit message from CVS: * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain): * tests/check/Makefile.am: * tests/check/gst-plugins-good.supp: Add a few libjpeg suppressions and initialize a variable to make smokeenc valgrind clean. Fixes bug #515701. 2008-02-11 21:24:30 +0000 Jan Schmidt gst/avi/gstavidemux.c: Revert patch which sends timestamps only on keyframes, as it breaks playback with current gst-... Original commit message from CVS: * gst/avi/gstavidemux.c: Revert patch which sends timestamps only on keyframes, as it breaks playback with current gst-ffmpeg. Fixes: #515562 2008-02-11 14:01:52 +0000 Sebastian Dröge Close some memory leaks spotted by the unit test. Fixes bug #515697. Original commit message from CVS: * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create): * tests/check/elements/multifile.c: (GST_START_TEST): Close some memory leaks spotted by the unit test. Fixes bug #515697. 2008-02-11 13:48:03 +0000 Sebastian Dröge ext/gconf/gconf.c: Use and unset the GError when pipeline creation fails instead of simply leaking it. Fixes bug #515... Original commit message from CVS: * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default): Use and unset the GError when pipeline creation fails instead of simply leaking it. Fixes bug #515704. 2008-02-11 09:13:33 +0000 Sebastian Dröge ext/lame/gstlame.c: Don't leak the allowed caps. Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_setup): Don't leak the allowed caps. * tests/check/pipelines/lame.c: (GST_START_TEST): Stop leaking all buffers. Fixes bug #515575. 2008-02-10 10:46:13 +0000 Sebastian Dröge gst/audiofx/: Fix long description of audiofx elements. Fixes bug #515457. Original commit message from CVS: * gst/audiofx/audioamplify.c: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiodynamic.c: * gst/audiofx/audioinvert.c: * gst/audiofx/audiopanorama.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: Fix long description of audiofx elements. Fixes bug #515457. 2008-02-09 01:45:32 +0000 Jan Schmidt Add a simple example application for the spectrum element, include it in the docs, and fix some documentation ambigui... Original commit message from CVS: * docs/plugins/Makefile.am: * gst/spectrum/gstspectrum.c: * tests/examples/spectrum/.cvsignore: * tests/examples/spectrum/Makefile.am: * tests/examples/spectrum/spectrum-example.c: Add a simple example application for the spectrum element, include it in the docs, and fix some documentation ambiguities. Fixes: #348085 2008-02-09 00:15:25 +0000 Jan Schmidt gst/: Fix includes order Original commit message from CVS: * gst/equalizer/Makefile.am: * gst/spectrum/Makefile.am: Fix includes order * tests/check/Makefile.am: Exclude v4l2src from the states test - it takes too long to start. * tests/check/elements/spectrum.c: Make the test run properly with CK_FORK=no 2008-02-08 15:32:36 +0000 Christian Schaller * gst-plugins-good.spec.in: add 3 new plugins to spec file Original commit message from CVS: add 3 new plugins to spec file 2008-02-08 15:27:51 +0000 Christian Schaller * ChangeLog: * gst/audiofx/Makefile.am: add missing header files for disting Original commit message from CVS: add missing header files for disting 2008-02-08 15:20:31 +0000 Julien Moutte gst/matroska/matroska-demux.c: Flag keyframe and delta units correctly when dealign with a Original commit message from CVS: 2008-02-08 Julien Moutte * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_blockgroup_or_simpleblock): Flag keyframe and delta units correctly when dealign with a BlockGroup. Fixes: #514397 2008-02-08 10:19:33 +0000 Jan Schmidt tests/check/elements/.cvsignore: Spell the new tests correctly in .cvsignore Original commit message from CVS: * tests/check/elements/.cvsignore: Spell the new tests correctly in .cvsignore 2008-02-08 10:09:33 +0000 Tim-Philipp Müller gst/multifile/gstmultifilesrc.c: Need to use gsize here for the size, fixes compiler warning. Original commit message from CVS: * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create): Need to use gsize here for the size, fixes compiler warning. * tests/examples/equalizer/.cvsignore: * tests/examples/equalizer/Makefile.am: * tests/examples/spectrum/.cvsignore: * tests/examples/spectrum/Makefile.am: Add missing files to fix the build. 2008-02-08 04:25:32 +0000 Jan Schmidt Move multifile plugin from -bad. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-multifile.xml: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: Move multifile plugin from -bad. Fixes: #490283 2008-02-08 03:44:12 +0000 David Schleef gst/multifile/: Use g_file_[sg]et_contents() instead of using stdio functions. Original commit message from CVS: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: Use g_file_[sg]et_contents() instead of using stdio functions. Should be less error prone. * tests/check/elements/multifile.c: Create a temporary directory using standard functions instead of creating a directory in the current dir. 2008-02-08 03:28:57 +0000 Jan Schmidt Move spectrum plugin from -bad. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-spectrum.xml: * gst/spectrum/Makefile.am: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/examples/Makefile.am: Move spectrum plugin from -bad. Move examples into tests/examples/spectrum. 2008-02-08 02:56:12 +0000 Jan Schmidt * ChangeLog: Mention bug 415627 fixed with previous commit Original commit message from CVS: Mention bug 415627 fixed with previous commit 2008-02-08 02:49:20 +0000 Jan Schmidt Move the equalizer plugin across from -bad Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/inspect/plugin-equalizer.xml: * gst/equalizer/Makefile.am: * tests/check/Makefile.am: * tests/examples/Makefile.am: Move the equalizer plugin across from -bad * tests/check/elements/.cvsignore: Add equalizer, audiosincwband and audiosincwlimit * tests/check/elements/equalizer.c: Fix compiler warnings 2008-02-08 02:48:54 +0000 Jan Schmidt docs/plugins/gst-plugins-bad-plugins.*: Remove equalizer plugin docs Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: Remove equalizer plugin docs * tests/check/Makefile.am: Add GST_OPTION_CFLAGS, to get -Werror -Wall into the tests as for other modules. * tests/check/elements/multifile.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: Fix compiler warnings from -Wall -Werror 2008-02-08 01:07:02 +0000 Jan Schmidt configure.ac: Only build with DISABLE_DEPRECATED during the CVS cycle. Pre-releases are treated like releases and bui... Original commit message from CVS: * configure.ac: Only build with DISABLE_DEPRECATED during the CVS cycle. Pre-releases are treated like releases and build without it. 2008-02-07 21:57:54 +0000 Jan Schmidt Move the lpwsinc and bpwsinc elements from gst-plugins-bad into the audiofx plugin, and rename to audiowsinclimit and... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-audiofx.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsincband.h: * gst/audiofx/audiowsinclimit.c: * gst/audiofx/audiowsinclimit.h: * tests/check/Makefile.am: * tests/check/elements/audiowsincband.c: * tests/check/elements/audiowsinclimit.c: Move the lpwsinc and bpwsinc elements from gst-plugins-bad into the audiofx plugin, and rename to audiowsinclimit and audiowsincband respectively. Fixes: #467666 2008-02-07 21:17:36 +0000 Tim-Philipp Müller Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without caps, and add a somewhat useful debug message. Plus test. Original commit message from CVS: * gst/icydemux/gsticydemux.c: (gst_icydemux_chain): * tests/check/elements/icydemux.c: Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without caps, and add a somewhat useful debug message. Plus test. 2008-02-07 19:13:56 +0000 Sébastien Moutte gst/rtsp/gstrtspsrc.c: Include unistd.h only if HAVE_UNISTD_H is defined Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: Include unistd.h only if HAVE_UNISTD_H is defined * win32/common/config.h.in: * win32/common/config.h: Define socklen_t as it seems it's not defined in default Visual Studio headers. * win32/vs6/libgstalpha.dsp: * win32/vs6/libgstapetag.dsp: * win32/vs6/libgstavi.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstvideomixer.dsp: Update project file dependencies and add new source files 2008-02-07 16:38:55 +0000 Bjarne Rosengren gst/matroska/ebml-write.c: Don't leak buffers when we don't push them downstream. Original commit message from CVS: Patch by: Bjarne Rosengren * gst/matroska/ebml-write.c: (gst_ebml_write_element_push): Don't leak buffers when we don't push them downstream. Fixes bug #514965. 2008-02-07 13:48:20 +0000 Stefan Kost gst/multifile/gstmultifilesink.c: Add a fixme comment. Original commit message from CVS: * gst/multifile/gstmultifilesink.c: Add a fixme comment. * gst/selector/gstoutputselector.c: Fix same leak as in input-selector. * tests/icles/output-selector-test.c: Improve the test. 2008-02-07 13:41:11 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Improve the docs. Original commit message from CVS: * gst/spectrum/gstspectrum.c: Improve the docs. 2008-02-07 10:17:14 +0000 Tim-Philipp Müller configure.ac: Bump requirements to (good) released versions to avoid confusion and make implicit core requirement exp... Original commit message from CVS: * configure.ac: Bump requirements to (good) released versions to avoid confusion and make implicit core requirement explicit. 2008-02-07 10:04:01 +0000 Sebastian Dröge gst/filter/gstlpwsinc.c: Fix typo in the long description of the element. Original commit message from CVS: * gst/filter/gstlpwsinc.c: Fix typo in the long description of the element. 2008-02-06 23:44:43 +0000 Jan Schmidt Rename audiochebyshevfreqband -> audiochebband and audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-audiofx.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiochebband.c: * gst/audiofx/audiochebband.h: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiocheblimit.h: * gst/audiofx/audiochebyshevfreqband.c: * gst/audiofx/audiochebyshevfreqband.h: * gst/audiofx/audiochebyshevfreqlimit.c: * gst/audiofx/audiochebyshevfreqlimit.h: * gst/audiofx/audiofx.c: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/audiochebband.c: * tests/check/elements/audiocheblimit.c: * tests/check/elements/audiochebyshevfreqband.c: * tests/check/elements/audiochebyshevfreqlimit.c: Rename audiochebyshevfreqband -> audiochebband and audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS surgery. Closes: #491811 2008-02-06 11:07:47 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.c: Fix memory leak and improve debugging a bit. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_create): Fix memory leak and improve debugging a bit. 2008-02-05 17:59:24 +0000 orjan gst/multipart/multipartmux.c: Fix caps memory leak. Fixes #514573. Original commit message from CVS: Patch by: orjan * gst/multipart/multipartmux.c: (gst_multipart_mux_collected): Fix caps memory leak. Fixes #514573. 2008-02-04 12:07:14 +0000 Edward Hervey gst/avi/gstavidemux.c: If there's no entries in the subindex, don't try to do anything stupid, just return. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex): If there's no entries in the subindex, don't try to do anything stupid, just return. 2008-02-02 19:47:50 +0000 John Millikin ext/flac/gstflacdec.c: Fix extraction of picture blocks with newer libflac versions again: Original commit message from CVS: Patch by: John Millikin * ext/flac/gstflacdec.c: (gst_flac_dec_scan_for_last_block), (gst_flac_extract_picture_buffer), (gst_flac_dec_metadata_callback): Fix extraction of picture blocks with newer libflac versions again: FLAC__METADATA_TYPE_PICTURE is an enum, not a define (#513628). 2008-02-02 18:06:19 +0000 Tim-Philipp Müller tests/check/Makefile.am: Add rtp-payloading test to VALGRIND_TO_FIX. Original commit message from CVS: * tests/check/Makefile.am: Add rtp-payloading test to VALGRIND_TO_FIX. * tests/check/elements/rtp-payloading.c: Add semicolons after GST_TEST_END so gst-indent gets the formatting right; make test less verbose in general, but more verbose in the error case (which should probably make the test fail anyway). 2008-02-01 18:29:21 +0000 Thijs Vermeir Add documentation for avisubtitle and change class to Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/avi/gstavisubtitle.c: Add documentation for avisubtitle and change class to Codec/Parser/Subtitle 2008-01-31 16:12:28 +0000 Jan Schmidt sys/v4l2/v4l2_calls.c: Treat ENOTTY (driver does not implement ioctl) the same as Original commit message from CVS: * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): Treat ENOTTY (driver does not implement ioctl) the same as EINVAL since it implies there are no available standards. * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format), (gst_v4l2src_get_nearest_size): Replace gst_v4l2src_get_size_limits with 2 calls to new function gst_v4l2src_get_nearest_size, and get it to use VIDIOC_S_FMT to probe if the driver does not support VIDIOC_TRY_FMT for whatever reason, and if we aren't yet actively capturing. * sys/v4l2/v4l2src_calls.h: Remove replaced function declaration. 2008-01-31 16:03:48 +0000 Jan Schmidt configure.ac: Bump plugins-base requirement to 0.10.16 for the gst_video_format_* Original commit message from CVS: * configure.ac: Bump plugins-base requirement to 0.10.16 for the gst_video_format_* API. 2008-01-31 09:50:31 +0000 Sebastian Dröge ext/soup/gstsouphttpsrc.c: Add changes to gstsouphttpsrc.c that were missing from last commit. Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: (_do_init), (gst_soup_http_src_base_init), (gst_soup_http_src_class_init), (gst_soup_http_src_init), (gst_soup_http_src_dispose), (gst_soup_http_src_set_property), (gst_soup_http_src_get_property), (gst_soup_http_src_unicodify), (gst_soup_http_src_cancel_message), (gst_soup_http_src_queue_message), (gst_soup_http_src_add_range_header), (gst_soup_http_src_session_unpause_message), (gst_soup_http_src_session_pause_message), (gst_soup_http_src_session_close), (gst_soup_http_src_got_headers_cb), (gst_soup_http_src_got_body_cb), (gst_soup_http_src_finished_cb), (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_response_cb), (gst_soup_http_src_parse_status), (gst_soup_http_src_create), (gst_soup_http_src_start), (gst_soup_http_src_stop), (gst_soup_http_src_unlock), (gst_soup_http_src_unlock_stop), (gst_soup_http_src_get_size), (gst_soup_http_src_is_seekable), (gst_soup_http_src_do_seek), (gst_soup_http_src_set_location), (gst_soup_http_src_set_proxy), (gst_soup_http_src_uri_get_type), (gst_soup_http_src_uri_get_protocols), (gst_soup_http_src_uri_get_uri), (gst_soup_http_src_uri_set_uri), (gst_soup_http_src_uri_handler_init), (plugin_init): Add changes to gstsouphttpsrc.c that were missing from last commit. 2008-01-31 08:57:16 +0000 Wouter Cloetens Make coding style more consistent, including class renaming. Original commit message from CVS: Patch by: Wouter Cloetens * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/inspect/plugin-soup.xml: (gst_soup_http_src_base_init), (gst_soup_http_src_class_init), (gst_soup_http_src_init), (gst_soup_http_src_dispose), (gst_soup_http_src_set_property), (gst_soup_http_src_get_property), (gst_soup_http_src_unicodify), (gst_soup_http_src_cancel_message), (gst_soup_http_src_queue_message), (gst_soup_http_src_add_range_header), (gst_soup_http_src_session_unpause_message), (gst_soup_http_src_session_pause_message), (gst_soup_http_src_session_close), (gst_soup_http_src_got_headers_cb), (gst_soup_http_src_got_body_cb), (gst_soup_http_src_finished_cb), (gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_response_cb), (gst_soup_http_src_parse_status), (gst_soup_http_src_create), (gst_soup_http_src_start), (gst_soup_http_src_stop), (gst_soup_http_src_unlock), (gst_soup_http_src_unlock_stop), (gst_soup_http_src_get_size), (gst_soup_http_src_is_seekable), (gst_soup_http_src_do_seek), (gst_soup_http_src_set_location), (gst_soup_http_src_set_proxy), (gst_soup_http_src_uri_get_type), (gst_soup_http_src_uri_get_protocols), (gst_soup_http_src_uri_get_uri), (gst_soup_http_src_uri_set_uri), (gst_soup_http_src_uri_handler_init), (plugin_init): * ext/soup/gstsouphttpsrc.h: Make coding style more consistent, including class renaming. 2008-01-31 00:03:26 +0000 Jan Schmidt configure.ac: Fix typo. Original commit message from CVS: * configure.ac: Fix typo. 2008-01-31 00:00:23 +0000 Jan Schmidt gst/alpha/: Re-write the 'alpha' plugin to be BaseTransform based, simplifying some stuff, and making buffer-alloc an... Original commit message from CVS: * gst/alpha/Makefile.am: * gst/alpha/gstalpha.c: Re-write the 'alpha' plugin to be BaseTransform based, simplifying some stuff, and making buffer-alloc and resizing work automatically. No longer crashes on odd frame widths and heights, although there seems to be a disagreement with ffmpegcolorspace about what size an AYUV frame with odd height should be. 2008-01-30 15:40:36 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.c: Update documentation a bit. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: Update documentation a bit. * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-glimagesink.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rawparse.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-switch.xml: * docs/plugins/inspect/plugin-videocrop.xml: Regenerate everything for the documentation changes we had. 2008-01-30 13:29:15 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.c: Let the proxy property default to the content of the $http_proxy environment variable. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_init): Let the proxy property default to the content of the $http_proxy environment variable. 2008-01-30 13:08:45 +0000 Wouter Cloetens tests/check/: Add missing files for the unit test. Original commit message from CVS: Patch by: Wouter Cloetens * tests/check/test-cert.pem: * tests/check/test-key.pem: Add missing files for the unit test. 2008-01-30 13:06:01 +0000 Wouter Cloetens docs/plugins/: Add souphttpsrc to the docs. Original commit message from CVS: Patch by: Wouter Cloetens * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: Add souphttpsrc to the docs. * configure.ac: * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init), (gst_souphttp_src_init), (gst_souphttp_src_dispose), (gst_souphttp_src_set_property), (gst_souphttp_src_get_property), (gst_souphttp_src_cancel_message), (gst_souphttp_src_queue_message), (gst_souphttp_src_add_range_header), (gst_souphttp_src_session_unpause_message), (gst_souphttp_src_session_pause_message), (gst_souphttp_src_session_close), (gst_souphttp_src_got_headers_cb), (gst_souphttp_src_got_body_cb), (gst_souphttp_src_finished_cb), (gst_souphttp_src_got_chunk_cb), (gst_souphttp_src_response_cb), (gst_souphttp_src_parse_status), (gst_souphttp_src_create), (gst_souphttp_src_start), (gst_souphttp_src_stop), (gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop), (gst_souphttp_src_get_size), (gst_souphttp_src_is_seekable), (gst_souphttp_src_do_seek), (gst_souphttp_src_set_location), (gst_souphttp_src_set_proxy), (plugin_init): * ext/soup/gstsouphttpsrc.h: Add support for libsoup2.4 and require it. Also implement redirection and manual proxy specification. Fixes bug #510708. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/souphttpsrc.c: Add unit test for souphttpsrc. 2008-01-29 18:43:32 +0000 Alessandro Decina ext/libpng/gstpngenc.*: Preallocate the output buffer so that g_memdup() and gst_buffer_merge() aren't needed anymore... Original commit message from CVS: Patch by: Alessandro Decina * ext/libpng/gstpngenc.c: (user_write_data), (gst_pngenc_chain): * ext/libpng/gstpngenc.h: Preallocate the output buffer so that g_memdup() and gst_buffer_merge() aren't needed anymore. This greatly improves performances and fixes #512544. 2008-01-29 18:24:28 +0000 Wim Taymans gst/avi/gstavidemux.c: GStreamer timestamps are PTS values while AVI only knows about DTS timestamps. Make sure we on... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data): GStreamer timestamps are PTS values while AVI only knows about DTS timestamps. Make sure we only copy the DTS as the buffer timestamp when we are dealing with a key frame. 2008-01-29 15:45:48 +0000 Stefan Kost tests/check/: Add add testsuite for the rtp-payloader that tries simulating dataflow. Needs more test data. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/rtp-payloading.c: Add add testsuite for the rtp-payloader that tries simulating dataflow. Needs more test data. 2008-01-29 15:27:02 +0000 Stefan Kost tests/check/elements/alphacolor.c: Remove two unused variables. Original commit message from CVS: * tests/check/elements/alphacolor.c: Remove two unused variables. 2008-01-28 12:17:02 +0000 Tim-Philipp Müller gst/rtsp/gstrtspsrc.c: Use g_ascii_strtoll() instead of atoll, which is only available in C99. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo): Use g_ascii_strtoll() instead of atoll, which is only available in C99. 2008-01-26 16:19:26 +0000 Sebastian Dröge gst/filter/: Don't implement get_unit_size() ourselves, the GstAudioFilter base class already does this for us. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): Don't implement get_unit_size() ourselves, the GstAudioFilter base class already does this for us. 2008-01-25 10:53:17 +0000 Thijs Vermeir gst/rtp/: Add MPEG2 video payloader Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtpmpvpay.h: Add MPEG2 video payloader 2008-01-23 17:05:32 +0000 Sebastian Dröge gst/level/gstlevel.c: Use #include instead of #include "math.h". Original commit message from CVS: * gst/level/gstlevel.c: Use #include instead of #include "math.h". 2008-01-21 19:41:45 +0000 Jan Schmidt tests/check/Makefile.am: Fix up some CFLAGS sets. Original commit message from CVS: * tests/check/Makefile.am: Fix up some CFLAGS sets. Don't include gconfvideosrc in the states test. * tests/check/elements/autodetect.c: (GST_START_TEST): Add some error strings to fail_unless arguments to fix some weird compiler errors on Solaris. 2008-01-21 19:35:58 +0000 Brian Cameron configure.ac: Detect video4linux headers on Solaris too. Original commit message from CVS: * configure.ac: Detect video4linux headers on Solaris too. * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2object.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize), (gst_v4l2_buffer_new): Make v4l2 build on Solaris. Patch by: Brian Cameron Fixes: #510505 2008-01-21 11:46:19 +0000 Stefan Kost docs/plugins/gst-plugins-good-plugins-docs.sgml: Update list from (still local) scanning script. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: Update list from (still local) scanning script. 2008-01-21 09:57:07 +0000 Stefan Kost docs/plugins/: Add symbols from -unused.txt to the right place. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-sections.txt: Add symbols from -unused.txt to the right place. * gst/dvdspu/gstdvdspu.c: * gst/dvdspu/gstdvdspu.h: Coherent namespace usage. * gst/spectrum/gstspectrum.c: Fix broken XML fragment in doc snippet even more. 2008-01-21 07:54:02 +0000 Stefan Kost docs/plugins/Makefile.am: Update include list. Original commit message from CVS: * docs/plugins/Makefile.am: Update include list. * docs/plugins/gst-plugins-bad-plugins-docs.sgml: Update xml includes. * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvbsrc.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstinterlace.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-switch.xml: * docs/plugins/inspect/plugin-timidity.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoparse.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: Regenerate files. * gst/spectrum/gstspectrum.c: Fix broken XML fragment in doc snippet. * tests/check/elements/.cvsignore: Add test binary to ignores. 2008-01-20 05:07:52 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.c: Report the size of the stream as the total size instead of the remaining Content-Length, w... Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (soup_got_headers): Report the size of the stream as the total size instead of the remaining Content-Length, which is wrong after a seek. 2008-01-19 14:59:08 +0000 Sebastian Dröge * ChangeLog: Add bug number to the latest entry Original commit message from CVS: Add bug number to the latest entry 2008-01-19 14:53:58 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Set variable to NULL after freeing it to prevent double frees or make failures by another... Original commit message from CVS: Based on a patch by: Victor STINNER * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Set variable to NULL after freeing it to prevent double frees or make failures by another use of it afterwards more obvious and fix use of it after the freeing. 2008-01-19 14:34:50 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.c: Correctly set duration on the GstBaseSrc segment when we know it to fix failing the durati... Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (soup_got_headers): Correctly set duration on the GstBaseSrc segment when we know it to fix failing the duration query. 2008-01-18 13:40:38 +0000 Thijs Vermeir gst/udp/gstmultiudpsink.c: use GST_WARNING for logging Original commit message from CVS: * gst/udp/gstmultiudpsink.c: use GST_WARNING for logging 2008-01-18 10:05:53 +0000 Sebastian Dröge gst/multifile/gstmultifilesrc.c: Fix memory leak spotted by the unit test. Original commit message from CVS: * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create): Fix memory leak spotted by the unit test. 2008-01-18 10:04:25 +0000 Thijs Vermeir gst/udp/gstmultiudpsink.c: Don't try to leave a multicast group with an invalid socket Original commit message from CVS: * gst/udp/gstmultiudpsink.c: Don't try to leave a multicast group with an invalid socket 2008-01-18 08:49:59 +0000 Sebastian Dröge tests/check/: Add some minimal tests for the equalizer plugin. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/equalizer.c: (setup_equalizer), (cleanup_equalizer), (GST_START_TEST), (equalizer_suite), (main): Add some minimal tests for the equalizer plugin. 2008-01-18 07:03:23 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Unparent all bands from the equalizer when finalizing to stop leaking them. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_finalize): Unparent all bands from the equalizer when finalizing to stop leaking them. 2008-01-18 05:32:26 +0000 Sebastian Dröge ext/soup/gstsouphttpsrc.c: Add support for WebDAV. Original commit message from CVS: * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_uri_get_protocols): Add support for WebDAV. 2008-01-18 05:24:39 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.*: Add support for seeking to souphttpsrc. Fixes bug #502335. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init), (gst_souphttp_src_init), (gst_souphttp_src_create), (gst_souphttp_src_is_seekable), (gst_souphttp_src_do_seek), (soup_add_range_header), (soup_got_headers), (soup_got_chunk): * ext/soup/gstsouphttpsrc.h: Add support for seeking to souphttpsrc. Fixes bug #502335. 2008-01-17 21:23:32 +0000 Tim-Philipp Müller ext/flac/gstflacdec.c: where the picture metadata defines and structs don't exist yet. Original commit message from CVS: * ext/flac/gstflacdec.c: Fix compilation against flac 1.1.2 (as on debian stable), where the picture metadata defines and structs don't exist yet. Fixes #509301. 2008-01-17 17:26:48 +0000 Zaheer Abbas Merali ext/lame/gstlame.*: Fix the case where you initially have stereo input, and so lame's mode is not set to mono, and th... Original commit message from CVS: * ext/lame/gstlame.c: * ext/lame/gstlame.h: Fix the case where you initially have stereo input, and so lame's mode is not set to mono, and then you get input with mono audio and soon after you get stereo input again. What happened before this commit is that it would keep the encoding mode as mono. It should change it back to the one requested by the app (or the default one) if not requested. 2008-01-17 11:13:16 +0000 Olivier Crete gst/udp/gstmultiudpsink.*: Add property to automatically join a multicast group or not. This can be useful when shari... Original commit message from CVS: Patch by: Olivier Crete * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_init), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal), (gst_multiudpsink_remove): * gst/udp/gstmultiudpsink.h: Add property to automatically join a multicast group or not. This can be useful when sharing a socket between multiple elements. Fixes #509531. 2008-01-16 21:53:41 +0000 Stefan Kost gst/videomixer/Makefile.am: Add controller flags. Original commit message from CVS: * gst/videomixer/Makefile.am: Add controller flags. 2008-01-16 20:17:08 +0000 Stefan Kost gst/videomixer/videomixer.c: Also commit the missing gst_object_sync_values(). Original commit message from CVS: * gst/videomixer/videomixer.c: Also commit the missing gst_object_sync_values(). 2008-01-16 08:11:46 +0000 Stefan Kost docs/plugins/Makefile.am: Remove duplicate entry. Original commit message from CVS: * docs/plugins/Makefile.am: Remove duplicate entry. 2008-01-15 16:52:10 +0000 Stefan Kost docs/plugins/: Add 3 more plugins to docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-gamma.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-video4linux2.xml: Add 3 more plugins to docs. 2008-01-15 16:04:44 +0000 Stefan Kost Revert previous change caused by a file that got stuck on an old revision. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-sections.txt: * sys/osxvideo/osxvideosink.h: Revert previous change caused by a file that got stuck on an old revision. 2008-01-15 15:40:58 +0000 Stefan Kost Re-add multipartdemux to the docs. Last round of section cleanup. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/multipart/Makefile.am: * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: * gst/multipart/multipartmux.c: * gst/multipart/multipartmux.h: Re-add multipartdemux to the docs. Last round of section cleanup. 2008-01-15 15:22:41 +0000 Stefan Kost Managed to resolve most unused declarations. Filed a bug for one left. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-sections.txt: * sys/osxaudio/gstosxaudiosink.h: * sys/osxvideo/osxvideosink.h: Managed to resolve most unused declarations. Filed a bug for one left. 2008-01-15 08:03:49 +0000 Stefan Kost docs/plugins/gst-plugins-good-plugins-sections.txt: Cleanup section file. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-sections.txt: Cleanup section file. 2008-01-15 07:42:51 +0000 Stefan Kost docs/plugins/: Update plugin docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: Update plugin docs. * gst/videomixer/Makefile.am: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: * gst/videomixer/videomixerpad.h: Split out header to fix warnings from the doc-build. 2008-01-14 12:35:23 +0000 Wim Taymans As found by: Tommi Myöhänen Original commit message from CVS: As found by: Tommi Myöhänen * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo): Use atoll to parse the rtptime with enough precision. Fixes #509329. 2008-01-14 12:11:43 +0000 Tim-Philipp Müller gst/: Initialise variables to work around (false) 'foo might be used uninitialized in this function' warnings by gcc-... Original commit message from CVS: * gst/avi/gstavisubtitle.c: (gst_avi_subtitle_extract_file): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send): Initialise variables to work around (false) 'foo might be used uninitialized in this function' warnings by gcc-3.3.3 (#509298). 2008-01-14 11:24:57 +0000 Sebastian Dröge ext/lame/gstlame.c: Use gst_util_uint64_scale instead of gst_util_uint64_scale_int as 8 * GST_SECOND is too large for... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_event): Use gst_util_uint64_scale instead of gst_util_uint64_scale_int as 8 * GST_SECOND is too large for int. 2008-01-14 09:17:47 +0000 Mark Nauwelaerts ext/lame/gstlame.c: Correctly set number of channels when using mono-encoding mode and fix the duration calculation o... Original commit message from CVS: Patch by: Mark Nauwelaerts * ext/lame/gstlame.c: (gst_lame_sink_setcaps), (gst_lame_sink_event): Correctly set number of channels when using mono-encoding mode and fix the duration calculation of the EOS buffer. 2008-01-12 02:32:35 +0000 David Schleef Ignore more files for the buildbot. Original commit message from CVS: * docs/plugins/.cvsignore: * tests/check/pipelines/.cvsignore: Ignore more files for the buildbot. 2008-01-11 21:08:59 +0000 Jan Schmidt Generate the image-type values correctly. Leave them out of the caps when outputting a "preview image" tag, since it ... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_extract_picture_buffer): * gst/id3demux/id3v2frames.c: (parse_picture_frame): Generate the image-type values correctly. Leave them out of the caps when outputting a "preview image" tag, since it only makes sense to have one of those - the type is irrelevant. * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_open): If we can, mark the mixer multiple open when we use it, in case (for some reason) the process wants to open it again elsewhere. 2008-01-11 19:16:53 +0000 Tim-Philipp Müller tests/check/elements/: It's "endianness", not "endianess". Fixes unit tests. Original commit message from CVS: * tests/check/elements/rganalysis.c: (test_buffer_const_float_mono), (test_buffer_const_float_stereo), (test_buffer_const_int16_mono), (test_buffer_const_int16_stereo), (test_buffer_square_float_mono), (test_buffer_square_float_stereo), (test_buffer_square_int16_mono), (test_buffer_square_int16_stereo): * tests/check/elements/rglimiter.c: (create_test_buffer): * tests/check/elements/rgvolume.c: (test_buffer_new): It's "endianness", not "endianess". Fixes unit tests. 2008-01-11 18:56:06 +0000 Edward Hervey * tests/check/pipelines/.cvignore: ignore some more Original commit message from CVS: ignore some more 2008-01-11 18:54:31 +0000 Edward Hervey * tests/check/elements/.gitignore: ignore some more Original commit message from CVS: ignore some more 2008-01-11 17:21:30 +0000 Olivier Crete gst/rtp/: Fix the clock rate to 90000 as required by the RFC. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps): * gst/rtp/gstrtptheorapay.c: Fix the clock rate to 90000 as required by the RFC. Fixes #508644. 2008-01-11 17:12:37 +0000 Tim-Philipp Müller tests/check/elements/icydemux.c: Don't use deprecated GST_PLUGIN_DEFINE_STATIC. Original commit message from CVS: * tests/check/elements/icydemux.c: (GST_START_TEST), (icydemux_suite): Don't use deprecated GST_PLUGIN_DEFINE_STATIC. 2008-01-10 12:25:44 +0000 Sebastian Dröge autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We... Original commit message from CVS: * autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We require GNU make in almost every Makefile anyway. * configure.ac: Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o at the same time is required for per target flags. 2008-01-09 15:28:29 +0000 Edward Hervey gst/videomixer/videomixer.c: Fix error from my last commit. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_init): Fix error from my last commit. 2008-01-09 15:20:19 +0000 Tommi Myöhänen gst/id3demux/id3v2frames.c: Make sure the ISO 639-X language code in ID3v2 COMM frames so we don't end up with non-UT... Original commit message from CVS: Based on patch by: Tommi Myöhänen * gst/id3demux/id3v2frames.c: (parse_comment_frame): Make sure the ISO 639-X language code in ID3v2 COMM frames is actually valid UTF-8 (or rather: ASCII), so we don't end up with non-UTF8 strings in tags if there's garbage in the language field. Also make sure the language code is always lower case. Fixes: #508291. 2008-01-09 13:55:28 +0000 Stefan Kost ChangeLog: Fix ChangeLog typo. Original commit message from CVS: * ChangeLog: Fix ChangeLog typo. 2008-01-09 13:50:09 +0000 Stefan Kost Makefile.am: Include lcov.mak to allow builging coverage reports. Guard check-torture target like in the other packages. Original commit message from CVS: * Makefile.am: Include lcov.mak to allow builging coverage reports. Guard check-torture target like in the other packages. 2008-01-09 12:33:58 +0000 Edward Hervey gst/videomixer/videomixer.c: Implement GstChildProxy interface. Original commit message from CVS: reviewed by: Edward Hervey * gst/videomixer/videomixer.c: (gst_videomixer_set_master_geometry), (_do_init), (gst_videomixer_child_proxy_get_child_by_index), (gst_videomixer_child_proxy_get_children_count), (gst_videomixer_child_proxy_init), (gst_videomixer_reset), (gst_videomixer_init), (gst_videomixer_request_new_pad), (gst_videomixer_release_pad), (gst_videomixer_fill_queues): Implement GstChildProxy interface. Send newsegment at the right moment Fixes #488879 2008-01-09 12:01:14 +0000 Edward Hervey gst/alpha/: Make the various properties of 'alpha' controllable. This allows doing niceties like fade-in/fade-out. Original commit message from CVS: * gst/alpha/Makefile.am: * gst/alpha/gstalpha.c: (gst_alpha_class_init), (gst_alpha_init), (gst_alpha_sink_event), (gst_alpha_chain), (gst_alpha_change_state), (plugin_init): Make the various properties of 'alpha' controllable. This allows doing niceties like fade-in/fade-out. 2008-01-09 11:11:01 +0000 Stefan Kost gst/rtp/: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on... Original commit message from CVS: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on disk). 2008-01-08 20:03:30 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: Also fix up pad templates to indicate that image/jpeg doesn't absolutely require the fra... Original commit message from CVS: * gst/matroska/matroska-mux.c: (COMMON_VIDEO_CAPS_NO_FRAMERATE), (videosink_templ): Also fix up pad templates to indicate that image/jpeg doesn't absolutely require the framerate property to be set (#504081). 2008-01-08 19:57:23 +0000 Wouter Cloetens gst/matroska/matroska-mux.*: Keep track of first and last timestamps for each incoming stream, so we can calculate th... Original commit message from CVS: Based on patch by: Wouter Cloetens * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps), (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad), (gst_matroska_mux_finish), (gst_matroska_mux_collected): * gst/matroska/matroska-mux.h: Keep track of first and last timestamps for each incoming stream, so we can calculate the total duration for live sources and other input where we can't query the duration from the start or where there's no constant framerate from which we can deduce the duration; also use calculated/observed duration if it is bigger than the previously queried duration. Furthermore, use gst_pad_query_peer_duration() and take into account that it may return TRUE but still a duration of CLOCK_TIME_NONE, which easily screws up comparisons when using unsigned integers. Fixes #504081. 2008-01-08 14:58:18 +0000 Sebastian Dröge Make elements GST_BUFFER_FLAG_GAP aware and call gst_base_transform_set_gap_aware for this. Original commit message from CVS: * configure.ac: * gst/audiofx/audioamplify.c: (gst_audio_amplify_clipping_method_get_type), (gst_audio_amplify_init), (gst_audio_amplify_transform_ip): * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_init), (gst_audio_dynamic_transform_ip): * gst/audiofx/audioinvert.c: (gst_audio_invert_init), (gst_audio_invert_transform_ip): * gst/audiofx/audiopanorama.c: (gst_audio_panorama_init), (gst_audio_panorama_transform): * gst/level/gstlevel.c: (gst_level_init): Make elements GST_BUFFER_FLAG_GAP aware and call gst_base_transform_set_gap_aware for this. Bump core requirement to CVS. * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_transform_ip): * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_transform_ip): Also sync GObject properties to the controller if operating in passthrough mode. 2008-01-07 16:41:00 +0000 Tim-Philipp Müller sys/directdraw/gstdirectdrawsink.c: FALSE is not a gpointer. Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c: (gst_directdraw_sink_window_thread): FALSE is not a gpointer. 2008-01-05 21:20:08 +0000 Julien Moutte sys/directdraw/gstdirectdrawsink.c: Make sure we create our internal window only when we need it. That will give a ch... Original commit message from CVS: 2008-01-05 Julien Moutte * sys/directdraw/gstdirectdrawsink.c: (gst_directdraw_sink_set_window_id), (gst_directdraw_sink_set_caps), (gst_directdraw_sink_change_state), (gst_directdraw_sink_buffer_alloc), (gst_directdraw_sink_draw_borders), (gst_directdraw_sink_show_frame), (gst_directdraw_sink_setup_ddraw), (gst_directdraw_sink_window_thread), (gst_directdraw_sink_get_ddrawcaps), (gst_directdraw_sink_surface_create): Make sure we create our internal window only when we need it. That will give a chance to the application to get the prepare-xwindow-id bus message. Draw black borders when keeping aspect ratio. Handle the case where our rendering window disappears (closed or errors) like other sinks do. Various 80 columns fixes, improve state change order. That element could need some more love. 2008-01-04 18:30:21 +0000 Sebastian Dröge ext/taglib/: Remove useless typedefs without new type name. Fixes a warning with gcc 4.3. Original commit message from CVS: * ext/taglib/gstapev2mux.h: * ext/taglib/gstid3v2mux.h: Remove useless typedefs without new type name. Fixes a warning with gcc 4.3. 2008-01-03 12:26:03 +0000 John Millikin ext/flac/gstflacdec.c: Emit metadata messages when a PICTURE block is encountered. Original commit message from CVS: Patch by: John Millikin * ext/flac/gstflacdec.c: (gst_flac_dec_setup_seekable_decoder), (gst_flac_dec_setup_stream_decoder), (gst_flac_normalize_picture_mime_type), (gst_flac_extract_picture_buffer), (gst_flac_dec_metadata_callback): Emit metadata messages when a PICTURE block is encountered. Fixes #506715. 2008-01-02 13:54:10 +0000 Thijs Vermeir gst/avi/gstavi.c: increase rank because no known issues anymore ... Original commit message from CVS: * gst/avi/gstavi.c: increase rank because no known issues anymore ... * gst/avi/gstavisubtitle.c: send subtitle name to the srcpad 2007-12-31 13:27:32 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Implement redirect for the DESCRIBE reply. Fixes #506025. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: Implement redirect for the DESCRIBE reply. Fixes #506025. 2007-12-29 16:48:33 +0000 Sebastian Dröge ext/flac/gstflacdec.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached() ... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_write): Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached() macro in GLib-2.15.x and don't abort() in any case but properly report the error. 2007-12-28 11:44:28 +0000 Tim-Philipp Müller ext/soup/: Use gst_tag_freeform_string_to_utf8() and post radio station info as tags on the bus. Original commit message from CVS: * ext/soup/Makefile.am: * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_get_property), (gst_souphttp_src_unicodify), (soup_got_headers): Use gst_tag_freeform_string_to_utf8() and post radio station info as tags on the bus. 2007-12-26 16:03:57 +0000 Tim-Philipp Müller Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached() macro in GLib-2.15.x (i... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_loop): * gst/wavparse/gstwavparse.c: (gst_wavparse_chain): * sys/ximage/gstximagesrc.c: (composite_pixel): Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached() macro in GLib-2.15.x (it's not really nice to abort in any case). Fixes #505745. 2007-12-20 17:07:22 +0000 Tim-Philipp Müller gst/: Ignore more. Original commit message from CVS: * gst/equalizer/.cvsignore: * gst/switch/.cvsignore: Ignore more. 2007-12-18 23:17:14 +0000 Tim-Philipp Müller tests/check/elements/avisubtitle.c: Small unit test fix (has no practical impact at the moment, since we're only feed... Original commit message from CVS: * tests/check/elements/avisubtitle.c: (check_correct_buffer): Small unit test fix (has no practical impact at the moment, since we're only feeding utf8 and hence just create a sub- buffer for the output). 2007-12-18 21:13:05 +0000 Thijs Vermeir Add seeking support for avi subtitle Original commit message from CVS: * gst/avi/gstavisubtitle.c: * tests/check/elements/avisubtitle.c: Add seeking support for avi subtitle 2007-12-18 17:40:34 +0000 Wim Taymans ext/flac/gstflacdec.*: Remove some unused vars. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders), (gst_flac_dec_update_metadata), (gst_flac_dec_metadata_callback), (gst_flac_dec_write): * ext/flac/gstflacdec.h: Remove some unused vars. Do more cleanup of leftover events and tags. Output tags after the segment event. Fixes #504018. 2007-12-18 14:31:36 +0000 Tim-Philipp Müller gst/avi/gstavisubtitle.c: Detect other UTF byte order markers and convert to UTF-8 as appropriate. Original commit message from CVS: * gst/avi/gstavisubtitle.c: (IS_BOM_UTF8), (IS_BOM_UTF16_BE), (IS_BOM_UTF16_LE), (IS_BOM_UTF32_BE), (IS_BOM_UTF32_LE), (gst_avi_subtitle_extract_file), (gst_avi_subtitle_parse_gab2_chunk): Detect other UTF byte order markers and convert to UTF-8 as appropriate. 2007-12-18 13:30:15 +0000 Tim-Philipp Müller gst/avi/gstavisubtitle.*: Refactor a bit; fix name extraction; don't assume all the data in the chunk is actually sub... Original commit message from CVS: * gst/avi/gstavisubtitle.c: (src_template), (gst_avi_subtitle_extract_utf8_file), (gst_avi_subtitle_parse_gab2_chunk), (gst_avi_subtitle_chain), (gst_avi_subtitle_base_init), (gst_avi_subtitle_class_init), (gst_avi_subtitle_init), (gst_avi_subtitle_change_state): * gst/avi/gstavisubtitle.h: Refactor a bit; fix name extraction; don't assume all the data in the chunk is actually subtitle data, there may be padding at the end; fix GST_ELEMENT_ERROR usage; store extracted subtitle file so it's there to send again after a seek (for future use). 2007-12-18 09:13:12 +0000 Thijs Vermeir Add avi subtitle element for bug #442034. Need seeking support and more support for character conversion. Original commit message from CVS: * gst/avi/Makefile.am: * gst/avi/gstavi.c: * gst/avi/gstavisubtitle.c: * gst/avi/gstavisubtitle.h: * tests/check/Makefile.am: * tests/check/elements/avisubtitle.c: * win32/common/config.h: Add avi subtitle element for bug #442034. Need seeking support and more support for character conversion. 2007-12-18 09:07:17 +0000 Tim-Philipp Müller Makefile.am: Include common/win32.mak for CRLF check of win32 project files (see #393626). Original commit message from CVS: * Makefile.am: Include common/win32.mak for CRLF check of win32 project files (see #393626). * win32/vs6/libgstpng.dsp: Fix line endings and do cvs admin -kb. 2007-12-17 21:12:28 +0000 David Schleef gst/multifile/gstmultifilesrc.*: When subsequent files are read, if the file doesn't exist, send an EOS instead of ca... Original commit message from CVS: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: When subsequent files are read, if the file doesn't exist, send an EOS instead of causing an error. 2007-12-16 23:43:46 +0000 Edward Hervey ext/jpeg/gstjpegdec.c: Actually drop the buffers which are outside the currently configured segment instead of just e... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): Actually drop the buffers which are outside the currently configured segment instead of just emitting a WARNING. 2007-12-14 18:49:34 +0000 Wim Taymans ext/flac/gstflacdec.*: Send segments from the streaming thread. Fixes #502187. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback), (gst_flac_dec_write): * ext/flac/gstflacdec.h: Send segments from the streaming thread. Fixes #502187. Fix segment seeking and a bunch of other seeking cases. 2007-12-14 10:17:10 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Parse WOAF frames and put the result into GST_TAG_CONTACT, which is where it would end up... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_url_link_frame): Parse WOAF frames and put the result into GST_TAG_CONTACT, which is where it would end up if the same information was put in a vorbis comment (don't think it's worth adding a new URI tag for this). Fixes #488112. 2007-12-11 22:29:18 +0000 Tim-Philipp Müller configure.ac: We need core/base 0.10.15 or later. Original commit message from CVS: * configure.ac: We need core/base 0.10.15 or later. 2007-12-11 16:47:12 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Fix regression in stream numbering. Fixes #502655. Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/avi/gstavimux.c: (gst_avi_mux_start_file): Fix regression in stream numbering. Fixes #502655. 2007-12-11 16:39:39 +0000 Wouter Cloetens ext/soup/gstsouphttpsrc.*: Do not try to unpause I/O in the "queued" state. Original commit message from CVS: Patch by: Wouter Cloetens * ext/soup/gstsouphttpsrc.c: (_do_init), (gst_souphttp_src_class_init), (gst_souphttp_src_init), (gst_souphttp_src_dispose), (gst_souphttp_src_set_property), (gst_souphttp_src_get_property), (unicodify), (gst_souphttp_src_unicodify), (gst_souphttp_src_create), (gst_souphttp_src_start), (gst_souphttp_src_stop), (gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop), (gst_souphttp_src_get_size), (gst_souphttp_src_is_seekable), (soup_got_headers), (soup_got_body), (soup_finished), (soup_got_chunk), (soup_response), (soup_parse_status), (gst_souphttp_src_uri_get_type), (gst_souphttp_src_uri_get_protocols), (gst_souphttp_src_uri_get_uri), (gst_souphttp_src_uri_set_uri), (gst_souphttp_src_uri_handler_init): * ext/soup/gstsouphttpsrc.h: Do not try to unpause I/O in the "queued" state. Reorganise a bunch of things and cleanups. Uses G_GUINT64_FORMAT instead of hard-coding %llu. See #502335. 2007-12-11 16:31:49 +0000 Wai-Ming Ho gst/rtp/gstrtph264pay.*: Use higher performance start-code searching. Original commit message from CVS: Patch by: Wai-Ming Ho * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_init), (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps), (next_start_code), (is_nal_equal), (gst_rtp_h264_pay_decode_nal), (encode_base64), (gst_rtp_h264_pay_parse_sps_pps), (gst_rtp_h264_pay_handle_buffer): * gst/rtp/gstrtph264pay.h: Use higher performance start-code searching. Parse NALs and store SPS, PPS and profile in the caps so that they can be used in the SDP. Fixes #502814. 2007-12-11 11:50:54 +0000 Tim-Philipp Müller sys/v4l2/: Init some structs to zero before we pass them to ioctl, which avoids valgrind warnings. Also fix a small ... Original commit message from CVS: * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list): Init some structs to zero before we pass them to ioctl, which avoids valgrind warnings. Also fix a small memory leak. 2007-12-11 11:05:57 +0000 Wouter Cloetens gst/multipart/multipartdemux.c: Copy timestamp from input to output. Not very perfect yet but better than nothing. Fi... Original commit message from CVS: Patch by: Wouter Cloetens * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain): Copy timestamp from input to output. Not very perfect yet but better than nothing. Fixes #503023. 2007-12-09 16:49:09 +0000 Sebastian Dröge ext/wavpack/gstwavpackdec.c: Also print a useful error message with the old Wavpack API if possible. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): Also print a useful error message with the old Wavpack API if possible. 2007-12-09 16:34:08 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackdec.c: More build fixes for old libwavpack versions: include config.h so that WAVPACK_OLD_API i... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: More build fixes for old libwavpack versions: include config.h so that WAVPACK_OLD_API is actually defined as detected; only use WavpackGetErrorMessage if it is available. This fixes the build on debian stable for me. 2007-12-09 16:21:02 +0000 Sebastian Dröge ext/wavpack/: Workaround the non-existance of WavpackGetChannelMask in Wavpack versions below 4.40.0. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_create_src_pad): Workaround the non-existance of WavpackGetChannelMask in Wavpack versions below 4.40.0. 2007-12-09 05:13:58 +0000 Sebastian Dröge configure.ac: And now do it right for real... Original commit message from CVS: * configure.ac: And now do it right for real... 2007-12-09 05:09:57 +0000 Sebastian Dröge configure.ac: Correctly reset $LIBS to not contain -lm. Original commit message from CVS: * configure.ac: Correctly reset $LIBS to not contain -lm. 2007-12-09 05:02:17 +0000 Kwang Yul Seo Fix compilation with MSVC by using gst_util_guint64_to_gdouble() and checking for rint() and implementing it ourself ... Original commit message from CVS: Based on a patch by: Kwang Yul Seo * configure.ac: * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_print_smpte_time): Fix compilation with MSVC by using gst_util_guint64_to_gdouble() and checking for rint() and implementing it ourself if it doesn't exist. 2007-12-09 04:29:08 +0000 Sebastian Dröge configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181. Original commit message from CVS: * configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181. 2007-12-08 16:47:33 +0000 Jan Schmidt sys/oss/gstosshelper.c: Verify that the format returned after the ioctl is the one we requested. It is valid for the ... Original commit message from CVS: * sys/oss/gstosshelper.c: Verify that the format returned after the ioctl is the one we requested. It is valid for the ioctl to succeed while substituting an alternate 'supported' sample format. 2007-12-07 20:07:49 +0000 Tim-Philipp Müller sys/oss/: Post decent (and translated) error message when we can't open the audio device for some reason. Original commit message from CVS: * sys/oss/gstossaudio.c: (plugin_init): * sys/oss/gstosssink.c: (gst_oss_sink_open): * sys/oss/gstosssrc.c: (gst_oss_src_open): Post decent (and translated) error message when we can't open the audio device for some reason. 2007-12-07 19:29:39 +0000 Jan Schmidt sys/oss/: Allow the AUDIODEV environment variable to redirect us to a different default OSS device, like sunaudiosink... Original commit message from CVS: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: Allow the AUDIODEV environment variable to redirect us to a different default OSS device, like sunaudiosink does on Solaris (makes audio play automatically on SunRays). 2007-12-06 12:45:50 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Fix compilation. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_transform_ip): Fix compilation. 2007-12-06 12:42:11 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Don't process buffers in passthrough mode. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_transform_ip): Don't process buffers in passthrough mode. 2007-12-06 12:37:43 +0000 Sebastian Dröge gst/filter/: The transform() methods are not called in passthrough mode so there's no need for checking if the elemen... Original commit message from CVS: * gst/filter/gstbpwsinc.c: (bpwsinc_transform): * gst/filter/gstlpwsinc.c: (lpwsinc_transform): The transform() methods are not called in passthrough mode so there's no need for checking if the element is in passthrough mode. 2007-12-06 12:29:26 +0000 Sebastian Dröge gst/filter/: Sync the GObject properties with the controller even in passthrough mode to get consistent property values. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (bpwsinc_transform): * gst/filter/gstlpwsinc.c: (lpwsinc_transform): Sync the GObject properties with the controller even in passthrough mode to get consistent property values. 2007-12-06 12:11:29 +0000 Sebastian Dröge gst/audiofx/: The transform_ip() methods should do nothing if in passthrough mode. Original commit message from CVS: * gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip): * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_transform_ip): * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_transform_ip): * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip): * gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip): The transform_ip() methods should do nothing if in passthrough mode. It might get non-writable buffers in that case but the buffer might as well be writable. * gst/audiofx/audiopanorama.c: (gst_audio_panorama_transform): The transform() methods won't be called in passthrough mode and otherwise the buffer is always writable so don't check here. 2007-12-06 11:46:22 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: Fix seeking in .wav files again (#501775). Some people seem to think they don't need to ... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_srcpad_event): Fix seeking in .wav files again (#501775). Some people seem to think they don't need to test their changes when they're just 'reflowing' some code. 2007-12-05 16:04:47 +0000 Wim Taymans gst/autodetect/gstautovideosink.*: Fix docs. Original commit message from CVS: * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose), (gst_auto_video_sink_init), (gst_auto_video_sink_create_element_with_pretty_name), (gst_auto_video_sink_find_best), (gst_auto_video_sink_set_property), (gst_auto_video_sink_get_property): * gst/autodetect/gstautovideosink.h: Fix docs. Use same error reporting code as autoaudiosink. Add property to filter sinks based on caps. Only select raw video sinks by default for backwards compat. API: GstAutoVideoSink::filter-caps 2007-12-05 16:02:15 +0000 Tommi Myöhänen gst/autodetect/gstautoaudiosink.*: Add property to filter sinks based on caps. Only select raw audio sinks by default... Original commit message from CVS: Patch by: Tommi Myöhänen * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose), (gst_auto_audio_sink_init), (gst_auto_audio_sink_find_best), (gst_auto_audio_sink_set_property), (gst_auto_audio_sink_get_property): * gst/autodetect/gstautoaudiosink.h: Add property to filter sinks based on caps. Only select raw audio sinks by default for backwards compat. Fixes #417420. API: GstAutoAudioSink::filter-caps 2007-11-29 11:40:15 +0000 Arek Korbik gst/videobox/gstvideobox.c: Initialise liboil in plugin_init() Original commit message from CVS: Patch by: Arek Korbik * gst/videobox/gstvideobox.c: (plugin_init): Initialise liboil in plugin_init() 2007-11-29 10:49:18 +0000 Wouter Cloetens configure.ac: Bump libsoup requirement as libsoup does not support async client operation prior to version 2.2.104 an... Original commit message from CVS: Patch by: Wouter Cloetens * configure.ac: Bump libsoup requirement as libsoup does not support async client operation prior to version 2.2.104 and it has some leaks. * ext/soup/gstsouphttpsrc.c: (gst_souphttp_src_class_init), (gst_souphttp_src_init), (gst_souphttp_src_dispose), (gst_souphttp_src_set_property), (gst_souphttp_src_create), (gst_souphttp_src_start), (gst_souphttp_src_stop), (gst_souphttp_src_unlock), (gst_souphttp_src_unlock_stop), (gst_souphttp_src_get_size), (soup_got_headers), (soup_got_body), (soup_finished), (soup_got_chunk), (soup_response), (soup_session_close): * ext/soup/gstsouphttpsrc.h: Implement unlock(). Picks up the size from the Content-Length header and emit a duration message. Don't leak the GMainContext object. Fixes #500099. 2007-11-29 10:34:18 +0000 Wim Taymans ext/libpng/gstpngdec.c: Post error before sending EOS. Fixes #499178. Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_task): Post error before sending EOS. Fixes #499178. 2007-11-28 21:54:50 +0000 Sébastien Moutte win32/vs6/: Add a project file for libgstpng Original commit message from CVS: * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstpng.dsp: Add a project file for libgstpng 2007-11-28 17:48:45 +0000 Edward Hervey gst/rtp/gstrtph263depay.c: Code beautification. Original commit message from CVS: * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_process): Code beautification. Added debug statements. Don't bit-shift everything, just do operations on last/first byte instead. 2007-11-27 11:11:08 +0000 Jayarama S. Santana gst/rtp/gstrtpmp4adepay.c: Fix wrong comparison in overrun check. Fixes #499239 some more. Original commit message from CVS: Patch by: Jayarama S. Santana * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_process): Fix wrong comparison in overrun check. Fixes #499239 some more. 2007-11-27 00:01:41 +0000 Edward Hervey gst/rtp/gstrtph263depay.*: Fix h263 depayloader so that ANY h263 decoder can handle the outgoing stream. Original commit message from CVS: * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_init), (gst_rtp_h263_depay_process): * gst/rtp/gstrtph263depay.h: Fix h263 depayloader so that ANY h263 decoder can handle the outgoing stream. 2007-11-26 19:17:10 +0000 Wim Taymans gst/rtp/gstrtpmp4adepay.*: Fix depayloading when multiple frames are inside one RTP packet. Original commit message from CVS: Based on Path by: Jayarama S. Santana * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process): * gst/rtp/gstrtpmp4adepay.h: Fix depayloading when multiple frames are inside one RTP packet. Fixes #499239. 2007-11-26 12:26:20 +0000 Stefan Kost gst/level/gstlevel.c: Add GAP-flag support. Original commit message from CVS: * gst/level/gstlevel.c: Add GAP-flag support. 2007-11-26 12:01:11 +0000 Edward Hervey gst/rtp/gstrtph263depay.c: Read the I flag for Mode A h263 rtp stream and set the Original commit message from CVS: * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_process): Read the I flag for Mode A h263 rtp stream and set the GST_BUFFER_FLAG_DELTA_UNIT accordingly. Fixes #499383 2007-11-26 10:08:20 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Use dispose and finalize. Dispose can be called multiple times. Original commit message from CVS: * gst/spectrum/gstspectrum.c: Use dispose and finalize. Dispose can be called multiple times. 2007-11-26 10:04:49 +0000 Stefan Kost gst/level/gstlevel.c: Remove some dead code and do cleanups. Original commit message from CVS: * gst/level/gstlevel.c: Remove some dead code and do cleanups. 2007-11-26 09:13:48 +0000 Stefan Kost tests/check/pipelines/simple-launch-lines.c: Improve the tests by allowing to set a target state. Original commit message from CVS: * tests/check/pipelines/simple-launch-lines.c: Improve the tests by allowing to set a target state. 2007-11-26 09:04:17 +0000 Sebastian Dröge tests/check/elements/wavpackenc.c: Don't check the caps of the output buffer if they're equal some other caps. The ca... Original commit message from CVS: * tests/check/elements/wavpackenc.c: (GST_START_TEST): Don't check the caps of the output buffer if they're equal some other caps. The caps can change in a backward compatible way and did at this point. 2007-11-24 14:55:04 +0000 Julien Moutte gst/qtdemux/qtdemux.c: Implement reverse playback support. Original commit message from CVS: 2007-11-24 Julien MOUTTE * gst/qtdemux/qtdemux.c: (gst_qtdemux_find_segment), (gst_qtdemux_move_stream), (gst_qtdemux_do_seek), (gst_qtdemux_seek_to_previous_keyframe), (gst_qtdemux_activate_segment), (gst_qtdemux_advance_sample), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop): Implement reverse playback support. 2007-11-21 09:56:54 +0000 Sebastian Dröge gst/filter/: Post a GST_MESSAGE_LATENCY if the latency changes. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (bpwsinc_set_property): * gst/filter/gstlpwsinc.c: (lpwsinc_set_property): Post a GST_MESSAGE_LATENCY if the latency changes. 2007-11-21 08:21:10 +0000 Stefan Kost gst/equalizer/: Remove preset iface again. We'll re-add this after its been released in -good. Original commit message from CVS: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: Remove preset iface again. We'll re-add this after its been released in -good. 2007-11-20 13:14:40 +0000 Sebastian Dröge ext/wavpack/gstwavpackcommon.c: Also set the channel layout on the Wavpack caps if we're having a mono layout. Of cou... Original commit message from CVS: * ext/wavpack/gstwavpackcommon.c: (gst_wavpack_set_channel_layout): Also set the channel layout on the Wavpack caps if we're having a mono layout. Of course only do it for "audio/x-wavpack". 2007-11-20 13:08:45 +0000 Sebastian Dröge ext/wavpack/: Add support for encoding, parsing and decoding multichannel files with up to 8 channels. This also impr... Original commit message from CVS: * ext/wavpack/gstwavpackcommon.c: (gst_wavpack_get_default_channel_mask), (gst_wavpack_set_channel_layout), (gst_wavpack_get_default_channel_positions), (gst_wavpack_get_channel_mask_from_positions), (gst_wavpack_set_channel_mapping): * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset), (gst_wavpack_enc_init), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_push_block), (gst_wavpack_enc_fix_channel_order), (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block), (gst_wavpack_enc_sink_event): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset), (gst_wavpack_parse_scan_to_find_sample), (gst_wavpack_parse_sink_event), (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_loop): * ext/wavpack/gstwavpackparse.h: Add support for encoding, parsing and decoding multichannel files with up to 8 channels. This also improves the robustness of parsing quite a bit. * ext/wavpack/gstwavpackstreamreader.c: (gst_wavpack_stream_reader_read_bytes), (gst_wavpack_stream_reader_get_pos), (gst_wavpack_stream_reader_set_pos_abs), (gst_wavpack_stream_reader_set_pos_rel), (gst_wavpack_stream_reader_push_back_byte), (gst_wavpack_stream_reader_get_length), (gst_wavpack_stream_reader_can_seek), (gst_wavpack_stream_reader_write_bytes): Improve debugging. 2007-11-20 12:20:38 +0000 Stefan Kost ext/libpng/gstpngdec.*: Don't release the png-memory from within the callback. Original commit message from CVS: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngdec.h: Don't release the png-memory from within the callback. 2007-11-20 12:11:13 +0000 René Stadler ext/libpng/gstpngenc.c: Don't leak buffer data memory. Fixes #498395. Original commit message from CVS: Patch by: René Stadler * ext/libpng/gstpngenc.c: Don't leak buffer data memory. Fixes #498395. 2007-11-20 11:46:28 +0000 René Stadler tests/check/pipelines/simple-launch-lines.c: Tests for #498395. Original commit message from CVS: Patch by: René Stadler * tests/check/pipelines/simple-launch-lines.c: Tests for #498395. 2007-11-20 11:41:13 +0000 Julien Moutte Fix build on Mac OS X 10.5 Original commit message from CVS: 2007-11-20 Julien MOUTTE * ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag), (gst_tag_lib_mux_adjust_event_offsets): * gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension): * sys/osxaudio/Makefile.am: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5 2007-11-19 20:30:19 +0000 Stefan Kost gst/equalizer/: Activate preset iface and upload two presets here. Original commit message from CVS: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: Activate preset iface and upload two presets here. 2007-11-16 05:52:55 +0000 David Schleef ext/cairo/gsttextoverlay.c: Change strcasecmp() to g_strcasecmp(). Fixes #497292. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: Change strcasecmp() to g_strcasecmp(). Fixes #497292. 2007-11-15 18:19:19 +0000 Jordi Jaen Pallares gst/rtp/gstrtpmp2tpay.*: Fill the MTU with as many packets as possible. Fixes #491323. Original commit message from CVS: Patch by: Jordi Jaen Pallares * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_finalize), (gst_rtp_mp2t_pay_flush), (gst_rtp_mp2t_pay_handle_buffer): * gst/rtp/gstrtpmp2tpay.h: Fill the MTU with as many packets as possible. Fixes #491323. 2007-11-15 17:47:43 +0000 Tommi Myöhänen gst/rtsp/gstrtspsrc.c: Fix some more leaks. Fixes #497007. Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Fix some more leaks. Fixes #497007. 2007-11-15 17:35:18 +0000 Tommi Myöhänen gst/rtsp/gstrtspsrc.c: Fix 3 pad leaks. Fixes #496983. Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_tcp): Fix 3 pad leaks. Fixes #496983. 2007-11-15 17:26:25 +0000 Wouter Cloetens Added HTTP source based on libsoup. Fixes #497020. Original commit message from CVS: Patch by: Wouter Cloetens * configure.ac: * ext/Makefile.am: * ext/soup/Makefile.am: * ext/soup/gstsouphttpsrc.c: (_do_init), (gst_souphttp_src_base_init), (gst_souphttp_src_class_init), (gst_souphttp_src_init), (gst_souphttp_src_dispose), (gst_souphttp_src_set_property), (gst_souphttp_src_get_property), (gst_souphttp_src_create), (gst_souphttp_src_start), (gst_souphttp_src_stop), (gst_souphttp_src_unlock), (gst_souphttp_src_set_location), (soup_got_chunk), (soup_response), (soup_session_close), (plugin_init): * ext/soup/gstsouphttpsrc.h: Added HTTP source based on libsoup. Fixes #497020. 2007-11-15 17:01:32 +0000 Tommi Myöhänen gst/rtp/gstrtph264depay.c: Fix small leak. Fixes #497017. Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps): Fix small leak. Fixes #497017. 2007-11-15 16:31:32 +0000 Wim Taymans gst/qtdemux/: Add suppport for theora in quicktime according to XiphQT. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state), (gst_qtdemux_prepare_current_sample), (gst_qtdemux_loop_state_movie), (qtdemux_parse_theora_extension), (qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps): * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: Add suppport for theora in quicktime according to XiphQT. 2007-11-15 12:22:10 +0000 Edgard Lima * ChangeLog: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2src_calls.c: Always copy buffers by default (handle safer with bugged drivers) and added a property to make it possible to use mma... Original commit message from CVS: Always copy buffers by default (handle safer with bugged drivers) and added a property to make it possible to use mmap effectively (no copy if possible) when application wants to. Fixes: #480557. 2007-11-14 21:39:47 +0000 Tim-Philipp Müller gst/id3demux/: We don't want the same string multiple times in a tag list for the same tag ever, for any tag, not jus... Original commit message from CVS: * gst/id3demux/id3tags.c: * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist): We don't want the same string multiple times in a tag list for the same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure this doesn't happen and remove special-case code for GST_TAG_GENRE. 2007-11-14 21:04:12 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID into ID3v2 TXXX frames (fixe... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: (add_musicbrainz_tag), (add_funcs): Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID into ID3v2 TXXX frames (fixes #347848). 2007-11-14 20:34:24 +0000 Tim-Philipp Müller gst/rtsp/gstrtspsrc.c: Don't leak sdp message contents (fixes #496773). Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Don't leak sdp message contents (fixes #496773). * gst/udp/gstudpsink.c: (gst_udpsink_finalize): Don't leak URI string. 2007-11-14 19:10:37 +0000 Julien Puydt ext/raw1394/: Implement GstPropertyProbe interface and add "device-name" property, so applications can use this to pr... Original commit message from CVS: Patch by: Julien Puydt * ext/raw1394/Makefile.am: * ext/raw1394/gst1394probe.c: (gst_1394_get_guid_array), (gst_1394_property_probe_get_properties), (gst_1394_property_probe_probe_property), (gst_1394_property_probe_needs_probe), (gst_1394_property_probe_get_values), (gst_1394_property_probe_interface_init), (gst_1394_type_add_property_probe_interface): * ext/raw1394/gst1394probe.h: (GST_1394_PROBE_H): * ext/raw1394/gstdv1394src.c: (_do_init), (gst_dv1394src_class_init), (gst_dv1394src_init), (gst_dv1394src_dispose), (gst_dv1394src_set_property), (gst_dv1394src_get_property), (gst_dv1394src_discover_avc_node), (gst_dv1394src_query), (gst_dv1394src_update_device_name): * ext/raw1394/gstdv1394src.h: Implement GstPropertyProbe interface and add "device-name" property, so applications can use this to probe for available devices in the same way they can already with v4lsrc and v4l2src (however horrible this property probe interface may be). Fixes #358841. 2007-11-14 17:03:18 +0000 Sebastian Dröge tests/check/elements/spectrum.c: Fix spectrum unit test for the latest spectrum changes. Original commit message from CVS: * tests/check/elements/spectrum.c: (GST_START_TEST): Fix spectrum unit test for the latest spectrum changes. 2007-11-14 15:29:05 +0000 Tommi Myöhänen gst/rtsp/gstrtspsrc.c: Don't leak event, don't leak range (fixes #496752). Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event), (gst_rtspsrc_parse_range): Don't leak event, don't leak range (fixes #496752). 2007-11-14 10:22:41 +0000 Arek Korbik gst/alpha/gstalphacolor.c: Detect RGBA/BGRA correctly on little endian systems. Original commit message from CVS: Patch by: Arek Korbik * gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps): Detect RGBA/BGRA correctly on little endian systems. 2007-11-13 17:19:13 +0000 Tim-Philipp Müller sys/v4l2/v4l2src_calls.c: but the corresponding ioctl() call fails even though the driver claims to support this form... Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format): If VIDIOC_ENUM_FRAMESIZES is defined (= recent kernel), but the corresponding ioctl() call fails even though the driver claims to support this format, just fall back to the pre-2.6.19 kernel routine that creates caps with suitable height and width ranges (see #448278). 2007-11-13 17:01:07 +0000 Mark Nauwelaerts gst/matroska/: Extract palette data for dvd subpicture streams and send it downstream as custom gstreamer dvd event (... Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_dvd_clut_change_event), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_subtitle_caps): * gst/matroska/matroska-ids.h: Extract palette data for dvd subpicture streams and send it downstream as custom gstreamer dvd event (fixes #453417). 2007-11-13 14:51:30 +0000 Tim-Philipp Müller ext/cairo/gsttextoverlay.c: Implement minimal parsing of the passed pango font description string, so passing a font ... Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_font_init): Implement minimal parsing of the passed pango font description string, so passing a font size works the same as with the pango textoverlay plugin; fixes #455086. (Maybe we could just use pangocairo here at some point). 2007-11-13 06:55:28 +0000 Stefan Kost gst/: Return the result in _activate_pull(). Don't ref element there. Original commit message from CVS: * gst/avi/gstavidemux.c: * gst/wavparse/gstwavparse.c: Return the result in _activate_pull(). Don't ref element there. 2007-11-13 06:23:51 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Ref the element when we should, but not when we its not needed. Reflow the event_handling... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers), (gst_wavparse_pad_convert), (gst_wavparse_pad_query), (gst_wavparse_srcpad_event): Ref the element when we should, but not when we its not needed. Reflow the event_handling to not leak the event. 2007-11-12 21:07:31 +0000 René Stadler gst/replaygain/rganalysis.c: Avoid slowdown from denormals when processing near-silence input data. Original commit message from CVS: Patch by: René Stadler * gst/replaygain/rganalysis.c: (yule_filter): Avoid slowdown from denormals when processing near-silence input data. Spotted by Gabriel Bouvigne. Fixes #494499. 2007-11-12 17:59:40 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Properly free QTDemuxSamples array. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state), (qtdemux_parse_samples): Properly free QTDemuxSamples array. Protect table write with a sensible check, some files apparently DO contain stts values starting with 0 :( 2007-11-12 17:21:59 +0000 Stefan Kost gst/: Drop EOS in _handle_src_event(). Fix the refcount in qtdemux that previous commit messed up. Original commit message from CVS: * gst/avi/gstavidemux.c: * gst/qtdemux/qtdemux.c: Drop EOS in _handle_src_event(). Fix the refcount in qtdemux that previous commit messed up. 2007-11-12 17:06:32 +0000 Stefan Kost gst/: Sync _handle_src_event() with oggdemux. In avidemux also ref the element when we should, but not when we its no... Original commit message from CVS: * gst/avi/gstavidemux.c: * gst/qtdemux/qtdemux.c: Sync _handle_src_event() with oggdemux. In avidemux also ref the element when we should, but not when we its not needed. 2007-11-11 21:12:10 +0000 Sebastian Dröge gst/: Change the meaning of the magnitude values given in the Original commit message from CVS: * gst/equalizer/demo.c: (draw_spectrum): * gst/spectrum/demo-audiotest.c: (draw_spectrum): * gst/spectrum/demo-osssrc.c: (draw_spectrum): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): Change the meaning of the magnitude values given in the GstMessages by spectrum to decibel instead of decibel+threshold. 2007-11-11 13:55:27 +0000 Sebastian Dröge gst/equalizer/: And continue to update docs. Also include some sample code for the n-band equalizer in the docs. Original commit message from CVS: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: And continue to update docs. Also include some sample code for the n-band equalizer in the docs. 2007-11-11 12:54:31 +0000 Sebastian Dröge gst/equalizer/: Update docs and property ranges to the real values. Original commit message from CVS: * gst/equalizer/gstiirequalizer10bands.c: (gst_iir_equalizer_10bands_class_init): * gst/equalizer/gstiirequalizer3bands.c: (gst_iir_equalizer_3bands_class_init): * gst/equalizer/gstiirequalizernbands.c: Update docs and property ranges to the real values. 2007-11-09 17:27:00 +0000 Sebastian Dröge gst/spectrum/gstspectrum.c: Now do the scaling right for real. Also initialize a previously uninitialized variable. Original commit message from CVS: * gst/spectrum/gstspectrum.c: Now do the scaling right for real. Also initialize a previously uninitialized variable. 2007-11-08 15:56:46 +0000 Stefan Kost gst/equalizer/demo.c: Make default volume a bit less. Improve layout by giving more space to the slider with big-numb... Original commit message from CVS: * gst/equalizer/demo.c: Make default volume a bit less. Improve layout by giving more space to the slider with big-numbers and enable fill. 2007-11-08 15:00:40 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Return FALSE if we can't handle a query instead of changing the format. Ignore fact when ... Original commit message from CVS: * gst/wavparse/gstwavparse.c: Return FALSE if we can't handle a query instead of changing the format. Ignore fact when dealing with mpeg audio. 2007-11-06 12:23:35 +0000 Sebastian Dröge gst/spectrum/demo-audiotest.c: Use autoaudiosink instead of alsasink and use a sine wave. Original commit message from CVS: * gst/spectrum/demo-audiotest.c: (main): Use autoaudiosink instead of alsasink and use a sine wave. * gst/spectrum/gstspectrum.c: Fix the magnitude calculation. 2007-11-03 19:50:11 +0000 Sebastian Dröge gst/equalizer/: Allow setting 0 as bandwidth and handle this correctly. Original commit message from CVS: * gst/equalizer/demo.c: (main): * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_class_init), (setup_filter): Allow setting 0 as bandwidth and handle this correctly. Also handle a bandwidth of rate/2 properly. * gst/equalizer/gstiirequalizernbands.c: (gst_iir_equalizer_nbands_class_init): Make it possible to generate a N-band equalizer with 1 bands. The previous limit of 2 was caused by a nowadays replaced calculation doing a division by zero if number of bands was 1. 2007-11-02 21:16:09 +0000 Ole André Vadla Ravnås Fix includes for MSVC and GLib-2.14.0 (#492388). Original commit message from CVS: Patch by: Ole André Vadla Ravnås * configure.ac: * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsink.h: Fix includes for MSVC and GLib-2.14.0 (#492388). * gst/udp/gstudpsrc.c: (gst_udpsrc_start): No more pipe define since GLib-2.14.0, need to use _pipe() directly. 2007-11-02 17:23:43 +0000 Edward Hervey gst/law/mulaw-decode.*: Calculate outgoing buffer duration if incoming buffer didn't have a valid duration. Original commit message from CVS: * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps), (gst_mulawdec_chain): * gst/law/mulaw-decode.h: Calculate outgoing buffer duration if incoming buffer didn't have a valid duration. 2007-10-30 21:37:49 +0000 Sebastian Dröge gst/equalizer/: Add small demo application based on the spectrum demo applications that gets white noise as input, pu... Original commit message from CVS: * gst/equalizer/Makefile.am: * gst/equalizer/demo.c: (on_window_destroy), (on_configure_event), (on_gain_changed), (on_bandwidth_changed), (on_freq_changed), (draw_spectrum), (message_handler), (main): Add small demo application based on the spectrum demo applications that gets white noise as input, pushes it through an equalizer and paints the spectrum. For every equalizer band it's possible to set gain, bandwidth and frequency. * gst/equalizer/gstiirequalizer.c: (setup_filter): Add some guarding against too large or too small frequencies and bandwidths. Also improve debugging a bit. 2007-10-30 21:18:45 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Replace filters with a bit better filters for which we can actually find documentati... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_band_get_property), (gst_iir_equalizer_band_class_init), (arg_to_scale), (setup_filter), (gst_iir_equalizer_compute_frequencies): Replace filters with a bit better filters for which we can actually find documentation, which don't change anything on zero gain, etc. Make the frequency property of the bands writable, rename the band-width property to bandwidth and change the meaning to the frequency difference between bandedges, change the meaning of the gain property to dB instead of a weird scale between -1 and 1 that has no real meaning. 2007-10-30 12:29:46 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Smarter combine_flow code that also deals with downstream elements returning UNEXPECTED when t... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment), (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie): Smarter combine_flow code that also deals with downstream elements returning UNEXPECTED when they receive data out of the segment boundaries. Fixes #491305. 2007-10-27 16:04:48 +0000 Tim-Philipp Müller gst/interleave/interleave.c: Let's not call every request pad we create "sink%d", that'll create problems if there's ... Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_request_new_pad): Let's not call every request pad we create "sink%d", that'll create problems if there's to be more than one pad. Fixes #490682. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/interleave.c: Add unit test for the above. 2007-10-26 15:03:06 +0000 Tim-Philipp Müller sys/v4l2/v4l2src_calls.c: Fix 'unused variable' compiler warning when compiling against older kernel headers. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: Fix 'unused variable' compiler warning when compiling against older kernel headers. 2007-10-26 12:10:43 +0000 Christian Schaller * gst-plugins-good.spec.in: update spec file Original commit message from CVS: update spec file 2007-10-25 23:42:52 +0000 David Schleef Improve documentation, write some tests for multifilesrc/sink for upcoming ->good review. Original commit message from CVS: * gst/multifile/Makefile.am: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * tests/check/Makefile.am: * tests/check/elements/multifile.c: Improve documentation, write some tests for multifilesrc/sink for upcoming ->good review. 2007-10-25 15:00:15 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc (add_funcs): Map new SORTNAME tags to ID3v2 TSOP, TSOA and TSOT frames (#414539). Original commit message from CVS: * ext/taglib/gstid3v2mux.cc (add_funcs): Map new SORTNAME tags to ID3v2 TSOP, TSOA and TSOT frames (#414539). 2007-10-24 07:01:47 +0000 Stefan Kost tests/check/pipelines/simple-launch-lines.c: Improve the tests a little more. Original commit message from CVS: * tests/check/pipelines/simple-launch-lines.c: Improve the tests a little more. 2007-10-23 08:38:50 +0000 Yun Zheng Hu sys/osxaudio/gstosxaudiosrc.c: Use default input device instead of default output device and only memcpy actual avail... Original commit message from CVS: patch by: Yun Zheng Hu * sys/osxaudio/gstosxaudiosrc.c: Use default input device instead of default output device and only memcpy actual available bytes. 2007-10-22 19:14:08 +0000 Edgard Lima sys/v4l2/v4l2src_calls.c: Fixes "v4l2src ! queue ! xvimagesink". The queue ask for buffer too early. It is temporary ... Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame): Fixes "v4l2src ! queue ! xvimagesink". The queue ask for buffer too early. It is temporary until we find something better. 2007-10-22 16:44:48 +0000 Tommi Myöhänen gst/rtsp/gstrtspsrc.c: Fix race when pausing a RTSP stream in interleaved. Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved): Fix race when pausing a RTSP stream in interleaved. Fixes #475784. 2007-10-22 09:53:16 +0000 Peter Kjellerstedt gst/rtp/gstrtpmp4vpay.c: Use correct unref function for buffers. #488844. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize): Use correct unref function for buffers. #488844. 2007-10-19 19:33:16 +0000 Stefan Kost Add some debug and sync tests with the fix. Original commit message from CVS: * gst/avi/gstavimux.c: * tests/check/elements/avimux.c: Add some debug and sync tests with the fix. 2007-10-18 17:04:14 +0000 Laurent Glayal gst/udp/gstudpsrc.c: When the socket is used by the app for other purposes, don't generate an error if there is activ... Original commit message from CVS: Based on patch by: Laurent Glayal * gst/udp/gstudpsrc.c: (gst_udpsrc_create): When the socket is used by the app for other purposes, don't generate an error if there is activaty on the socket that is not data related. Fixes #487488. 2007-10-18 14:55:38 +0000 Wim Taymans sys/v4l2/v4l2src_calls.c: Add some more debug info. Generate an error when we run out of buffers for some reason. See... Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize), (gst_v4l2src_grab_frame): Add some more debug info. Generate an error when we run out of buffers for some reason. See #480557. 2007-10-18 08:27:56 +0000 Anders Skargren gst/rtp/gstrtph264pay.c: Set marker bit correctly. Original commit message from CVS: Patch by: Anders Skargren * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer): Set marker bit correctly. 2007-10-18 06:20:21 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.c: Add a missing break. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property): Add a missing break. 2007-10-18 06:14:42 +0000 Sebastian Dröge gst/equalizer/gstiirequalizer.*: Move bandwidth property to the separate bands and add float64 support. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_band_get_property), (gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_init), (gst_iir_equalizer_band_get_type), (gst_iir_equalizer_class_init), (setup_filter), (gst_iir_equalizer_setup): * gst/equalizer/gstiirequalizer.h: Move bandwidth property to the separate bands and add float64 support. 2007-10-17 15:08:02 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Use allowed name for the GstStructure. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Use allowed name for the GstStructure. 2007-10-17 11:47:23 +0000 Tim-Philipp Müller Use new gst_bus_pop_filtered(). Original commit message from CVS: * ext/gconf/gstswitchsink.c: * gst/autodetect/gstautoaudiosink.c: Use new gst_bus_pop_filtered(). 2007-10-13 12:03:44 +0000 Tim-Philipp Müller sys/v4l2/: When probing the formats and sizes a camera supports, make sure the best ones (highest resolution, prefere... Original commit message from CVS: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: When probing the formats and sizes a camera supports, make sure the best ones (highest resolution, prefered format) end up at the beginning of the probed caps and the less desirable ones at the end. This is important because the order within the caps matters for things like fixation and negotiation, ie. what format is chosen in the end. With recent kernels, the current probing code will end up querying the supported sizes from lowest resolution to highest resolution, adding them to the probed caps in that order, resulting to v4l2src fixating to the lowest possible resolution if downstream does not express a size preference. Also make up a somewhat random ranking of prefered output formats for the same reason. Fixes #485828. 2007-10-11 17:55:29 +0000 Jason Kivlighn gst/id3demux/id3v2frames.c: Extract license/copyright URIs from ID3v2 WCOP frames (Fixes #447000). Original commit message from CVS: Based on patch by: Jason Kivlighn * gst/id3demux/id3v2frames.c: Extract license/copyright URIs from ID3v2 WCOP frames (Fixes #447000). * tests/check/elements/id3demux.c: * tests/files/Makefile.am: * tests/files/id3-447000-wcop.tag: Add simple unit test. 2007-10-11 16:41:44 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: Add support for license/copyright URI tags (ID3v2 WCOP frame). Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Add support for license/copyright URI tags (ID3v2 WCOP frame). Prerequisite for #447000. 2007-10-08 17:44:42 +0000 Jan Schmidt gst/rtsp/gstrtspsrc.c: Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise a GstClockTime. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush): Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise a GstClockTime. 2007-10-08 11:58:51 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new playback segment in order to configure it pr... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek), (gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_change_state): More seeking fixes, mostly passing around the new playback segment in order to configure it properly. Also reset base_time of udp sources when setting them back to PLAYING as a temporary hack until core supports seek in live sources properly. 2007-10-08 10:34:03 +0000 Wim Taymans gst/rtp/gstrtpmp4adepay.c: Fix caps as to not confuse autopluggers. Original commit message from CVS: * gst/rtp/gstrtpmp4adepay.c: Fix caps as to not confuse autopluggers. 2007-10-06 16:13:14 +0000 Tim-Philipp Müller gst/id3demux/: Port ID3 tag demuxer over to the new GstTagDemux in -base (now would be a good time to test re-importi... Original commit message from CVS: * gst/id3demux/gstid3demux.c: * gst/id3demux/gstid3demux.h: * gst/id3demux/id3tags.c: * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: Port ID3 tag demuxer over to the new GstTagDemux in -base (now would be a good time to test re-importing your music collection). 2007-10-06 15:13:09 +0000 Tim-Philipp Müller gst/apetag/: Port APE tag demuxer over to the new GstTagDemux in -base. Original commit message from CVS: * gst/apetag/Makefile.am: * gst/apetag/gstapedemux.c: * gst/apetag/gstapedemux.h: * gst/apetag/gsttagdemux.c: * gst/apetag/gsttagdemux.h: Port APE tag demuxer over to the new GstTagDemux in -base. 2007-10-05 13:18:19 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Improve flushing behaviour. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_internal_src_query), (gst_rtspsrc_handle_src_query), (new_session_pad), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_send_cmd): Improve flushing behaviour. Set state of the udp sources to PAUSE/PLAYING correctly. Handle events and queries for UDP and TCP transport now. 2007-10-04 07:29:48 +0000 Stefan Kost gst/rtp/: Add log category. Original commit message from CVS: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: Add log category. 2007-10-04 07:24:02 +0000 Timo Hotti tests/check/: Add unit tests for payloaders/depayloaders. Original commit message from CVS: Patch by: Timo Hotti * tests/check/Makefile.am: * tests/check/pipelines/simple-launch-lines.c: Add unit tests for payloaders/depayloaders. 2007-10-02 10:49:03 +0000 Stefan Kost gst/avi/gstavimux.*: Also save codec data for audio streams. Fixes #482495. Original commit message from CVS: * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: Also save codec data for audio streams. Fixes #482495. 2007-10-02 10:23:04 +0000 Stefan Kost gst/avi/gstavimux.c: Fix "Index entry has invalid stream nr 1". Original commit message from CVS: * gst/avi/gstavimux.c: Fix "Index entry has invalid stream nr 1". Add support for muxing aac - work in progress (see #482495). 2007-10-01 16:34:56 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth), (gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved): * gst/rtsp/gstrtspsrc.h: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't have an API for that yet. 2007-10-01 13:57:28 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Use shiny new function in -base to get the default clock-rate. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved): Use shiny new function in -base to get the default clock-rate. Update some docs. 2007-09-29 12:50:36 +0000 Sébastien Moutte win32/MANIFEST: Add files to win32 manifest. Original commit message from CVS: * win32/MANIFEST: Add files to win32 manifest. * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstqtdemux.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: Update project files. 2007-09-28 14:56:19 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense to try to skew compensate, also some servers send the first batch of data in a burst. 2007-09-27 15:00:30 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Fix setting the discont flag on the first buffer pushed downstream for formats with pr... Original commit message from CVS: * gst/matroska/matroska-demux.c: Fix setting the discont flag on the first buffer pushed downstream for formats with private codec data that needs to be deserialised into buffers (such as vorbis and FLAC when in a matroska container). 2007-09-27 11:10:12 +0000 Antoine Tremblay gst/rtp/gstrtpmp4vpay.*: Free the config string. Fixes #480707. Original commit message from CVS: Patch by: Antoine Tremblay * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init), (gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush), (gst_rtp_mp4v_pay_handle_buffer): * gst/rtp/gstrtpmp4vpay.h: Free the config string. Fixes #480707. Clean up the timestamp code a little. 2007-09-26 20:12:52 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: Set timestamps on RTP buffers in interleaved mode. Mark first buffers with a DISCONT. Remove flush hack now that sync for live sources has been figured out. 2007-09-26 14:28:20 +0000 Wim Taymans gst/udp/gstudpsrc.c: Update documentation. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Update documentation. 2007-09-26 14:26:39 +0000 Wim Taymans gst/qtdemux/gstrtpxqtdepay.*: Fail if we don't know the quicktime format. Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_change_state): * gst/qtdemux/gstrtpxqtdepay.h: Fail if we don't know the quicktime format. 2007-09-26 13:40:35 +0000 Tim-Philipp Müller ext/lame/gstlame.c: Fix up case where there is no peer, in which case _get_allowed_caps() will return NULL. Original commit message from CVS: * ext/lame/gstlame.c: Fix up case where there is no peer, in which case _get_allowed_caps() will return NULL. 2007-09-26 13:19:17 +0000 Tim-Philipp Müller ext/flac/gstflacenc.*: Save the flow return from the last gst_pad_push() and make sure we pass the right flow return ... Original commit message from CVS: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: Save the flow return from the last gst_pad_push() and make sure we pass the right flow return value upstream in the case of failure; minor clean-ups. 2007-09-25 19:09:33 +0000 Tim-Philipp Müller Add support for the new GST_TAG_COMPOSER (#459809). Original commit message from CVS: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * gst/apetag/gstapedemux.c: Add support for the new GST_TAG_COMPOSER (#459809). 2007-09-25 17:18:34 +0000 Tim-Philipp Müller gst/law/: Compulsive clean-ups: use boilerplate macros, add debug categories, fix up things to conform to symbol nome... Original commit message from CVS: * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: * gst/law/alaw.c: * gst/law/mulaw-conversion.h: Compulsive clean-ups: use boilerplate macros, add debug categories, fix up things to conform to symbol nomenklatura, etc. 2007-09-25 16:05:29 +0000 Laurent Glayal gst/law/: Use static tables for A-Law decoding and encoding; this makes Original commit message from CVS: Based on patch by: Laurent Glayal * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: Use static tables for A-Law decoding and encoding; this makes A-Law decoding and encoding less CPU-intensive, but increases the binary size a bit. Leaving old code around for now, selectable by a define in the code. Fixes #435435. 2007-09-25 13:20:27 +0000 Tim-Philipp Müller ext/lame/gstlame.c: Use GST_PTR_FORMAT to print caps in debug statement. Original commit message from CVS: * ext/lame/gstlame.c: Use GST_PTR_FORMAT to print caps in debug statement. 2007-09-25 08:51:36 +0000 Sebastian Dröge configure.ac: Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and Original commit message from CVS: * configure.ac: Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and AG_GST_ARG_ENABLE_EXPERIMENTAL instead of duplicating those macros in configure.ac. 2007-09-25 05:03:58 +0000 Sebastian Dröge gst/qtdemux/qtdemux.c: Add fourccs for MPEG2 HDV streams. Fixes #479960. Original commit message from CVS: Patch by: * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add fourccs for MPEG2 HDV streams. Fixes #479960. 2007-09-24 10:53:36 +0000 Stefan Kost Massive leak fixing, plus code cleanups. Original commit message from CVS: * ext/audioresample/gstaudioresample.c: * ext/x264/gstx264enc.c: * gst/dvdspu/gstdvdspu.c: * gst/dvdspu/gstdvdspu.h: * gst/festival/gstfestival.c: * gst/h264parse/gsth264parse.c: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtsparse.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/nuvdemux/gstnuvdemux.c: * sys/dshowsrcwrapper/gstdshowaudiosrc.c: * sys/dshowsrcwrapper/gstdshowvideosrc.c: * sys/vcd/vcdsrc.c: Massive leak fixing, plus code cleanups. 2007-09-24 10:26:21 +0000 Thomas Vander Stichele ext/lame/gstlame.c: Allow fixing the sample rate lame converts to by negotiating fixed sample rate on the src pad caps. Original commit message from CVS: * ext/lame/gstlame.c: Allow fixing the sample rate lame converts to by negotiating fixed sample rate on the src pad caps. Add docs for it. * tests/check/Makefile.am: * tests/check/pipelines/lame.c: Add a check for it. 2007-09-23 18:57:14 +0000 Stefan Kost sys/oss/gstosshelper.c: Use GST_WARNING instead of a g_critical. This situation is not caused by the application. Original commit message from CVS: * sys/oss/gstosshelper.c: Use GST_WARNING instead of a g_critical. This situation is not caused by the application. 2007-09-22 18:15:12 +0000 Thomas Vander Stichele po/: Updated translations. Original commit message from CVS: * po/LINGUAS: * po/nl.po: Updated translations. 2007-09-22 18:13:58 +0000 Thomas Vander Stichele po/eu.po: Added Basque translation. Original commit message from CVS: translated by: Mikel Olasagasti * po/eu.po: Added Basque translation. 2007-09-22 18:13:10 +0000 Thomas Vander Stichele po/: Added Chinese (traditional and Hong Kong) translation. Original commit message from CVS: translated by: Abel Cheung * po/zh_HK.po: * po/zh_TW.po: Added Chinese (traditional and Hong Kong) translation. 2007-09-22 18:10:42 +0000 Thomas Vander Stichele po/pl.po: Added Polish translation. Original commit message from CVS: translated by: Jakub Bogusz * po/pl.po: Added Polish translation. 2007-09-22 18:09:59 +0000 Thomas Vander Stichele po/fi.po: Added Finnish translation. Original commit message from CVS: translated by: Ilkka Tuohela * po/fi.po: Added Finnish translation. 2007-09-22 18:09:09 +0000 Thomas Vander Stichele po/es.po: Added Spanish translation. Original commit message from CVS: translated by: Jorge González González * po/es.po: Added Spanish translation. 2007-09-22 18:08:13 +0000 Thomas Vander Stichele po/da.po: Added Danish translation. Original commit message from CVS: translated by: Mogens Jaeger * po/da.po: Added Danish translation. 2007-09-22 18:06:55 +0000 Thomas Vander Stichele po/zh_CN.po: Added Chinese (simplified) translation. Original commit message from CVS: translated by: Funda Wang * po/zh_CN.po: Added Chinese (simplified) translation. 2007-09-22 18:05:37 +0000 Thomas Vander Stichele po/bg.po: Added Bulgarian translation. Original commit message from CVS: translated by: Alexander Shopov * po/bg.po: Added Bulgarian translation. 2007-09-22 08:12:57 +0000 Thomas Vander Stichele * common: * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: fix header and comments Original commit message from CVS: fix header and comments 2007-09-21 11:34:34 +0000 Wim Taymans gst/rtp/gstrtpamrdepay.c: Set outgoing packet duration because we can. Fixes #478244 some more. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process): Set outgoing packet duration because we can. Fixes #478244 some more. 2007-09-20 13:35:34 +0000 Stefan Kost ext/cairo/gsttextoverlay.c: Add info about static leak. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: Add info about static leak. * tests/check/Makefile.am: * tests/check/generic/states.c: Improved state change unit test. 2007-09-19 18:19:49 +0000 Stefan Kost Ignore registries in any format. Original commit message from CVS: * docs/plugins/.cvsignore: * tests/check/.cvsignore: Ignore registries in any format. 2007-09-19 16:24:09 +0000 Wim Taymans gst/rtp/gstrtpL16pay.c: Removed some unused code. Original commit message from CVS: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer): Removed some unused code. * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer): * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer): * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet), (gst_rtp_theora_pay_flush_packet): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet): Try to preserve the incomming buffer duration on the outgoing packets. Fixes #478244. 2007-09-19 10:22:40 +0000 Tim-Philipp Müller ext/taglib/: Work around compiler warnings with g++-4.2 when assigning a string constant to a gchar * (partially fixe... Original commit message from CVS: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: Work around compiler warnings with g++-4.2 when assigning a string constant to a gchar * (partially fixes #478092). 2007-09-18 16:44:46 +0000 Tim-Philipp Müller configure.ac: We require core CVS now for gst_base_src_set_do_timestamp(). Original commit message from CVS: * configure.ac: We require core CVS now for gst_base_src_set_do_timestamp(). 2007-09-18 13:55:06 +0000 Stefan Kost gst/spectrum/: Handling window resize. Original commit message from CVS: * gst/spectrum/demo-audiotest.c: * gst/spectrum/demo-osssrc.c: Handling window resize. 2007-09-18 11:45:06 +0000 Stefan Kost ChangeLog: Add missing newline. Original commit message from CVS: * ChangeLog: Add missing newline. * gst/librfb/rfbdecoder.c: Fix the build (missing stdlib.h). * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: Use basetransform segment so that it is correctly managed on flushes and start/stop. Report message timestamp as stream time, which is what an application can understand. (Yes these are adapted from wim recent level element changes) 2007-09-17 17:35:13 +0000 Jan Schmidt gst/: Fix compiler warnings shown with Forte. Original commit message from CVS: * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_handle_message): Fix compiler warnings shown with Forte. 2007-09-17 02:05:14 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to configure for some reason. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams), (gst_rtspsrc_dup_printf): Give meaningfull error when all streams failed to configure for some reason. 2007-09-16 19:13:58 +0000 Wim Taymans gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver. Original commit message from CVS: * gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver. 2007-09-14 09:40:49 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the element driving the pipeline is responsible f... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_loop), (gst_wavparse_chain): Don't push EOS from the chain function, the element driving the pipeline is responsible for this. The bug this was meant to fix seems to be queue not forwarding EOS in all cases (see #476514). 2007-09-13 17:31:16 +0000 Wim Taymans gst/level/gstlevel.*: Use basetransform segment so that it is correctly managed on flushes and start/stop. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start), (gst_level_transform_ip): * gst/level/gstlevel.h: Use basetransform segment so that it is correctly managed on flushes and start/stop. Report message timestamp as stream time, which is what an application can understand. 2007-09-13 15:04:15 +0000 Sebastian Dröge Update my mail address. Original commit message from CVS: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstapev2mux.h: * ext/taglib/gsttaglibmux.c: * tests/check/elements/apev2mux.c: Update my mail address. 2007-09-13 12:37:56 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Add EOS logic for the push-based mode too. Fixes #476514. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos), (gst_wavparse_loop), (gst_wavparse_chain): Add EOS logic for the push-based mode too. Fixes #476514. 2007-09-12 22:01:59 +0000 Wim Taymans gst/law/: Fix law encoder timestamps. Original commit message from CVS: * gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain): * gst/law/alaw-encode.h: * gst/law/mulaw-encode.c: (gst_mulawenc_init), (gst_mulawenc_chain): * gst/law/mulaw-encode.h: Fix law encoder timestamps. 2007-09-12 09:13:39 +0000 Stefan Kost ext/gconf/gstgconfaudiosink.c: Fix warning when building without debug. Original commit message from CVS: * ext/gconf/gstgconfaudiosink.c: Fix warning when building without debug. * sys/oss/gstossmixertrack.c: Use const like in alsamixertrack.c (fixes warnings). 2007-09-12 08:38:21 +0000 Peter Kjellerstedt gst/: Printf format fixes (#476128). Original commit message from CVS: Patch by: Peter Kjellerstedt * gst-libs/gst/app/gstappsink.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvparse.c: * gst/interleave/deinterleave.c: * gst/switch/gstswitch.c: Printf format fixes (#476128). 2007-09-11 15:37:55 +0000 Wim Taymans sys/v4l2/v4l2src_calls.c: Fix framerate detection code some more. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format_and_size): Fix framerate detection code some more. Handle the case where there is a weird step in the stepwise framerates. Don't overwrite the min interval with the framerate, use a temp variable instead. Use max in the Continuous framerate intervals instead of step, which is 1 according to the docs. Fixes #475424. 2007-09-10 19:53:28 +0000 Wim Taymans gst/udp/gstudpsrc.c: Make udpsrc timestamp outgoing buffers based on when they were received. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create): Make udpsrc timestamp outgoing buffers based on when they were received. Also make it output a segment in time. 2007-09-10 06:49:32 +0000 Stefan Kost gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups. Original commit message from CVS: * gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups. 2007-09-09 18:08:36 +0000 Tim-Philipp Müller configure.ac: Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old flac versions, 's good for cross-compilation ... Original commit message from CVS: * configure.ac: Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old flac versions, 's good for cross-compilation karma. 2007-09-07 18:04:41 +0000 Haakon Sporsheim gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add padding between the structure fields, si... Original commit message from CVS: Patch by: Haakon Sporsheim * gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add padding between the structure fields, since that would lead to us sending RTP packets with broken headers (as is currently the case when compiling with MSVC). Also see similar fixes in libgstrtp in gst-plugins-base. (#474616; #471194) 2007-09-07 16:04:14 +0000 Wim Taymans sys/v4l2/v4l2src_calls.c: Don't overwrite our GValue with 0 but instead use the previously computed value. Fixes #471... Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format_and_size): Don't overwrite our GValue with 0 but instead use the previously computed value. Fixes #471823 some more. 2007-09-07 15:54:38 +0000 Sebastian Dröge gst/spectrum/gstspectrum.c: Use the correct parameter order for the memset calls. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_start), (gst_spectrum_transform_ip): Use the correct parameter order for the memset calls. Thanks to Christian Schaller for noticing. 2007-09-06 12:00:36 +0000 Tim-Philipp Müller docs/plugins/gst-plugins-good-plugins.hierarchy: No tabs in this file please, or gtk-doc will end up documenting rath... Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins.hierarchy: No tabs in this file please, or gtk-doc will end up documenting rather absurd class hierarchies. 2007-09-06 10:48:56 +0000 Tim-Philipp Müller ext/gconf/gstswitchsink.c: If the new kid element fails to change state for some reason forward the error message it ... Original commit message from CVS: * ext/gconf/gstswitchsink.c: If the new kid element fails to change state for some reason (e.g. esdsink not being able to connect to the sound server), forward the error message it posted on the bus instead of just posting a generic 'Internal state change error: please file a bug' error message. Fixes #471364. 2007-09-06 07:21:22 +0000 Sebastian Dröge Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message... Original commit message from CVS: * configure.ac: * gst/spectrum/Makefile.am: * gst/spectrum/demo-audiotest.c: (draw_spectrum), (message_handler), (main): * gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler): * gst/spectrum/gstspectrum.c: (gst_spectrum_base_init), (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_dispose), (gst_spectrum_set_property), (gst_spectrum_get_property), (gst_spectrum_start), (gst_spectrum_setup), (gst_spectrum_message_new), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message contents, average all FFTs done in one interval for better results, use a better windowing function, allow posting the phase in the message and actually do an FFT with the requested number of bands instead of interpolating. * tests/check/elements/spectrum.c: (GST_START_TEST), (spectrum_suite): Improve the units tests by checking for a 11025Hz sine wave and add unit tests for all 4 supported sample types. 2007-09-05 16:23:21 +0000 Tim-Philipp Müller gst/qtdemux/: Don't assume tags are encoded as UTF-8 (#473670). Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: Don't assume tags are encoded as UTF-8 (#473670). 2007-09-05 14:43:16 +0000 Tim-Philipp Müller sys/v4l2/: Implement LATENCY queries in the crudest way possible so I don't have to use sync=false any longer when te... Original commit message from CVS: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/v4l2src_calls.c: Implement LATENCY queries in the crudest way possible so I don't have to use sync=false any longer when testing with videosinks. 2007-09-05 09:25:23 +0000 Tim-Philipp Müller configure.ac: Fix build. Original commit message from CVS: * configure.ac: Fix build. 2007-09-05 00:12:46 +0000 Wim Taymans sys/v4l2/v4l2src_calls.c: Add some more debugging in the framerate function. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format_and_size): Add some more debugging in the framerate function. Iterate stepwise framerate up to and _including_ the max and if nothing was added to the list, add a dummy 0/1 to 100/1 framerate so that we don't end up with an empty list. 2007-09-04 22:42:21 +0000 Wim Taymans gst/udp/gstmultiudpsink.c: Add property do configure destination address/port pairs Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_set_clients_string), (gst_multiudpsink_get_clients_string), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_add_internal), (gst_multiudpsink_add), (gst_multiudpsink_clear_internal), (gst_multiudpsink_clear): Add property do configure destination address/port pairs API:GstMultiUDPSink::clients 2007-09-04 18:30:22 +0000 Wim Taymans tests/examples/: Added some RTP example scripts for sending and receiving RTP streams. Original commit message from CVS: * tests/examples/Makefile.am: * tests/examples/rtp/Makefile.am: * tests/examples/rtp/client-H263p-AMR.sh: * tests/examples/rtp/client-H263p-PCMA.sdp: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H264-PCMA.sdp: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Added some RTP example scripts for sending and receiving RTP streams. 2007-09-04 16:40:05 +0000 Wim Taymans sys/v4l2/gstv4l2src.c: Restructure the setcaps function so that we can also compute the expected GStreamer output siz... Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info), (gst_v4l2src_set_caps), (gst_v4l2src_get_mmap): Restructure the setcaps function so that we can also compute the expected GStreamer output size of the video frames. Set frame_byte_size correctly so that read-based devices have a chance of working correctly. When grabbing a frame, discard frames that are not of the expected size. Some cameras don't output the right framesize for the first buffer. Try only a couple of times to get a valid frame, else error out. * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities), (gst_v4l2_fill_lists), (gst_v4l2_get_input): Add some more debug info when scanning the device. * sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new), (gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate), (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init): Add some more debug info when dequeing a frame. 2007-09-04 14:37:22 +0000 Stefan Kost gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and improve debugs logs. Original commit message from CVS: * gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and improve debugs logs. 2007-09-04 07:58:36 +0000 Stefan Kost gst/wavparse/gstwavparse.*: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int... Original commit message from CVS: * gst/wavparse/gstwavparse.c: * gst/wavparse/gstwavparse.h: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff out of loops. 2007-09-03 07:44:34 +0000 Stefan Kost gst/avi/gstavidemux.c: Implement seek-query. Original commit message from CVS: * gst/avi/gstavidemux.c: Implement seek-query. 2007-08-29 21:43:08 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP packet not wait for preroll. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_dup_printf): Use new basesink async property to make sparse RTCP packet not wait for preroll. 2007-08-27 14:44:19 +0000 Jan Schmidt gst/audiofx/Makefile.am: Dist the right file. Original commit message from CVS: * gst/audiofx/Makefile.am: Dist the right file. 2007-08-23 16:27:36 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values in the POSIX locale instead of the curre... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf), (gst_rtspsrc_get_float), (gst_rtspsrc_play): Make sure we generate and parse floating point values in the POSIX locale instead of the current locale. 2007-08-22 15:01:29 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Fix method detection again. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: Fix method detection again. Keep track of when we must send a Range header. Use segment values for Range, Speed and Scale headers. Parse Speed and Scale headers to update the segment values. 2007-08-22 08:22:50 +0000 Mark Nauwelaerts sys/v4l2/v4l2src_calls.c: Handle optional v4l2 ioctls gracefully. Original commit message from CVS: patch by: Mark Nauwelaerts * sys/v4l2/v4l2src_calls.c: Handle optional v4l2 ioctls gracefully. 2007-08-20 16:52:03 +0000 Wim Taymans gst/rtp/: Added an H263 depayloader. Fixes #369392. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init), (gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init), (gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps), (gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property), (gst_rtp_h263_depay_get_property), (gst_rtp_h263_depay_change_state), (gst_rtp_h263_depay_plugin_init): * gst/rtp/gstrtph263depay.h: Added an H263 depayloader. Fixes #369392. * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps), (gst_rtp_h263p_depay_process): * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type), (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush): Make the H263+ pay/depayloader support H263-1998 and H263-2000 payloads. Also alow plain H263 on the h263p payloaders. Fixes #465040. 2007-08-19 19:16:33 +0000 Sebastian Dröge gst/filter/: Add small comparision with the chebyshev filters in the docs. Original commit message from CVS: * gst/filter/gstbpwsinc.c: * gst/filter/gstlpwsinc.c: Add small comparision with the chebyshev filters in the docs. 2007-08-19 19:11:04 +0000 Sebastian Dröge gst/audiofx/: Add small comparision with the windowed sinc filters in the docs. Original commit message from CVS: * gst/audiofx/audiochebyshevfreqband.c: * gst/audiofx/audiochebyshevfreqlimit.c: Add small comparision with the windowed sinc filters in the docs. 2007-08-19 19:01:45 +0000 Sebastian Dröge tests/check/elements/: Also test everything in 32 bit float mode. Original commit message from CVS: * tests/check/elements/bpwsinc.c: (GST_START_TEST), (bpwsinc_suite): * tests/check/elements/lpwsinc.c: (GST_START_TEST), (lpwsinc_suite): Also test everything in 32 bit float mode. 2007-08-19 18:47:19 +0000 Sebastian Dröge tests/check/elements/: Also test 32 bit float mode and the type 2 variants of the filters. Original commit message from CVS: * tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST), (audiochebyshevfreqband_suite): * tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST), (audiochebyshevfreqlimit_suite): Also test 32 bit float mode and the type 2 variants of the filters. 2007-08-18 19:44:55 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Refactor the udp and interleaved loop function a bit. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop): Refactor the udp and interleaved loop function a bit. 2007-08-17 17:08:11 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_connection_send), (gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain), (gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455808. 2007-08-17 15:30:39 +0000 Wim Taymans gst/debug/rndbuffersize.c: Fix debug statement. Original commit message from CVS: * gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop): Fix debug statement. 2007-08-17 15:28:40 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Fix stray %u in debug line as spotted by Saur on IRC. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos): Fix stray %u in debug line as spotted by Saur on IRC. 2007-08-17 15:05:17 +0000 Sebastian Dröge Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GOb... Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (bpwsinc_set_property), (bpwsinc_get_property): * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property), (lpwsinc_get_property): * gst/filter/gstlpwsinc.h: * tests/check/elements/lpwsinc.c: (GST_START_TEST): Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GObject properties so automatically generated UIs can use sliders and change frequency properties to floats to save a bit of memory, even ints would in theory be enough. Also rename frequency to cutoff for consistency reasons. * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-gstrtpmanager.xml: Regenerated for the above changes. 2007-08-17 14:43:33 +0000 Sebastian Dröge gst/audiofx/: Use generator macros for the process functions for the different sample types, add lower upper boundari... Original commit message from CVS: * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_class_init): * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_class_init): Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GObject properties so automatically generated UIs can use sliders and add a note about the number of poles as a too high number of poles combined with very low or very high frequencies will produce only noise. * docs/plugins/gst-plugins-good-plugins.args: Regenerated for the property changes. 2007-08-17 14:15:19 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Improve timeout handling. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property), (gst_rtspsrc_flush), (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Improve timeout handling. Use the same socket for sending and receiving RTCP packets so that some servers can track clients better. Improve connection closed handling. Try to reconnect. Don't overwrite our content base with NULL. Improve debugging. Improve range parsing and handling. Remove flushing hack now that core does the right thing. 2007-08-17 13:59:15 +0000 Wim Taymans gst/udp/gstmultiudpsink.*: Add support for getting and setting the socket to use. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_init), (gst_multiudpsink_set_property), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_close), (gst_multiudpsink_add): * gst/udp/gstmultiudpsink.h: Add support for getting and setting the socket to use. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_get_property): Add support for getting the currently used socket. 2007-08-16 19:22:48 +0000 Sebastian Dröge gst/filter/gstbpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ... Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (gst_bpwsinc_init), (process_32), (process_64), (bpwsinc_build_kernel), (bpwsinc_push_residue), (bpwsinc_transform), (bpwsinc_start), (bpwsinc_query), (bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property): * gst/filter/gstbpwsinc.h: Implement latency query and only forward those samples downstream that actually contain the data we want, i.e. drop kernel_length/2 in the beginning and append kernel_length/2 (created by convolving the filter kernel with zeroes) to the end. * tests/check/elements/bpwsinc.c: (GST_START_TEST): Adjust the unit test for this slightly changed behaviour. * gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel): Reset residue length only when actually creating a residue. 2007-08-16 17:02:07 +0000 Sebastian Dröge gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements. Original commit message from CVS: reviewed by: Stefan Kost * gst/audiofx/Makefile.am: * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_mode_get_type), (gst_audio_chebyshev_freq_band_base_init), (gst_audio_chebyshev_freq_band_dispose), (gst_audio_chebyshev_freq_band_class_init), (gst_audio_chebyshev_freq_band_init), (generate_biquad_coefficients), (calculate_gain), (generate_coefficients), (gst_audio_chebyshev_freq_band_set_property), (gst_audio_chebyshev_freq_band_get_property), (gst_audio_chebyshev_freq_band_setup), (process), (process_64), (process_32), (gst_audio_chebyshev_freq_band_transform_ip), (gst_audio_chebyshev_freq_band_start): * gst/audiofx/audiochebyshevfreqband.h: * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_mode_get_type), (gst_audio_chebyshev_freq_limit_base_init), (gst_audio_chebyshev_freq_limit_dispose), (gst_audio_chebyshev_freq_limit_class_init), (gst_audio_chebyshev_freq_limit_init), (generate_biquad_coefficients), (calculate_gain), (generate_coefficients), (gst_audio_chebyshev_freq_limit_set_property), (gst_audio_chebyshev_freq_limit_get_property), (gst_audio_chebyshev_freq_limit_setup), (process), (process_64), (process_32), (gst_audio_chebyshev_freq_limit_transform_ip), (gst_audio_chebyshev_freq_limit_start): * gst/audiofx/audiochebyshevfreqlimit.h: * gst/audiofx/audiofx.c: (plugin_init): Add Chebyshev lowpass/highpass and bandpass/bandreject elements. Fixes #464800. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/audiochebyshevfreqband.c: (setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband), (GST_START_TEST), (audiochebyshevfreqband_suite), (main): * tests/check/elements/audiochebyshevfreqlimit.c: (setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit), (GST_START_TEST), (audiochebyshevfreqlimit_suite), (main): Add unit tests for the chebyshev filters. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-wavpack.xml: And add docs for the chebyshev filters. While doing that also run make update in docs/plugins. 2007-08-16 12:15:06 +0000 Stefan Kost Make ro memory to share. Original commit message from CVS: * ext/annodex/gstcmmltag.c: * gst/rtp/gstrtpvorbispay.c: Make ro memory to share. 2007-08-16 11:49:01 +0000 Wim Taymans gst/udp/gstudpsrc.c: Improve UDP performance by avoiding a select() when we have data available immediatly. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Improve UDP performance by avoiding a select() when we have data available immediatly. 2007-08-16 11:47:19 +0000 Wim Taymans gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals. Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT), (gst_rtp_dec_class_init): * gst/rtsp/gstrtpdec.h: Add (dummy) SSRC management signals. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (find_stream), (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc), (on_timeout), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add connection-speed property. Add find_stream helper functions. Handle stream EOS based on BYE messages or SSRC timeout. Returns SUCCESS from the state change function as we hide our async elements from the parent. 2007-08-16 09:48:27 +0000 Sebastian Dröge gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ... Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32), (process_64), (lpwsinc_build_kernel), (lpwsinc_push_residue), (lpwsinc_transform), (lpwsinc_start), (lpwsinc_query), (lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property): * gst/filter/gstlpwsinc.h: Implement latency query and only forward those samples downstream that actually contain the data we want, i.e. drop kernel_length/2 in the beginning and append kernel_length/2 (created by convolving the filter kernel with zeroes) to the end. * tests/check/elements/lpwsinc.c: (GST_START_TEST): Adjust the unit test for this slightly changed behaviour. 2007-08-16 07:40:48 +0000 Stefan Kost gst/debug/rndbuffersize.c: Fix da leak. Original commit message from CVS: * gst/debug/rndbuffersize.c: Fix da leak. 2007-08-14 13:50:43 +0000 Stefan Kost gst/debug/: Add new test element and clean-up the others a little. Original commit message from CVS: * gst/debug/Makefile.am: * gst/debug/breakmydata.c: * gst/debug/gstdebug.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/rndbuffersize.c: * gst/debug/testplugin.c: Add new test element and clean-up the others a little. 2007-08-13 13:50:39 +0000 Sebastian Dröge Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: * gst/filter/gstbpwsinc.c: * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: * gst/filter/gstlpwsinc.h: Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other docs via make update in docs/plugins. 2007-08-12 20:55:01 +0000 Sebastian Dröge tests/check/elements/bpwsinc.c: Make one test constraint a bit stricter. Original commit message from CVS: * tests/check/elements/bpwsinc.c: (GST_START_TEST): Make one test constraint a bit stricter. 2007-08-12 20:53:11 +0000 Sebastian Dröge tests/check/: Add unit tests for bpwsinc, testing fundamental functionality again. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/bpwsinc.c: (setup_bpwsinc), (cleanup_bpwsinc), (GST_START_TEST), (bpwsinc_suite), (main): Add unit tests for bpwsinc, testing fundamental functionality again. 2007-08-12 20:19:37 +0000 Sebastian Dröge tests/check/: Add unit tests for lpwsinc, testing fundamental functionality. Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/lpwsinc.c: (setup_lpwsinc), (cleanup_lpwsinc), (GST_START_TEST), (lpwsinc_suite), (main): Add unit tests for lpwsinc, testing fundamental functionality. 2007-08-12 15:41:57 +0000 Sebastian Dröge gst/filter/: Improve debugging a bit. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel): * gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel): Improve debugging a bit. 2007-08-12 14:35:41 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Fix parsing of mp4a version 0 atoms. Fixes #465774. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_node): Fix parsing of mp4a version 0 atoms. Fixes #465774. 2007-08-12 12:46:20 +0000 Sebastian Dröge gst/filter/: Reset the residue in BaseTransform::start to get a clean residue on stream changes. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (bpwsinc_start): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (lpwsinc_start): Reset the residue in BaseTransform::start to get a clean residue on stream changes. 2007-08-11 15:58:30 +0000 Sebastian Dröge gst/filter/: Fix processing with buffer sizes that are larger than the filter kernel size. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (process_32), (process_64): * gst/filter/gstlpwsinc.c: (process_32), (process_64): Fix processing with buffer sizes that are larger than the filter kernel size. 2007-08-10 17:08:01 +0000 Stefan Kost gst/rtp/gstrtpilbcdepay.c: Include stdlib. Original commit message from CVS: * gst/rtp/gstrtpilbcdepay.c: Include stdlib. 2007-08-10 16:10:47 +0000 Wim Taymans gst/rtp/gstrtpmpvdepay.c: Set the mpegversion in the caps so that autoplugging does not get confused. Original commit message from CVS: * gst/rtp/gstrtpmpvdepay.c: Set the mpegversion in the caps so that autoplugging does not get confused. 2007-08-10 05:51:40 +0000 Sebastian Dröge gst/filter/gstbpwsinc.c: Fix a segfault with more than one channel and don't rebuild the kernel & residue with every ... Original commit message from CVS: * gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel): Fix a segfault with more than one channel and don't rebuild the kernel & residue with every buffer. 2007-08-10 05:35:25 +0000 Sebastian Dröge gst/filter/gstbpwsinc.*: Add support for a bandreject mode and allow specifying the window function that should be used. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type), (gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init), (gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property), (bpwsinc_get_property): * gst/filter/gstbpwsinc.h: Add support for a bandreject mode and allow specifying the window function that should be used. * gst/filter/gstlpwsinc.c: And another small formatting fix. 2007-08-10 05:20:06 +0000 Sebastian Dröge gst/filter/gstbpwsinc.*: Apply the same changes to the bandpass filter: Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init), (gst_bpwsinc_init), (process_32), (process_64), (bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size), (bpwsinc_transform), (bpwsinc_set_property), (bpwsinc_get_property): * gst/filter/gstbpwsinc.h: Apply the same changes to the bandpass filter: - Support double input - Fix processing for input with >1 channels - Specify frequency in Hz - Specify actual filter kernel length - Use transform instead of transform_ip as we're working out of place anyway - Factor out filter kernel generation and update the filter kernel when the properties are set Fix bandpass filter kernel generation to actually generate a bandpass filter by creating a highpass instead of a second lowpass. * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): Small formatting fix. 2007-08-10 04:44:43 +0000 Sebastian Dröge gst/filter/gstlpwsinc.*: Specify the actual filter length instead of a weird 2N+1. Setting the property will round to... Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32), (process_64), (lpwsinc_build_kernel), (lpwsinc_set_property), (lpwsinc_get_property): * gst/filter/gstlpwsinc.h: Specify the actual filter length instead of a weird 2N+1. Setting the property will round to the next odd number. Also remove now obsolete FIXMEs. 2007-08-10 04:32:47 +0000 Sebastian Dröge gst/filter/gstlpwsinc.*: Allow choosing between hamming and blackman window. The blackman window provides a better st... Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_window_get_type), (gst_lpwsinc_class_init), (gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property), (lpwsinc_get_property): * gst/filter/gstlpwsinc.h: Allow choosing between hamming and blackman window. The blackman window provides a better stopband attenuation but a bit slower rolloff. 2007-08-10 04:21:39 +0000 Sebastian Dröge gst/filter/gstlpwsinc.*: Add a highpass mode. Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_mode_get_type), (gst_lpwsinc_class_init), (process_32), (process_64), (lpwsinc_build_kernel), (lpwsinc_set_property), (lpwsinc_get_property): * gst/filter/gstlpwsinc.h: Add a highpass mode. 2007-08-10 04:06:53 +0000 Sebastian Dröge gst/filter/gstlpwsinc.c: Fix processing if the input has more than one channel. Original commit message from CVS: * gst/filter/gstlpwsinc.c: (process_32), (process_64), (lpwsinc_build_kernel): Fix processing if the input has more than one channel. 2007-08-09 19:23:33 +0000 Sebastian Dröge gst/filter/gstbpwsinc.c: "this" is a C++ keyword, use "self" instead. Original commit message from CVS: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose), (gst_bpwsinc_init), (bpwsinc_setup), (bpwsinc_transform_ip), (bpwsinc_set_property), (bpwsinc_get_property): "this" is a C++ keyword, use "self" instead. Add TODOs and FIXMEs and remove two wrong FIXMEs. * gst/filter/gstlpwsinc.c: Add FIXMEs and a new TODO. 2007-08-09 18:08:05 +0000 Sebastian Dröge gst/filter/gstlpwsinc.*: Add double support, replace "this" with "self" as the former is a C++ keyword. Original commit message from CVS: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose), (gst_lpwsinc_class_init), (gst_lpwsinc_init), (process_32), (process_64), (lpwsinc_build_kernel), (lpwsinc_setup), (lpwsinc_get_unit_size), (lpwsinc_transform), (lpwsinc_set_property), (lpwsinc_get_property): * gst/filter/gstlpwsinc.h: Add double support, replace "this" with "self" as the former is a C++ keyword. Implement the frequency property in Hz instead of fraction of sampling frequency. Remove some unecessary FIXMEs and add some TODOs, add some required locking and refactor the kernel generation into a separate function that is also called when the properties change now. And use BaseTransform::transform instead of transform_ip as the convolution is done out of place anyway. Should be done in place later. 2007-08-09 17:39:47 +0000 Sebastian Dröge Port the stereo element to GStreamer 0.10. Original commit message from CVS: * configure.ac: * gst/stereo/Makefile.am: * gst/stereo/gststereo.c: (gst_stereo_base_init), (gst_stereo_class_init), (gst_stereo_init), (gst_stereo_transform_ip), (gst_stereo_set_property), (gst_stereo_get_property): * gst/stereo/gststereo.h: Port the stereo element to GStreamer 0.10. 2007-08-09 10:54:05 +0000 Thomas Vander Stichele po/: Updated translations. Original commit message from CVS: * po/hu.po: * po/uk.po: * po/vi.po: Updated translations. 2007-08-08 20:47:33 +0000 Sebastian Dröge gst/filter/: Use GstAudioFilter as base class and don't leak the memory of the filter kernel and residue. Original commit message from CVS: * gst/filter/Makefile.am: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose), (gst_bpwsinc_base_init), (gst_bpwsinc_class_init), (gst_bpwsinc_init), (bpwsinc_setup): * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose), (gst_lpwsinc_base_init), (gst_lpwsinc_class_init), (gst_lpwsinc_init), (lpwsinc_setup): * gst/filter/gstlpwsinc.h: Use GstAudioFilter as base class and don't leak the memory of the filter kernel and residue. 2007-08-08 17:47:05 +0000 Michael Smith gst/videobox/gstvideobox.c: Render right border in the correct location. Original commit message from CVS: * gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420): Render right border in the correct location. 2007-08-08 10:54:50 +0000 Olivier Crete gst/rtp/: Make mode property a string. Fixes #464475. Original commit message from CVS: Patch by: Olivier Crete * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps): * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps): Make mode property a string. Fixes #464475. 2007-08-05 14:58:20 +0000 Stefan Kost ext/flac/gstflacenc.c: Widen caps to match decoder a bit and add more FIXMEs. Original commit message from CVS: * ext/flac/gstflacenc.c: Widen caps to match decoder a bit and add more FIXMEs. 2007-08-05 14:53:36 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Fix ODML index tag numbering. Fixes #463624. Original commit message from CVS: patch by: Mark Nauwelaerts * gst/avi/gstavimux.c: Fix ODML index tag numbering. Fixes #463624. 2007-08-03 16:08:56 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Fix default clock-rate for realmedia. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_udp_sink): Fix default clock-rate for realmedia. Fix parsing of transport. Don't try to link NULL pads. 2007-07-30 17:17:04 +0000 Tim-Philipp Müller po/POTFILES.skip: Add POTFILES.skip with list of source files that aren't disted at the moment but contain translatab... Original commit message from CVS: * po/POTFILES.skip: Add POTFILES.skip with list of source files that aren't disted at the moment but contain translatable strings. Should hopefully pacify broken tools and make it clearer that these files are left out intentionally (#461600). 2007-07-30 12:41:58 +0000 Edward Hervey gst/qtdemux/qtdemux.c: If the buffer was entirely clipped ... don't try sending it :) Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie): If the buffer was entirely clipped ... don't try sending it :) 2007-07-27 16:56:45 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: If we don't hav a session manager, set the caps on outgoing buffers ourselves. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports): If we don't hav a session manager, set the caps on outgoing buffers ourselves. Force PAUSE/PLAY methods for now until the extensions can overwrite. Append final bit of the transport string even when it does not contain a placeholder. 2007-07-27 11:21:20 +0000 Wim Taymans gst/rtsp/: Clean up the interface list. Original commit message from CVS: * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free), (gst_rtsp_ext_list_connect): * gst/rtsp/gstrtspext.h: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_send_cb): Clean up the interface list. Allow connecting to interface signals for the extensions. Remove old extension code. Free list on cleanup. Allow extensions to send additional RTSP messages. 2007-07-27 10:38:34 +0000 Jan Schmidt ext/gconf/gconf.c: Handle a NULL gconf key gracefully by rendering the default element. Original commit message from CVS: * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default): Handle a NULL gconf key gracefully by rendering the default element. 2007-07-27 10:11:18 +0000 Wim Taymans gst/rtsp/gstrtspext.h: Fix include path for extension interface. Original commit message from CVS: * gst/rtsp/gstrtspext.h: Fix include path for extension interface. 2007-07-26 19:45:30 +0000 Sebastian Dröge gst/audiofx/audioamplify.h: Also remove a now unecessary variable here. Original commit message from CVS: * gst/audiofx/audioamplify.h: Also remove a now unecessary variable here. 2007-07-26 19:41:07 +0000 Sebastian Dröge gst/audiofx/: Don't save format information ourselves, this is already saved in Original commit message from CVS: * gst/audiofx/audioamplify.c: (gst_audio_amplify_init), (gst_audio_amplify_setup), (gst_audio_amplify_transform_ip): * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init), (gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip): * gst/audiofx/audiodynamic.h: * gst/audiofx/audioinvert.c: (gst_audio_invert_init), (gst_audio_invert_setup), (gst_audio_invert_transform_ip): * gst/audiofx/audioinvert.h: Don't save format information ourselves, this is already saved in GstAudioFilter. 2007-07-26 15:48:47 +0000 Wim Taymans gst/rtsp/: Use rank to filter out extensions. Original commit message from CVS: * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter), (gst_rtsp_ext_list_stream_select): * gst/rtsp/gstrtspext.h: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Use rank to filter out extensions. Add url to stream_select interface call. 2007-07-25 18:50:08 +0000 Wim Taymans gst/rtsp/: Use shiny new RTSP and SDP library. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/base64.c: * gst/rtsp/base64.h: * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter), (gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get), (gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send), (gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp), (gst_rtsp_ext_list_setup_media), (gst_rtsp_ext_list_configure_stream), (gst_rtsp_ext_list_get_transports), (gst_rtsp_ext_list_stream_select): * gst/rtsp/gstrtspext.h: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream), (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_flush), (gst_rtspsrc_do_seek), (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_mcast), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string), (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspconnection.c: * gst/rtsp/rtspconnection.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspext.h: * gst/rtsp/rtspextwms.c: * gst/rtsp/rtspextwms.h: * gst/rtsp/rtspmessage.c: * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsprange.c: * gst/rtsp/rtsprange.h: * gst/rtsp/rtsptransport.c: * gst/rtsp/rtsptransport.h: * gst/rtsp/rtspurl.c: * gst/rtsp/rtspurl.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.c: * gst/rtsp/sdpmessage.h: * gst/rtsp/test.c: Use shiny new RTSP and SDP library. Implement RTSP extensions using the new interface. Remove a lot of old code. 2007-07-24 14:31:56 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'. 2007-07-24 05:07:59 +0000 Sebastian Dröge ext/wavpack/gstwavpackdec.c: Don't unref the outgoing buffer twice when dropping it because it's outside of the segment. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): Don't unref the outgoing buffer twice when dropping it because it's outside of the segment. 2007-07-24 04:57:20 +0000 Sebastian Dröge Use the new buffer clipping function from gstaudio here and require gst-plugins-base CVS. Original commit message from CVS: * configure.ac: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event): Use the new buffer clipping function from gstaudio here and require gst-plugins-base CVS. * tests/check/elements/wavpackdec.c: (GST_START_TEST): For framed Wavpack buffers we require a valid timestamp. 2007-07-23 18:03:54 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Clip raw audio and video when we can, keep track of current output segment. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment), (gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie), (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps): Clip raw audio and video when we can, keep track of current output segment. Don't leak buffers and events when there is no output pad. Improve debugging here and there. 2007-07-23 09:02:07 +0000 Stefan Kost configure.ac: Sync liboil check with plugins-base. Original commit message from CVS: * configure.ac: Sync liboil check with plugins-base. 2007-07-20 11:37:37 +0000 Stefan Kost gst/equalizer/: Better algorith for the center frequencies. Subtract band filters from input for negative gains. Rewo... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_child_proxy_get_child_by_index), (gst_iir_equalizer_child_proxy_get_children_count), (gst_iir_equalizer_child_proxy_interface_init), (gst_iir_equalizer_class_init), (arg_to_scale), (setup_filter), (gst_iir_equalizer_compute_frequencies): * gst/equalizer/gstiirequalizer10bands.c: (gst_iir_equalizer_10bands_class_init): * gst/equalizer/gstiirequalizer3bands.c: (gst_iir_equalizer_3bands_class_init): * gst/equalizer/gstiirequalizernbands.c: Better algorith for the center frequencies. Subtract band filters from input for negative gains. Rework the gain mapping. 2007-07-20 07:41:58 +0000 Stefan Kost ext/annodex/Makefile.am: Fix CFLAGS/LIBS. Original commit message from CVS: * ext/annodex/Makefile.am: Fix CFLAGS/LIBS. * ext/cdio/gstcdiocddasrc.c: * ext/libpng/gstpngdec.c: (gst_pngdec_task): Include stdlib * ext/cairo/Makefile.am: * gst/videofilter/Makefile.am: * tests/examples/level/Makefile.am: Use $(LIBM) instead of -lm 2007-07-18 11:55:13 +0000 Stefan Kost sys/v4l2/gstv4l2src.c: Add another example pipeline. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: Add another example pipeline. 2007-07-18 11:42:33 +0000 Alexander Eichner sys/v4l2/gstv4l2src.c: Use define here. Original commit message from CVS: Patch by: Alexander Eichner * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): Use define here. * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_set_frequency_and_notify): Don't touch the property - its still disabled. * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits): * sys/v4l2/v4l2src_calls.h: Improve fallback format negotionation. Fixes #451388 2007-07-18 10:33:39 +0000 Stefan Kost tests/check/elements/videocrop.c: Fix the test. Original commit message from CVS: * tests/check/elements/videocrop.c: (GST_START_TEST): Fix the test. 2007-07-18 09:21:23 +0000 Stefan Kost More docs. More logs in pngdec. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-png.xml: * ext/jpeg/gstjpegdec.c: * ext/libpng/gstpngdec.c: (gst_pngdec_task), (gst_pngdec_sink_setcaps): More docs. More logs in pngdec. 2007-07-18 07:51:11 +0000 Stefan Kost gst/multifile/gstmultifilesrc.c: Add example to the docs. Fix buffer-offset-end and add some debug. Original commit message from CVS: * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create): Add example to the docs. Fix buffer-offset-end and add some debug. 2007-07-18 07:35:32 +0000 Stefan Kost Add stdlib include (free, atoi, exit). Original commit message from CVS: * examples/app/appsrc_ex.c: * examples/switch/switcher.c: * ext/neon/gstneonhttpsrc.c: * ext/timidity/gstwildmidi.c: * ext/x264/gstx264enc.c: * gst/mve/mveaudioenc.c: (mve_compress_audio): * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/spectrum/demo-audiotest.c: * gst/spectrum/demo-osssrc.c: * sys/dvb/gstdvbsrc.c: Add stdlib include (free, atoi, exit). 2007-07-17 11:35:29 +0000 Stefan Kost sys/v4l2/gstv4l2src.c: Initialize num_buffers with minimum value. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): Initialize num_buffers with minimum value. * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame): Handle frame-size query failure gracefully. 2007-07-16 12:11:36 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Fix parsing of esds atoms inside mp4a atoms so that we can set correct codec_info for AAC audi... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_node): Fix parsing of esds atoms inside mp4a atoms so that we can set correct codec_info for AAC audio. Fixes #457097 along with a whole other bunch of qt/aac files. 2007-07-16 09:16:03 +0000 Sebastian Dröge ext/wavpack/gstwavpackdec.c: Fix buffer clipping to correctly clip to the segment stop. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_clip_outgoing_buffer): Fix buffer clipping to correctly clip to the segment stop. 2007-07-13 16:31:27 +0000 Jan Schmidt Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and... Original commit message from CVS: * configure.ac: * tests/Makefile.am: Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and we weren't actually _using_ the information for libcheck ourselves anyway. 2007-07-12 11:21:01 +0000 Stefan Kost configure.ac: Use pkg-config to locate check. Original commit message from CVS: * configure.ac: Use pkg-config to locate check. 2007-07-11 23:43:25 +0000 Tim-Philipp Müller gst/: Fix build against core CVS. Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_process): * gst/vmnc/vmncdec.c: (vmnc_make_buffer): Fix build against core CVS. 2007-07-11 22:31:06 +0000 Tim-Philipp Müller Fix build against core CVS. Original commit message from CVS: * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_chain): * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): * gst/debug/gstnavigationtest.c: (gst_navigationtest_transform): * gst/effectv/gstaging.c: (gst_agingtv_transform): * gst/effectv/gstdice.c: (gst_dicetv_transform): * gst/effectv/gstedge.c: (gst_edgetv_transform): * gst/effectv/gstquark.c: (gst_quarktv_transform): * gst/effectv/gstrev.c: (gst_revtv_transform): * gst/effectv/gstshagadelic.c: (gst_shagadelictv_transform): * gst/effectv/gstvertigo.c: (gst_vertigotv_transform): * gst/effectv/gstwarp.c: (gst_warptv_transform): * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_wvpk_header), (gst_matroska_demux_check_subtitle_buffer), (gst_matroska_decode_buffer): * gst/videofilter/gstvideoflip.c: (gst_video_flip_transform): Fix build against core CVS. 2007-07-10 10:16:38 +0000 Edward Hervey gst/id3demux/gstid3demux.c: Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We don't have enough gra... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_chain): Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We don't have enough granularity to convert that boolean into a GstFlowReturn. 2007-07-06 15:00:47 +0000 Michael Smith gst/law/: Fix capsnego bogosity in *law decoders. Original commit message from CVS: * gst/law/alaw-decode.c: (alawdec_sink_setcaps), (gst_alawdec_class_init), (gst_alawdec_init), (gst_alawdec_chain), (gst_alawdec_change_state): * gst/law/alaw-decode.h: * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps), (gst_mulawdec_class_init), (gst_mulawdec_init), (gst_mulawdec_chain), (gst_mulawdec_change_state): * gst/law/mulaw-decode.h: Fix capsnego bogosity in *law decoders. 2007-07-06 14:35:59 +0000 Michael Smith ext/jpeg/gstsmokeenc.*: Remove stupidity in get/set caps functions. Original commit message from CVS: * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init), (gst_smokeenc_setcaps), (gst_smokeenc_chain), (gst_smokeenc_change_state): * ext/jpeg/gstsmokeenc.h: Remove stupidity in get/set caps functions. Fix some refcounting problems. 2007-07-06 11:42:53 +0000 Jan Schmidt ext/libpng/gstpngdec.c: Remove endianness-flipping hack that seems to have been required only because of a bug in ffm... Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set): Remove endianness-flipping hack that seems to have been required only because of a bug in ffmpegcolorspace. Partially Fixes: #451908 2007-07-05 08:44:11 +0000 Stefan Kost docs/plugins/Makefile.am: Simplify --extra-dir as gtkdoc scans recursively. Original commit message from CVS: * docs/plugins/Makefile.am: Simplify --extra-dir as gtkdoc scans recursively. 2007-07-03 09:59:46 +0000 Tommi Myöhänen gst/rtp/gstrtpilbcpay.c: Set the encoding-name in the rtp caps to all uppercase, as required by the caps spec. Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps): Set the encoding-name in the rtp caps to all uppercase, as required by the caps spec. Some small cleanups in the error paths. Fixes #453037. 2007-07-03 08:01:18 +0000 Stefan Kost gst/multifile/: Add .h files to be able to add it to the docs. Original commit message from CVS: * gst/multifile/Makefile.am: * gst/multifile/gstmultifile.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: Add .h files to be able to add it to the docs. 2007-07-03 07:16:26 +0000 Stefan Kost gst/replaygain/gstrgvolume.h: Fix GObject macros. Original commit message from CVS: * gst/replaygain/gstrgvolume.h: Fix GObject macros. 2007-06-28 19:00:43 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.*: Use a GSList for the GArray that is used like a list anyway. Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_index_get_last_entry), (gst_wavpack_parse_index_get_entry_from_sample), (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset), (gst_wavpack_parse_scan_to_find_sample): * ext/wavpack/gstwavpackparse.h: Use a GSList for the GArray that is used like a list anyway. 2007-06-28 13:25:05 +0000 Tim-Philipp Müller ext/gdk_pixbuf/gstgdkpixbuf.c: Add state change function where we set 0/1 as default framerate in case our setcaps fu... Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state): Add state change function where we set 0/1 as default framerate in case our setcaps function isn't called, like it might not in a filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by gdkpixbufdec trying to create caps with a 0/0 framerate. Also post an error message on the bus if gst_pad_push() fails when called from our sink event handler (+1 for flow returns for event functions in 0.11) instead of failing silently. 2007-06-27 11:36:24 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Cast stack args to the proper types. Fixes #451249. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps): Cast stack args to the proper types. Fixes #451249. 2007-06-27 11:04:47 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: For container formats we only need to activate one of the streams so that we correctly signal ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (new_session_pad), (gst_rtspsrc_setup_streams): * gst/rtsp/gstrtspsrc.h: For container formats we only need to activate one of the streams so that we correctly signal no-more-pads. Fixes #451015. 2007-06-25 12:46:08 +0000 Stefan Kost docs/plugins/: Update docs with caps info. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Update docs with caps info. 2007-06-25 12:13:09 +0000 Tim-Philipp Müller po/POTFILES.in: Add more files with translatable strings (#450878). Original commit message from CVS: * po/POTFILES.in: Add more files with translatable strings (#450878). 2007-06-22 20:23:18 +0000 Jens Granseuer gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.). Original commit message from CVS: Patch by: Jens Granseuer * gst/equalizer/gstiirequalizer.c: * gst/equalizer/gstiirequalizer10bands.c: * gst/equalizer/gstiirequalizer3bands.c: * gst/equalizer/gstiirequalizernbands.c: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_push_sorted): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): * gst/switch/gstswitch.c: (gst_switch_chain): Build fixes for gcc-2.9x (no mid-block variable declarations etc.). Fixes #450185. 2007-06-22 14:26:36 +0000 Jan Schmidt MAINTAINERS: Updating all the maintainers files Original commit message from CVS: * MAINTAINERS: Updating all the maintainers files 2007-06-22 10:12:15 +0000 Edward Hervey Fix memory leaks. Original commit message from CVS: * ext/flac/gstflactag.c: (gst_flac_tag_init): * gst/interleave/deinterleave.c: (deinterleave_init), (deinterleave_sink_link): * gst/interleave/interleave.c: (interleave_init): * gst/median/gstmedian.c: (gst_median_init): * gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init): Fix memory leaks. * tests/check/elements/id3demux.c: (pad_added_cb): Remove unused variable. 2007-06-21 10:48:10 +0000 Damien Carbery ext/gconf/gconf.h: Make the prototype of gst_gconf_get_key_for_sink_profile match the implementation. Original commit message from CVS: * ext/gconf/gconf.h: Make the prototype of gst_gconf_get_key_for_sink_profile match the implementation. Patch by: Damien Carbery Fixes: #449747 2007-06-20 12:56:12 +0000 Michael Smith gst/rtp/gstrtpdepay.c: Fix description - rtpdepay is not a payloader. Original commit message from CVS: * gst/rtp/gstrtpdepay.c: Fix description - rtpdepay is not a payloader. 2007-06-20 10:15:00 +0000 Stefan Kost gst/equalizer/gstiirequalizer.c: Document parameter mapping. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: Document parameter mapping. 2007-06-20 08:56:17 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Fix leaking buffers. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_event), (gst_spectrum_transform_ip): Fix leaking buffers. * tests/check/Makefile.am: * tests/check/elements/spectrum.c: (setup_spectrum), (cleanup_spectrum), (GST_START_TEST), (spectrum_suite), (main): Add simple test for spectrum element. 2007-06-20 08:26:21 +0000 Stefan Kost gst/qtdemux/: Add MJPG to the variants of motion jpeg. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_samples), (qtdemux_video_caps): * gst/qtdemux/qtdemux_fourcc.h: Add MJPG to the variants of motion jpeg. 2007-06-19 16:40:40 +0000 Tim-Philipp Müller tests/check/: Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the error flags are included and it errors... Original commit message from CVS: * tests/check/Makefile.am: * tests/check/elements/audiopanorama.c: (GST_START_TEST): * tests/check/elements/videocrop.c: (GST_START_TEST): * tests/check/elements/videofilter.c: * tests/check/elements/wavpackdec.c: (GST_START_TEST): * tests/check/elements/wavpackparse.c: (GST_START_TEST): Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the error flags are included and it errors out on compiler warnings for CVS builds; remove unused variables in various unit tests. 2007-06-19 14:48:03 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Use threadsafe inet_ntop to convert an ip number to a string. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_close), (rtsp_connection_free): Use threadsafe inet_ntop to convert an ip number to a string. Fixes #447961. Don't leak fd (and ip) when freeing a connection without first closing it. 2007-06-19 14:11:49 +0000 Christian Schaller * gst/qtdemux/LEGAL: add 'LEGAL' file describing why this is in -good and under what circumstances it might need to move. Original commit message from CVS: add 'LEGAL' file describing why this is in -good and under what circumstances it might need to move. 2007-06-19 10:41:49 +0000 Jan Schmidt configure.ac: Back to CVS Original commit message from CVS: * configure.ac: Back to CVS * gst-plugins-good.doap: Add 0.10.6 to the doap file. === release 0.10.6 === 2007-06-19 10:24:55 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-quicktime.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * win32/common/config.h: Release 0.10.6 Original commit message from CVS: Release 0.10.6 2007-06-18 17:53:20 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/ja.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2007-06-17 12:35:03 +0000 Tim-Philipp Müller gst/rtsp/rtspconnection.c: Revert previous commit again, since we are frozen (sorry). Original commit message from CVS: * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_free): Revert previous commit again, since we are frozen (sorry). 2007-06-17 12:24:58 +0000 Peter Kjellerstedt gst/rtsp/rtspconnection.c: inet_ntoa() uses a static buffer internally, so we need to copy the returned string if we ... Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_free): inet_ntoa() uses a static buffer internally, so we need to copy the returned string if we want to store it for later (#447961). 2007-06-15 09:13:55 +0000 Jan Schmidt win32/vs6/: Mark *.dsp & *.dsw as binary files and convert to DOS line endings, as they don't load into VS6 correctly... Original commit message from CVS: * win32/vs6/autogen.dsp: * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstalaw.dsp: * win32/vs6/libgstalpha.dsp: * win32/vs6/libgstalphacolor.dsp: * win32/vs6/libgstapetag.dsp: * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstauparse.dsp: * win32/vs6/libgstautodetect.dsp: * win32/vs6/libgstavi.dsp: * win32/vs6/libgstcutter.dsp: * win32/vs6/libgstdirectdraw.dsp: * win32/vs6/libgstdirectsound.dsp: * win32/vs6/libgsteffectv.dsp: * win32/vs6/libgstflx.dsp: * win32/vs6/libgstgoom.dsp: * win32/vs6/libgsticydemux.dsp: * win32/vs6/libgstid3demux.dsp: * win32/vs6/libgstinterleave.dsp: * win32/vs6/libgstjpeg.dsp: * win32/vs6/libgstlevel.dsp: * win32/vs6/libgstmatroska.dsp: * win32/vs6/libgstmedian.dsp: * win32/vs6/libgstmonoscope.dsp: * win32/vs6/libgstmulaw.dsp: * win32/vs6/libgstmultipart.dsp: * win32/vs6/libgstqtdemux.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstsmpte.dsp: * win32/vs6/libgstspeex.dsp: * win32/vs6/libgstudp.dsp: * win32/vs6/libgstvideobalance.dsp: * win32/vs6/libgstvideobox.dsp: * win32/vs6/libgstvideocrop.dsp: * win32/vs6/libgstvideoflip.dsp: * win32/vs6/libgstvideomixer.dsp: * win32/vs6/libgstwaveform.dsp: * win32/vs6/libgstwavenc.dsp: * win32/vs6/libgstwavparse.dsp: Mark *.dsp & *.dsw as binary files and convert to DOS line endings, as they don't load into VS6 correctly otherwise. 2007-06-15 08:32:52 +0000 Vincent Torri gst/rtsp/rtspconnection.c: Fix the MingW build. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect): Fix the MingW build. Patch By: Vincent Torri Fixes: #446981 2007-06-14 14:03:41 +0000 Jan Schmidt tests/: Hush the buildbots up Original commit message from CVS: * tests/check/elements/.cvsignore: * tests/icles/.cvsignore: Hush the buildbots up 2007-06-14 12:14:24 +0000 Jan Schmidt Make sure to dist everything needed for win32 builds. Original commit message from CVS: * configure.ac: * sys/Makefile.am: * sys/directdraw/Makefile.am: * sys/directsound/Makefile.am: * sys/waveform/Makefile.am: Make sure to dist everything needed for win32 builds. 2007-06-14 10:23:20 +0000 Edward Hervey gst/qtdemux/qtdemux.c: For AMR-NB streams, export the AMRSpecificBox as codec_data on the caps. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): For AMR-NB streams, export the AMRSpecificBox as codec_data on the caps. Fixes #447458 2007-06-13 17:11:24 +0000 Wim Taymans gst/rtp/gstrtph264depay.c: Make sure we allocate enough memory for the codec_data. Original commit message from CVS: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps): Make sure we allocate enough memory for the codec_data. Fixes #447210. 2007-06-12 21:05:22 +0000 Sébastien Moutte win32/MANIFEST: Add videocrop project file to the win32 manifest. Original commit message from CVS: * win32/MANIFEST: Add videocrop project file to the win32 manifest. * win32/vs6/gst_plugins_good.dsw: Add qtdemux,videocrop and waveform projects to the workspace. * win32/vs6/libgstqtdemux.dsp: Add zlib to the link list of qtdemux. * win32/vs6/libgstvideocrop.dsp: Add a project file for videocrop. 2007-06-12 20:22:26 +0000 Jan Schmidt po/POTFILES.in: Add qtdemux for translation Original commit message from CVS: * po/POTFILES.in: Add qtdemux for translation 2007-06-12 20:15:29 +0000 Jan Schmidt Move videocrop and osxvideo from -bad. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-videocrop.xml: * gst-plugins-good.spec.in: * sys/Makefile.am: * tests/check/Makefile.am: * tests/icles/Makefile.am: * tests/icles/videocrop-test.c: Move videocrop and osxvideo from -bad. 2007-06-12 19:35:08 +0000 Jan Schmidt Move qtdemux from -bad. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-quicktime.xml: * win32/MANIFEST: Move qtdemux from -bad. * gst-plugins-good.spec.in: Update spec file to reflect moving of qtdemux and wavpack 2007-06-12 19:01:41 +0000 Jan Schmidt * ChangeLog: * win32/MANIFEST: Fix typo in the changelog and commit the manifest too Original commit message from CVS: Fix typo in the changelog and commit the manifest too 2007-06-12 18:52:33 +0000 Jan Schmidt win32/MANIFEST Original commit message from CVS: * win32/MANIFEST * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-directdraw.xml: * docs/plugins/inspect/plugin-directsound.xml: * docs/plugins/inspect/plugin-waveform.xml: Move the waveform plugin from -bad too. Update the inspect xml files to mention Plugins Good instead of Plugins Bad. 2007-06-12 13:33:56 +0000 Andy Wingo * ChangeLog: * sys/v4l2/v4l2src_calls.c: Return a copy of the pool buffer if all mmap buffers have been dequeued. Original commit message from CVS: (gst_v4l2src_grab_frame): Return a copy of the pool buffer if all mmap buffers have been dequeued. 2007-06-12 11:23:01 +0000 Andy Wingo sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize) (gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type) Original commit message from CVS: 2007-06-12 Andy Wingo * sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize) (gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type) (gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with finalization and resuscitation. No longer public. (gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init) (gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type) (gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate) (gst_v4l2_buffer_pool_destroy): Make the pool follow common miniobject semantics, and be threadsafe. (gst_v4l2src_queue_frame): Remove this function, as we just call the ioctls directly in the two places where we queue buffers. (gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer directly. (gst_v4l2src_capture_init): Use the new buffer_pool_new function to allocate the pool, which also preallocates the GstBuffers. (gst_v4l2src_capture_start): Call buffer_pool_activate instead of queueing the frames directly. * sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a real MiniObject instead of rolling our own refcounting and finalizing. Give it a lock. (struct _GstV4l2Buffer): Remove one intermediary object, having the buffers hold the struct v4l2_buffer directly. * sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to capture_init so that it can set them on the buffers that it will create. (gst_v4l2src_get_read): For better or for worse, include the timestamping and offsetting code here; really we should be using bufferalloc though. (gst_v4l2src_get_mmap): Just make grab_frame return one of our preallocated, mmap'd buffers. 2007-06-11 11:41:56 +0000 daniel fischer sys/ximage/gstximagesrc.c: Actually use the display_name property so that we can dump any available X display. Fixes ... Original commit message from CVS: Patch by: daniel fischer * sys/ximage/gstximagesrc.c: (gst_ximage_src_start), (gst_ximage_src_get_caps): Actually use the display_name property so that we can dump any available X display. Fixes #445905. 2007-06-11 10:21:13 +0000 Tommi Myöhänen gst/rtp/: Add missing rate fields to caps. Fixes #441118. Original commit message from CVS: Patch by: Tommi Myöhänen * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps): Add missing rate fields to caps. Fixes #441118. 2007-06-10 21:14:11 +0000 Sébastien Moutte win32/: Add DirectSound and DirectDraw sinks project files to workspace and solution files. Original commit message from CVS: * win32/vs6/gst_plugins_good.dsw: * win32/vs8/gst-plugins-good.sln: Add DirectSound and DirectDraw sinks project files to workspace and solution files. 2007-06-10 10:53:26 +0000 Josh Coalson Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887. Original commit message from CVS: Patch by: Josh Coalson , updated by Alexis Ballier : * configure.ac: * ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders), (gst_flac_dec_setup_seekable_decoder), (gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek), (gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof), (gst_flac_dec_read_seekable), (gst_flac_dec_read_stream): * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.c: (gst_flac_enc_init), (gst_flac_enc_finalize), (gst_flac_enc_set_metadata), (gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality), (gst_flac_enc_seek_callback), (gst_flac_enc_write_callback), (gst_flac_enc_tell_callback), (gst_flac_enc_sink_event), (gst_flac_enc_chain), (gst_flac_enc_set_property), (gst_flac_enc_get_property), (gst_flac_enc_change_state): * ext/flac/gstflacenc.h: Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887. 2007-06-09 15:41:52 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.c: Remove workaround for bug #421543. This is fixed in core 0.10.13 and not necessary anymo... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps): Remove workaround for bug #421543. This is fixed in core 0.10.13 and not necessary anymore as we need at least that core version. 2007-06-09 15:33:32 +0000 Sebastian Dröge ext/wavpack/: Improve discont handling by checking if the next Wavpack block has the expected, following block index. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset), (gst_wavpack_parse_push_buffer): * ext/wavpack/gstwavpackparse.h: Improve discont handling by checking if the next Wavpack block has the expected, following block index. 2007-06-08 20:23:07 +0000 Thomas Vander Stichele * tests/check/elements/.gitignore: moap ignore Original commit message from CVS: moap ignore 2007-06-08 20:20:56 +0000 Thomas Vander Stichele gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details): Fix element description. Original commit message from CVS: * gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details): Fix element description. 2007-06-08 20:19:55 +0000 Thomas Vander Stichele move wavpack plugin. See #352605. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-wavpack.xml: * ext/Makefile.am: * tests/check/Makefile.am: move wavpack plugin. See #352605. 2007-06-08 19:45:43 +0000 Thomas Vander Stichele * docs/plugins/Makefile.am: the alphabet tripping up people since 10929BC Original commit message from CVS: the alphabet tripping up people since 10929BC 2007-06-08 17:37:02 +0000 Jan Schmidt Add DirectDraw & DirectSound plugins to the build and docs. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * sys/Makefile.am: * win32/MANIFEST: Add DirectDraw & DirectSound plugins to the build and docs. 2007-06-08 16:31:15 +0000 Jan Schmidt Rename the keep-aspect-ratio property to force-aspect-ratio to make it consistent with xvimagesink and ximagesink. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.args: * sys/directdraw/gstdirectdrawsink.c: (gst_directdraw_sink_class_init): Rename the keep-aspect-ratio property to force-aspect-ratio to make it consistent with xvimagesink and ximagesink. 2007-06-08 10:43:26 +0000 Tim-Philipp Müller ext/: When operating in pull mode, error out correct on not-linked. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_loop): * ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task): When operating in pull mode, error out correct on not-linked. 2007-06-08 08:12:43 +0000 Tim-Philipp Müller tests/icles/videocrop-test.c: Default to xvimagesink instead of autovideosink while autovideosink/ghostpads/whatever ... Original commit message from CVS: * tests/icles/videocrop-test.c: (main): Default to xvimagesink instead of autovideosink while autovideosink/ghostpads/whatever don't handle the way we use it in the way we expect it to. 2007-06-06 10:19:17 +0000 Andy Wingo * ChangeLog: * sys/v4l2/v4l2src_calls.c: sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format) Original commit message from CVS: 2007-06-06 Andy Wingo * sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format) (gst_v4l2src_probe_caps_for_format_and_size): Only probe for format and size if the ioctls are defined; should fix compilation on Linux < 2.16.19. 2007-06-06 08:53:12 +0000 Tim-Philipp Müller gst/videobox/gstvideobox.c: Printf fixes in debug statements; use LOG level for debug statements that are printed for... Original commit message from CVS: * gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420): Printf fixes in debug statements; use LOG level for debug statements that are printed for each and every frame; convert c++ comments to C-style comments; not much point using g_try_malloc() if we then not even check the return value. 2007-06-05 16:32:19 +0000 Tim-Philipp Müller configure.ac: Bump requirements to released versions (core and base 0.10.13). Original commit message from CVS: * configure.ac: Bump requirements to released versions (core and base 0.10.13). * gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify): Use gst_tag_utf8_from_freeform_string() from libgsttag instead of own implementation. 2007-06-05 14:17:25 +0000 Andy Wingo sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add some useless comments. Original commit message from CVS: 2007-06-05 Andy Wingo * sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add some useless comments. * sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue frames before calling STREAMON, that might leave them in a state where they can't be dequeued if we go back to NULL without calling STREAMON, according to the docs. (gst_v4l2src_capture_start): Enqueue buffers here instead, right before we call STREAMON. (gst_v4l2src_capture_deinit): Remove crack to work around dequeue failures. (For me this code hung.) The pool refcounting is still crack; added a note to that effect. 2007-06-05 09:11:41 +0000 Wim Taymans gst/multipart/multipartmux.c: Add support for mapping gst structure names to the MIME type equivalent. Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init), (gst_multipart_mux_get_mime), (gst_multipart_mux_collected): Add support for mapping gst structure names to the MIME type equivalent. Implemented for audio/x-mulaw->audio/basic. Fixes #442874. 2007-06-03 11:21:44 +0000 Sebastian Dröge gst/wavenc/gstwavenc.*: Properly write wav files with width!=depth by having the depth most significant bytes set and... Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf), (gst_wavenc_sink_setcaps), (gst_wavenc_format_samples), (gst_wavenc_chain), (gst_wavenc_change_state): * gst/wavenc/gstwavenc.h: Properly write wav files with width!=depth by having the depth most significant bytes set and all others zero. Fixes #442535. 2007-06-01 13:52:17 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Add include to make buildbot happy. Original commit message from CVS: * gst/rtsp/rtspconnection.c: Add include to make buildbot happy. 2007-06-01 13:07:11 +0000 Peter Kjellerstedt gst/rtsp/: Improves version checking, allowing an RTSP server to reply with "505 Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (add_date_header), (rtsp_connection_send), (parse_response_status), (parse_request_line), (parse_line), (rtsp_connection_receive): * gst/rtsp/rtspdefs.c: (rtsp_version_as_text): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init_request), (rtsp_message_init_response), (rtsp_message_remove_header), (rtsp_message_append_headers), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Improves version checking, allowing an RTSP server to reply with "505 RTSP Version not supported. Adds a Date header to all messages. Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we want to be able to send a response even if something in the request was invalid. EINVAL is only used when passing wrong arguments to functions. Do not handle an invalid method in parse_request_line(). Defer this to the caller so it can respond with "405 Method Not Allowed". Improves parsing of the timeout parameter to the Session header, allowing whitespace after the semicolon. Avoids a compiler warning due to variables shadowing a function argument. 2007-06-01 11:16:17 +0000 Daniel Charles gst/rtp/: Add support for AMR-WB. Original commit message from CVS: Based on Patch by: Daniel Charles * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init), (gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init), (gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer): * gst/rtp/gstrtpamrpay.h: Add support for AMR-WB. Small cleanups such as using BOILERPLATE. 2007-05-31 15:57:07 +0000 Wim Taymans gst/rtsp/rtspextwms.c: Fix compile warning when debug is disabled as spotted bu Saur on IRC. Original commit message from CVS: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream): Fix compile warning when debug is disabled as spotted bu Saur on IRC. 2007-05-30 14:57:44 +0000 Andy Wingo sys/v4l2/gstv4l2object.*: Revert some unintended changes. Original commit message from CVS: 2007-05-30 Andy Wingo * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some unintended changes. 2007-05-30 14:40:53 +0000 Andy Wingo sys/v4l2/v4l2src_calls.*: Store the format list in the order that the driver gives it to us. Original commit message from CVS: 2007-05-30 Andy Wingo * sys/v4l2/v4l2src_calls.h: * sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store the format list in the order that the driver gives it to us. (gst_v4l2src_probe_caps_for_format_and_size) (gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps based on the capabilities of the device. (gst_v4l2src_grab_frame): Update for object variable renaming. (gst_v4l2src_set_capture): Update to be strict in its parameters, as in the set_caps below. (gst_v4l2src_capture_init): Update for object variable renaming, and reflow. (gst_v4l2src_capture_start, gst_v4l2src_capture_stop) (gst_v4l2src_capture_deinit): Update for object variable renaming. (gst_v4l2src_update_fps, gst_v4l2src_set_fps) (gst_v4l2src_get_fps): Remove; these functions don't have much meaning outside of an atomic set_caps method. (gst_v4l2src_buffer_new): Don't set buffer duration, it is not known. * sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove call to update_fps; not sure about this change. (gst_v4l2_tuner_set_norm): Work around the fact that for the moment we don't have an update_fps_func. * sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2 structures in the object, just store what we need. Do store the probed caps of the device. Don't store the current frame rate. * sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the update_fps_function, for now. Update for new object variable naming. (gst_v4l2src_set_property, gst_v4l2src_get_property): Update for new object variable naming. (gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps. (gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_.... (gst_v4l2src_get_caps): Rework to probe the device for supported frame sizes and frame rates. (gst_v4l2src_set_caps): Rework to be strict in the given parameters: if someone asks us to have a certain size and rate, that is what we configure. (gst_v4l2src_get_read): Update for object variable naming. Don't leak buffers on short reads. (gst_v4l2src_get_mmap): Update for object variable naming, and add comments. (gst_v4l2src_create): Update for object variable naming. 2007-05-30 14:38:59 +0000 Tim-Philipp Müller gst/avi/gstavidemux.*: Parse subtitle text streams instead of erroring out (#442034). Still needs a parser for the su... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_base_init), (gst_avi_demux_reset), (gst_avi_demux_parse_stream): * gst/avi/gstavidemux.h: Parse subtitle text streams instead of erroring out (#442034). Still needs a parser for the subtitles to actually show up. 2007-05-30 12:46:32 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Make _push_event() return TRUE if the event could be pushed on at least one pad and not only i... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_push_event), (gst_avi_demux_loop): Make _push_event() return TRUE if the event could be pushed on at least one pad and not only if it could be pushed on all pads, otherwise we'll end up posting an error message on EOS if one or more source pads are not connected. 2007-05-28 16:39:09 +0000 Wim Taymans gst/rtsp/rtsptransport.c: Use renamed RTP bin. Original commit message from CVS: * gst/rtsp/rtsptransport.c: Use renamed RTP bin. 2007-05-28 15:01:33 +0000 Dejan Sakelšak gst/videobox/gstvideobox.c: Add AYUV->AYUV and AYUV->I420 formats. Original commit message from CVS: Based on patch by: Dejan Sakelšak * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_set_property), (gst_video_box_transform_caps), (video_box_recalc_transform), (gst_video_box_set_caps), (gst_video_box_get_unit_size), (gst_video_box_apply_alpha), (gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor), (UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv), (gst_video_box_i420_i420), (gst_video_box_transform), (plugin_init): Add AYUV->AYUV and AYUV->I420 formats. Fix negotiation and I420->AYUV conversion. Fixes #429329. 2007-05-26 15:25:18 +0000 Wim Taymans ext/speex/gstspeexdec.c: Use different variables for nested for loops so that the outer loop functions properly and s... Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data): Use different variables for nested for loops so that the outer loop functions properly and speex files with multiple frames per buffer work properly. Fixes #441408. 2007-05-25 20:51:36 +0000 Tim-Philipp Müller gst/id3demux/gstid3demux.c: Don't leak newsegment events. Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event): Don't leak newsegment events. 2007-05-25 20:33:10 +0000 Tim-Philipp Müller gst/wavparse/Makefile.am: Add '-lm' to LIBS for ceil(), don't assume one of our dependencies drags it in. Original commit message from CVS: * gst/wavparse/Makefile.am: Add '-lm' to LIBS for ceil(), don't assume one of our dependencies drags it in. 2007-05-25 16:02:51 +0000 Tim-Philipp Müller ext/flac/gstflacenc.*: Collect headers, add "streamheader" field to output caps and set Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_init), (notgst_value_array_append_buffer), (gst_flac_enc_process_stream_headers), (gst_flac_enc_write_callback), (gst_flac_enc_chain), (gst_flac_enc_change_state): * ext/flac/gstflacenc.h: Collect headers, add "streamheader" field to output caps and set BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux produces output according to the official FLAC-to-Ogg mapping instead of completely broken files. Fixes #426044. 2007-05-25 10:44:12 +0000 Jan Schmidt gst/: Handle and adjust new-segment events so that downstream really sees a stream with the tag pieces stripped off t... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_reset), (gst_id3demux_send_new_segment), (gst_id3demux_chain), (gst_id3demux_sink_event): * gst/id3demux/gstid3demux.h: * gst/apetag/gsttagdemux.c: (gst_tag_demux_reset), (gst_tag_demux_chain), (gst_tag_demux_sink_event), (gst_tag_demux_send_new_segment): Handle and adjust new-segment events so that downstream really sees a stream with the tag pieces stripped off the front and back. Fixes strangeness in seeking when mp3 decoders use the new-segment byte position to estimate their current playback position timestamp and then the arriving buffers don't match up. 2007-05-25 10:23:49 +0000 Jan Schmidt gst/autodetect/gstautoaudiosink.c: Don't unnecessarily perform a READY->NULL->READY transition on the detected audio ... Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect): Don't unnecessarily perform a READY->NULL->READY transition on the detected audio sink when starting up. Fixes: #440127 2007-05-24 17:00:21 +0000 Tim-Philipp Müller ext/flac/gstflacenc.c: Don't crash in chain function if setcaps hasn't been called. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps), (gst_flac_enc_chain): Don't crash in chain function if setcaps hasn't been called. 2007-05-24 08:35:23 +0000 Vincent Torri sys/directdraw/gstdirectdrawsink.*: Fix more warnings when compiling with MingW (#439914). Original commit message from CVS: Patch by: Vincent Torri * sys/directdraw/gstdirectdrawsink.c: (gst_directdraw_sink_buffer_alloc), (gst_directdraw_sink_show_frame), (gst_directdraw_sink_check_primary_surface), (gst_directdraw_sink_check_offscreen_surface), (EnumModesCallback2), (gst_directdraw_sink_get_ddrawcaps), (gst_directdraw_sink_surface_create): * sys/directdraw/gstdirectdrawsink.h: Fix more warnings when compiling with MingW (#439914). 2007-05-24 08:14:00 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Init value to avoid infinte loops. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods): Init value to avoid infinte loops. 2007-05-24 08:10:42 +0000 Peter Kjellerstedt gst/rtsp/: Fix for new API. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth), (gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_play): (rtsp_connection_send), (rtsp_connection_receive): * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send): Fix for new API. * gst/rtsp/rtspconnection.c: (add_auth_header), Only add authorisation and session headers when sending messages. * gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init), (rtsp_message_init_request), (rtsp_message_init_response), (rtsp_message_unset), (rtsp_message_add_header), (rtsp_message_remove_header), (rtsp_message_get_header), (rtsp_message_append_headers), (dump_key_value), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Add support for multiple headers of the same type by storing the parsed headers in a GArray instaed of a hashtable. 2007-05-23 22:44:12 +0000 Sébastien Moutte docs/plugins/gst-plugins-bad-plugins.args: Remove directsoundsink property doc as this sink use the mixer interface now. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.args: Remove directsoundsink property doc as this sink use the mixer interface now. * docs/plugins/gst-plugins-bad-plugins.interfaces: Add interfaces implemented by Windows sinks. * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: Remove directsoundsink property and implement the mixer interface. * win32/vs6/gst_plugins_bad.dsw: * win32/vs6/libgstdirectsound.dsp: Update project files. * gst-libs/gst/dshow/gstdshow.cpp: * gst-libs/gst/dshow/gstdshow.h: * gst-libs/gst/dshow/gstdshowfakesink.cpp: * gst-libs/gst/dshow/gstdshowfakesink.h: * gst-libs/gst/dshow/gstdshowfakesrc.cpp: * gst-libs/gst/dshow/gstdshowfakesrc.h: * gst-libs/gst/dshow/gstdshowinterface.cpp: * gst-libs/gst/dshow/gstdshowinterface.h: * win32/common/libgstdshow.def: * win32/vs6/libgstdshow.dsp: Add a new gst library which allow to create internal Direct Show graph (pipelines) to wrap Windows sources, decoders or encoders. It includes a DirectShow fake source and sink and utility functions. * sys/dshowsrcwrapper/gstdshowaudiosrc.c: * sys/dshowsrcwrapper/gstdshowaudiosrc.h: * sys/dshowsrcwrapper/gstdshowsrcwrapper.c: * sys/dshowsrcwrapper/gstdshowsrcwrapper.h: * sys/dshowsrcwrapper/gstdshowvideosrc.c: * sys/dshowsrcwrapper/gstdshowvideosrc.h: * win32/vs6/libdshowsrcwrapper.dsp: Add a new plugin to wrap DirectShow sources on Windows. It gets data from any webcam, dv cam, micro. We could add tv tunner card later. 2007-05-22 11:14:13 +0000 Stefan Kost configure.ac: Depend on gstreamer-0.10.12.1. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _Gs... Original commit message from CVS: * configure.ac: Depend on gstreamer-0.10.12.1. * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBand, object, _GstIirEqualizerBandClass, parent_class, gst_iir_equalizer_band_set_property, gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type, gst_iir_equalizer_child_proxy_get_child_by_index, gst_iir_equalizer_child_proxy_get_children_count, gst_iir_equalizer_child_proxy_interface_init, setup_filter, gst_iir_equalizer_compute_frequencies, gst_iir_equalizer_set_property, gst_iir_equalizer_get_property, plugin_init): * gst/equalizer/gstiirequalizer.h (audiofilter): * gst/equalizer/gstiirequalizernbands.c (ARG_NUM_BANDS, gst_iir_equalizer_nbands_base_init, gst_iir_equalizer_nbands_init, gst_iir_equalizer_nbands_set_property): Use new locking macros. * gst/filter/gstbpwsinc.c (bpwsinc_set_caps): Add fixme. * gst/spectrum/gstspectrum.c (SPECTRUM_WINDOW_BASE, SPECTRUM_WINDOW_LEN, gst_spectrum_init, gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip): Use new locking macros. Turn two fixed values into #defines. 2007-05-22 11:03:30 +0000 Edward Hervey docs/plugins/Makefile.am: Also look for .m (objectivec) files. Original commit message from CVS: * docs/plugins/Makefile.am: Also look for .m (objectivec) files. * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * sys/osxvideo/osxvideosink.m: Add documentation for element and properties. 2007-05-21 14:01:16 +0000 Stefan Kost ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa... Original commit message from CVS: * ChangeLog: ChangeLog surgery. * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBand, object, _GstIirEqualizerBandClass, parent_class, gst_iir_equalizer_band_set_property, gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type, gst_iir_equalizer_child_proxy_get_child_by_index, gst_iir_equalizer_child_proxy_get_children_count, gst_iir_equalizer_child_proxy_interface_init, setup_filter, gst_iir_equalizer_compute_frequencies, plugin_init): * tests/icles/equalizer-test.c: Add fixme and comment for example. 2007-05-21 12:43:37 +0000 Stefan Kost * gst/spectrum/gstspectrum.c: gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip): Original commit message from CVS: * gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip): Use lock to protect from concurrent access. 2007-05-21 11:37:16 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.c: Specify and use properties as unsigned int that are an unsigned int. Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init), (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property): Specify and use properties as unsigned int that are an unsigned int. 2007-05-21 11:17:21 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.*: Fixup docs, make the bitrate property an int as it should be and allow to set the differ... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init), (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property): * ext/wavpack/gstwavpackenc.h: Fixup docs, make the bitrate property an int as it should be and allow to set the different extra processing modes instead of only allowing none and the default one. 2007-05-21 10:07:05 +0000 Wim Taymans gst/udp/gstudpsrc.c: Since we depend on 0.10.13 -core, override the unlock_stop vmethod for safer shutdown. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop): Since we depend on 0.10.13 -core, override the unlock_stop vmethod for safer shutdown. 2007-05-21 10:03:42 +0000 Wim Taymans gst/rtsp/gstrtpdec.*: Added signal for backwards compat. Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init): * gst/rtsp/gstrtpdec.h: Added signal for backwards compat. 2007-05-21 09:32:26 +0000 René Stadler Use audioconvert for converting from non-native endianness floats in auparse instead of doing it ourself. Fixes #424527. Original commit message from CVS: Patch by: René Stadler * configure.ac: * gst/auparse/gstauparse.c: (gst_au_parse_reset), (gst_au_parse_parse_header), (gst_au_parse_chain): * gst/auparse/gstauparse.h: Use audioconvert for converting from non-native endianness floats in auparse instead of doing it ourself. Fixes #424527. This needs the audioconvert from plugins-base CVS. 2007-05-21 09:29:30 +0000 Wim Taymans gst/rtp/gstrtph263ppay.c: Fix enum registration. Original commit message from CVS: * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type), (gst_rtp_h263p_pay_flush): Fix enum registration. 2007-05-21 08:57:18 +0000 Antoine Tremblay gst/rtp/gstrtph263ppay.*: Add new fragmentation mode base on GOB headers. Fixes #438940. Original commit message from CVS: Patch by: Antoine Tremblay * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type), (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init), (gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property), (gst_rtp_h263p_pay_flush): * gst/rtp/gstrtph263ppay.h: Add new fragmentation mode base on GOB headers. Fixes #438940. 2007-05-20 21:31:58 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example pipelines of wavpackenc. As the wavpack stuff n... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example pipelines of wavpackenc. As the wavpack stuff now needs input with 32 bit width (and random depth) this is needed now. The example pipelines for the parser and decoder are still fine. 2007-05-20 14:59:46 +0000 Tim-Philipp Müller sys/directdraw/gstdirectdrawsink.c: Bunch of small fixes: remove static function that doesn't exist; declare another ... Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_finalize), (gst_directdraw_sink_buffer_alloc), (gst_directdraw_sink_get_ddrawcaps), (gst_directdraw_sink_surface_create): Bunch of small fixes: remove static function that doesn't exist; declare another one that does; printf format fix; use right macro when specifying debug category; remove a bunch of unused variables; #if 0 out an unused chunk of code (partially fixes #439914). 2007-05-20 14:14:49 +0000 Tim-Philipp Müller gst/: Printf format fixes (#439910, #439911). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample): * gst/switch/gstswitch.c: (gst_switch_chain): Printf format fixes (#439910, #439911). 2007-05-20 14:05:42 +0000 Tim-Philipp Müller gst/rtsp/gstrtspsrc.c: Printf format fix. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp): Printf format fix. 2007-05-19 10:01:45 +0000 René Stadler Add replaygain playback elements (#412710). Original commit message from CVS: Patch by: René Stadler * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-replaygain.xml: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init), (gst_rg_analysis_start), (gst_rg_analysis_set_caps), (gst_rg_analysis_transform_ip), (gst_rg_analysis_event), (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags), (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result), (gst_rg_analysis_album_result): * gst/replaygain/gstrganalysis.h: * gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init), (gst_rg_limiter_class_init), (gst_rg_limiter_init), (gst_rg_limiter_set_property), (gst_rg_limiter_get_property), (gst_rg_limiter_transform_ip): * gst/replaygain/gstrglimiter.h: * gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init), (gst_rg_volume_class_init), (gst_rg_volume_init), (gst_rg_volume_set_property), (gst_rg_volume_get_property), (gst_rg_volume_dispose), (gst_rg_volume_change_state), (gst_rg_volume_sink_event), (gst_rg_volume_tag_event), (gst_rg_volume_reset), (gst_rg_volume_update_gain), (gst_rg_volume_determine_gain): * gst/replaygain/gstrgvolume.h: * gst/replaygain/replaygain.c: (plugin_init): * gst/replaygain/replaygain.h: * gst/replaygain/rganalysis.h: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/rganalysis.c: (send_eos_event), (GST_START_TEST): * tests/check/elements/rglimiter.c: (setup_rglimiter), (cleanup_rglimiter), (set_playing_state), (create_test_buffer), (verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main): * tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume), (cleanup_rgvolume), (set_playing_state), (set_null_state), (send_eos_event), (send_tag_event), (test_buffer_new), (fail_unless_target_gain), (fail_unless_result_gain), (fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main): Add replaygain playback elements (#412710). 2007-05-18 13:27:39 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Don't crash when an unsupported transport error was returned by the server, just try to config... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Don't crash when an unsupported transport error was returned by the server, just try to configure the next stream. Fixes #439255. 2007-05-18 11:39:12 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP connection. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: Add TCP timeout property and use it for all TCP connection. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_write), (rtsp_connection_next_timeout), (rtsp_connection_reset_timeout): Make connect and writes cancelable and make them use the timeout. 2007-05-18 10:36:12 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Refactor timeout handling. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_setup_streams): Refactor timeout handling. Also send keep-alive when dealing with TCP transport. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_free), (rtsp_connection_next_timeout), (rtsp_connection_reset_timeout): * gst/rtsp/rtspconnection.h: Use a timer to handle the session timeouts, add some methods to deal with timeouts. 2007-05-17 14:56:39 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Ignore streams that fail the setup command, we will retry with a different transport later on. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_setup_streams): Ignore streams that fail the setup command, we will retry with a different transport later on. * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp), (rtsp_ext_wms_configure_stream): Fix encoding name case. 2007-05-17 10:59:00 +0000 Edward Hervey sys/osxvideo/osxvideosink.*: Remove the event-loop-in-separate-thread modifications, because MacOSX is $#@(*%$# ! For... Original commit message from CVS: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Remove the event-loop-in-separate-thread modifications, because MacOSX is $#@(*%$# ! For those wondering, the event handling needs to be done in the main thread after all.. 2007-05-17 09:41:48 +0000 Edward Hervey sys/osxvideo/osxvideosink.*: Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now. Original commit message from CVS: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now. Use a separate thread/task for the cocoa event_loop, else it wouldn't stop. 2007-05-16 16:50:23 +0000 Edward Hervey ext/libpng/gstpngdec.c: Fix build on macosx. Original commit message from CVS: * ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data): Fix build on macosx. 2007-05-16 16:30:03 +0000 Sebastian Dröge ext/raw1394/gstdv1394src.c: Replace direct comparison of a string with the string literal "" with a comparison of the... Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri): Replace direct comparison of a string with the string literal "" with a comparison of the first character with '\0'. Fixes #438926. 2007-05-15 17:22:58 +0000 Tim-Philipp Müller Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save and restore the various flags in the directdraw/dir... Original commit message from CVS: * configure.ac: * sys/directdraw/Makefile.am: * sys/directsound/Makefile.am: Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save and restore the various flags in the directdraw/directsound detection section. Apparently improves cross-compiling for win32 with mingw32 under some circumstances (#437539). 2007-05-15 11:18:33 +0000 Stefan Kost gst/debug/breakmydata.c (gst_break_my_data_init): One more try. This should be the proper fix now. Original commit message from CVS: * gst/debug/breakmydata.c (gst_break_my_data_init): One more try. This should be the proper fix now. 2007-05-15 06:41:58 +0000 Stefan Kost gst/debug/breakmydata.c: Ooops, no // comments please. Original commit message from CVS: * gst/debug/breakmydata.c: Ooops, no // comments please. 2007-05-15 06:34:48 +0000 Stefan Kost gst/debug/breakmydata.c: Fix gst_buffer_is_writable() assertion. Original commit message from CVS: * gst/debug/breakmydata.c: (gst_break_my_data_class_init), (gst_break_my_data_init): Fix gst_buffer_is_writable() assertion. 2007-05-15 02:56:23 +0000 David Schleef sys/v4l2/gstv4l2src.c: Add support for Bayer images as video/x-raw-bayer. Fixes #314160. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: Add support for Bayer images as video/x-raw-bayer. Fixes #314160. 2007-05-14 17:10:12 +0000 Wim Taymans gst/rtp/: Update theora pay/depayloader in a similar to vorbis. Original commit message from CVS: * gst/rtp/gstrtptheoradepay.c: (decode_base64), (gst_rtp_theora_depay_parse_configuration): * gst/rtp/gstrtptheorapay.c: (encode_base64), (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_handle_buffer): Update theora pay/depayloader in a similar to vorbis. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_parse_configuration): Update docs. 2007-05-14 16:19:58 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: When we try to execute a method that is not supported by the server, don't error out but remov... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send): When we try to execute a method that is not supported by the server, don't error out but remove the method from the accepted methods so that we never try to perform this method again. 2007-05-14 14:47:26 +0000 Wim Taymans gst/rtp/gstrtpvorbisdepay.c: Remove annoying _dump_mem. Original commit message from CVS: * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process): Remove annoying _dump_mem. 2007-05-14 11:11:42 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Parse range correctly. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range): Parse range correctly. * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri): The baseurl now always has a '/' at the start. 2007-05-14 09:01:05 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Factor out caps configuration and configure more stuff such as the time ranges and speed/scale... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play): Factor out caps configuration and configure more stuff such as the time ranges and speed/scale values. * gst/rtsp/rtsptransport.c: Add Copyright after non-trival fixes. 2007-05-13 19:57:45 +0000 David Schleef gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403. Original commit message from CVS: * gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403. By Ali Sabil . 2007-05-13 15:47:13 +0000 Sébastien Moutte gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 can build in_data += (filter->width / 8). Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_transform_ip): Use guint8 * instead of gpointer then vs6 can build in_data += (filter->width / 8). 2007-05-12 16:37:50 +0000 Peter Kjellerstedt gst/rtsp/: Make channel guint8 where possible. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_receive): * gst/rtsp/rtspmessage.c: (rtsp_message_init_data), (rtsp_message_get_header): * gst/rtsp/rtspmessage.h: Make channel guint8 where possible. Make rtsp_message_init_data() take the channel as a guint8. * gst/rtsp/rtspdefs.c: Fixed a typo: Timout -> Timeout * gst/rtsp/rtspdefs.h: Make RTSP_CHECK() behave as a statement. * gst/rtsp/sdpmessage.c: Avoid a compiler warning in INIT_ARRAY(). Fixes #437692. 2007-05-12 16:27:51 +0000 Peter Kjellerstedt gst/rtsp/rtspurl.*: Add support for query parameters to RTSP URLs. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free), (rtsp_url_get_request_uri): * gst/rtsp/rtspurl.h: Add support for query parameters to RTSP URLs. 2007-05-12 16:26:06 +0000 Peter Kjellerstedt gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse(). Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode), (parse_range), (range_as_text), (rtsp_transport_mode_as_text), (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text), (rtsp_transport_parse), (rtsp_transport_as_text): * gst/rtsp/rtsptransport.h: Add validation to rtsp_transport_parse(). Add rtsp_transport_as_text() to generate an RTSP header from an RTSPTransport. Change ssrc to guint (was a string) since that is what it is, even though it is sent as a hex string. Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is incorrect, which can be seen when looking at the examples in the RFC). Fixes #437670. 2007-05-11 16:11:04 +0000 Eric Anholt * ChangeLog: * sys/ximage/gstximagesrc.c: sys/ximage/gstximagesrc.c (gst_ximage_src_open_display, gst_ximage_src_ximage_get): Original commit message from CVS: Patch by: Eric Anholt * sys/ximage/gstximagesrc.c (gst_ximage_src_open_display, gst_ximage_src_ximage_get): Use union of all damage between frames to make it faster. Fixes bug #342463. Also fix crasher when cursor is at bottom right of window. 2007-05-11 16:01:45 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: Skip LIST chunks before the fmt chunk (fixes #437499). Also fix streaming mode regression... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Skip LIST chunks before the fmt chunk (fixes #437499). Also fix streaming mode regression for file from #343837 with 'bext' chunk before the 'fmt' chunk. 2007-05-11 15:09:39 +0000 Wim Taymans gst/rtsp/: Preliminary seek support. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspdefs.h: Preliminary seek support. Activate internal pads so that we can receive events on them. Don't try to parse a range string when it's NULL. 2007-05-11 15:04:38 +0000 Wim Taymans gst/rtp/README: Update README with new RTP variables that will be used for synchronisation. Original commit message from CVS: * gst/rtp/README: Update README with new RTP variables that will be used for synchronisation. * gst/rtp/gstrtpvorbisdepay.c: (decode_base64), (gst_rtp_vorbis_depay_parse_configuration), (gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbispay.c: (encode_base64), (gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_handle_buffer): Update vorbis pay and depayloader to draft-04. 2007-05-11 11:24:13 +0000 Wim Taymans gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP. Original commit message from CVS: * gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP. 2007-05-11 10:31:27 +0000 Zaheer Abbas Merali sys/ximage/gstximagesrc.c (gst_ximage_src_start, gst_ximage_src_ximage_get): Original commit message from CVS: * sys/ximage/gstximagesrc.c (gst_ximage_src_start, gst_ximage_src_ximage_get): * sys/ximage/gstximagesrc.h (last_ximage): When using Damage actually keep the last frame, and not assume that the buffer we get already has the last frame on it. Copy the cursor over if we specify a non-zero start x and start y. 2007-05-11 09:12:55 +0000 Wim Taymans gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified. Original commit message from CVS: * gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified. 2007-05-10 14:02:07 +0000 Stefan Kost gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde... Original commit message from CVS: * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows, gst_qtdemux_loop_state_movie, gst_qtdemux_loop, qtdemux_parse_segments, qtdemux_parse_trak): * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth, rtp_session_get_rtcp_bandwidth, rtp_session_get_cname, rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone, rtp_session_get_location, rtp_session_get_tool, rtp_session_process_bye, session_report_blocks): * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp, rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb): More format arg fixing (spotted by Ali Sabil ). * gst/switch/Makefile.am: Add require libraries(spotted by Ali Sabil ). 2007-05-10 01:21:19 +0000 David Schleef gst/level/gstlevel.c: Revert last change. Original commit message from CVS: * gst/level/gstlevel.c: Revert last change. 2007-05-09 21:30:53 +0000 Sébastien Moutte gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 know the size of data pointed when moving the pointer. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_calculate_##TYPE), (gst_level_transform_ip): Use guint8 * instead of gpointer then vs6 know the size of data pointed when moving the pointer. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer): Move instructions after variables declaration. * win32/vs6/autogen.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: Update vs6 project files. 2007-05-09 11:23:39 +0000 Wim Taymans gst/rtsp/: Add code to parse time ranges. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open): * gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range), (parse_clock_range), (parse_smpte_range), (rtsp_range_parse), (rtsp_range_free): * gst/rtsp/rtsprange.h: Add code to parse time ranges. Report DURATION on the stream when possible. 2007-05-08 15:49:01 +0000 Tim-Philipp Müller gst/videomixer/videomixer.c: Fix strides calculation for AYUV (it's just width*4) (#436910). Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv), (gst_videomixer_fill_checker), (gst_videomixer_fill_color), (gst_videomixer_collected): Fix strides calculation for AYUV (it's just width*4) (#436910). 2007-05-06 21:32:40 +0000 Sebastian Dröge gst/audiofx/: Sync the GObject properties before each processing step to properly work with the controller. Original commit message from CVS: * gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip): * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip): * gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip): Sync the GObject properties before each processing step to properly work with the controller. 2007-05-04 15:17:14 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can catch more error cases. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state): Let more error state trickle down so that we can catch more error cases. Handle keep-alive a little smarter by selecting a method the server actually supports. Fix a race in UDP streaming shutdown. 2007-05-04 13:04:31 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Ignore errors when trying to use the keep-alive messages. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive): Ignore errors when trying to use the keep-alive messages. 2007-05-04 12:31:32 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_mcast), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_stream_configure_transport): Send RTCP messages back to the server over the TCP connection. * gst/rtsp/rtspconnection.c: (rtsp_connection_write), (rtsp_connection_send), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Factor out and expose lowlevel _write and _read methods. Implement sending data messages to the server. 2007-05-03 15:55:06 +0000 Wim Taymans gst/multipart/multipartmux.c: Fix timestamps on outgoing buffers. Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected): Fix timestamps on outgoing buffers. 2007-05-03 14:39:09 +0000 Wim Taymans gst/multipart/multipartmux.c: Emit NEWSEGMENT events before pushing the first buffer. Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected), (gst_multipart_mux_change_state): Emit NEWSEGMENT events before pushing the first buffer. 2007-05-03 13:48:54 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Refactor transport configuration code. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize), (gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_mcast), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event), (gst_rtspsrc_loop_udp), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play), (gst_rtspsrc_pause): Refactor transport configuration code. Create internal pads for TCP transport so that we can implement events and queries. Handle events and queries. Parse range from the SDP. Fix race in pause handler where the connection could still be flushing. 2007-05-02 19:32:58 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (new_session_pad), (request_pt_map), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Fix race when multiple udp sources post timeouts, just act on the first received timeout. Protect stream list with a recursive lock to fix some races. Flush connection when we need to do a reconnect or stop. Make state lock recursive. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_close): Some small cleanups. 2007-05-02 18:31:16 +0000 Sebastian Dröge ext/wavpack/gstwavpack.c: Call bindtextdomain() to get localized strings. Original commit message from CVS: * ext/wavpack/gstwavpack.c: (plugin_init): Call bindtextdomain() to get localized strings. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain): * ext/wavpack/gstwavpackparse.h: Handle DISCONT buffers by correctly setting the DISCONT flag on outgoing buffers when necessary. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event) Send newsegment from the streaming thread. 2007-05-02 18:25:09 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Only set DISCONT when there actually is a discont or when we just started. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Only set DISCONT when there actually is a discont or when we just started. 2007-05-02 18:01:52 +0000 Sebastian Dröge ext/flac/gstflac.c: Call bindtextdomain() to get localized strings. Original commit message from CVS: * ext/flac/gstflac.c: (plugin_init): Call bindtextdomain() to get localized strings. 2007-05-02 17:19:36 +0000 Wim Taymans gst/wavparse/gstwavparse.*: Be a bit more clever when dealing with VBR files with FACT tags, we don't want to timesta... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data): * gst/wavparse/gstwavparse.h: Be a bit more clever when dealing with VBR files with FACT tags, we don't want to timestamp buffers in that case but the estimated BPS can be used for seeking. Only send close segment in the streaming thread. 2007-05-02 17:08:09 +0000 Sebastian Dröge ext/flac/gstflacdec.c: Correctly post an error on the bus if something went wrong in the loop function. This fixes a ... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_loop): Correctly post an error on the bus if something went wrong in the loop function. This fixes a few cases where the task was paused and nothing happened anymore. 2007-05-02 16:58:06 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Remove old workaround that was needed when seeking after the last sample. With the fix... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event): Remove old workaround that was needed when seeking after the last sample. With the fixed error handling this works now as expected without pushing the last sample although it wasn't requested. 2007-05-02 16:45:43 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Handle segment seeks in the seek event handler, correctly work with stop position == -... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event): Handle segment seeks in the seek event handler, correctly work with stop position == -1 and instead of stopping the task on seek just pause it. 2007-05-02 16:19:58 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Add handling for segment seeks. Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_loop): Add handling for segment seeks. 2007-05-02 15:13:04 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Correctly handle errors, especially in the loop function. Before it was easy to get th... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer), (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop), (gst_wavpack_parse_chain): Correctly handle errors, especially in the loop function. Before it was easy to get the task paused but no error being posted on the bus. 2007-05-02 14:27:28 +0000 Wim Taymans gst/rtsp/test.c: Fix compilation of deprecated test just because I'm too lazy to delete it. Original commit message from CVS: * gst/rtsp/test.c: (main): Fix compilation of deprecated test just because I'm too lazy to delete it. 2007-05-02 13:32:57 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_open), (gst_rtspsrc_handle_message): * gst/rtsp/gstrtspsrc.h: Fix sending RTCP to the right place. Fix bug in reffing the wrong UDP element. Use new pad names for the session manager. Implement handling server requests in interleaved and UDP modes. Handle session keep-alive in UDP modes. Remove GCond for handling UDP timeouts. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_send), (rtsp_connection_read), (read_body), (rtsp_connection_receive), (rtsp_connection_close): * gst/rtsp/rtspconnection.h: Store connection IP address for later. Add timeout args to all operations that might block forever. Parse session timeout. Only close sockets when not already closed. * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Add timeout return value and error string. * gst/rtsp/rtspmessage.c: (rtsp_message_init_response): Add small comment. 2007-05-01 16:13:58 +0000 Sjoerd Simons gst/rtp/gstrtpmp4vpay.*: Handle NEWSEGMENT and FLUSH events. Fixes #434824. Original commit message from CVS: Patch by: Sjoerd Simons * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init), (gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event): * gst/rtp/gstrtpmp4vpay.h: Handle NEWSEGMENT and FLUSH events. Fixes #434824. 2007-04-30 11:15:58 +0000 Tim-Philipp Müller docs/plugins/gst-plugins-good-plugins-docs.sgml: Remove v4l2src from docs, since it breaks the docs build, and the pl... Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: Remove v4l2src from docs, since it breaks the docs build, and the plugin is only built if --enable-experimental is used anyway. * docs/plugins/Makefile.am: Spaces => tab. 2007-04-29 14:43:37 +0000 Wim Taymans gst/udp/gstmultiudpsink.c: Add code to drop membership of a multicast group. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (leave_multicast), (gst_multiudpsink_add), (gst_multiudpsink_remove): Add code to drop membership of a multicast group. * gst/udp/gstudpsink.c: (gst_udpsink_update_uri), (gst_udpsink_set_uri): Implement URI handler. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_parse_rtpinfo): Use URI handler to make udpsink instace. Improve code to configure port and destination. 2007-04-29 13:56:18 +0000 Thomas Vander Stichele * sys/directdraw/gstdirectdrawsink.c: * sys/osxvideo/osxvideosink.m: 80 char police Original commit message from CVS: 80 char police 2007-04-29 13:53:16 +0000 Thomas Vander Stichele autogen.sh: Require automake 1.7 Original commit message from CVS: * autogen.sh: Require automake 1.7 * ext/alsaspdif/Makefile.am: * ext/divx/Makefile.am: * ext/ivorbis/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/neon/Makefile.am: * ext/sdl/Makefile.am: * ext/swfdec/Makefile.am: * ext/theora/Makefile.am: * ext/wavpack/Makefile.am: * ext/xvid/Makefile.am: * gst/modplug/Makefile.am: Fix up Makefile.am accordingly. 2007-04-29 13:49:02 +0000 Thomas Vander Stichele docs/plugins/inspect/: Add jack and update. Original commit message from CVS: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-glimagesink.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: Add jack and update. 2007-04-29 12:19:21 +0000 Wim Taymans gst/udp/gstmultiudpsink.c: Fix multicast detection. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add): Fix multicast detection. Don't try to join a multicast group if the address is not multicast. * gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri): Small debug improvement. 2007-04-27 16:44:17 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play), (gst_rtspsrc_handle_message): Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent. 2007-04-27 15:30:39 +0000 Wim Taymans gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg. Original commit message from CVS: * gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg. 2007-04-26 15:08:20 +0000 Edward Hervey docs/plugins/: Add documentation for osxaudio plugin. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-osxaudio.xml: Add documentation for osxaudio plugin. 2007-04-26 14:31:32 +0000 Edward Hervey docs/plugins/: Add documentation for osxvideo Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/inspect/plugin-osxvideo.xml: Add documentation for osxvideo 2007-04-26 10:08:27 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Protect state changes with a lock. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: Protect state changes with a lock. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (parse_line): * gst/rtsp/rtspconnection.h: Remove some unused stuff. 2007-04-26 08:48:30 +0000 Wim Taymans gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. F... Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. Fixes #433530. 2007-04-26 08:39:49 +0000 Wim Taymans gst/wavparse/gstwavparse.*: Apply DISCONT to buffers. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): * gst/wavparse/gstwavparse.h: Apply DISCONT to buffers. Only apply timestamp to the first sample after a DISCONT, too many VBR files cause random jitter in the timestamps. Fixes #433119. 2007-04-25 15:55:32 +0000 Wim Taymans gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin. Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init), (gst_rtp_dec_init), (gst_rtp_dec_set_property), (gst_rtp_dec_get_property): * gst/rtsp/gstrtpdec.h: Add dummy latency property to be backwards compat with rtpbin. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_parse_rtpinfo): * gst/rtsp/gstrtspsrc.h: Add latency property and configure in the session manager. Don't set invalid clock-base and seqnum-base on caps, some servers sometimes don't send them. 2007-04-25 15:31:53 +0000 Tim-Philipp Müller gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always ... Original commit message from CVS: * gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init), (gst_alpha_color_transform_caps), (gst_alpha_color_set_caps): Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always catch it if those caps aren't a subset of our template caps, also see #421543). Fixes #429319 in a way. Also, don't leak the pad template in the transform_caps function. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/alphacolor.c: (setup_alphacolor), (cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32), (create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4), (GST_START_TEST), (alphacolor_suite): Add some basic unit tests for alphacolor. 2007-04-25 15:08:22 +0000 Tim-Philipp Müller ext/libpng/gstpngdec.c: If we get a fatal flow return in the loop function, first post the error message and only the... Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_task): If we get a fatal flow return in the loop function, first post the error message and only then send the EOS event downstream, otherwise applications might get an eos message before the error message and think everything was ok (related to #429319). 2007-04-25 10:07:12 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Read the channel byte as an unsigned byte. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (rtsp_connection_receive): Read the channel byte as an unsigned byte. 2007-04-25 09:47:48 +0000 Wim Taymans gst/rtp/: Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init), (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init), (gst_rtp_gsm_depay_setcaps): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps): * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps): * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init), (gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps), (gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property), (gst_ilbc_depay_get_property): * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps): * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init), (gst_rtp_pcma_depay_setcaps): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init), (gst_rtp_pcmu_depay_setcaps): Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282. 2007-04-25 08:36:46 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Parse server address from SDP. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize), (gst_rtspsrc_stream_free), (request_pt_map), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: Parse server address from SDP. Hook up a udpsink to send RTCP back to the server. * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtsp/rtsptransport.h: Add some docs. 2007-04-25 06:52:09 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Make header field check conditional. Fixes #433135 Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Make header field check conditional. Fixes #433135 2007-04-24 09:12:42 +0000 Tim-Philipp Müller Add minimal docs blurb to alphacolor; split out headers into separate header file for gtk-doc. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-alphacolor.xml: * gst/alpha/Makefile.am: * gst/alpha/gstalphacolor.c: * gst/alpha/gstalphacolor.h: Add minimal docs blurb to alphacolor; split out headers into separate header file for gtk-doc. 2007-04-20 17:25:50 +0000 Tim-Philipp Müller gst/debug/progressreport.c: Don't try to post NULL message (in case we can't query upstream position or duration). Original commit message from CVS: * gst/debug/progressreport.c: (gst_progress_report_report): Don't try to post NULL message (in case we can't query upstream position or duration). 2007-04-18 12:36:37 +0000 Michael Smith gst/cutter/gstcutter.*: Fix some of the most obvious bugs in cutter. Now doesn't leak everything if input is silent. Original commit message from CVS: * gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain), (gst_cutter_get_caps): * gst/cutter/gstcutter.h: Fix some of the most obvious bugs in cutter. Now doesn't leak everything if input is silent. 2007-04-18 09:48:25 +0000 Sebastian Dröge gst/wavenc/gstwavenc.*: everything else results in a invalid block align and invalid files. Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf), (gst_wavenc_sink_setcaps), (gst_wavenc_change_state): * gst/wavenc/gstwavenc.h: Wav apparently only supports width==GST_ROUND_UP(depth), everything else results in a invalid block align and invalid files. 2007-04-17 16:39:02 +0000 Snaik gst/smpte/barboxwipes.c: Add missing break statement for BOX_HORIZONTAL case. Original commit message from CVS: Patch by: Snaik * gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw): Add missing break statement for BOX_HORIZONTAL case. 2007-04-17 10:14:43 +0000 Vincent Torri gst/wavparse/gstwavparse.c: Use correct format strings for integer types. Original commit message from CVS: Patch by: Vincent Torri * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Use correct format strings for integer types. 2007-04-17 02:51:02 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Use gst_riff_create_audio_template_caps () instead of the local caps. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_create_sourcepad): Use gst_riff_create_audio_template_caps () instead of the local caps. This makes updates of the local caps unecessary whenever libgstriff gets support for new formats. 2007-04-16 21:29:40 +0000 Brian Cameron sys/sunaudio/: Fix and/or update copyright attributions (#430228). Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudio.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixertrack.h: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: * sys/sunaudio/gstsunaudiosrc.c: * sys/sunaudio/gstsunaudiosrc.h: Fix and/or update copyright attributions (#430228). 2007-04-14 17:18:14 +0000 Sébastien Moutte docs/plugins/inspect/: Add xml doc files for Windows sinks Original commit message from CVS: * docs/plugins/inspect/plugin-directdraw.xml: * docs/plugins/inspect/plugin-directsound.xml: * docs/plugins/inspect/plugin-waveform.xml: Add xml doc files for Windows sinks * win32/vs6/libgstqtdemux.dsp: * win32/vs6/libgstmpegvideoparse.dsp: * win32/vs6/gst_plugins_bad.dsw: Update projects files. 2007-04-13 09:32:21 +0000 Wim Taymans docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs. * gst/rtsp/URLS: Add some more example urls. * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT), (gst_rtp_dec_chain_rtp): Better debugging. * gst/rtsp/gstrtspsrc.c: (request_pt_map), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo): Remove unused code. 2007-04-13 08:19:35 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Relax the audio/mpeg caps again and add FIXME: comment. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Relax the audio/mpeg caps again and add FIXME: comment. 2007-04-13 06:20:28 +0000 Stefan Kost gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix type for 'rate' header field. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): * gst/wavparse/gstwavparse.h: More sanity check for the header fields. Fix type for 'rate' header field. 2007-04-12 16:06:31 +0000 Tim-Philipp Müller gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are not UTF-8, try to interpret them accordi... Original commit message from CVS: * gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8), (gst_icydemux_unicodify): If the metadata strings we get in the stream are not UTF-8, try to interpret them according to the character encodings specified in the GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and only fall back to locale/ISO-8859-1 if those aren't set or don't work. Should fix #428901. 2007-04-12 14:20:56 +0000 Wim Taymans gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS. Original commit message from CVS: * gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS. 2007-04-12 11:41:11 +0000 Thomas Vander Stichele gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_... Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24): * gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__): Add a simple hashing implementation that we can use to generate a 24-bit ident value based on the codebooks for vorbis and theora. * gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers, gst_rtp_theora_pay_handle_buffer): * gst/rtp/gstrtpvorbisdepay.c (gst_rtp_vorbis_depay_parse_configuration, gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet, gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet, gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer): Use the hashing function, ensuring that the same codebooks result in the same ident and thus the same SDP description. Various log fixes/changes. 2007-04-12 11:37:50 +0000 jerry tan sys/sunaudio/gstsunaudiosrc.c: it is the application's responsibility to make sure it open the device once. Original commit message from CVS: Patch by: jerry tan * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open): remove the call of ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the application's responsibility to make sure it open the device once. Remove a careless error if AUDIODEV is set. Fixes #392620. 2007-04-12 10:52:02 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations. Original commit message from CVS: * gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations. 2007-04-12 08:21:28 +0000 Wim Taymans gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals. Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT), (gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp): * gst/rtsp/gstrtpdec.h: Make backward compat with rtpbin by adding the request-pt-map signals. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_stream_configure_caps), (gst_rtspsrc_activate_streams): * gst/rtsp/gstrtspsrc.h: Implement request-pt-map signals instead of setting caps on the buffers for the session manager. 2007-04-11 10:25:25 +0000 Wim Taymans gst/udp/gstudp.c: Register GstNetBuffer in plugin_init so that the type can be used from multiple threads without races. Original commit message from CVS: * gst/udp/gstudp.c: (plugin_init): Register GstNetBuffer in plugin_init so that the type can be used from multiple threads without races. 2007-04-11 10:19:06 +0000 Christian Schaller * gst-plugins-good.spec.in: update to spec file Original commit message from CVS: update to spec file 2007-04-11 09:53:38 +0000 Wim Taymans gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (qtdemux_parse_samples), (qtdemux_parse_segments), (qtdemux_parse_trak), (qtdemux_parse_tree): * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mdhd): Handle version 1 mdhd atoms to get extended precision durations. Fixes #426972. 2007-04-10 17:06:05 +0000 Wim Taymans gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): Fix depayloader clock_rate and some cleanups. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): * gst/rtp/gstrtph264depay.h: Don't push codec_data in the adapter because it might get flushed when we get a discont. * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): Handle multiple AU per packet. * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_plugin_init): Disable rank, this one does not work. Remove timestamping, base class does that. 2007-04-10 12:01:33 +0000 Stefan Kost gst/auparse/gstauparse.c: limit caps to the formats we announce in the template Original commit message from CVS: * gst/auparse/gstauparse.c: (gst_au_parse_parse_header): limit caps to the formats we announce in the template * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data): fix some crashers/asserts when dealing with broken files 2007-04-10 10:01:14 +0000 Peter Kjellerstedt gst/: Fix some compiler warnings. Fixes #428182. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index): * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send): Fix some compiler warnings. Fixes #428182. 2007-04-06 12:54:16 +0000 Wim Taymans gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session), (free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init), (gst_rtp_dec_init), (gst_rtp_dec_finalize), (gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp), (gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property), (gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock), (gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp), (create_rtcp), (gst_rtp_dec_request_new_pad), (gst_rtp_dec_release_pad): * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtsp.c: (plugin_init): Morph RTPDec into something compatible with RTPBin as a fallback. Various other style fixes. * gst/rtsp/gstrtspsrc.c: (find_stream_by_id), (find_stream_by_udpsrc), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps), (new_session_pad), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Implement RTPBin session manager handling. Don't try to add empty properties to caps. Implement fallback session manager, handling. Don't combine errors from RTCP streams, just ignore them. * gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager): * gst/rtsp/rtsptransport.h: Implement fallback session manager. Make RTPBin the default one when available. 2007-04-05 15:05:24 +0000 Wim Taymans gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better. Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_change_state): * gst/qtdemux/gstrtpxqtdepay.h: Try to recover from packet loss a little better. 2007-04-05 13:56:44 +0000 Wim Taymans gst/rtp/gstrtpmp4adepay.c: This element is ready to be autoplugged. Original commit message from CVS: * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init): This element is ready to be autoplugged. 2007-04-05 11:26:25 +0000 Julien Moutte gst/avi/gstavidemux.c: Don't leave the offsets defined by upstream element on the compressed data buffer we are pushi... Original commit message from CVS: 2007-04-05 Julien MOUTTE * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry): Don't leave the offsets defined by upstream element on the compressed data buffer we are pushing downstream. Make them GST_BUFFER_OFFSET_NONE. 2007-04-04 12:39:41 +0000 Stefan Kost gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only. Original commit message from CVS: * gst/avi/README: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data): Don't abort on out-of-memory. Use stream-nr as unsigned integer only. 2007-04-03 09:55:45 +0000 Wim Taymans gst/smpte/barboxwipes.c: Original commit message from CVS: * gst/smpte/barboxwipes.c: Fix error as spotted by Snaik 2007-03-30 17:19:34 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only works with plugins-base CVS, using an o... Original commit message from CVS: * gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only works with plugins-base CVS, using an older version doesn't have any disadvantages though. 2007-03-30 15:59:27 +0000 Sebastian Dröge Revert last change as we don't want plugins-good to depend on plugins-base CVS now. Original commit message from CVS: * configure.ac: * gst/auparse/gstauparse.c: (gst_au_parse_reset), (gst_au_parse_parse_header), (gst_au_parse_chain): * gst/auparse/gstauparse.h: Revert last change as we don't want plugins-good to depend on plugins-base CVS now. 2007-03-30 04:50:11 +0000 Sebastian Dröge ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_chain): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept this and let audioconvert convert to accepted formats instead of doing it in the element for n*8 depths. This also adds support for non-n*8 depths and prevents some useless memory allocations. Fixes #421598 Also add a workaround for bug #421542 in wavpackenc for now... * tests/check/elements/wavpackdec.c: (GST_START_TEST): * tests/check/elements/wavpackenc.c: (GST_START_TEST): * tests/check/elements/wavpackparse.c: (GST_START_TEST): Consider the change above in the unit tests and test if the correct caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in the wavpackparse unit test. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps): Set caps on the src pad as soon as possible. * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.h: Fix indention. gst-indent is now called by cicl. 2007-03-29 18:51:33 +0000 René Stadler configure.ac: Require gst-plugins-base CVS for audioconvert with non-native float support and width/depth fix in libg... Original commit message from CVS: * configure.ac: Require gst-plugins-base CVS for audioconvert with non-native float support and width/depth fix in libgstriff. Patch by: René Stadler * gst/auparse/gstauparse.c: (gst_au_parse_reset), (gst_au_parse_parse_header), (gst_au_parse_chain): * gst/auparse/gstauparse.h: Don't swap the floats ourself if they're not in native endianness. Instead let audioconvert handle this. Fixes #339838. 2007-03-29 14:40:35 +0000 Wim Taymans gst/rtp/: Flush adapter on disconts. Original commit message from CVS: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process), (gst_rtp_h263p_depay_change_state): * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process), (gst_rtp_h264_depay_change_state): * gst/rtp/gstrtph264depay.h: * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process): * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): Flush adapter on disconts. 2007-03-29 14:03:21 +0000 Wim Taymans gst/rtp/: Use more efficient adapter and rtpbuffer methods when possible. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process): * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process): * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process): * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush): * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process): * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process): Use more efficient adapter and rtpbuffer methods when possible. 2007-03-29 12:14:22 +0000 Sebastian Dröge gst/wavenc/gstwavenc.c: Correctly handle width!=depth input. Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf), (gst_wavenc_sink_setcaps): Correctly handle width!=depth input. * gst/wavparse/gstwavparse.c: Already export in the caps that width==8 uses unsigned samples and everything else uses signed samples. 2007-03-29 09:59:23 +0000 Laurent Glayal gst/udp/: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable. Original commit message from CVS: Patch by: Laurent Glayal * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init), (gst_dynudpsink_init), (gst_dynudpsink_set_property), (gst_dynudpsink_get_property), (gst_dynudpsink_init_send), (gst_dynudpsink_close): * gst/udp/gstdynudpsink.h: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop): * gst/udp/gstudpsrc.h: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable. Add a closefd property to instruct the udp elements to close the custom file descriptors when going to READY. Fixes #423304. API:GstUDPSrc::closefd property API:GstDynUDPSink::closefd property 2007-03-29 08:08:49 +0000 Laurent Glayal gst/rtp/: Added H264 payloader. Fixes #423782. Original commit message from CVS: Patch by: Laurent Glayal * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init), (gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init), (gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state), (gst_rtp_h264_pay_plugin_init): * gst/rtp/gstrtph264pay.h: Added H264 payloader. Fixes #423782. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): Small fixes. 2007-03-28 22:27:36 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32. Original commit message from CVS: * gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32. 2007-03-28 22:23:43 +0000 Sebastian Dröge gst/wavparse/gstwavparse.c: Add support for wav files containing audio/x-raw-int with random depths between 1 and 32 ... Original commit message from CVS: * gst/wavparse/gstwavparse.c: Add support for wav files containing audio/x-raw-int with random depths between 1 and 32 bits. 2007-03-28 18:40:12 +0000 Stefan Kost gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792. Original commit message from CVS: Based on patch by: Stefan Kost * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init), (gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init), (gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property), (gst_rtp_mp4a_depay_get_property), (gst_rtp_mp4a_depay_change_state), (gst_rtp_mp4a_depay_plugin_init): * gst/rtp/gstrtpmp4adepay.h: Added MP4A-LATM depayloader. Fixes #417792. * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process): Fixup depayloader, setting codec_data, using more efficient adaptor and rtpbuffer handling. * gst/rtsp/URLS: Add url to test above. 2007-03-28 15:17:27 +0000 Edward Hervey gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample), (gst_qtdemux_chain), (qtdemux_parse_samples): * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts): * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video). Use the offset present in 'ctts' to calculate the PTS for each packet and set the PTS on outgoing buffers. Fixes #423283 2007-03-25 15:34:42 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (find_stream_by_setup), (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_stream_configure_caps), (gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo): * gst/rtsp/gstrtspsrc.h: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field in the sdp can override the defaults. Parse RTP-Info field to get the seqnum and timebase fields that should go in the caps. Delay configuring caps after we got the RTP-Info from the PLAY reply from the server. 2007-03-24 19:46:59 +0000 Tim-Philipp Müller gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging input caps into 1-channel output caps (I... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps): Remove 'channel-positions' field when munging input caps into 1-channel output caps (I guess technically we should set the position for each channel on the output caps if it's non-NONE, but I'll save that as a task for another day). 2007-03-22 22:14:29 +0000 Tim-Philipp Müller gst/interleave/deinterleave.c: Don't leak input buffer in chain function; maintain our own list of source pads - ther... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads), (gst_deinterleave_remove_pads), (gst_deinterleave_process), (gst_deinterleave_chain): Don't leak input buffer in chain function; maintain our own list of source pads - there are no guarantees about the order of the list in the GstElement struct, and we want a very specific order; lastly, some more debugging. 2007-03-22 16:25:56 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite plugging loops with ranks is no clean solution... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init): Revert last commit, preventing infinite plugging loops with ranks is no clean solution and in general there's no reason why one wants to parse framed wavpack data again. 2007-03-22 15:52:51 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.c: Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wa... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block): Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init): Accept framed and non-framed input, wavpackparse doesn't care. To prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse ! ..." pipelines. 2007-03-22 11:08:03 +0000 Sebastian Dröge ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We can and should use it. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): Revert to use gst_pad_alloc_buffer() here. We can and should use it. Thanks to Jan and Mike for noticing my mistake. 2007-03-22 09:44:17 +0000 Christophe Dehais ext/gconf/gconf.c: Accept complex pipeline descriptions as an audio profile instead of just a single element. Fixes #... Original commit message from CVS: Patch by: Christophe Dehais * ext/gconf/gconf.c: (gst_gconf_render_bin_with_default): Accept complex pipeline descriptions as an audio profile instead of just a single element. Fixes #420658. 2007-03-22 00:17:41 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc struct directly and not as a pointer to sav... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init), (gst_wavpack_enc_init), (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block): * ext/wavpack/gstwavpackenc.h: Put the write helpers into the GstWavpackEnc struct directly and not as a pointer to save two small, but useless mallocs. This also makes it possible to drop the finalize method. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer): For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing buffers the same way wavpackenc does it. 2007-03-21 23:50:09 +0000 Sebastian Dröge ext/wavpack/gstwavpackdec.c: Don't use gst_pad_alloc_buffer() as we might clip the buffer later and Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): Don't use gst_pad_alloc_buffer() as we might clip the buffer later and BaseTransform-based elements will likely break because of wrong unit-size. Also plug a possible memleak that happens when decoding fails for some reason. 2007-03-21 12:53:57 +0000 Jan Schmidt ext/lame/gstlame.c: Disable the bitrate checking when the user has requested Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_setup): Disable the bitrate checking when the user has requested Free Format mode, as all bitrates less than the maximum are valid then. 2007-03-21 11:49:32 +0000 Tim-Philipp Müller gst/apetag/gsttagdemux.c: Rename registered type in preparation of GstTagDemux moving to Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type): Rename registered type in preparation of GstTagDemux moving to -base at some point in the future. 2007-03-19 10:29:19 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: Streaming mode fixes: don't unref buffer we don't own any longer; remove bogus adapter fl... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Streaming mode fixes: don't unref buffer we don't own any longer; remove bogus adapter flush. Fixes #419338. 2007-03-18 04:21:28 +0000 David Schleef REQUIREMENTS: Change the format to key/value, add a bunch of information, remove a bunch of requirements that are for... Original commit message from CVS: * REQUIREMENTS: Change the format to key/value, add a bunch of information, remove a bunch of requirements that are for other GStreamer packages. 2007-03-18 02:00:54 +0000 David Schleef REQUIREMENTS: Fix a few things. This file really needs a good once-over. Original commit message from CVS: * REQUIREMENTS: Fix a few things. This file really needs a good once-over. 2007-03-16 18:38:18 +0000 Edward Hervey sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView in the message. Original commit message from CVS: * sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView in the message. 2007-03-16 16:27:20 +0000 Edward Hervey sys/osxvideo/osxvideosink.m: Emit 'have-ns-view' message when working in embedded mode. The message will contain a po... Original commit message from CVS: * sys/osxvideo/osxvideosink.m: Emit 'have-ns-view' message when working in embedded mode. The message will contain a pointer to the newly created NSView. 2007-03-16 09:57:40 +0000 Stefan Kost gst/equalizer/gstiirequalizer10bands.c: A 10 band EQ should be initialized to 1 bands and not to 3. Original commit message from CVS: * gst/equalizer/gstiirequalizer10bands.c: (gst_iir_equalizer_10bands_init): A 10 band EQ should be initialized to 1 bands and not to 3. 2007-03-15 12:05:01 +0000 Edward Hervey sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory. Original commit message from CVS: * sys/Makefile.am: Don't forget to distribute the sys/osxaudio/ directory. 2007-03-15 11:39:53 +0000 Edward Hervey Activate osxaudio in gst-plugins-good with proper build setup. Original commit message from CVS: * configure.ac: * sys/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudiosink.c: (gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init), (gst_osx_audio_sink_getcaps), (gst_osx_audio_sink_create_ringbuffer), (plugin_init): * sys/osxaudio/gstosxaudiosrc.c: (gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init), (gst_osx_audio_src_create_ringbuffer): * sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type), (gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init), (gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start), (gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop): * sys/osxaudio/gstosxringbuffer.h: Activate osxaudio in gst-plugins-good with proper build setup. Add inlined documentation. Fix debug statements Fix ringbuffer when pausing. Fixes #323471 2007-03-14 22:21:26 +0000 Philippe Kalaf gst/rtp/: Ported mulaw and alaw payloaders to use new base class Original commit message from CVS: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: Ported mulaw and alaw payloaders to use new base class 2007-03-14 16:30:19 +0000 Edward Hervey sys/osxvideo/: Fix leaks when running a NSApp. Original commit message from CVS: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Fix leaks when running a NSApp. Accept any kind of resolutions. Works in fullscreen. Can maximize. Only thing left before being able to move this to -good is documentation and embedded window support. 2007-03-14 15:25:10 +0000 Thomas Vander Stichele po/: Update translations. Original commit message from CVS: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/it.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update translations. 2007-03-14 14:49:45 +0000 Tim-Philipp Müller configure.ac: Fix string replace error (AG_AG_GST_* => AG_GST_*). Original commit message from CVS: * configure.ac: Fix string replace error (AG_AG_GST_* => AG_GST_*). 2007-03-14 14:48:08 +0000 Stefan Kost gst/equalizer/: Add 3 and 10 band version and add missing gst_object_sync_values. Original commit message from CVS: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (_do_init), (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_get_type), (gst_iir_equalizer_child_proxy_get_child_by_index), (gst_iir_equalizer_child_proxy_get_children_count), (gst_iir_equalizer_child_proxy_interface_init), (setup_filter), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_transform_ip), (plugin_init): * gst/equalizer/gstiirequalizer10bands.c: (gst_iir_equalizer_10bands_base_init), (gst_iir_equalizer_10bands_class_init), (gst_iir_equalizer_10bands_init), (gst_iir_equalizer_10bands_set_property), (gst_iir_equalizer_10bands_get_property): * gst/equalizer/gstiirequalizer10bands.h: * gst/equalizer/gstiirequalizer3bands.c: (gst_iir_equalizer_3bands_base_init), (gst_iir_equalizer_3bands_class_init), (gst_iir_equalizer_3bands_init), (gst_iir_equalizer_3bands_set_property), (gst_iir_equalizer_3bands_get_property): * gst/equalizer/gstiirequalizer3bands.h: * gst/equalizer/gstiirequalizernbands.c: (gst_iir_equalizer_nbands_base_init), (gst_iir_equalizer_nbands_init): Add 3 and 10 band version and add missing gst_object_sync_values. * gst/spectrum/gstspectrum.c: (gst_spectrum_event), (gst_spectrum_transform_ip): Add some comments about float support. 2007-03-12 17:56:54 +0000 Tim-Philipp Müller gst/apetag/gsttagdemux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END her... Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event): Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END here as well. 2007-03-12 17:24:23 +0000 Jan Schmidt gst/id3demux/gstid3demux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END. Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event): Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END. 2007-03-12 15:49:02 +0000 Jan Schmidt * ChangeLog: I'm too lazy to comment this Original commit message from CVS: Add Patch by: line for wim, since he's away 2007-03-12 13:28:29 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is the image format a vari... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_picture_frame): Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is the image format a variable-length NUL-terminated string; in versions before that the image format is a fixed-length string of 3 characters (see #348644 for a sample tag). Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'. 2007-03-11 22:23:04 +0000 Sébastien Moutte sys/directdraw/gstdirectdrawsink.*: Handle display mode changes during playback. Original commit message from CVS: * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: Handle display mode changes during playback. 2007-03-10 16:07:31 +0000 Sébastien Moutte win32/MANIFEST: Add new project files to MANIFEST. Original commit message from CVS: * win32/MANIFEST: Add new project files to MANIFEST. * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: Update project files. 2007-03-10 12:30:48 +0000 Tim-Philipp Müller Printf format fixes; also add some missing quotes in translated strings. Fixes #416728 and #416727. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index): * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame): Printf format fixes; also add some missing quotes in translated strings. Fixes #416728 and #416727. 2007-03-09 20:12:08 +0000 Jan Schmidt gst/autodetect/gstautoaudiosink.c: Tim and I can't think of any reason the child audio sink needs to be set back to N... Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best): Tim and I can't think of any reason the child audio sink needs to be set back to NULL after successfully determining that it can reach READY - it gets immediately set back to READY by the caller anyway, causing an unnecessary close/open of any audio devices involved. 2007-03-09 19:51:27 +0000 Tim-Philipp Müller po/: Add ja.po file from #377306. Original commit message from CVS: * po/LINGUAS: * po/ja.po: Add ja.po file from #377306. 2007-03-09 19:44:30 +0000 Tim-Philipp Müller sys/sunaudio/: Actually translate sunaudio mixer track labels instead of just marking the strings as translatable (#3... Original commit message from CVS: * sys/sunaudio/gstsunaudio.c: (plugin_init): * sys/sunaudio/gstsunaudiomixertrack.c: (gst_sunaudiomixer_track_new): Actually translate sunaudio mixer track labels instead of just marking the strings as translatable (#377306); clean up weird label string mapping code that serves no apparent purpose. Also set the 'untranslated-label' property when creating mixer tracks if the GstMixerTrack base class supports this. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/sunaudio.c: (GST_START_TEST), (sunaudio_suite): Very minimalistic unit test for sunaudiomixer element (compiles, but not actually tested on a system where sunaudiomixer is available). 2007-03-09 18:49:37 +0000 Jan Schmidt tests/check/Makefile.am: Re-enable the states test and see if it works on the buildbots. Original commit message from CVS: * tests/check/Makefile.am: Re-enable the states test and see if it works on the buildbots. 2007-03-09 17:32:32 +0000 Wim Taymans ext/dv/gstdvdec.*: Infer pixel-aspect-ratio from the video frame format if it isn't provided by the container, as hap... Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps), (gst_dvdec_src_negotiate), (gst_dvdec_chain), (gst_dvdec_change_state): * ext/dv/gstdvdec.h: Infer pixel-aspect-ratio from the video frame format if it isn't provided by the container, as happens when playing DV from AVI or Quicktime containers. Patch by: Wim Taymans Fixes #380944 2007-03-09 17:05:17 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have dataflow and timeouts will later be handled... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams): When activated, remove the udpsrc timeout, we have dataflow and timeouts will later be handled by the jitterbuffer. 2007-03-09 16:53:39 +0000 Wim Taymans ext/taglib/gstid3v2mux.cc: Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag. Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag. Fixes #414496. 2007-03-09 15:04:45 +0000 Wim Taymans gst/avi/gstavidemux.c: Fix stream position reporting after a seek. Fixes #416445. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_push_event), (gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_chain): Fix stream position reporting after a seek. Fixes #416445. 2007-03-09 08:58:26 +0000 Stefan Kost gst/equalizer/: Refactor plugin into a base class and a first subclass (nband eq). The nband eq uses GstChildProxy an... Original commit message from CVS: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (_do_init), (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_band_get_property), (gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_get_type), (gst_iir_equalizer_child_proxy_get_child_by_index), (gst_iir_equalizer_child_proxy_get_children_count), (gst_iir_equalizer_child_proxy_interface_init), (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_finalize), (setup_filter), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_setup), (plugin_init): * gst/equalizer/gstiirequalizer.h: * gst/equalizer/gstiirequalizernbands.c: (gst_iir_equalizer_nbands_base_init), (gst_iir_equalizer_nbands_class_init), (gst_iir_equalizer_nbands_init), (gst_iir_equalizer_nbands_set_property), (gst_iir_equalizer_nbands_get_property): * gst/equalizer/gstiirequalizernbands.h: Refactor plugin into a base class and a first subclass (nband eq). The nband eq uses GstChildProxy and is controlable. More subclasses will follow. 2007-03-08 16:01:42 +0000 René Stadler gst/avi/gstavidemux.c: Make avidemux accept optional header chunks in any order. Original commit message from CVS: Patch by: René Stadler * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_push_event), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_chain): Make avidemux accept optional header chunks in any order. Fixes #415446. 2007-03-08 12:23:57 +0000 Jan Schmidt tests/check/Makefile.am: Disable the states check until the remaining Valgrind errors are fixed or suppressed. Original commit message from CVS: * tests/check/Makefile.am: Disable the states check until the remaining Valgrind errors are fixed or suppressed. 2007-03-08 10:24:43 +0000 Sebastian Dröge tests/check/elements/.cvsignore: Add audiodynamic check to .cvsignore Original commit message from CVS: * tests/check/elements/.cvsignore: Add audiodynamic check to .cvsignore 2007-03-08 10:02:12 +0000 Sebastian Dröge gst/audiofx/: Add new audiodynamic element which can act as a compressor or expander. Supported are hard-knee and sof... Original commit message from CVS: reviewed by: Stefan Kost * gst/audiofx/Makefile.am: * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_characteristics_get_type), (gst_audio_dynamic_mode_get_type), (gst_audio_dynamic_set_process_function), (gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init), (gst_audio_dynamic_init), (gst_audio_dynamic_set_property), (gst_audio_dynamic_get_property), (gst_audio_dynamic_setup), (gst_audio_dynamic_transform_hard_knee_compressor_int), (gst_audio_dynamic_transform_hard_knee_compressor_float), (gst_audio_dynamic_transform_soft_knee_compressor_int), (gst_audio_dynamic_transform_soft_knee_compressor_float), (gst_audio_dynamic_transform_hard_knee_expander_int), (gst_audio_dynamic_transform_hard_knee_expander_float), (gst_audio_dynamic_transform_soft_knee_expander_int), (gst_audio_dynamic_transform_soft_knee_expander_float), (gst_audio_dynamic_transform_ip): * gst/audiofx/audiodynamic.h: * gst/audiofx/audiofx.c: (plugin_init): Add new audiodynamic element which can act as a compressor or expander. Supported are hard-knee and soft-knee operation modes with user-specified ratio and threshold. Attack and release parameters are not yet implemented but will follow. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-audiofx.xml: Integrate audiodynamic into the docs. * tests/check/Makefile.am: * tests/check/elements/audiodynamic.c: (setup_dynamic), (cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main): Add unit test for audiodynamic. 2007-03-07 19:48:03 +0000 Jan Schmidt ext/raw1394/gstdv1394src.c: Free handles that we allocated when exiting via the error paths. Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (gst_dv1394src_start): Free handles that we allocated when exiting via the error paths. 2007-03-07 12:07:07 +0000 Sebastian Dröge ext/wavpack/: Use a general wavpack debug category for common code. Original commit message from CVS: * ext/wavpack/gstwavpack.c: (plugin_init): * ext/wavpack/gstwavpackcommon.c: Use a general wavpack debug category for common code. * ext/wavpack/gstwavpackstreamreader.c: (gst_wavpack_stream_reader_set_pos_abs), (gst_wavpack_stream_reader_set_pos_rel), (gst_wavpack_stream_reader_write_bytes): Use the general wavpack debug category here too and add debug output to the functions that should not be called at all by the wavpack library. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init): * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init): Change debugging category names to conform to the conventions. 2007-03-07 11:37:23 +0000 Edward Hervey gst/qtdemux/qtdemux.*: Share qtdemux debug category across all files, otherwise all debugging in files other than qtd... Original commit message from CVS: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: Share qtdemux debug category across all files, otherwise all debugging in files other than qtdemux.c would end up in the default category. 2007-03-07 11:24:05 +0000 Stefan Kost gst/level/gstlevel.*: Resolve message timestamps against the playback segment. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_set_caps), (gst_level_start), (gst_level_event), (gst_level_transform_ip): * gst/level/gstlevel.h: Resolve message timestamps against the playback segment. 2007-03-07 11:23:20 +0000 Stefan Kost gst/spectrum/gstspectrum.*: One FIXME less, by resolving message timestamps against the playback segment. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_start), (gst_spectrum_event), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: One FIXME less, by resolving message timestamps against the playback segment. 2007-03-06 23:21:41 +0000 Tim-Philipp Müller * ChangeLog: Fix ChangeLog message Original commit message from CVS: Fix ChangeLog message 2007-03-06 23:19:30 +0000 Tim-Philipp Müller gst/id3demux/gstid3demux.c: Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the caps passed to ... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad), (gst_id3demux_sink_activate): Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the caps passed to it (previouslly one code path assumes it takes ownership while another one assumes it doesn't). * configure.ac: * tests/files/Makefile.am: * tests/files/id3-407349-1.tag: * tests/files/id3-407349-2.tag: Add directory where data for unit tests can be stored. * tests/Makefile.am: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/id3demux.c: (pad_added_cb), (error_cb), (read_tags_from_file), (run_check_for_file), (check_date_1977_06_23), (GST_START_TEST), (id3demux_suite): Add unit test for id3demux, and in particular for bug #407349. Only testing pull-mode for now; push mode doesn't work yet because the test files are smaller than ID3_TYPE_FIND_MIN_SIZE. 2007-03-06 22:14:59 +0000 Tim-Philipp Müller tests/check/Makefile.am: Add missing backslash at end of line. Original commit message from CVS: * tests/check/Makefile.am: Add missing backslash at end of line. 2007-03-06 18:36:09 +0000 Jan Schmidt * ChangeLog: * common: Trigger rebuild. Original commit message from CVS: Trigger rebuild. 2007-03-06 18:16:49 +0000 Tim-Philipp Müller gst/id3demux/: Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interp... Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list): * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_obsolete_tdat_frame): Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interpreted as a year, whereas it is month and day in DDMM format. Instead, parse TDAT frames and fix up the date in the GST_TAG_DATE tag later if we also extracted a year. Fixes #407349. 2007-03-06 14:53:04 +0000 Jan Schmidt ext/gconf/gstswitchsink.c: Fix up the dispose logic so it doesn't leak, and fix setting of the child state so that we... Original commit message from CVS: * ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose), (gst_switch_commit_new_kid): Fix up the dispose logic so it doesn't leak, and fix setting of the child state so that we don't set a child to our current state just as we are changing it to something else. 2007-03-06 13:57:55 +0000 Wim Taymans gst/spectrum/gstspectrum.c: Fix and cleanup default property values. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_set_property), (gst_spectrum_transform_ip): Fix and cleanup default property values. Add FIXMEs for stuff that looks rather wrong. 2007-03-06 13:21:23 +0000 Wim Taymans gst/goom/gstgoom.*: Document, fix and improve goom adapter behaviour. Original commit message from CVS: * gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer), (gst_goom_chain): * gst/goom/gstgoom.h: Document, fix and improve goom adapter behaviour. Fixes #407006. 2007-03-05 18:43:29 +0000 Jan Schmidt ext/esd/esdsink.c: Unref static pad template after using it. Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_open): Unref static pad template after using it. 2007-03-05 17:17:04 +0000 Jan Schmidt ext/gconf/gstswitchsink.c: Fix up the reference counting of the child elements. Original commit message from CVS: * ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose), (gst_switch_commit_new_kid): Fix up the reference counting of the child elements. 2007-03-05 17:08:32 +0000 Wim Taymans gst/rtp/: Fix encoding-name case. Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps): * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers): Fix encoding-name case. 2007-03-05 16:39:29 +0000 Wim Taymans gst/rtp/: Fix speex (de)payloader. Fixes #358040. Original commit message from CVS: * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init), (gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init), (gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps), (gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer), (gst_rtp_speex_pay_change_state): * gst/rtp/gstrtpspeexpay.h: Fix speex (de)payloader. Fixes #358040. 2007-03-05 15:42:58 +0000 Jan Schmidt ext/gconf/gstswitchsink.c: Install fakesink in NULL by fixing some broken logic. This obviates the need to manually s... Original commit message from CVS: * ext/gconf/gstswitchsink.c: (gst_switch_sink_reset), (gst_switch_commit_new_kid), (gst_switch_sink_set_child): Install fakesink in NULL by fixing some broken logic. This obviates the need to manually set _IS_SINK. Add some comments and remove a little cruft while I'm at it. 2007-03-05 14:46:43 +0000 Wim Taymans ext/gconf/gstswitchsink.c: Mark us as a sink when we have no fakesink in NULL. Fixes #414887. Original commit message from CVS: * ext/gconf/gstswitchsink.c: (gst_switch_sink_reset): Mark us as a sink when we have no fakesink in NULL. Fixes #414887. 2007-03-05 08:30:52 +0000 Stefan Kost gst/spectrum/: Remove two obsolete and confusing comments. Original commit message from CVS: * gst/spectrum/demo-audiotest.c: (message_handler): * gst/spectrum/demo-osssrc.c: (message_handler): Remove two obsolete and confusing comments. 2007-03-04 18:52:12 +0000 Tim-Philipp Müller po/POTFILES.in: Update. Original commit message from CVS: * po/POTFILES.in: Update. 2007-03-04 17:33:34 +0000 Jan Schmidt tests/check/Makefile.am: Gah! Also disable gconfvideosink from the tests, otherwise it will instantiate autovideosink... Original commit message from CVS: * tests/check/Makefile.am: Gah! Also disable gconfvideosink from the tests, otherwise it will instantiate autovideosink, and dfbvideosink and leak on the buildbots. 2007-03-04 17:13:19 +0000 Jan Schmidt ext/cdio/gstcdiocddasrc.c: Make sure we always destroy our libcdio handle. Original commit message from CVS: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open), (gst_cdio_cdda_src_finalize): Make sure we always destroy our libcdio handle. 2007-03-04 17:05:58 +0000 Jan Schmidt tests/check/Makefile.am: Disable autovideosink so the buildbots don't barf over memory leaked in the directfb sink. Original commit message from CVS: * tests/check/Makefile.am: Disable autovideosink so the buildbots don't barf over memory leaked in the directfb sink. 2007-03-04 15:28:30 +0000 Jan Schmidt sys/ximage/gstximagesrc.c: Chain up in dispose Original commit message from CVS: * sys/ximage/gstximagesrc.c: (gst_ximage_src_dispose): Chain up in dispose 2007-03-04 15:07:15 +0000 Jan Schmidt gst/multipart/multipartdemux.c: Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new. Original commit message from CVS: * gst/multipart/multipartdemux.c: (gst_multipart_demux_init), (gst_multipart_find_pad_by_mime): Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new. 2007-03-04 14:56:53 +0000 Jan Schmidt sys/ximage/gstximagesrc.c: Catch the case where no clock has been set. Original commit message from CVS: * sys/ximage/gstximagesrc.c: (gst_ximage_src_create): Catch the case where no clock has been set. 2007-03-04 13:52:03 +0000 Jan Schmidt Fix a bunch of leaks shown by the newly-added states test. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_finalize): * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init), (gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init), (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose), (gst_gconf_audio_src_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init), (gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init), (gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose), (gst_gconf_video_src_finalize), (do_toggle_element): * ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init), (gst_switch_sink_reset), (gst_switch_sink_set_child): * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init), (gst_shout2send_init), (gst_shout2send_finalize): * gst/debug/testplugin.c: (gst_test_class_init), (gst_test_finalize): * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (gst_flxdec_dispose): * gst/multipart/multipartmux.c: (gst_multipart_mux_finalize): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize): * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context): * gst/rtsp/rtspextwms.h: * gst/smpte/gstsmpte.c: (gst_smpte_class_init), (gst_smpte_finalize): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize): * gst/udp/gstudpsink.c: (gst_udpsink_class_init), (gst_udpsink_finalize): * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose), (gst_wavparse_sink_activate): * sys/oss/gstosssink.c: (gst_oss_sink_finalise): * sys/oss/gstosssrc.c: (gst_oss_src_class_init), (gst_oss_src_finalize): * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_finalize): * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get): Fix a bunch of leaks shown by the newly-added states test. 2007-03-04 13:41:00 +0000 Jan Schmidt ext/dv/gstdvdec.c: Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_init): Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new. 2007-03-03 13:06:21 +0000 Loïc Minier Don't mix tabs and spaces (#414168). Original commit message from CVS: Patch by: Loïc Minier * ext/libcaca/Makefile.am: * gst/debug/Makefile.am: Don't mix tabs and spaces (#414168). 2007-03-02 21:35:11 +0000 Stefan Kost tests/check/generic/.cvsignore: Ignore files to please buildbot. Original commit message from CVS: * tests/check/generic/.cvsignore: Ignore files to please buildbot. 2007-03-02 21:01:19 +0000 Stefan Kost gst/wavparse/gstwavparse.c: Unbreak my previous commit (swapped nominator & denominator). Tim, thanks for spotting. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Unbreak my previous commit (swapped nominator & denominator). Tim, thanks for spotting. 2007-03-02 16:08:17 +0000 Wim Taymans ext/cdio/gstcdiocddasrc.c: Small code cleanups. Original commit message from CVS: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices), (gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open), (gst_cdio_cdda_src_finalize): Small code cleanups. Don't use pad_alloc as the base class cannot deal with the error codes. 2007-03-02 13:40:06 +0000 Wim Taymans gst/udp/gstudpsrc.c: Fix doc. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create): Fix doc. 2007-03-02 13:29:25 +0000 René Stadler gst/wavparse/gstwavparse.c: Handle rounding better to not drop last sample frame. Fixes #356692 Original commit message from CVS: Patch by: René Stadler * gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Handle rounding better to not drop last sample frame. Fixes #356692 2007-03-02 13:19:57 +0000 Jan Schmidt tests/check/Makefile.am: Disable cacasink from the states check too - it also calls exit(1) on us when it can't find ... Original commit message from CVS: * tests/check/Makefile.am: Disable cacasink from the states check too - it also calls exit(1) on us when it can't find a terminal to talk to. 2007-03-02 12:56:13 +0000 Thijs Vermeir gst/udp/gstudpsrc.*: Add support to strip proprietary headers. Fixes #350296. Original commit message from CVS: Patch by: Thijs Vermeir * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property): * gst/udp/gstudpsrc.h: Add support to strip proprietary headers. Fixes #350296. 2007-03-02 12:52:56 +0000 Wim Taymans gst/rtp/gstrtpmp2tdepay.c: Fix compilation. Original commit message from CVS: * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process): Fix compilation. 2007-03-02 12:16:16 +0000 Thijs Vermeir gst/rtp/gstrtpmp2tdepay.*: Add support to strip off proprietary headers. Fixes #350278. Original commit message from CVS: Patch by: Thijs Vermeir * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process), (gst_rtp_mp2t_depay_set_property), (gst_rtp_mp2t_depay_get_property): * gst/rtp/gstrtpmp2tdepay.h: Add support to strip off proprietary headers. Fixes #350278. 2007-03-02 11:22:35 +0000 Wim Taymans ext/hal/hal.c: Fix compilation. Original commit message from CVS: * ext/hal/hal.c: Fix compilation. 2007-03-02 10:54:49 +0000 Wim Taymans sys/sunaudio/gstsunaudiosrc.*: Remove device-name from GstSunAudioSrc. Fixes #412597. Original commit message from CVS: * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_class_init), (gst_sunaudiosrc_init), (gst_sunaudiosrc_get_property), (gst_sunaudiosrc_open): * sys/sunaudio/gstsunaudiosrc.h: Remove device-name from GstSunAudioSrc. Fixes #412597. 2007-03-01 21:50:36 +0000 Sebastian Dröge ext/hal/: Having NULL as UDI previously selected the default sink/src. Change this back but mention it in the debug o... Original commit message from CVS: * ext/hal/gsthalaudiosink.c: (do_toggle_element): * ext/hal/gsthalaudiosrc.c: (do_toggle_element): Having NULL as UDI previously selected the default sink/src. Change this back but mention it in the debug output. * ext/hal/hal.c: (gst_hal_get_alsa_element), (gst_hal_get_oss_element), (gst_hal_get_string), (gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink), (gst_hal_get_audio_src): * ext/hal/hal.h: Refactor a bit, check all error conditions, greatly improve debugging and fix some possible memory leaks. Also implement OSS support and allow specifying an UDI that points to a real device. For this the child device which supports ALSA (preferred) or OSS is used. As a side effect this makes it impossible now to get a alsasink in halaudiosrc and a alsasrc in halaudiosink. 2007-03-01 18:47:28 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Errors from the udp sources are not fatal unless all of them are in error. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (find_stream_by_channel), (find_stream_by_udpsrc), (gst_rtspsrc_handle_message): Errors from the udp sources are not fatal unless all of them are in error. 2007-03-01 18:14:42 +0000 Jan Schmidt tests/check/Makefile.am: Disable aasink in the states test. I suspect this is the element that is calling exit(1) whe... Original commit message from CVS: * tests/check/Makefile.am: Disable aasink in the states test. I suspect this is the element that is calling exit(1) when it can't proceed. 2007-03-01 17:26:30 +0000 Jan Schmidt tests/check/Makefile.am: Draw plugins in from the build tree sys/ dir, rather than picking up the already installed v... Original commit message from CVS: * tests/check/Makefile.am: Draw plugins in from the build tree sys/ dir, rather than picking up the already installed versions. 2007-03-01 10:44:36 +0000 Zaheer Abbas Merali sys/ximage/gstximagesrc.c: Error out correctly when getting xcontext fails. Original commit message from CVS: 2007-03-01 Zaheer Abbas Merali * sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display): Error out correctly when getting xcontext fails. 2007-03-01 09:29:34 +0000 Wim Taymans gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc... Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state): Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc relies on it. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_change_state): Don't error out when we don't get an error from the state change function. 2007-03-01 01:48:59 +0000 Sebastian Dröge ext/hal/: Check if the device UDI is set before trying to query HAL about it and give a useful error message if it wa... Original commit message from CVS: * ext/hal/gsthalaudiosink.c: (do_toggle_element): * ext/hal/gsthalaudiosrc.c: (do_toggle_element): Check if the device UDI is set before trying to query HAL about it and give a useful error message if it wasn't set. * ext/hal/hal.c: (gst_hal_get_string): Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL gives an assertion failure in D-Bus when running with DBUS_FATAL_WARNINGS=1. 2007-02-28 19:29:42 +0000 Thomas Vander Stichele * win32/common/config.h: update config to trunk Original commit message from CVS: update config to trunk 2007-02-28 19:29:25 +0000 Thomas Vander Stichele configure.ac: Convert to new AG_GST style. Original commit message from CVS: * configure.ac: Convert to new AG_GST style. 2007-02-28 18:41:38 +0000 Jan Schmidt ext/lame/gstlame.c: Display sensible defaults and limits for the vbr-min/max/mean properties. Fix the 'hard-limit' VB... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init), (gst_lame_setup): Display sensible defaults and limits for the vbr-min/max/mean properties. Fix the 'hard-limit' VBR min property - it's supposed to be a boolean 0/1 value. 2007-02-28 16:01:08 +0000 Jan Schmidt ext/lame/gstlame.c: Initialise the variables so gcc doesn't complain about possibly uninitialised uses, even though t... Original commit message from CVS: * ext/lame/gstlame.c: Initialise the variables so gcc doesn't complain about possibly uninitialised uses, even though they can't actually happen. 2007-02-28 12:59:43 +0000 Thomas Vander Stichele tests/check/: add test for states Original commit message from CVS: * tests/check/Makefile.am: * tests/check/generic/states.c: (GST_START_TEST), (states_suite): add test for states 2007-02-28 10:58:10 +0000 Wim Taymans tests/check/elements/.cvsignore: Add new videofilter check to .cvsignore. Original commit message from CVS: * tests/check/elements/.cvsignore: Add new videofilter check to .cvsignore. 2007-02-28 10:54:55 +0000 Wim Taymans gst/avi/gstavidemux.c: Fix combined flow return. Fixes #412608. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop), (gst_avi_demux_chain): Fix combined flow return. Fixes #412608. 2007-02-28 10:41:14 +0000 Wim Taymans gst/videofilter/Makefile.am: Dist header.. Original commit message from CVS: * gst/videofilter/Makefile.am: Dist header.. 2007-02-28 10:29:08 +0000 Wim Taymans gst/videofilter/gstgamma.h: Add header too. Original commit message from CVS: * gst/videofilter/gstgamma.h: Add header too. 2007-02-28 10:17:15 +0000 Mark Nauwelaerts gst/videofilter/: Port gamma filter to 0.10. Fixes #412704. Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/videofilter/Makefile.am: * gst/videofilter/gstgamma.c: (gst_gamma_base_init), (gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property), (gst_gamma_get_property), (gst_gamma_calculate_tables), (oil_tablelookup_u8), (gst_gamma_set_caps), (gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init): Port gamma filter to 0.10. Fixes #412704. * tests/check/Makefile.am: * tests/check/elements/videofilter.c: (setup_filter), (cleanup_filter), (check_filter), (GST_START_TEST), (videobalance_suite), (videoflip_suite), (gamma_suite), (main): Add unit tests for videofilters. 2007-02-28 10:06:27 +0000 Wim Taymans gst/rtsp/URLS: Add another interesting test url. Original commit message from CVS: * gst/rtsp/URLS: Add another interesting test url. * gst/rtsp/rtspmessage.c: (rtsp_message_get_header): Don't allow getting header fields from data packets. 2007-02-27 23:43:08 +0000 Michael Smith ext/shout2/gstshout2.*: Add a property for username. Original commit message from CVS: * ext/shout2/gstshout2.c: (gst_shout2send_class_init), (gst_shout2send_init), (gst_shout2send_start), (gst_shout2send_set_property), (gst_shout2send_get_property): * ext/shout2/gstshout2.h: Add a property for username. 2007-02-27 12:02:03 +0000 Christian Schaller * sys/directdraw/gstdirectdrawplugin.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: * sys/directsound/gstdirectsoundplugin.c: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: update copyright statements Original commit message from CVS: update copyright statements 2007-02-27 11:59:21 +0000 Christian Schaller * ChangeLog: * sys/osxaudio/gstosxaudio.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudioelement.h: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosink.h: update copyright statement Original commit message from CVS: update copyright statement 2007-02-27 11:30:19 +0000 Edward Hervey sys/osxvideo/: Disable the cocoa event loop since it's a huge memory leak. Should only matter if the sink isn't used ... Original commit message from CVS: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Disable the cocoa event loop since it's a huge memory leak. Should only matter if the sink isn't used within an NSApp (which has already got a coca event loop). Remove all unused code. 2007-02-26 12:07:14 +0000 Jan Schmidt gst/rtsp/Makefile.am: Fix make check too. Original commit message from CVS: * gst/rtsp/Makefile.am: Fix make check too. 2007-02-26 10:00:28 +0000 Jan Schmidt gst/rtsp/base64.*: Commit missing files for base64 encoding. Original commit message from CVS: * gst/rtsp/base64.c: (util_base64_encode): * gst/rtsp/base64.h: Commit missing files for base64 encoding. 2007-02-24 22:57:49 +0000 Loïc Minier Fix build with LDFLAGS='-Wl,-z,defs' (#410997) Original commit message from CVS: Patch by: Loïc Minier * configure.ac: * ext/annodex/Makefile.am: * ext/jpeg/Makefile.am: * ext/speex/Makefile.am: * gst/alpha/Makefile.am: * gst/cutter/Makefile.am: * gst/debug/Makefile.am: * gst/effectv/Makefile.am: * gst/goom/Makefile.am: * gst/level/Makefile.am: * gst/smpte/Makefile.am: * gst/videofilter/Makefile.am: Fix build with LDFLAGS='-Wl,-z,defs' (#410997) 2007-02-24 22:52:47 +0000 Tim-Philipp Müller Fix build with LDFLAGS='-Wl,-z,defs'. Original commit message from CVS: * configure.ac: * ext/gsm/Makefile.am: * ext/ladspa/Makefile.am: * ext/wavpack/Makefile.am: * gst/equalizer/Makefile.am: * gst/filter/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/replaygain/Makefile.am: * gst/speed/Makefile.am: Fix build with LDFLAGS='-Wl,-z,defs'. 2007-02-23 19:12:52 +0000 Jan Schmidt gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed fr... Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/rtspconnection.c: (append_auth_header), (rtsp_connection_send), (rtsp_connection_set_auth): g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed from GPL courtesy of Mike Smith. 2007-02-23 18:12:27 +0000 Jan Schmidt gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream), (rtsp_auth_method_to_string), (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth), (gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (append_auth_header), (rtsp_connection_send), (rtsp_connection_free), (rtsp_connection_set_auth): * gst/rtsp/rtspconnection.h: * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri): * gst/rtsp/rtspurl.h: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work on hosts that require authentication. 2007-02-22 17:53:26 +0000 Edgard Lima * ChangeLog: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/v4l2_calls.c: Fix segfault when oppening a radio device. Original commit message from CVS: Fix segfault when oppening a radio device. 2007-02-22 14:35:28 +0000 Stefan Kost Fix level for multi-channel case. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_set_caps), (gst_level_transform_ip): * sys/v4l2/README: * tests/check/elements/level.c: (GST_START_TEST): Fix level for multi-channel case. 2007-02-21 16:02:33 +0000 Thomas Vander Stichele ext/lame/gstlame.c: Fix up bitrate checking macro. Make it give us a Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_setcaps), (gst_lame_set_property), (gst_lame_setup): Fix up bitrate checking macro. Make it give us a GST_ELEMENT_WARNING message so the application has a chance of reporting this to the user. Move the checking to _setup, so we are sure it runs in the READY state, when we hope to have a pipeline and a bus that is not flushing. This fixes e.g. using 96 kbit/sec as a bitrate. 2007-02-21 10:18:12 +0000 Stefan Kost gst/level/gstlevel.*: Use function pointer for process function and add process functions for float audio. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps), (gst_level_transform_ip): * gst/level/gstlevel.h: Use function pointer for process function and add process functions for float audio. 2007-02-20 21:34:00 +0000 Sébastien Moutte sys/directsound/gstdirectsoundsink.*: Remove include of unused headers. Original commit message from CVS: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: Remove include of unused headers. * sys/waveform/gstwaveformplugin.c: * sys/waveform/gstwaveformsink.c: * sys/waveform/gstwaveformsink.h: * win32/vs6/libgstwaveform.dsp: Add a new waveform plugin which includes an audio sink element using the WaveForm win32 API. * win32/MANIFEST: Add the new project file form waveform plugin. 2007-02-19 12:22:43 +0000 Stefan Kost sys/v4l2/v4l2src_calls.c: Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO, fixes #407369 Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init): Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO, fixes #407369 2007-02-18 18:00:51 +0000 Sébastien Moutte sys/directdraw/: Prepare the plugin to move to good: Original commit message from CVS: * sys/directdraw/gstdirectdrawplugin.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: Prepare the plugin to move to good: Remove unused/untested code (rendering to an extern surface, yuv format rendering).Use GST_(DEBUG/*)_OBJECT macros Rename all functions from gst_directdrawsink to gst_directdraw_sink. Add gtk doc section Fix a bug in gst_directdraw_sink_show_frame, memcpy line by line respecting destination surface stride. * sys/directsound/gstdirectsoundplugin.c: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: Prepare the plugin to move to good: Rename all functions from gst_directsoundsink to gst_directsound_sink. Add gtk doc section * win32/common/config.h.in: * win32/MANIFEST: Add config.h.in 2007-02-18 13:24:26 +0000 Wim Taymans gst/rtp/: Added simple mpeg transport stream payloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init), (gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer), (gst_rtp_mp2t_pay_plugin_init): * gst/rtp/gstrtpmp2tpay.h: Added simple mpeg transport stream payloader. 2007-02-16 12:32:01 +0000 Wim Taymans gst/rtsp/URLS: Add example H264 rtsp url. Original commit message from CVS: * gst/rtsp/URLS: Add example H264 rtsp url. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): Don't convert values to lowercase or we might mess up base64 encoded properties. 2007-02-16 12:30:22 +0000 Wim Taymans gst/rtp/README: Fix case of string params. Original commit message from CVS: * gst/rtp/README: Fix case of string params. * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process): Fix depayloader, support more packet types. Add sync codes to make sure the packetizer can do its job. * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process): Fix caps case again. 2007-02-15 12:26:28 +0000 Wim Taymans gst/rtp/gstrtph264depay.c: Set right caps on output buffers. Original commit message from CVS: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process): Set right caps on output buffers. 2007-02-14 17:04:47 +0000 Wim Taymans gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling _init() on it. Original commit message from CVS: * gst/rtsp/sdpmessage.c: (sdp_parse_line): As spotted by: Peter Kjellerstedt : Clear stack allocated SDPMedia struct before calling _init() on it. Clarify this in the docs as well. 2007-02-14 17:01:25 +0000 Jan Schmidt ext/gconf/gstgconfaudiosink.c: Don't reset the profile when going switching states, as it makes the element non-reusa... Original commit message from CVS: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset), (do_change_child): Don't reset the profile when going switching states, as it makes the element non-reusable. 2007-02-14 15:24:50 +0000 jp.liu gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793. Original commit message from CVS: * gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init), (sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init), (sdp_key_init), (sdp_attribute_init), (sdp_message_init), (sdp_message_uninit), (sdp_message_free), (sdp_media_init), (sdp_media_uninit), (sdp_media_free), (sdp_message_add_media), (sdp_parse_line): * gst/rtsp/sdpmessage.h: Based on patch by: jp.liu Fix memory management of SDP messages. Fixes #407793. 2007-02-14 12:07:01 +0000 zhangfei gao gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). Fixes #407780. Original commit message from CVS: Patch by: zhangfei gao * gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps): Allow muxing video/x-h264 (was already in the caps). Fixes #407780. 2007-02-14 10:09:12 +0000 jp.liu gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797. Original commit message from CVS: Patch by: jp.liu * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of password field in url. Fixes #407797. 2007-02-14 09:55:47 +0000 Wim Taymans gst/wavparse/gstwavparse.*: Update docs. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (gst_wavparse_pad_convert), (gst_wavparse_pad_query), (gst_wavparse_srcpad_event), (gst_wavparse_change_state), (plugin_init): * gst/wavparse/gstwavparse.h: Update docs. Use boilerplate. Various code cleanups. When the bitrate is not known (bps == 0 or compressed formats) let downstream element guestimate the duration and position and don't generate timestamps or durations. Fixes #405213. Fix EOS and ERROR conditions in chain mode, we just need to forward the error flowreturn upstream. 2007-02-13 16:01:29 +0000 Jan Schmidt Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. ... Original commit message from CVS: * ext/gconf/Makefile.am: * ext/gconf/gconf.c: (gst_gconf_get_string), (gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string), (gst_gconf_render_bin_with_default): * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init), (gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init), (gst_gconf_audio_sink_dispose), (do_change_child), (gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property), (cb_change_child), (gst_gconf_audio_sink_change_state): * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init), (gst_switch_sink_class_init), (gst_switch_sink_reset), (gst_switch_sink_init), (gst_switch_sink_dispose), (gst_switch_commit_new_kid), (gst_switch_sink_set_child), (gst_switch_sink_set_property), (gst_switch_sink_handle_event), (gst_switch_sink_get_property), (gst_switch_sink_change_state): * ext/gconf/gstswitchsink.h: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose), (gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset), (gst_auto_audio_sink_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose), (gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset), (gst_auto_video_sink_detect): Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. The end goal of this is an audio sink that can be changed on the fly, but at the moment it still only changes on the next READY transition. 2007-02-13 11:57:18 +0000 Stefan Kost gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif 2007-02-13 09:46:26 +0000 Stefan Kost Add crossreferences to glib/gobject/gstream docs. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: Add crossreferences to glib/gobject/gstream docs. 2007-02-12 23:35:16 +0000 Tim-Philipp Müller gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS (but no LIBS, since we only use define... Original commit message from CVS: * gst/monoscope/Makefile.am: * gst/monoscope/gstmonoscope.c: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS (but no LIBS, since we only use defines from the headers). 2007-02-12 23:27:31 +0000 Jonathan Matthew gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in streaming mode due to Original commit message from CVS: Based on patch by: Jonathan Matthew * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init), (gst_wavparse_stream_data): Fix massive memory leak when operating in streaming mode due to GST_BUFFER_MALLOCDATA() not being set on newly-created buffers. Fixes #407057. 2007-02-12 15:29:44 +0000 Stefan Kost gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs t... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs to questionable parts. 2007-02-12 12:57:22 +0000 Stefan Kost sys/v4l2/: More FIXME comments and messaging changes. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps), (gst_v4l2src_get_caps): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init): More FIXME comments and messaging changes. 2007-02-12 12:43:00 +0000 Stefan Kost gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR. Original commit message from CVS: * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init), (gst_goom_change_state): * gst/goom/gstgoom.h: Improved docs and use GST_DEBUG_FUNCPTR. * gst/level/gstlevel.c: (gst_level_class_init): Use GST_DEBUG_FUNCPTR. * gst/monoscope/gstmonoscope.c: (gst_monoscope_init), (gst_monoscope_chain), (gst_monoscope_change_state): Improved docs source cleanups. 2007-02-12 10:29:57 +0000 Tim-Philipp Müller gst/debug/: Add code for a pushfilesrc element that implements a pushfile:// URI handler, to make debugging push-mode... Original commit message from CVS: * gst/debug/Makefile.am: * gst/debug/gstdebug.c: (plugin_init): * gst/debug/gstpushfilesrc.c: * gst/debug/gstpushfilesrc.h: Add code for a pushfilesrc element that implements a pushfile:// URI handler, to make debugging push-mode operation of demuxer/decoders that support both easier in connection with seek/playbin/etc. The element isn't registered at the moment. 2007-02-11 15:26:49 +0000 Sébastien Moutte Makefile.am: Add win32 MANIFEST Original commit message from CVS: * Makefile.am: Add win32 MANIFEST * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: Clear unused code and add comments. Remove yuv from template caps, it only supports RGB actually. Implement XOverlay interface and remove window and fullscreen properties. Add debug logs. Test for blit capabilities to return only the current colorspace if the hardware can't blit for one colorspace to another. * sys/directsound/gstdirectsoundsink.c: Add some debugs. * win32/MANIFEST: Add VS7 project files and solution. * win32/vs6/gst_plugins_bad.dsw: * win32/vs6/libgstdirectdraw.dsp: * win32/vs6/libgstdirectsound.dsp: * win32/vs6/libgstqtdemux.dsp: Update project files. 2007-02-11 12:57:47 +0000 Sébastien Moutte gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. Original commit message from CVS: * gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. * gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp): Use gst_guint64_to_gdouble for conversion. * gst/rtsp/rtspconnection.c:(rtsp_connection_send): Move variables declaration before the first instruction. * gst/rtsp/rtspdefs.c:(rtsp_strresult): Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported. And don't include netdb.h for G_OS_WIN32 * gst/rtsp/sdpmessage.c:(sdp_parse_line): This initialization SDPMedia nmedia = {.media = NULL }; is not supported by VS6 then use an other way to initialize SDPMedia structure. * gst/udp/gstdynudpsink.h: * gst/udp/gstdynudpnetutils.h: Do not include for G_OS_WIN32 * gst/udp/gstudpsrc.c: Define socklen_t as int for G_OS_WIN32 * win/common/config.h.in: Undef HAVE_NETINET_IN_H * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstautogen.dsp: * win32/vs6/libgstaudiofx.dsp: * win32/vs6/libgstudp.dsp: Add and update project files. * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-enumtypes.h: Add a copy of udp enumtypes to win32/common as in core and base. 2007-02-11 10:53:21 +0000 Stefan Kost configure.ac: Activate monoscope when building with --enable-experimental. Fix Original commit message from CVS: * configure.ac: Activate monoscope when building with --enable-experimental. Fix --enable-external configure switch description. * sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init): * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose): Help gst-indent. 2007-02-09 16:24:45 +0000 Tim-Philipp Müller ext/lame/gstlame.*: On receiving EOS, we try to push a last buffer with the remaining samples. Don't do that if we go... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain), (gst_lame_change_state): * ext/lame/gstlame.h: On receiving EOS, we try to push a last buffer with the remaining samples. Don't do that if we got an unclean flow return on the last gst_pad_push(), downstream might not handle this very gracefully (see #403168). * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Pass flow returns upstream (helps #403168). 2007-02-09 09:24:58 +0000 Tim-Philipp Müller gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer in order to avoid compiler warnings on s... Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header): Explicitly cast result of pointer arithmetic to integer in order to avoid compiler warnings on some 64-bit systems. Should fix #406018. 2007-02-08 11:09:15 +0000 Tim-Philipp Müller gst/debug/progressreport.c: Some more docs. Original commit message from CVS: * gst/debug/progressreport.c: Some more docs. 2007-02-07 21:09:45 +0000 Tim-Philipp Müller docs/plugins/inspect/plugin-rtp.xml: Update for new elements. Original commit message from CVS: * docs/plugins/inspect/plugin-rtp.xml: Update for new elements. * gst/debug/progressreport.h: Commit newly-created header file as well. 2007-02-07 20:39:16 +0000 Tim-Philipp Müller Make progressreport element post messages with the current progress on the bus. Also add some basic docs for it. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * gst/debug/Makefile.am: * gst/debug/progressreport.c: (gst_progress_report_post_progress), (gst_progress_report_do_query), (gst_progress_report_report): Make progressreport element post messages with the current progress on the bus. Also add some basic docs for it. 2007-02-07 13:08:34 +0000 Tim-Philipp Müller ext/hal/hal.*: Some small cleanups; deal with errors when parsing the HAL ALSA capabilities a bit better. Original commit message from CVS: * ext/hal/hal.c: (gst_hal_get_string): * ext/hal/hal.h: Some small cleanups; deal with errors when parsing the HAL ALSA capabilities a bit better. 2007-02-06 16:29:30 +0000 Tim-Philipp Müller gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time. Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type): Let's try this again and use the right cast this time. 2007-02-06 16:24:57 +0000 Tim-Philipp Müller gst/smpte/gstsmpte.c: Add cast to avoid compiler warnings with older GLib versions where the nick/name members in GEn... Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type): Add cast to avoid compiler warnings with older GLib versions where the nick/name members in GEnumValue are not declared as constant strings. 2007-02-06 15:56:14 +0000 Tim-Philipp Müller ext/gconf/: In gconfaudiosink, get the right key as the old key in do_toggle (ie. one dependent on the profile select... Original commit message from CVS: * ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile), (gst_gconf_render_bin_from_key), (gst_gconf_get_default_audio_sink): * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile), (do_toggle_element), (gst_gconf_audio_sink_set_property), (gst_gconf_audio_sink_get_property): In gconfaudiosink, get the right key as the old key in do_toggle (ie. one dependent on the profile selected). Log some more stuff so we can see what's actually going on. 2007-02-06 11:16:49 +0000 Sebastian Dröge gst/audiofx/: Some small cleanups and port both elements to the new GstAudioFilter base class to save a few lines of ... Original commit message from CVS: * gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init), (gst_audio_amplify_class_init), (gst_audio_amplify_init), (gst_audio_amplify_set_process_function), (gst_audio_amplify_setup): * gst/audiofx/audioamplify.h: * gst/audiofx/audioinvert.c: (gst_audio_invert_base_init), (gst_audio_invert_class_init), (gst_audio_invert_setup): * gst/audiofx/audioinvert.h: Some small cleanups and port both elements to the new GstAudioFilter base class to save a few lines of common code. * gst/audiofx/Makefile.am: Link against libgstaudio for the above changes 2007-02-03 23:35:26 +0000 Tim-Philipp Müller Fix up to use the newly ported (actually working) GstAudioFilter. Original commit message from CVS: * configure.ac: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_init), (setup_filter), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup), (plugin_init): * gst/equalizer/gstiirequalizer.h: Fix up to use the newly ported (actually working) GstAudioFilter. Bump core/base requirements to CVS for this. * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/equalizer-test.c: (check_bus), (equalizer_set_band_value), (equalizer_set_all_band_values), (equalizer_set_band_value_and_wait), (equalizer_set_all_band_values_and_wait), (do_slider_fiddling), (main): Add brain-dead interactive test for equalizer. 2007-02-02 18:36:28 +0000 Tim-Philipp Müller gst/equalizer/gstiirequalizer.c: Rename "values" property to "band-values" and change type into a Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_filter_inplace): Rename "values" property to "band-values" and change type into a GValueArray, so it's more easily bindable and the range of the values passed in is defined and checked etc.; also do some locking. 2007-02-02 17:39:21 +0000 James Doc Livingston Port equalizer plugin to 0.10 (#403572). Original commit message from CVS: Patch by: James "Doc" Livingston * configure.ac: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type), (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup), (plugin_init): Port equalizer plugin to 0.10 (#403572). 2007-01-31 08:32:59 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Fix a off by one that leads to the duration reported as one sample less than it is Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_create_src_pad): Fix a off by one that leads to the duration reported as one sample less than it is 2007-01-30 17:19:33 +0000 Edward Hervey configure.ac: Check for an Objective C compiler Original commit message from CVS: * configure.ac: Check for an Objective C compiler * sys/Makefile.am: * sys/osxvideo/Makefile.am: * sys/osxvideo/cocoawindow.h: * sys/osxvideo/cocoawindow.m: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Port of osxvideo plugin to 0.10. Do NOT consider 100% stable ! Fixes #402470 2007-01-29 10:59:48 +0000 Wim Taymans tests/check/elements/.cvsignore: Some more ignores. Original commit message from CVS: * tests/check/elements/.cvsignore: Some more ignores. 2007-01-28 18:28:33 +0000 Tim-Philipp Müller gst/videocrop/gstvideocrop.c: Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY. Original commit message from CVS: * gst/videocrop/gstvideocrop.c: (gst_video_crop_get_image_details_from_caps), (gst_video_crop_transform_packed_complex): Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY. * tests/icles/videocrop-test.c: (check_bus_for_errors), (test_with_caps), (main): Block streaming thread before changing filter caps while the pipeline is running so that we don't get random not-negotiated errors just because GStreamer can't handle that yet. 2007-01-27 16:08:15 +0000 Tim-Philipp Müller tests/icles/videocrop-test.c: Catch errors while the test is running. Original commit message from CVS: * tests/icles/videocrop-test.c: (test_with_caps): Catch errors while the test is running. 2007-01-26 12:21:41 +0000 charles ext/shout2/gstshout2.*: Properly handle tags in shout2send. Fixes #399825. Original commit message from CVS: Patch by: charles * ext/shout2/gstshout2.c: (gst_shout2send_init), (set_shout_metadata), (gst_shout2send_event): * ext/shout2/gstshout2.h: Properly handle tags in shout2send. Fixes #399825. 2007-01-25 23:27:59 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Fix the SEEKING query. We can seek if we are in pull mode, not the other way around. A... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query): Fix the SEEKING query. We can seek if we are in pull mode, not the other way around. Also set the correct format in the seeking query and handle the case where the headers are not read yet and we can't say anything about our seeking capabilities. 2007-01-25 21:55:49 +0000 Sebastian Dröge ext/wavpack/: Fix spelling in 2 places: It's called Wavpack, not WavePack. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): Fix spelling in 2 places: It's called Wavpack, not WavePack. 2007-01-25 14:40:15 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the rules in the SDP to caps document. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_activate_streams): Convert SDP fields to upper/lowercase following the rules in the SDP to caps document. 2007-01-25 14:22:53 +0000 Wim Taymans gst/rtp/: Fix case of encoding-name and key/value pairs to match the document. Original commit message from CVS: * gst/rtp/README: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Fix case of encoding-name and key/value pairs to match the document. This is to make interoperation with SDP case-insensitive as required by the relevant RFCs. 2007-01-25 12:05:11 +0000 Edward Hervey gst/: Use proper print statements. Original commit message from CVS: * gst/multifile/gstmultifilesink.c: (gst_multi_file_sink_class_init): * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init): * gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer), (gst_mve_video_palette), (gst_mve_video_code_map), (gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create), (gst_mve_demux_chain): * gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk): * gst/mve/mveaudioenc.c: (mve_compress_audio): * gst/mve/mvevideodec16.c: (ipvideo_copy_block): * gst/mve/mvevideodec8.c: (ipvideo_copy_block): * gst/mve/mvevideoenc16.c: (mve_encode_frame16): * gst/mve/mvevideoenc8.c: (mve_encode_frame8): Use proper print statements. Fixes build on mac os x. oo look at me my name is edward i'm hacking on macos wooo 2007-01-25 11:02:01 +0000 Wim Taymans configure.ac: Bump required -core/-base to CVS Original commit message from CVS: * configure.ac: Bump required -core/-base to CVS 2007-01-25 10:54:19 +0000 Wim Taymans gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter. Original commit message from CVS: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer): * gst/rtp/gstrtpL16pay.h: Fill up to MTU using adapter. Timestamp rtp packets. 2007-01-25 10:36:35 +0000 Edward Hervey Use G_GSIZE_FORMAT in print statements for portability. Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_collected): * sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls): Use G_GSIZE_FORMAT in print statements for portability. Fixes build on macosx. 2007-01-24 18:20:14 +0000 Wim Taymans gst/rtp/: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init), (gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init), (gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps), (gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property), (gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state), (gst_rtp_L16_depay_plugin_init): * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type), (gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init), (gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize), (gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer), (gst_rtp_L16_pay_plugin_init): * gst/rtp/gstrtpL16pay.h: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side. 2007-01-24 16:25:55 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Only unblock the udp pads when we linked and activated them all. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp): * gst/rtsp/gstrtspsrc.h: Only unblock the udp pads when we linked and activated them all. Fixes #395688. 2007-01-24 15:18:34 +0000 Wim Taymans gst/rtp/: Added simple AC3 depayloader (RFC 4184). Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init), (gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init), (gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process), (gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property), (gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init): * gst/rtp/gstrtpac3depay.h: Added simple AC3 depayloader (RFC 4184). * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps): Fix a leak. 2007-01-24 12:41:03 +0000 Sebastian Dröge gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme... Original commit message from CVS: reviewed by: Stefan Kost * gst/audiofx/Makefile.am: * gst/audiofx/audioamplify.c: (gst_audio_amplify_clipping_method_get_type), (gst_audio_amplify_base_init), (gst_audio_amplify_class_init), (gst_audio_amplify_init), (gst_audio_amplify_set_process_function), (gst_audio_amplify_set_property), (gst_audio_amplify_get_property), (gst_audio_amplify_set_caps), (gst_audio_amplify_transform_int_clip), (gst_audio_amplify_transform_int_wrap_negative), (gst_audio_amplify_transform_int_wrap_positive), (gst_audio_amplify_transform_float_clip), (gst_audio_amplify_transform_float_wrap_negative), (gst_audio_amplify_transform_float_wrap_positive), (gst_audio_amplify_transform_ip): * gst/audiofx/audioamplify.h: * gst/audiofx/audiofx.c: (plugin_init): Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" element, but provides different modes for clipping and allows unlimited amplification. It's mainly targeted for creative sound design and not as a replacement of the "volume" element. Fixes #397162 * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-audiofx.xml: Add docs for audioamplify and integrate them into the build system * tests/check/Makefile.am: * tests/check/elements/audioamplify.c: (setup_amplify), (cleanup_amplify), (GST_START_TEST), (amplify_suite), (main): Add fairly extensive unit test suite for audioamplify 2007-01-24 12:26:41 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Unblock pads after adding the pads to the element so that autopluggers get a change to link so... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked): Unblock pads after adding the pads to the element so that autopluggers get a change to link something. Possibly fixes #395688. 2007-01-24 12:22:51 +0000 Wim Taymans gst/rtp/: Fix caps with payload numbers. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process): * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Fix caps with payload numbers. Add some fixed payload numbers to caps when possible. 2007-01-24 11:29:00 +0000 Wim Taymans gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader. Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader. 2007-01-23 18:16:09 +0000 Sebastian Dröge gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can b... Original commit message from CVS: reviewed by: Stefan Kost * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audioinvert.c: (gst_audio_invert_base_init), (gst_audio_invert_class_init), (gst_audio_invert_init), (gst_audio_invert_set_property), (gst_audio_invert_get_property), (gst_audio_invert_set_caps), (gst_audio_invert_transform_int), (gst_audio_invert_transform_float), (gst_audio_invert_transform_ip): * gst/audiofx/audioinvert.h: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can be used for example for a wide-stereo effect. Fixes #396057 * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-audiofx.xml: Add docs for the audioinvert element and add them to the build system. * tests/check/Makefile.am: * tests/check/elements/audioinvert.c: (setup_invert), (cleanup_invert), (GST_START_TEST), (invert_suite), (main): Add unit test suite for the audioinvert element. 2007-01-23 17:36:32 +0000 Wim Taymans gst/rtp/gstrtpmp4gdepay.c: Parse config params as string and int. Original commit message from CVS: * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int), (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process): Parse config params as string and int. Parse and use AU header length 2007-01-23 17:27:39 +0000 Wim Taymans gst/smpte/: constify some static structs. Original commit message from CVS: * gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw), (gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw): * gst/smpte/gstmask.c: (_gst_mask_register): * gst/smpte/gstmask.h: * gst/smpte/gstsmpte.c: (gst_smpte_update_mask): * gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line), (gst_smpte_paint_triangle_clock): constify some static structs. Don't update the mask if nothing changed to the params. Make sure we never draw outside of the picture. Fixes #398325. 2007-01-22 13:06:43 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Error out properly when pull_range fails while we're reading the headers, instead of just paus... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull): Error out properly when pull_range fails while we're reading the headers, instead of just pausing the task silently. Fixes #399338. 2007-01-19 13:06:07 +0000 Tim-Philipp Müller gst/smpte/gstsmpte.c: Some more sanity checks to make sure the input formats match and the input pads are actually ne... Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_collected): Some more sanity checks to make sure the input formats match and the input pads are actually negotiated, in case someone tries to feed buffers from fakesrc or filesrc. Fixes #398299. Also const-ify an array, just because we can. 2007-01-19 10:35:13 +0000 Edward Hervey gst/smpte/gstsmpte.c: Ignore previous commit, that was only valid for widths and heights that are multiples of 4. Original commit message from CVS: * gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected): Ignore previous commit, that was only valid for widths and heights that are multiples of 4. Copy over size/stride macros from jpegdec. This allows the element to work with any width,height... ... but puts in evidence that the actual transformations only work with width/height that are multiples of 4. 2007-01-19 09:48:47 +0000 Edward Hervey gst/smpte/gstsmpte.c: Allocate buffers of the right size. Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_collected): Allocate buffers of the right size. The proper size of a I420 buffer in bytes is: width * height * 3 ------------------ 2 2007-01-18 18:37:39 +0000 Tim-Philipp Müller gst/smpte/gstsmpte.c: Proxy getcaps on sink pads too, so that we either end up with the same dimensions on all pads o... Original commit message from CVS: * gst/smpte/gstsmpte.c: (gst_smpte_init): Proxy getcaps on sink pads too, so that we either end up with the same dimensions on all pads or error out if that's not possible (seems to work even!). Fixes #398086, I think. 2007-01-18 11:29:17 +0000 Tim-Philipp Müller docs/plugins/: Remove ladspa from docs; add hierarchy info for GstAudioPanorama; fix integer properties with -1 as mi... Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: Remove ladspa from docs; add hierarchy info for GstAudioPanorama; fix integer properties with -1 as minimum value. * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Update to CVS. 2007-01-18 11:23:36 +0000 Stefan Kost gst/audiofx/audiopanorama.c: Fix doc section name (Fixes #397946) Original commit message from CVS: * gst/audiofx/audiopanorama.c: Fix doc section name (Fixes #397946) 2007-01-18 10:33:50 +0000 Tim-Philipp Müller * ChangeLog: Remove bogus ChangeLog entry Original commit message from CVS: Remove bogus ChangeLog entry 2007-01-17 14:30:50 +0000 Stefan Kost sys/v4l2/: Fix EIO handing when capturing. Add new property to specify the number of buffers to enque (and remove the... Original commit message from CVS: * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_install_properties_helper), (gst_v4l2_object_set_property_helper), (gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_init), (gst_v4l2src_set_property), (gst_v4l2src_get_property), (gst_v4l2src_set_caps): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init), (gst_v4l2src_capture_start), (gst_v4l2src_capture_deinit): Fix EIO handing when capturing. Add new property to specify the number of buffers to enque (and remove the borked num-buffers usage). 2007-01-16 08:29:11 +0000 Sebastian Dröge gst/audiofx/audiopanorama.c: Use a function array for process methods, add more docs and define the startindex of enums. Original commit message from CVS: Patch by: Sebastian Dröge * gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init), (gst_audio_panorama_set_process_function): Use a function array for process methods, add more docs and define the startindex of enums. 2007-01-14 17:55:33 +0000 Mark Nauwelaerts Add support for more than one audio stream; write better AVIX header; refactor code a bit; don't announce vorbis caps... Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/avi/gstavimux.c: (gst_avi_mux_finalize), (gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init), (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps), (gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad), (gst_avi_mux_riff_get_avi_header), (gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header), (gst_avi_mux_write_avix_index), (gst_avi_mux_add_index), (gst_avi_mux_bigfile), (gst_avi_mux_start_file), (gst_avi_mux_stop_file), (gst_avi_mux_handle_event), (gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer), (gst_avi_mux_change_state): * gst/avi/gstavimux.h: * tests/check/elements/avimux.c: (teardown_src_pad): Add support for more than one audio stream; write better AVIX header; refactor code a bit; don't announce vorbis caps on our audio sink pads since we don't support it anyway. Closes #379298. 2007-01-13 19:12:32 +0000 Andy Wingo gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads): Use fixed caps on src pads. Original commit message from CVS: 2007-01-13 Andy Wingo * gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads): Use fixed caps on src pads. (gst_deinterleave_remove_pads): Remove src pads, not sink pads. I seem to have reverse midas disease! (gst_deinterleave_process): Proxy timestamps, offsets, durations, and set caps on outgoing buffers. Fixes #395597, I think. 2007-01-13 18:01:41 +0000 Andy Wingo gst/interleave/interleave.c (gst_interleave_init): Init the activation mode properly. Original commit message from CVS: 2007-01-13 Andy Wingo * gst/interleave/interleave.c (gst_interleave_init): Init the activation mode properly. (gst_interleave_src_setcaps, gst_interleave_src_getcaps) (gst_interleave_init): Set a setcaps and getcaps function on the src pad, so that we can implement pull-mode negotiation. (gst_interleave_sink_setcaps): Renamed from gst_interleave_setcaps, as it only does the sink logic now. Implement both for pull-mode and push-mode. (gst_interleave_process): Set caps on our outgoing buffer. (gst_interleave_src_activate_pull): Fix some more bogus casts. What is up with this. 2007-01-13 15:52:18 +0000 Sebastian Dröge gst/audiofx/audiopanorama.*: Add 'method' property and provide a simple (non-psychoacustic) processing method (#394859). Original commit message from CVS: Patch by: Sebastian Dröge * gst/audiofx/audiopanorama.c: (gst_audio_panorama_method_get_type), (gst_audio_panorama_class_init), (gst_audio_panorama_init), (gst_audio_panorama_set_process_function), (gst_audio_panorama_set_property), (gst_audio_panorama_get_property), (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s_int_simple), (gst_audio_panorama_transform_s2s_int_simple), (gst_audio_panorama_transform_m2s_float_simple), (gst_audio_panorama_transform_s2s_float_simple): * gst/audiofx/audiopanorama.h: Add 'method' property and provide a simple (non-psychoacustic) processing method (#394859). * tests/check/elements/audiopanorama.c: (GST_START_TEST), (panorama_suite): Tests for new method. 2007-01-12 18:28:13 +0000 Christian Schaller * gst-plugins-good.spec.in: comment out LADSPA plugin for now Original commit message from CVS: comment out LADSPA plugin for now 2007-01-12 17:16:51 +0000 Wim Taymans gst/qtdemux/: Add X-QT depayloader that will eventually share code with the demuxer. Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_base_init), (gst_rtp_xqt_depay_class_init), (gst_rtp_xqt_depay_init), (gst_rtp_xqt_depay_finalize), (gst_rtp_quicktime_parse_sd), (gst_rtp_xqt_depay_setcaps), (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_set_property), (gst_rtp_xqt_depay_get_property), (gst_rtp_xqt_depay_change_state), (gst_rtp_xqt_depay_plugin_init): * gst/qtdemux/gstrtpxqtdepay.h: * gst/qtdemux/qtdemux.c: (gst_qtdemux_base_init), (gst_qtdemux_loop_state_header), (gst_qtdemux_loop), (qtdemux_parse_moov), (qtdemux_parse_container), (qtdemux_parse_node), (gst_qtdemux_add_stream), (qtdemux_parse_trak), (qtdemux_audio_caps): * gst/qtdemux/qtdemux.h: * gst/qtdemux/quicktime.c: (plugin_init): Add X-QT depayloader that will eventually share code with the demuxer. Make new plugin entry point with quicktime releated stuff. 2007-01-12 12:10:19 +0000 Tim-Philipp Müller gst/qtdemux/Makefile.am: Dist all new files. Original commit message from CVS: * gst/qtdemux/Makefile.am: Dist all new files. 2007-01-12 10:27:25 +0000 Wim Taymans docs/plugins/: Activate docs for jack, sdl and qtdemux. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-qtdemux.xml: Activate docs for jack, sdl and qtdemux. 2007-01-12 10:22:16 +0000 Wim Taymans gst/qtdemux/: Cleanup and refactor to make the code more readable. Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc), (gst_qtdemux_loop_state_header), (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_sink_activate_pull), (qtdemux_inflate), (qtdemux_parse_moov), (qtdemux_parse_container), (qtdemux_parse_node), (qtdemux_tree_get_child_by_type), (qtdemux_tree_get_sibling_by_type), (gst_qtdemux_add_stream), (qtdemux_parse_samples), (qtdemux_parse_segments), (qtdemux_parse_trak), (qtdemux_tag_add_str), (qtdemux_tag_add_num), (qtdemux_tag_add_date), (qtdemux_tag_add_gnre), (qtdemux_parse_udta), (qtdemux_redirects_sort_func), (qtdemux_process_redirects), (qtdemux_parse_redirects), (qtdemux_parse_tree), (gst_qtdemux_handle_esds), (qtdemux_video_caps), (qtdemux_audio_caps): * gst/qtdemux/qtdemux.h: * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mvhd), (qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd), (qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref), (qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss), (qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco), (qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd), (qtdemux_dump_unknown), (qtdemux_node_dump_foreach), (qtdemux_node_dump): * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: (qtdemux_type_get): * gst/qtdemux/qtdemux_types.h: * gst/qtdemux/qtpalette.h: Cleanup and refactor to make the code more readable. Move debugging/tables into separate files. Add 2/4/16 color palletee support. Fix raw 15 bit RGB handling. Use more FOURCC constants. Add some docs. 2007-01-11 19:51:04 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.c: Minor clean-up: use enum values instead of hardcoded constants (#395536). Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type), (gst_wavpack_enc_correction_mode_get_type), (gst_wavpack_enc_joint_stereo_mode_get_type): Minor clean-up: use enum values instead of hardcoded constants (#395536). 2007-01-11 16:59:40 +0000 Tim-Philipp Müller gst/: Set correct caps on outgoing pulled buffers, or things blow up after recent core changes. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range): * gst/id3demux/gstid3demux.c: (gst_id3demux_read_range): Set correct caps on outgoing pulled buffers, or things blow up after recent core changes. 2007-01-11 11:05:04 +0000 Jonas Holmberg gst/multipart/multipartmux.c: Return FLOW errors ASAP. Fixes #394977. Original commit message from CVS: Based on patch by: Jonas Holmberg * gst/multipart/multipartmux.c: (gst_multipart_mux_init), (gst_multipart_mux_request_new_pad), (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected), (gst_multipart_mux_change_state): Return FLOW errors ASAP. Fixes #394977. Misc cleanups. 2007-01-11 09:30:59 +0000 Lutz Mueller gst/rtsp/gstrtspsrc.c: Check for stream pad before activating. Original commit message from CVS: Patch by: Lutz Mueller * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams): Check for stream pad before activating. 2007-01-10 15:19:48 +0000 Peter Kjellerstedt gst/rtsp/: Allow url to be NULL to be able to use it for server connections. Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/rtsp/COPYING.MIT: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_send), (read_line), (parse_request_line), (parse_line), (rtsp_connection_read), (rtsp_connection_close): * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult), (rtsp_method_as_text), (rtsp_header_as_text), (rtsp_status_as_text), (rtsp_find_header_field), (rtsp_find_method): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send), (rtsp_ext_wms_configure_stream): * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init), (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_get_body), (dump_mem), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_media_get_attribute_val_n), (read_string), (read_string_del), (sdp_parse_line), (sdp_message_parse_buffer), (print_media), (sdp_message_dump): Allow url to be NULL to be able to use it for server connections. Can now send responses as well as requests. No longer hangs in an endless loop if EOF is received. Can now convert a status code to a text string. Return RTSP_HDR_INVALID for unknown headers. Return RTSP_INVALID for unknown methods. Copy CSeq and Session headers from the request. Only free memory corresponding to the currently set message type. Added const to function arguments as appropriate. Avoid a compiler warning when initializing nmedia. Use guint rather than gint to avoid compiler warnings. Fix crasher in wms extension. Factor out stream setup from open_connection. Delay activation of streams when actual data is received from the server, this prepares us to do proper protocol switching. Added new license. Fixes #380895. 2007-01-10 09:47:43 +0000 Sebastian Dröge Some small docs fixes (#394851). Original commit message from CVS: Patch by: Sebastian Dröge * docs/plugins/Makefile.am: * gst/audiofx/audiopanorama.c: Some small docs fixes (#394851). 2007-01-09 12:25:26 +0000 Wim Taymans gst/avi/gstavidemux.c: Fix docs. Original commit message from CVS: * gst/avi/gstavidemux.c: Fix docs. 2007-01-09 12:23:48 +0000 Wim Taymans gst/rtp/: Added RFC 2250 MPEG Video Depayloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init), (gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process), (gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property), (gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init): * gst/rtp/gstrtpmpvdepay.h: Added RFC 2250 MPEG Video Depayloader. * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps), (gst_rtp_h263p_depay_process): Fix Header file. Small cleanups. * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize), (gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init), (gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize), (gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process), (gst_rtp_mp4v_depay_change_state): Remove usused code. Remove Adapter from state Change. Added debug. * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init), (gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): * gst/rtp/gstrtpmpadepay.h: Subclass base depayloader. Added debug. Support static payload type assignment as well. * gst/rtp/gstrtpmpapay.c: Fix caps. 2007-01-08 12:45:10 +0000 Vincent Torri ext/jpeg/: These libjpeg callbacks should return a 'boolean' (unsigned char apparently) and not a 'gboolean' (which m... Original commit message from CVS: Patch by: Vincent Torri * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/smokecodec.c: These libjpeg callbacks should return a 'boolean' (unsigned char apparently) and not a 'gboolean' (which maps to gint). Fixes warnings when compiling with MingW (#393427). * gst/rtsp/rtspconnection.c: (rtsp_connection_read): Use ioctlsocket on win32. * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Some printf format fixes for win32. 2007-01-07 22:03:54 +0000 Andy Wingo New elements interleave and deinterleave, implement channel interleaving and deinterleaving. Original commit message from CVS: 2007-01-07 Andy Wingo * configure.ac: * gst/interleave/Makefile.am: * gst/interleave/plugin.h: * gst/interleave/plugin.c: * gst/interleave/interleave.c: * gst/interleave/deinterleave.c: New elements interleave and deinterleave, implement channel interleaving and deinterleaving. The interleaver can operate in pull or push mode but the deinterleaver is more like a demuxer and can only operate in push mode. 2007-01-07 10:44:12 +0000 Sébastien Moutte gst/cutter/gstcutter.c: Use gst_guint64_to_gdouble for conversion. Original commit message from CVS: * gst/cutter/gstcutter.c: (gst_cutter_chain): Use gst_guint64_to_gdouble for conversion. * win32/vs6/libgstmatroska.dsp: Add zlib to the link. * win32/vs6/libgstvideobox.dsp: Update liboil library name (project is linked to liboil-0.3-0.lib now). 2007-01-05 18:32:03 +0000 Tim-Philipp Müller Check for zlib and if available pass it explicitly to the linker when linking qtdemux. If not available (or --disable... Original commit message from CVS: * configure.ac: * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: (qtdemux_parse_moov): Check for zlib and if available pass it explicitly to the linker when linking qtdemux. If not available (or --disable-external has been specified!), disable the bits in qtdemux that use it. Fixes build on MingW (#392856). 2007-01-05 17:23:04 +0000 Tim-Philipp Müller gst/matroska/Makefile.am: If zlib is available and used, we must link it explicitly for things to work on MingW (fixe... Original commit message from CVS: * gst/matroska/Makefile.am: If zlib is available and used, we must link it explicitly for things to work on MingW (fixes #392855). 2007-01-05 16:07:12 +0000 Tim-Philipp Müller tests/icles/videocrop-test.c: Call g_thread_init() right at the beginning. Remove superfluous gst_init() - we've alre... Original commit message from CVS: * tests/icles/videocrop-test.c: (main): Call g_thread_init() right at the beginning. Remove superfluous gst_init() - we've already been inited via the GOption stuff. 2007-01-04 11:02:29 +0000 Tim-Philipp Müller ext/esd/esdsink.c: Don't return bogus values when esd_get_delay() fails for some reason (#392189). Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_delay): Don't return bogus values when esd_get_delay() fails for some reason (#392189). 2007-01-04 09:44:57 +0000 Vincent Torri Add directsoundsink to build and dist it, so it gets built when compiling with MingW on win32 and the required header... Original commit message from CVS: Patch by: Vincent Torri * configure.ac: * sys/Makefile.am: * sys/directsound/Makefile.am: * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_reset): Add directsoundsink to build and dist it, so it gets built when compiling with MingW on win32 and the required headers and libraries are available (fixes: #392638). Also simplify DirectDraw check a bit. * tests/check/elements/.cvsignore: Fix CVS ignore for neonhttpsrc test binary. 2007-01-03 19:54:33 +0000 Vincent Torri Add directdrawsink to build and dist it, so it gets built when compiling with MingW on win32 and the required headers... Original commit message from CVS: Patch by: Vincent Torri * configure.ac: * sys/Makefile.am: * sys/directdraw/Makefile.am: Add directdrawsink to build and dist it, so it gets built when compiling with MingW on win32 and the required headers and libraries are available (fixes: #392313). * sys/directdraw/gstdirectdrawsink.c: (gst_directdrawsink_center_rect), (gst_directdrawsink_show_frame), (gst_directdrawsink_setup_ddraw), (gst_directdrawsink_surface_create): Comment out some unused things and fix some printf format issues in order to avoid warnings when buildling with MingW (#392313). 2007-01-03 16:41:10 +0000 Jens Granseuer Fix build with gcc-2.x (declare variables at the beginning of a block etc.). Fixes #391971. Original commit message from CVS: Patch by: Jens Granseuer * ext/xvid/gstxvidenc.c: (gst_xvidenc_encode), (gst_xvidenc_get_property): * gst/filter/gstbpwsinc.c: (bpwsinc_transform_ip): * gst/filter/gstfilter.c: (plugin_init): * gst/filter/gstiir.c: (iir_transform_ip): * gst/filter/gstlpwsinc.c: (lpwsinc_transform_ip): * gst/modplug/gstmodplug.cc: * gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_header_load), (gst_nuv_demux_stream_extend_header): Fix build with gcc-2.x (declare variables at the beginning of a block etc.). Fixes #391971. 2006-12-30 20:01:35 +0000 Thomas Vander Stichele ext/lame/gstlame.c: warn when outgoing sample rate is different from incoming Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_setcaps), (gst_lame_chain): warn when outgoing sample rate is different from incoming 2006-12-30 12:44:01 +0000 Tim-Philipp Müller tests/check/elements/videocrop.c: When we can't create an element needed for the test, print a message detailing whic... Original commit message from CVS: * tests/check/elements/videocrop.c: (GST_START_TEST), (videocrop_test_cropping_init_context): When we can't create an element needed for the test, print a message detailing which element it actually is that's missing (#390673). 2006-12-24 11:36:31 +0000 Tim-Philipp Müller sys/ximage/gstximagesrc.c: Fix presumably copy'n'pasto for 16bpp depth. Original commit message from CVS: * sys/ximage/gstximagesrc.c: (composite_pixel): Fix presumably copy'n'pasto for 16bpp depth. 2006-12-24 11:24:59 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: The "signed" field in audio caps is of boolean type, trying to use gst_structure_get_int... Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_setcaps): The "signed" field in audio caps is of boolean type, trying to use gst_structure_get_int() to extract it will fail. Fixing this makes matroskamux accept raw audio input (#387121) (use at your own risk though, due to the matroska spec being not entirely useful in this respect). Also fix up raw audio structures in template caps so that they represent what our setcaps function will actually accept, so that converters know what to convert to. Finally, don't fail if there isn't an "endianness" field in 8-bit PCM caps. 2006-12-22 10:15:24 +0000 Stefan Kost tests/check/elements/: reapply consistent pad (de)activation Original commit message from CVS: * tests/check/elements/mpeg2enc.c: (setup_mpeg2enc), (cleanup_mpeg2enc): * tests/check/elements/rganalysis.c: (cleanup_rganalysis): * tests/check/elements/wavpackdec.c: (setup_wavpackdec), (cleanup_wavpackdec): * tests/check/elements/wavpackenc.c: (setup_wavpackenc), (cleanup_wavpackenc): * tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc): reapply consistent pad (de)activation 2006-12-22 10:15:23 +0000 Stefan Kost tests/check/elements/: reapply consistent pad (de)activation Original commit message from CVS: * tests/check/elements/audiopanorama.c: (cleanup_panorama): * tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux): * tests/check/elements/cmmldec.c: (setup_cmmldec), (teardown_cmmldec): * tests/check/elements/cmmlenc.c: (setup_cmmlenc), (teardown_cmmlenc): * tests/check/elements/level.c: (setup_level), (cleanup_level): reapply consistent pad (de)activation 2006-12-21 17:03:39 +0000 Jan Schmidt configure.ac: Back to CVS Original commit message from CVS: * configure.ac: Back to CVS * gst-plugins-good.doap: Add 0.10.5 doap entry === release 0.10.4 === 2006-12-21 15:45:02 +0000 Jan Schmidt configure.ac: releasing 0.10.4, "Black Bugs" Original commit message from CVS: === release 0.10.4 === 2006-12-21 Jan Schmidt * configure.ac: releasing 0.10.4, "Black Bugs" === release 0.10.5 === 2006-12-21 15:40:55 +0000 Jan Schmidt configure.ac: releasing 0.10.5, "The Path of Thorns" Original commit message from CVS: === release 0.10.5 === 2006-12-21 Jan Schmidt * configure.ac: releasing 0.10.5, "The Path of Thorns" 2006-12-21 14:03:42 +0000 Stefan Kost tests/check/elements/mpeg2enc.c: (setup_mpeg2enc) Original commit message from CVS: * tests/check/elements/mpeg2enc.c: (setup_mpeg2enc) (cleanup_mpeg2enc): * tests/check/elements/rganalysis.c: (cleanup_rganalysis): * tests/check/elements/wavpackdec.c: (setup_wavpackdec), (cleanup_wavpackdec): * tests/check/elements/wavpackenc.c: (setup_wavpackenc), (cleanup_wavpackenc): * tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc): revert my freeze breakage 2006-12-21 12:48:32 +0000 Stefan Kost tests/check/elements/: revert my freeze breakage Original commit message from CVS: * tests/check/elements/audiopanorama.c: (cleanup_panorama): * tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux): * tests/check/elements/cmmldec.c: (setup_cmmldec), (teardown_cmmldec): * tests/check/elements/cmmlenc.c: (setup_cmmlenc), (teardown_cmmlenc): * tests/check/elements/level.c: (setup_level), (cleanup_level): revert my freeze breakage 2006-12-21 08:20:10 +0000 Stefan Kost tests/check/elements/: consistent pad (de)activation Original commit message from CVS: * tests/check/elements/mpeg2enc.c: (setup_mpeg2enc), (cleanup_mpeg2enc): * tests/check/elements/rganalysis.c: (cleanup_rganalysis): * tests/check/elements/wavpackdec.c: (setup_wavpackdec), (cleanup_wavpackdec): * tests/check/elements/wavpackenc.c: (setup_wavpackenc), (cleanup_wavpackenc): * tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc): consistent pad (de)activation 2006-12-21 08:15:23 +0000 Stefan Kost tests/check/elements/: consistent pad (de)activation Original commit message from CVS: * tests/check/elements/audiopanorama.c: (cleanup_panorama): * tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux): * tests/check/elements/cmmldec.c: (setup_cmmldec), (teardown_cmmldec): * tests/check/elements/cmmlenc.c: (setup_cmmlenc), (teardown_cmmlenc): * tests/check/elements/level.c: (setup_level), (cleanup_level): consistent pad (de)activation 2006-12-18 17:11:49 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Don't post BUFFERING messages in streaming mode if the stream headers are behind the movie dat... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_post_progress), (gst_qtdemux_chain): Don't post BUFFERING messages in streaming mode if the stream headers are behind the movie data; instead, post "progress" element messages as a temporary solution. Apps might get confused and do silly things to the pipeline state if they see buffering messages from different sources and don't realize they come from different sources (#387160). 2006-12-18 16:46:17 +0000 Jan Schmidt Disable LADPSA, as it has moved to the -bad module for the duration. Original commit message from CVS: * configure.ac: * ext/Makefile.am: Disable LADPSA, as it has moved to the -bad module for the duration. 2006-12-18 15:51:54 +0000 Wim Taymans ext/ladspa/gstsignalprocessor.c: Reset flow_state back to _OK after a flush stop so that we exit our error state afte... Original commit message from CVS: * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps), (gst_signal_processor_event): Reset flow_state back to _OK after a flush stop so that we exit our error state after the flush. Fixes #374213 2006-12-18 15:49:08 +0000 Jan Schmidt ChangeLog surgery on one of Stefan's commits from August: Original commit message from CVS: ChangeLog surgery on one of Stefan's commits from August: * ext/Makefile.am: Quietly (accidentally) enable LADSPA for building by default, despite the fact that it doesn't meet the plugin checklist. -- Added by Jan Schmidt 18 Dec 2006 2006-12-18 13:40:34 +0000 Jan Schmidt gst/qtdemux/qtdemux.c: Don't output g_warning for an unsupported format, just send a Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_chain), (gst_qtdemux_add_stream): Don't output g_warning for an unsupported format, just send a GST_ELEMENT_WARNING and don't add the pad. Fix the case where it doesn't check for a NULL pad in streaming mode. Fixes #387137 2006-12-18 12:27:32 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix crash dereferencing NULL pointer if there's no stco atom. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Fix crash dereferencing NULL pointer if there's no stco atom. Fixes #387122. 2006-12-18 10:02:56 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.h: Use local copy of md5.h, as it disappeared in recent wavpack installs. Original commit message from CVS: * ext/wavpack/gstwavpackenc.h: Use local copy of md5.h, as it disappeared in recent wavpack installs. Patch by: Sebastian Dröge Fixes: #387076 2006-12-17 19:42:05 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2006-12-17 06:11:39 +0000 David Schleef sys/osxvideo/osxvideosink.*: Decent effort at porting to 0.10. Needs cleanup on OS/X. Original commit message from CVS: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Decent effort at porting to 0.10. Needs cleanup on OS/X. 2006-12-17 05:07:07 +0000 Vijay Santhanam sys/osxvideo/: Preliminary patch for porting osxvideosink Original commit message from CVS: Patch by: Vijay Santhanam * sys/osxvideo/Makefile.am: * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: Preliminary patch for porting osxvideosink 2006-12-16 16:21:26 +0000 Sjoerd Simons gst/videomixer/videomixer.c: Introduce some locking around the videomixer state so that it does not crash when adding... Original commit message from CVS: Patch by: Sjoerd Simons * gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property), (gst_videomixer_set_master_geometry), (gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free), (gst_videomixer_reset), (gst_videomixer_init), (gst_videomixer_finalize), (gst_videomixer_request_new_pad), (gst_videomixer_release_pad), (gst_videomixer_collected), (gst_videomixer_change_state): Introduce some locking around the videomixer state so that it does not crash when adding/removing pads. Fixes #383043. 2006-12-16 15:25:23 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: We don't support seeking in streaming mode, so don't even try. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types), (gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event): We don't support seeking in streaming mode, so don't even try. Implement seeking query so apps can query seekability properly (see #365414). Fix duration query. 2006-12-16 11:42:56 +0000 Tim-Philipp Müller configure.ac: Make sure libcaca can actually be used instead of just checking for /usr/bin/caca-config, so we don't w... Original commit message from CVS: * configure.ac: Make sure libcaca can actually be used instead of just checking for /usr/bin/caca-config, so we don't wrongly try to build cacasink when cross-compiling (fixes #384587). 2006-12-15 10:54:28 +0000 Thomas Vander Stichele adding doap file Original commit message from CVS: * Makefile.am: * gst-plugins-good.doap: * gst-plugins-good.spec.in: adding doap file 2006-12-14 16:20:15 +0000 Tim-Philipp Müller configure.ac: libflac-1.1.3 changed API again, but we can't build against it yet, so make sure our check doesn't use ... Original commit message from CVS: * configure.ac: libflac-1.1.3 changed API again, but we can't build against it yet, so make sure our check doesn't use libflac-1.1.3 and add a comment to this effect. 2006-12-14 14:25:17 +0000 Tim-Philipp Müller gst/effectv/gstquark.c: Add some NULL pointer checks (possibly related to #385623). Original commit message from CVS: * gst/effectv/gstquark.c: (gst_quarktv_transform), (gst_quarktv_planetable_clear): Add some NULL pointer checks (possibly related to #385623). 2006-12-14 10:15:24 +0000 Roland Kay ext/lame/gstlame.*: Fix leak (by calling lame_init_params() before lame_close()); handle Original commit message from CVS: Based on patch by: Roland Kay * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain), (gst_lame_setup): * ext/lame/gstlame.h: Fix leak (by calling lame_init_params() before lame_close()); handle NULL return from lame_init() more gracefully. Fixes #385311. 2006-12-13 17:12:22 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Add AMR-WB to the list of supported formats. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (gst_qtdemux_handle_src_query), (qtdemux_parse_trak), (qtdemux_audio_caps): Add AMR-WB to the list of supported formats. 2006-12-12 18:45:58 +0000 Tim-Philipp Müller gst/: In streaming mode, if the first buffer we get doesn't have an offset, fix it up to be 0, otherwise trimming won... Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag), (gst_tag_demux_chain): * gst/id3demux/gstid3demux.c: (gst_id3demux_chain): In streaming mode, if the first buffer we get doesn't have an offset, fix it up to be 0, otherwise trimming won't work later on and we'll be typefinding application/x-id3, which may result in decodebin plugging an endless number of id3demux elements as a consequence. Fixes #385031. 2006-12-11 21:21:16 +0000 Jan Schmidt sys/sunaudio/gstsunaudiosink.c: Ignore the buffer_time the sound device reports. Turns out it is sometimes completely... Original commit message from CVS: * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare): Ignore the buffer_time the sound device reports. Turns out it is sometimes completely bogus and we're better off without it. 2006-12-11 17:33:26 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix non-working redirects from inetfilm.com (handle 'alis' reference data type as well). Fixes... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree): Fix non-working redirects from inetfilm.com (handle 'alis' reference data type as well). Fixes #378613. 2006-12-11 13:59:33 +0000 Tim-Philipp Müller gst/matroska/: Try harder to extract the framerate for video tracks correctly and save it directly instead of convert... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_video_caps): * gst/matroska/matroska-ids.c: (gst_matroska_track_init_video_context): * gst/matroska/matroska-ids.h: Try harder to extract the framerate for video tracks correctly and save it directly instead of converting it back and forth a few times. Mostly makes a difference for very small framerates (<1). Fixes #380199. 2006-12-11 11:41:18 +0000 Tim-Philipp Müller ext/gconf/gstgconfaudiosrc.*: Remove gconf notify hook when the gconfaudiosrc element is destroyed, otherwise the cal... Original commit message from CVS: * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose), (do_toggle_element): * ext/gconf/gstgconfaudiosrc.h: Remove gconf notify hook when the gconfaudiosrc element is destroyed, otherwise the callback may be called on an already-destroyed instance and bad things happen. Should fix #378184. Also ignore gconf key changes when the source is already running. 2006-12-09 19:27:28 +0000 Sebastian Dröge gst/apetag/gstapedemux.c: We need to be able to read and parse any possible floating point string format ("1,234" or ... Original commit message from CVS: Patch by: Sebastian Dröge * gst/apetag/gstapedemux.c: (ape_demux_parse_tags): We need to be able to read and parse any possible floating point string format ("1,234" or "1.234") irrespective of the current locale. g_strod() will parse the former only in certain locales though, so we really need to canonicalise the separator to '.' and then use g_ascii_strtod() to make sure we can parse either version at all times. Fixes #382982 for real. 2006-12-09 16:17:33 +0000 Jan Schmidt sys/sunaudio/: Use the sunaudio debug category. Original commit message from CVS: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiosrc.c: Use the sunaudio debug category. * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize), (gst_sunaudiosink_class_init), (gst_sunaudiosink_init), (gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property), (gst_sunaudiosink_open), (gst_sunaudiosink_close), (gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay), (gst_sunaudiosink_write), (gst_sunaudiosink_delay), (gst_sunaudiosink_reset): * sys/sunaudio/gstsunaudiosink.h: Uses the sunaudio debug category for all debug output Implements the _delay() callback to synchronise video playback better Change the segtotal and segsize values back to the parent class defaults (taken from buffer_time and latency_times of 200ms and 10ms respectively) Measure the samples written to the device vs. played. Keep track of segments in the device by writing empty eof frames, and sleep using a GCond when we get too far ahead and risk overrunning the sink's ringbuffer. Fixes: #360673 2006-12-08 21:12:47 +0000 Jan Schmidt * ChangeLog: Correct the attribution of the previous commit. The patch in question was written by Brian Cameron. Original commit message from CVS: Correct the attribution of the previous commit. The patch in question was written by Brian Cameron. 2006-12-08 17:06:43 +0000 René Stadler gst/qtdemux/qtdemux.c: Fix caps for 24 bit raw PCM audio (2). Original commit message from CVS: Patch by: René Stadler * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (gst_qtdemux_handle_src_query), (qtdemux_parse_trak), (qtdemux_audio_caps): Fix caps for 24 bit raw PCM audio (2). Fixes #383471. 2006-12-08 16:38:18 +0000 Sebastian Dröge gst/audiofx/audiopanorama.*: Fix audiopanorame with float samples. Fixes #383726. Original commit message from CVS: Patch by: Sebastian Dröge * gst/audiofx/audiopanorama.c: (gst_audio_panorama_init), (gst_audio_panorama_set_caps), (gst_audio_panorama_transform): * gst/audiofx/audiopanorama.h: Fix audiopanorame with float samples. Fixes #383726. 2006-12-08 15:12:01 +0000 Padraig O'Briain sys/sunaudio/: Implement reset functions to unblock the src/sink more quickly on state change requests. Original commit message from CVS: * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_reset): * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open), (gst_sunaudiosrc_reset): Implement reset functions to unblock the src/sink more quickly on state change requests. Patch by: Padraig O'Briain 2006-12-08 14:42:42 +0000 Jerry Tan sys/sunaudio/gstsunaudiomixer.c: Construct the correct mixer device name when the AUDIODEV env var is set. Original commit message from CVS: * sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_change_state): Construct the correct mixer device name when the AUDIODEV env var is set. Patch by: Jerry Tan Fixes: #383596 2006-12-08 14:32:51 +0000 Jerry Tan sys/sunaudio/gstsunaudiosrc.c: Apply patch to open the mixer control and set the MULTIPLE_OPEN ioctl. On solaris, the... Original commit message from CVS: * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open): Apply patch to open the mixer control and set the MULTIPLE_OPEN ioctl. On solaris, the mixer device doesn't need opening non-blocking - it can be opened by multiple processes by default, but needs the ioctl for multiple opens within 1 process. Patch by: Jerry Tan Fixes: #349015 2006-12-07 17:30:03 +0000 Wim Taymans gst/smpte/: Port to 0.10 some more. Original commit message from CVS: * gst/smpte/gstmask.h: * gst/smpte/gstsmpte.c: (gst_smpte_class_init), (gst_smpte_setcaps), (gst_smpte_init), (gst_smpte_reset), (gst_smpte_collected), (gst_smpte_set_property), (gst_smpte_get_property), (gst_smpte_change_state), (plugin_init): * gst/smpte/gstsmpte.h: Port to 0.10 some more. Added duration property to specify the duration of the transition. Make framerate a fraction. Deprecate fps property, we only use negotiated fps. Added docs. Fix collectpad usage. Reset state in READY. Send NEWSEGMENT event. Fix racy updates of object properties. Added debug category. Fixes #383323. 2006-12-07 11:35:41 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Handle more H263 variants. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (gst_qtdemux_handle_src_query), (qtdemux_parse_trak), (qtdemux_video_caps): Handle more H263 variants. 2006-12-06 15:06:04 +0000 Sjoerd Simons gst/videomixer/videomixer.c: Don't reset xpos and ypos in the setcaps function because causes unexpected behaviour. Original commit message from CVS: Patch by: Sjoerd Simons * gst/videomixer/videomixer.c: (gst_videomixer_set_master_geometry), (gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free): Don't reset xpos and ypos in the setcaps function because causes unexpected behaviour. Fixes #382179. 2006-12-06 14:45:30 +0000 Wim Taymans gst/multipart/multipartmux.c: Keep track of the buffer timestamp in the collectdata member instead of modifying the b... Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_compare_pads), (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected): Keep track of the buffer timestamp in the collectdata member instead of modifying the buffer without making the metadata writable first. Fixes #382277. 2006-12-06 14:33:54 +0000 Rob Taylor gst/udp/gstudpsrc.c: If using multicast in udpsrc, bind to the multicast address rather than Original commit message from CVS: Patch by: Rob Taylor * gst/udp/gstudpsrc.c: (gst_udpsrc_start): If using multicast in udpsrc, bind to the multicast address rather than IN_ADDR_ANY. This allows the simultanous use of multiple udpsrcs listening on different multicat addresses. Without this all udpsrcs will receive all packets from all subscribed multicast addresses. Fixes #383001. 2006-12-06 13:35:52 +0000 Jonathan Matthew ext/taglib/gstid3v2mux.cc: Don't attempt to write a NULL frame into the ID3 tag set when the createFrame method retur... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Don't attempt to write a NULL frame into the ID3 tag set when the createFrame method returned NULL. Fixes: #381857 Patch by: Jonathan Matthew 2006-12-06 13:16:59 +0000 Sebastian Dröge gst/apetag/gstapedemux.c: Use g_strtod() instead of sscanf to parse doubles, so that it will try parsing in the C loc... Original commit message from CVS: * gst/apetag/gstapedemux.c: (ape_demux_parse_tags): Use g_strtod() instead of sscanf to parse doubles, so that it will try parsing in the C locale if the current locale fails. Fixes: #382982 Patch by: Sebastian Dröge 2006-12-01 10:31:46 +0000 Sergey Scobich win32/MANIFEST: Fix compilation on win32 under VS8 Original commit message from CVS: * win32/MANIFEST: Fix compilation on win32 under VS8 Patch by: Sergey Scobich Partially fixes #381175 2006-11-30 16:48:51 +0000 Stefan Kost gst/avi/gstavimux.c: accept all mpegversions,fixes #380825 spotted by: Jerome Alet Original commit message from CVS: * gst/avi/gstavimux.c: accept all mpegversions,fixes #380825 spotted by: Jerome Alet 2006-11-30 16:46:13 +0000 Stefan Kost sys/v4l2/v4l2src_calls.c: cleanup the error message a bit more Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame), (gst_v4l2src_get_capture), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init), (gst_v4l2src_buffer_finalize): cleanup the error message a bit more 2006-11-30 15:08:08 +0000 René Stadler gst/replaygain/gstrganalysis.c: Call the base class handler. Fixes #380610. Original commit message from CVS: Patch by: René Stadler * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_event): Call the base class handler. Fixes #380610. 2006-11-28 12:30:10 +0000 Wim Taymans ext/libcaca/gstcacasink.c: Fix width and height properties. Original commit message from CVS: * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init): Fix width and height properties. * ext/libcaca/gstcacasink.h: Fix compilation on newer libcaca that require us to include a new header. Fixes #379918. 2006-11-28 11:52:27 +0000 Wim Taymans gst/rtsp/: Add method so that extensions can choose to disable the setup of a stream. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspext.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream), (rtsp_ext_wms_get_context): Add method so that extensions can choose to disable the setup of a stream. Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792. 2006-11-27 17:16:26 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Remove some asserts and replace them with a proper error message. Fixes #379261. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (gst_qtdemux_handle_src_query), (qtdemux_parse_trak): Remove some asserts and replace them with a proper error message. Fixes #379261. 2006-11-27 16:30:49 +0000 Wim Taymans * ChangeLog: mention bug fix Original commit message from CVS: mention bug fix 2006-11-27 16:29:07 +0000 Jonas Holmberg gst/multipart/multipartmux.c: Push header in a separate buffer instead of memcpy:ing all data Original commit message from CVS: Patch by: Jonas Holmberg * gst/multipart/multipartmux.c: (gst_multipart_mux_collected): Push header in a separate buffer instead of memcpy:ing all data Change LF => CRLF in headers Move trailing LF to header 2006-11-27 16:26:50 +0000 Wim Taymans gst/rtp/gstrtpmpadepay.c: Small buffer overflow fix and improve debugging. Original commit message from CVS: * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_chain): Small buffer overflow fix and improve debugging. 2006-11-24 08:58:53 +0000 Stefan Kost ext/esd/: remove obsolete _factory_init protos Original commit message from CVS: * ext/esd/esdmon.h: * ext/esd/esdsink.h: remove obsolete _factory_init protos 2006-11-24 07:46:54 +0000 Stefan Kost gst/avi/gstavidemux.c: remove dead code, tweak debugs statements, add comments, use _uint64_scale instead _uint64_sca... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_peek_chunk), (gst_avi_demux_parse_subindex), (gst_avi_demux_read_subindexes_push), (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_next_data_buffer), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): remove dead code, tweak debugs statements, add comments, use _uint64_scale instead _uint64_scale_int when using guint64 values, small optimizations, reflow some error handling 2006-11-22 17:39:13 +0000 Edward Hervey po/.cvsignore: We never put .pot files in cvs. Let's ignore them all. Original commit message from CVS: * po/.cvsignore: We never put .pot files in cvs. Let's ignore them all. 2006-11-21 12:57:50 +0000 Christian Schaller * gst-plugins-good.spec.in: enalbe LADSPA plugin in spec file Original commit message from CVS: enalbe LADSPA plugin in spec file 2006-11-19 18:46:03 +0000 Tim-Philipp Müller po/POTFILES.in: ... but better exclude files that aren't disted. Original commit message from CVS: * po/POTFILES.in: ... but better exclude files that aren't disted. 2006-11-19 16:32:49 +0000 Tim-Philipp Müller po/POTFILES.in: Add v4l2 source files to list of files with translations, so the strings are actually extracted (howe... Original commit message from CVS: * po/POTFILES.in: Add v4l2 source files to list of files with translations, so the strings are actually extracted (however bad they still may be). 2006-11-19 16:30:19 +0000 Tim-Philipp Müller gst/videobox/gstvideobox.c: Minor clean-ups: const-ify static array, remove trailing comma from use GST_DEBUG_FUNCPTR. Original commit message from CVS: * gst/videobox/gstvideobox.c: (gst_video_box_class_init): Minor clean-ups: const-ify static array, remove trailing comma from last enum (gcc-2.9x trips over that), use GST_DEBUG_FUNCPTR. 2006-11-19 13:41:53 +0000 René Stadler gst/id3demux/id3v2frames.c: Make sure that g_free always gets called on the same pointer that was returned by g_mallo... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame): Make sure that g_free always gets called on the same pointer that was returned by g_malloc. Fixes #376594. Do not leak memory if decompressed size is wrong. Remove unneeded check of return value of g_malloc. Patch by: René Stadler 2006-11-18 18:14:34 +0000 Tim-Philipp Müller sys/v4l2/v4l2src_calls.c: Add missing curly brackets. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_deinit): Add missing curly brackets. 2006-11-17 14:54:01 +0000 Edgard Lima * ChangeLog: * sys/v4l2/v4l2src_calls.c: Fix capture_deinit. Original commit message from CVS: Fix capture_deinit. 2006-11-16 15:36:48 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: Use GST_DEBUG_FUNCPTR; activate request pad before returning it. Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init), (gst_matroska_mux_request_new_pad): Use GST_DEBUG_FUNCPTR; activate request pad before returning it. * tests/check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad), (GST_START_TEST): Activate pads before using them. 2006-11-16 15:04:55 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Initialise variable to get rid of bogus compiler warning. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan): Initialise variable to get rid of bogus compiler warning. 2006-11-16 07:26:17 +0000 Ville Syrjala gst/rtp/: Specify H.263 variant and version in the caps (fixes #361637) Original commit message from CVS: Patch by: Ville Syrjala * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: Specify H.263 variant and version in the caps (fixes #361637) 2006-11-15 17:44:01 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Don't set a data pointer to NULL and a size > 0 when we deal with empty packets. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (read_body): Don't set a data pointer to NULL and a size > 0 when we deal with empty packets. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_take_body): Check that we can't create invalid empty packets. 2006-11-15 12:35:46 +0000 Sebastian Dröge ext/wavpack/: Some small clean-ups: use enums instead of hard-coded numbers, const-ify element details, re-factor som... Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_init), (gst_wavpack_dec_change_state): * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init), (gst_wavpack_enc_class_init), (gst_wavpack_enc_reset), (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_change_state): * ext/wavpack/gstwavpackparse.c: Some small clean-ups: use enums instead of hard-coded numbers, const-ify element details, re-factor some code into _reset() functions (#352605). 2006-11-15 12:08:20 +0000 Mark Nauwelaerts gst/matroska/matroska-mux.*: Add basic tag writing support; implement releasing pads (#374658). Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/matroska/matroska-mux.c: (gst_matroska_mux_add_interfaces), (gst_matroska_mux_class_init), (gst_matroska_pad_free), (gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event), (gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad), (gst_matroska_mux_track_header), (gst_matroska_mux_start), (gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish): * gst/matroska/matroska-mux.h: Add basic tag writing support; implement releasing pads (#374658). 2006-11-15 11:19:13 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Handle opaque/unspecified A_AAC audio codec ID (fixes #374737). Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_audio_caps): Handle opaque/unspecified A_AAC audio codec ID (fixes #374737). 2006-11-15 00:12:19 +0000 David Schleef gst/matroska/matroska-mux.c: Add Dirac fourcc. Original commit message from CVS: * gst/matroska/matroska-mux.c: Add Dirac fourcc. 2006-11-14 20:07:22 +0000 Sergey Scobich win32/vs8/: Make end-of-line returns unixy, so that when the files are checked out on win32 the line returns will be ... Original commit message from CVS: Patch by: Sergey Scobich * win32/vs8/gst-plugins-good.sln: * win32/vs8/libgst1394.vcproj: * win32/vs8/libgstaasink.vcproj: * win32/vs8/libgstalaw.vcproj: * win32/vs8/libgstalpha.vcproj: * win32/vs8/libgstalphacolor.vcproj: * win32/vs8/libgstannodex.vcproj: * win32/vs8/libgstapetag.vcproj: * win32/vs8/libgstaudiofx.vcproj: * win32/vs8/libgstauparse.vcproj: * win32/vs8/libgstautodetect.vcproj: * win32/vs8/libgstavi.vcproj: * win32/vs8/libgstcacasink.vcproj: * win32/vs8/libgstcdio.vcproj: * win32/vs8/libgstcutter.vcproj: * win32/vs8/libgstdv.vcproj: * win32/vs8/libgsteffectv.vcproj: * win32/vs8/libgstflac.vcproj: * win32/vs8/libgstflxdec.vcproj: * win32/vs8/libgstgoom.vcproj: * win32/vs8/libgsticydemux.vcproj: * win32/vs8/libgstid3demux.vcproj: * win32/vs8/libgstjpeg.vcproj: * win32/vs8/libgstladspa.vcproj: * win32/vs8/libgstlevel.vcproj: * win32/vs8/libgstmatroska.vcproj: * win32/vs8/libgstmikmod.vcproj: * win32/vs8/libgstmng.vcproj: * win32/vs8/libgstmonoscope.vcproj: * win32/vs8/libgstmulaw.vcproj: * win32/vs8/libgstmultipart.vcproj: * win32/vs8/libgstpng.vcproj: * win32/vs8/libgstrtp.vcproj: * win32/vs8/libgstrtsp.vcproj: * win32/vs8/libgstshout2.vcproj: * win32/vs8/libgstsmpte.vcproj: * win32/vs8/libgstspeex.vcproj: * win32/vs8/libgsttaglib.vcproj: * win32/vs8/libgstudp.vcproj: * win32/vs8/libgstvideobalance.vcproj: * win32/vs8/libgstvideobox.vcproj: * win32/vs8/libgstvideoflip.vcproj: * win32/vs8/libgstvideomixer.vcproj: * win32/vs8/libgstwavenc.vcproj: * win32/vs8/libgstwavparse.vcproj: Make end-of-line returns unixy, so that when the files are checked out on win32 the line returns will be 0d 0a and not 0d 0d 0a. Hopefully fixes #366492. 2006-11-14 15:55:32 +0000 Wim Taymans gst/avi/gstavidemux.c: Disable init_frames delay timestamp adjustment, it does not seem to be needed at all. Fixes #3... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index): Disable init_frames delay timestamp adjustment, it does not seem to be needed at all. Fixes #369621. 2006-11-14 11:43:40 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Don't parse extra sample params for raw pcm. Fixes #374914. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (gst_qtdemux_handle_src_query), (qtdemux_parse_trak): Don't parse extra sample params for raw pcm. Fixes #374914. 2006-11-14 10:29:37 +0000 Wim Taymans ext/lame/gstlame.*: Make lame timestamp flushed eos buffer by some additional timestamp accounting. Fixes #374760. Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain), (gst_lame_change_state): * ext/lame/gstlame.h: Make lame timestamp flushed eos buffer by some additional timestamp accounting. Fixes #374760. 2006-11-13 18:31:18 +0000 Mark Nauwelaerts gst/videomixer/videomixer.c: Fix memleak by unref'ing collectpads instance (when finalizing) Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/videomixer/videomixer.c: (gst_videomixer_set_master_geometry), (gst_videomixer_pad_sink_setcaps), (gst_videomixer_class_init), (gst_videomixer_collect_free), (gst_videomixer_reset), (gst_videomixer_init), (gst_videomixer_finalize), (gst_videomixer_request_new_pad), (gst_videomixer_release_pad), (gst_videomixer_collected), (gst_videomixer_change_state): Fix memleak by unref'ing collectpads instance (when finalizing) Implement releasing a request pad. Fixes #374479. 2006-11-10 20:08:42 +0000 Sergey Scobich win32/vs8/: Add VS8 project files (note that many of the plugins in ext are disabled by default). Fixes #366492. Original commit message from CVS: Patch by: Sergey Scobich * win32/vs8/gst-plugins-good.sln: * win32/vs8/libgst1394.vcproj: * win32/vs8/libgstaasink.vcproj: * win32/vs8/libgstalaw.vcproj: * win32/vs8/libgstalpha.vcproj: * win32/vs8/libgstalphacolor.vcproj: * win32/vs8/libgstannodex.vcproj: * win32/vs8/libgstapetag.vcproj: * win32/vs8/libgstaudiofx.vcproj: * win32/vs8/libgstauparse.vcproj: * win32/vs8/libgstautodetect.vcproj: * win32/vs8/libgstavi.vcproj: * win32/vs8/libgstcacasink.vcproj: * win32/vs8/libgstcdio.vcproj: * win32/vs8/libgstcutter.vcproj: * win32/vs8/libgstdv.vcproj: * win32/vs8/libgsteffectv.vcproj: * win32/vs8/libgstflac.vcproj: * win32/vs8/libgstflxdec.vcproj: * win32/vs8/libgstgoom.vcproj: * win32/vs8/libgsticydemux.vcproj: * win32/vs8/libgstid3demux.vcproj: * win32/vs8/libgstjpeg.vcproj: * win32/vs8/libgstladspa.vcproj: * win32/vs8/libgstlevel.vcproj: * win32/vs8/libgstmatroska.vcproj: * win32/vs8/libgstmikmod.vcproj: * win32/vs8/libgstmng.vcproj: * win32/vs8/libgstmonoscope.vcproj: * win32/vs8/libgstmulaw.vcproj: * win32/vs8/libgstmultipart.vcproj: * win32/vs8/libgstpng.vcproj: * win32/vs8/libgstrtp.vcproj: * win32/vs8/libgstrtsp.vcproj: * win32/vs8/libgstshout2.vcproj: * win32/vs8/libgstsmpte.vcproj: * win32/vs8/libgstspeex.vcproj: * win32/vs8/libgsttaglib.vcproj: * win32/vs8/libgstudp.vcproj: * win32/vs8/libgstvideobalance.vcproj: * win32/vs8/libgstvideobox.vcproj: * win32/vs8/libgstvideoflip.vcproj: * win32/vs8/libgstvideomixer.vcproj: * win32/vs8/libgstwavenc.vcproj: * win32/vs8/libgstwavparse.vcproj: Add VS8 project files (note that many of the plugins in ext are disabled by default). Fixes #366492. 2006-11-10 19:18:33 +0000 David Schleef gst/multifile/Makefile.am: Let's not depend on a file that doesn't exist. Original commit message from CVS: * gst/multifile/Makefile.am: Let's not depend on a file that doesn't exist. 2006-11-10 18:51:10 +0000 David Schleef Revive multifile[src|sink]. Original commit message from CVS: * configure.ac: * gst/multifile/Makefile.am: * gst/multifile/gstmultifile.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multifile/multifile.vproj: Revive multifile[src|sink]. 2006-11-10 08:09:05 +0000 Stefan Kost sys/v4l2/v4l2src_calls.c: we do not translate debug messages Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame): we do not translate debug messages 2006-11-08 12:04:03 +0000 Stefan Kost gst/flx/gstflxdec.c: fix categorisation, make short desc more explicit, remove unused code Original commit message from CVS: * gst/flx/gstflxdec.c: (gst_flxdec_class_init): fix categorisation, make short desc more explicit, remove unused code Fixes #372021 2006-11-08 01:30:39 +0000 Christian Schaller gst/rtp/: Fix element descriptions. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: Fix element descriptions. 2006-11-08 01:29:51 +0000 Christian Schaller gst/rtp/: Fix description. Original commit message from CVS: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_handle_buffer): Fix description. Small cleanup in the payloader. 2006-11-08 01:28:00 +0000 Christian Schaller gst/rtp/: Add theora pay/depayloaders. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init), (gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init), (gst_rtp_theora_depay_finalize), (gst_rtp_theora_depay_parse_configuration), (gst_rtp_theora_depay_setcaps), (gst_rtp_theora_depay_switch_codebook), (gst_rtp_theora_depay_process), (gst_rtp_theora_depay_set_property), (gst_rtp_theora_depay_get_property), (gst_rtp_theora_depay_change_state), (gst_rtp_theora_depay_plugin_init): * gst/rtp/gstrtptheoradepay.h: * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init), (gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init), (gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet), (gst_rtp_theora_pay_init_packet), (gst_rtp_theora_pay_flush_packet), (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id), (gst_rtp_theora_pay_handle_buffer), (gst_rtp_theora_pay_plugin_init): * gst/rtp/gstrtptheorapay.h: Add theora pay/depayloaders. 2006-11-07 01:43:06 +0000 Christian Schaller gst/rtp/Makefile.am: We depend on gsttag to generate the vorbis comments. Original commit message from CVS: * gst/rtp/Makefile.am: We depend on gsttag to generate the vorbis comments. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_parse_configuration), (gst_rtp_vorbis_depay_setcaps), (gst_rtp_vorbis_depay_switch_codebook), (gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbisdepay.h: Parse configuration string in the depayloader. Implement selecting and switching to a new codebook. Receiving vorbis over RTP now works. * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet), (gst_rtp_vorbis_pay_init_packet), (gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_handle_buffer): * gst/rtp/gstrtpvorbispay.h: Set timestamps on outgoing buffers and RTP packets. Fix configuration string, prepend number of Packet headers. Fix encoding of ident string. Add delivery-method to caps. Streaming vorbis over RTP now works. 2006-11-06 20:52:10 +0000 Christian Schaller gst/rtp/gstrtpvorbispay.*: Generate a valid configuration string in the caps based on the vorbis headers. Original commit message from CVS: * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id), (gst_rtp_vorbis_pay_handle_buffer): * gst/rtp/gstrtpvorbispay.h: Generate a valid configuration string in the caps based on the vorbis headers. 2006-11-02 20:13:26 +0000 Sebastian Dröge Fix enum nicks; only emit no-more-pads once; add support for very fast encoding mode in upcoming 4.40.0 release (#369... Original commit message from CVS: Patch by: Sebastian Dröge * configure.ac: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type), (gst_wavpack_enc_correction_mode_get_type), (gst_wavpack_enc_joint_stereo_mode_get_type), (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config): Fix enum nicks; only emit no-more-pads once; add support for very fast encoding mode in upcoming 4.40.0 release (#369539). 2006-11-02 14:43:11 +0000 Tim-Philipp Müller ext/cdio/: Move CD-TEXT utility function into common file so it can also be used by a future cdioparanoiasrc. Original commit message from CVS: * ext/cdio/gstcdio.c: (gst_cdio_get_cdtext): * ext/cdio/gstcdio.h: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open): Move CD-TEXT utility function into common file so it can also be used by a future cdioparanoiasrc. 2006-11-01 19:48:26 +0000 Edgard Lima * ChangeLog: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to xoverlay code that is still not implemented. Original commit message from CVS: Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to xoverlay code that is still not implemented. 2006-11-01 13:59:49 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: We require a -base more recent than 0.10.9, so it's safe to use Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_picture_frame): We require a -base more recent than 0.10.9, so it's safe to use GST_TYPE_TAG_IMAGE_TYPE unconditionally now. * ext/dv/gstdvdec.c: (gst_dvdec_sink_event): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event): Use _newsegment_full() now that we depend on a recent enough core. * gst/wavparse/gstwavparse.c: Remove cruft that we don't need any longer now that we depend on a recent enough -base. 2006-11-01 10:19:18 +0000 Sergey Scobich sys/: Wait until the window is created before using it; guard unistd.h includes with HAVE_UNISTD_H. (#366523) Original commit message from CVS: Patch by: Sergey Scobich * sys/directdraw/gstdirectdrawsink.c: (gst_directdrawsink_window_thread), (gst_directdrawsink_create_default_window): * sys/directdraw/gstdirectdrawsink.h: * sys/directsound/gstdirectsoundsink.c: Wait until the window is created before using it; guard unistd.h includes with HAVE_UNISTD_H. (#366523) * win32/vs8/libgstdirectdraw.vcproj: * win32/vs8/libgstdirectsound.vcproj: Update project files. 2006-10-31 10:52:31 +0000 Wim Taymans gst/rtp/: Fix and activate ILBC pay and depayloaders. Fixes #368162. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init), (gst_rtpilbcpay_setcaps): Fix and activate ILBC pay and depayloaders. Fixes #368162. 2006-10-31 10:31:18 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Handle unbounded length streams a bit better. Fixes #367696. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (gst_qtdemux_handle_src_query), (qtdemux_parse_tree), (qtdemux_parse_trak): Handle unbounded length streams a bit better. Fixes #367696. 2006-10-31 09:44:39 +0000 Wim Taymans ext/speex/gstspeexdec.c: Some small cleanups, use _scale. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_sink_event), (speex_dec_chain_parse_header): Some small cleanups, use _scale. 2006-10-31 09:29:36 +0000 Wim Taymans gst/avi/gstavidemux.c: Use higher precision scale function. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query): Use higher precision scale function. 2006-10-30 16:18:18 +0000 Michal Benes gst/matroska/matroska-demux.c: Fix several issues with encoded/compressed/encrypted/signed tracks; also, remove super... Original commit message from CVS: Patch by: Michal Benes * gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp), (gst_matroska_demux_read_track_encodings), (gst_matroska_decode_buffer): Fix several issues with encoded/compressed/encrypted/signed tracks; also, remove superfluous newline characters from some debug statements. (#366155) 2006-10-30 09:24:53 +0000 Wim Taymans ext/jpeg/: Various cleanups, capsnego and leak fixes. Original commit message from CVS: * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init), (gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain), (gst_smokedec_change_state): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init), (gst_smokeenc_init), (gst_smokeenc_finalize), (gst_smokeenc_getcaps), (gst_smokeenc_setcaps), (gst_smokeenc_resync), (gst_smokeenc_chain), (gst_smokeenc_set_property), (gst_smokeenc_get_property), (gst_smokeenc_change_state): Various cleanups, capsnego and leak fixes. 2006-10-30 08:17:08 +0000 Mark Nauwelaerts gst/videomixer/videomixer.c: Fix videomixer so that it can handle any combination of framerates. Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/videomixer/videomixer.c: (gst_videomixer_update_queues): Fix videomixer so that it can handle any combination of framerates. Fixes #367221. 2006-10-28 16:37:20 +0000 Wim Taymans gst/avi/gstavidemux.c: Fix position query for audio. also fixes timestamps in streaming mode and bug #364958. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_file_header), (gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data), (gst_avi_demux_chain): Fix position query for audio. also fixes timestamps in streaming mode and bug #364958. Small cleanups. 2006-10-27 17:10:42 +0000 Wim Taymans ext/libpng/gstpngenc.*: Fix strides. Fixes #364856. Original commit message from CVS: * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps), (gst_pngenc_chain): * ext/libpng/gstpngenc.h: Fix strides. Fixes #364856. Cleanup capsnego. Set caps on outgoing buffers. 2006-10-18 17:06:21 +0000 Ville Syrjala gst/rtp/: Add static payload numbers in addition to the dynamic ones. Original commit message from CVS: Patch by: Ville Syrjala * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush), (gst_rtp_pcma_pay_handle_buffer): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush): Add static payload numbers in addition to the dynamic ones. Fixes #361639. 2006-10-18 16:18:55 +0000 Wim Taymans gst/rtsp/: Reuse already existing enum for lower transport. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_open), (gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri): * gst/rtsp/rtspconnection.c: (rtsp_connection_create): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspurl.c: (rtsp_url_parse): * gst/rtsp/rtspurl.h: Reuse already existing enum for lower transport. Add rtspt and rtspu protocols. Send redirect to rtspt when udp times out. 2006-10-18 14:00:44 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Fix seeking some more, mostly for speed changes. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_data): Fix seeking some more, mostly for speed changes. 2006-10-18 11:28:05 +0000 Tim-Philipp Müller * ChangeLog: ChangeLog surgery: fix Fredrik's e-mail address Original commit message from CVS: ChangeLog surgery: fix Fredrik's e-mail address 2006-10-18 11:04:09 +0000 Fredrik Persson sys/v4l2/gstv4l2tuner.*: Fix _set_channel(): remove useless g_object_notify() for "channel" property that doesn't exi... Original commit message from CVS: Patch by: Fredrik Persson * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: Fix _set_channel(): remove useless g_object_notify() for "channel" property that doesn't exist any longer and therefore now also useless redirect (#338818). 2006-10-17 15:16:47 +0000 Tim-Philipp Müller Activate pads before adding them to running element. Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_create_src_pad): * gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_create_pads): * tests/check/elements/wavpackparse.c: (wavpackparse_found_pad): Activate pads before adding them to running element. 2006-10-17 14:57:17 +0000 Josep Torra Valles gst/qtdemux/qtdemux.c: Make compile with Forte compiler, mostly don't do pointer arithmetic with void pointers (#3626... Original commit message from CVS: Patch by: Josep Torra Valles * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event), (next_entry_size), (qtdemux_inflate), (qtdemux_parse_moov), (qtdemux_parse_tree), (qtdemux_parse_trak), (qtdemux_tag_add_str), (qtdemux_tag_add_num), (qtdemux_tag_add_date), (qtdemux_tag_add_gnre): Make compile with Forte compiler, mostly don't do pointer arithmetic with void pointers (#362626). 2006-10-17 14:37:49 +0000 Wim Taymans sys/oss/gstosssink.c: Some drivers do not support unsetting the non-blocking flag once the device is opened. In those... Original commit message from CVS: * sys/oss/gstosssink.c: (gst_oss_sink_prepare): Some drivers do not support unsetting the non-blocking flag once the device is opened. In those cases, close/open the device in non-blocking mode. Fixes #362673. 2006-10-17 13:44:14 +0000 Stefan Kost sys/v4l2/: dear stefan, framespersecond is not frameperiod, reverting but adding comment Original commit message from CVS: * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps), (gst_v4l2src_get_fps): dear stefan, framespersecond is not frameperiod, reverting but adding comment 2006-10-17 11:28:50 +0000 Stefan Kost sys/v4l2/: Numerator is numerator and denominator is denominator. Say that aloud 5 times and retry after next beer. Original commit message from CVS: * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps), (gst_v4l2src_get_fps): Numerator is numerator and denominator is denominator. Say that aloud 5 times and retry after next beer. 2006-10-17 10:59:55 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.*: Avoid void pointer usage, better use guint8 * instead. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_moov), (qtdemux_parse), (qtdemux_node_dump_foreach), (qtdemux_dump_mvhd), (qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd), (qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref), (qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss), (qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco), (qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd), (qtdemux_dump_unknown), (qtdemux_tree_get_child_by_type), (qtdemux_tree_get_sibling_by_type): * gst/qtdemux/qtdemux.h: Avoid void pointer usage, better use guint8 * instead. 2006-10-16 18:22:47 +0000 Josep Torra Valles Fix a bunch of problems discovered by the Forte compiler, mostly type mixups and pointer arithmetics with void pointe... Original commit message from CVS: Patch by: Josep Torra Valles * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform): * ext/esd/esdsink.c: (gst_esdsink_write): * ext/flac/gstflacdec.c: (gst_flac_dec_length), (gst_flac_dec_read_seekable), (gst_flac_dec_chain), (gst_flac_dec_send_newsegment): * ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback), (gst_flac_enc_tell_callback): * ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode), (smokecodec_parse_header), (smokecodec_decode): * gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index): * gst/debug/efence.c: (gst_fenced_buffer_alloc): * gst/goom/Makefile.am: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward): * gst/rtsp/gstrtspsrc.c: * gst/rtsp/rtspconnection.c: (rtsp_connection_read): * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_change_state): * sys/sunaudio/gstsunaudiomixertrack.h: Fix a bunch of problems discovered by the Forte compiler, mostly type mixups and pointer arithmetics with void pointers. Fixes #362603. 2006-10-13 14:45:11 +0000 Tim-Philipp Müller ext/lame/gstlame.c: Round up not allowed bitrates to the next higher allowed one (Closes: #361140). Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_set_property): Round up not allowed bitrates to the next higher allowed one (Closes: #361140). 2006-10-13 14:19:24 +0000 Tim-Philipp Müller Add docs for lame and lame to docs. Specify allowed bitrates in the properties description (#361140). Canonicalise ob... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-ugly-plugins-docs.sgml: * docs/plugins/gst-plugins-ugly-plugins-sections.txt: * ext/lame/gstlame.c: (gst_lame_class_init): * ext/lame/gstlame.h: Add docs for lame and lame to docs. Specify allowed bitrates in the properties description (#361140). Canonicalise object property names (ie. use hyphen instead of underscore). * docs/plugins/inspect/plugin-a52dec.xml: * docs/plugins/inspect/plugin-amrnb.xml: * docs/plugins/inspect/plugin-asf.xml: * docs/plugins/inspect/plugin-dvdlpcmdec.xml: * docs/plugins/inspect/plugin-dvdread.xml: * docs/plugins/inspect/plugin-dvdsub.xml: * docs/plugins/inspect/plugin-iec958.xml: * docs/plugins/inspect/plugin-lame.xml: * docs/plugins/inspect/plugin-mad.xml: * docs/plugins/inspect/plugin-mpeg2dec.xml: * docs/plugins/inspect/plugin-mpegaudioparse.xml: * docs/plugins/inspect/plugin-mpegstream.xml: * docs/plugins/inspect/plugin-siddec.xml: Update version to CVS. 2006-10-13 10:00:27 +0000 Tim-Philipp Müller Add i18n magic to lame plugin. Throw decent error message when we fail to setup the encoder (#361140, 361151); misc. ... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_setcaps), (gst_lame_set_property), (gst_lame_get_property), (gst_lame_chain), (plugin_init): * po/POTFILES.in: Add i18n magic to lame plugin. Throw decent error message when we fail to setup the encoder (#361140, 361151); misc. minor clean-ups. 2006-10-12 19:02:51 +0000 Tim-Philipp Müller ext/speex/: Miscellaneous clean-ups, among other things: speexenc => enc to enhance code readability; change speexenc... Original commit message from CVS: * ext/speex/gstspeex.c: (plugin_init): * ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats), (gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init), (gst_speex_enc_class_init), (gst_speex_enc_finalize), (gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src), (gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types), (gst_speex_enc_src_query), (gst_speex_enc_sink_query), (gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer), (gst_speex_enc_set_last_msg), (gst_speex_enc_setup), (gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer), (gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent), (gst_speex_enc_chain), (gst_speex_enc_get_property), (gst_speex_enc_set_property), (gst_speex_enc_change_state): * ext/speex/gstspeexenc.h: Miscellaneous clean-ups, among other things: speexenc => enc to enhance code readability; change speexenc => speex_enc; in chain function unref input buffer in case of error; take reference in event function; use boilerplate macro; use gst_pad_query_peer_* convenience functions. 2006-10-12 18:35:10 +0000 Tim-Philipp Müller ext/speex/gstspeexenc.c: Fix some mem leaks. Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speexenc_finalize), (gst_speexenc_set_last_msg), (gst_speexenc_setup), (gst_speexenc_set_header_on_caps): Fix some mem leaks. 2006-10-11 16:21:53 +0000 Wim Taymans gst/rtsp/URLS: Added some other URL. Original commit message from CVS: * gst/rtsp/URLS: Added some other URL. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp), (gst_rtspsrc_handle_request), (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Work on fallback to TCP connection when the UDP socket times out. Handler server requests, just reply with OK for now. * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Added some more Real extension headers. * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of urls with a ':' that is not part of the hostname:port part of the url. 2006-10-11 13:49:26 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Add some fourcc for DV format. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add some fourcc for DV format. 2006-10-11 13:24:42 +0000 Tim-Philipp Müller gst/: Activate pad before adding it to the already-running element. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad): * gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad): * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad): Activate pad before adding it to the already-running element. * tests/check/elements/icydemux.c: (icydemux_found_pad): Activate newly-created pad too. 2006-10-11 08:34:14 +0000 Sebastien Cote gst/udp/gstudpsrc.c: Fix some leaks in caps and uris. Fixes #361252. Original commit message from CVS: Patch by: Sebastien Cote * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri), (gst_udpsrc_start): Fix some leaks in caps and uris. Fixes #361252. 2006-10-10 18:54:05 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Printf format fixes. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc), (gst_qtdemux_loop_state_header): Printf format fixes. * sys/dvb/gstdvbsrc.c: Use "_stdint.h". 2006-10-10 09:57:19 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Reorganise some stuff. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_init), (gst_qtdemux_push_event), (gst_qtdemux_do_seek), (gst_qtdemux_change_state), (extract_initial_length_and_fourcc), (gst_qtdemux_loop_state_header), (gst_qtdemux_activate_segment), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (gst_qtdemux_post_buffering), (gst_qtdemux_chain), (gst_qtdemux_add_stream), (qtdemux_process_redirects), (qtdemux_parse_tree), (qtdemux_parse_trak): Reorganise some stuff. Parse RTSP redirection URLS. 2006-10-10 08:29:07 +0000 Tim-Philipp Müller gst/wavparse/Makefile.am: Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489). Original commit message from CVS: * gst/wavparse/Makefile.am: Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489). 2006-10-09 07:01:19 +0000 Jan Schmidt sys/v4l2/gstv4l2xoverlay.*: Fix build as per the patch in #338818 comment 36. Original commit message from CVS: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/gstv4l2xoverlay.h: Fix build as per the patch in #338818 comment 36. 2006-10-08 20:05:13 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: inspect updates Original commit message from CVS: inspect updates 2006-10-07 21:15:40 +0000 Tim-Philipp Müller gst/rtsp/gstrtspsrc.c: Activate pads before adding them to the source. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport): Activate pads before adding them to the source. 2006-10-07 11:37:59 +0000 Tim-Philipp Müller docs/plugins/: Add/update docs stuff. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-swfdec.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-wavpack.xml: Add/update docs stuff. 2006-10-06 17:00:14 +0000 Wim Taymans Activate pads before adding. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain): * gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad): Activate pads before adding. 2006-10-06 16:03:23 +0000 Wim Taymans gst/multipart/multipartdemux.c: Activate pads before adding. Original commit message from CVS: * gst/multipart/multipartdemux.c: (gst_multipart_demux_init), (gst_multipart_find_pad_by_mime): Activate pads before adding. * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): BOILERPLATE sets parent_class for us. 2006-10-06 15:56:01 +0000 René Stadler Add ReplayGain analysis element (#357069). Original commit message from CVS: Patch by: René Stadler * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_base_init), (gst_rg_analysis_class_init), (gst_rg_analysis_init), (gst_rg_analysis_set_property), (gst_rg_analysis_get_property), (gst_rg_analysis_start), (gst_rg_analysis_set_caps), (gst_rg_analysis_transform_ip), (gst_rg_analysis_event), (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags), (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result), (gst_rg_analysis_album_result), (plugin_init): * gst/replaygain/gstrganalysis.h: * gst/replaygain/rganalysis.c: (yule_filter), (butter_filter), (apply_filters), (reset_filters), (accumulator_add), (accumulator_clear), (accumulator_result), (rg_analysis_new), (rg_analysis_set_sample_rate), (rg_analysis_destroy), (rg_analysis_analyze_mono_float), (rg_analysis_analyze_stereo_float), (rg_analysis_analyze_mono_int16), (rg_analysis_analyze_stereo_int16), (rg_analysis_analyze), (rg_analysis_track_result), (rg_analysis_album_result), (rg_analysis_reset_album), (rg_analysis_reset): * gst/replaygain/rganalysis.h: Add ReplayGain analysis element (#357069). * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/rganalysis.c: (get_expected_gain), (setup_rganalysis), (cleanup_rganalysis), (set_playing_state), (send_eos_event), (send_tag_event), (poll_eos), (poll_tags), (fail_unless_track_gain), (fail_unless_track_peak), (fail_unless_album_gain), (fail_unless_album_peak), (fail_if_track_tags), (fail_if_album_tags), (fail_unless_num_tracks), (test_buffer_const_float_mono), (test_buffer_const_float_stereo), (test_buffer_const_int16_mono), (test_buffer_const_int16_stereo), (test_buffer_square_float_mono), (test_buffer_square_float_stereo), (test_buffer_square_int16_mono), (test_buffer_square_int16_stereo), (push_buffer), (GST_START_TEST), (rganalysis_suite), (main): Unit tests for the new replaygain element. 2006-10-06 15:49:39 +0000 Wim Taymans ext/faad/gstfaad.c: Some cleanups. Original commit message from CVS: * ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chain), (gst_faad_close_decoder): Some cleanups. Added some more debugging. Don't ever ignore unlinked, we're not a demuxer. * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream): Activate pad before adding it to the element. 2006-10-06 12:55:53 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to share channels and udp ports. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel), (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_configure_transports), (gst_rtspsrc_open), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Rework how the transport string is constructed, try to share channels and udp ports. Make most of the stuff less dependant on RTP as we are also going to use it for RDT. Add support for transport specific session managers. * gst/rtsp/rtspconnection.c: (rtsp_connection_flush): Implement _flush(). * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Add generic error return code. * gst/rtsp/rtspext.h: Add support for pluggable tranport strings. * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send), (rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp), (rtsp_ext_wms_get_context): Detect WMServer and activate the extension. * gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime), (rtsp_transport_get_manager), (rtsp_transport_parse): * gst/rtsp/rtsptransport.h: Added methods to get mime/manager for certain transports. 2006-10-06 11:31:11 +0000 Tim-Philipp Müller gst/spectrum/gstspectrum.c: Fix mem leak, avoid unnecessary memcpy. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip): Fix mem leak, avoid unnecessary memcpy. 2006-10-06 02:29:35 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Removed cruft code that was just commented out. Removed some obsolete debug logs statements. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_init), (gst_spectrum_transform_ip): Removed cruft code that was just commented out. Removed some obsolete debug logs statements. 2006-10-05 18:14:46 +0000 Tim-Philipp Müller Another batch of printf format fixes. Original commit message from CVS: * ext/dts/gstdtsdec.c: (gst_dtsdec_chain): * ext/musicbrainz/gsttrm.c: (gst_trm_setcaps): * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps): * gst/qtdemux/qtdemux.c: (gst_qtdemux_chain), (qtdemux_parse), (qtdemux_parse_trak): * gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip): Another batch of printf format fixes. 2006-10-05 16:37:33 +0000 Tim-Philipp Müller Printf format fixes. Original commit message from CVS: * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_update_font_height): * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain): * ext/libpng/gstpngdec.c: (user_endrow_callback): * gst/auparse/gstauparse.c: (gst_au_parse_parse_header): * gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_stream_data): * gst/cutter/gstcutter.c: (gst_cutter_chain): * gst/debug/efence.c: (gst_efence_buffer_alloc), (gst_fenced_buffer_copy): * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame): * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream): * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_handle_message): * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): * sys/ximage/ximageutil.c: (ximageutil_xcontext_get): Printf format fixes. 2006-10-04 22:37:07 +0000 Tim-Philipp Müller gst/videocrop/gstvideocrop.*: Handle packed YUV formats (UYVY, YUY2, YUYV) separately; also, fix passthrough mode; la... Original commit message from CVS: * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init), (gst_video_crop_init), (gst_video_crop_get_image_details_from_caps), (gst_video_crop_transform_packed_complex), (gst_video_crop_transform_packed_simple), (gst_video_crop_transform), (gst_video_crop_transform_caps), (gst_video_crop_set_caps), (gst_videocrop_clear_negotiated_caps_locked), (gst_video_crop_set_property): * gst/videocrop/gstvideocrop.h: Handle packed YUV formats (UYVY, YUY2, YUYV) separately; also, fix passthrough mode; lastly, clear negotiated basetransform caps when the cropping changes in order to force renegotiation. 2006-10-04 20:05:07 +0000 Tim-Philipp Müller tests/icles/: Visual test for videocrop, shows that packed yuv doesn't work right yet. --with-ffmpegcolorspace option... Original commit message from CVS: * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/videocrop-test.c: (quit_mainloop), (tick_cb), (test_with_caps), (video_crop_get_test_caps), (main): Visual test for videocrop, shows that packed yuv doesn't work right yet. --with-ffmpegcolorspace option doesn't work yet for unknown reasons (another basetransform issue?) 2006-10-04 17:53:12 +0000 Wim Taymans gst/rtsp/Makefile.am: Dist new .h file too. Original commit message from CVS: * gst/rtsp/Makefile.am: Dist new .h file too. 2006-10-04 17:24:40 +0000 Wim Taymans gst/rtsp/: Factor out extension in separate module. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps), (gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream), (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_handle_message): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspext.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp), (rtsp_ext_wms_get_context): * gst/rtsp/rtspextwms.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode), (rtsp_transport_parse): * gst/rtsp/rtsptransport.h: Factor out extension in separate module. Fix getcaps to filter against the padtemplate. Use Content-Base if the server gives one. Rework the transport parsing a bit for future extensions. Added some Real Header field definitions. 2006-10-04 10:29:11 +0000 Thomas Vander Stichele docs/plugins/: added v4l2 stubs Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: added v4l2 stubs * gst-plugins-good.spec.in: add v4l2 2006-10-04 10:24:49 +0000 Tim-Philipp Müller gst/apetag/gstapedemux.c: Extract disc/album/medium number and count and try harder to extract track number/count. Original commit message from CVS: * gst/apetag/gstapedemux.c: (ape_demux_parse_tags): Extract disc/album/medium number and count and try harder to extract track number/count. 2006-10-03 18:36:29 +0000 Thomas Vander Stichele * tests/icles/.gitignore: moap ignore Original commit message from CVS: moap ignore 2006-10-03 18:35:34 +0000 Thomas Vander Stichele * tests/icles/Makefile.am: add icle for v4l2 Original commit message from CVS: add icle for v4l2 2006-10-03 18:15:58 +0000 Thomas Vander Stichele add build stuff for v4l2, needs --enable-experimental until the last bits are resolved Original commit message from CVS: * configure.ac: * sys/Makefile.am: add build stuff for v4l2, needs --enable-experimental until the last bits are resolved 2006-10-03 13:47:10 +0000 Thomas Vander Stichele * sys/v4l2/gstv4l2object.c: comment out the notifies for removed properties Original commit message from CVS: comment out the notifies for removed properties 2006-10-03 13:30:48 +0000 Thomas Vander Stichele sys/v4l2/gstv4l2object.c: comment out the properties that are already part of the tuner interface. Original commit message from CVS: * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_install_properties_helper): comment out the properties that are already part of the tuner interface. 2006-10-03 13:18:59 +0000 Zaheer Abbas Merali sys/v4l2/gstv4l2src.c: Improve docs. Original commit message from CVS: 2006-10-03 Zaheer Abbas Merali * sys/v4l2/gstv4l2src.c: Improve docs. 2006-10-02 16:14:06 +0000 Christian Schaller * gst-plugins-good.spec.in: stop removing gdkpixbuf plugin from package Original commit message from CVS: stop removing gdkpixbuf plugin from package 2006-09-29 15:39:41 +0000 Tim-Philipp Müller tests/check/Makefile.am: Disable autodetect test temporarily, so that the build bots update -bad and the ranks of unr... Original commit message from CVS: * tests/check/Makefile.am: Disable autodetect test temporarily, so that the build bots update -bad and the ranks of unreliable video sinks in there. * tests/check/elements/autodetect.c: (GST_START_TEST): Skip test if no usable videosink is found. 2006-09-29 15:37:29 +0000 Wim Taymans gst/rtsp/URLS: Add some more URLs. Original commit message from CVS: * gst/rtsp/URLS: Add some more URLs. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add timeout property to control UDP timeouts. Fix error messages. Also start a loop function when operating in UDP mode so that we can do some more stuff async. Handle element messages from udpsrc to detect timeouts. If a timeout happens we currently generate an error. API: rtspsrc::timeout property. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create): Really implement the timeout in microseconds and not milliseconds. 2006-09-29 11:09:40 +0000 Wim Taymans gst/udp/gstudpsrc.*: Added property to post a message on timeout. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop): * gst/udp/gstudpsrc.h: Added property to post a message on timeout. Updated docs. When restarting the select, initialize the fdsets again. Init control sockets so we don't accidentally close a random socket. API: GstUDPSrc::timeout property 2006-09-29 08:15:05 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Fix flag registration. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type): Fix flag registration. * gst/rtsp/rtspconnection.c: (rtsp_connection_read): Reading 0 also means 'no more commands' 2006-09-29 08:09:24 +0000 Antoine Tremblay gst/udp/gstudpsrc.c: Fix possible infinite loop when shutting down, a read can also return 0 to indicate no more mess... Original commit message from CVS: Patch by: Antoine Tremblay * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Fix possible infinite loop when shutting down, a read can also return 0 to indicate no more messages are available. Fixes #358156. 2006-09-28 17:08:47 +0000 Wim Taymans sys/v4l2/: Framerate can be 0/1 too. Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_all_caps), (gst_v4l2src_get_caps): * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): Framerate can be 0/1 too. Init framerate to 0/1 before querying it so that we can detect devices that don't know about a framerate. Add some more debugging info. 2006-09-28 14:31:41 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Add support for 'yv12' fourcc. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add support for 'yv12' fourcc. 2006-09-27 17:47:57 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * tests/icles/v4l2src-test.c: Removed set-undef-fps. Original commit message from CVS: Removed set-undef-fps. 2006-09-27 17:04:22 +0000 Wim Taymans sys/v4l2/: Renamed some properties to match the tuner interface naming. Original commit message from CVS: * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_install_properties_helper), (gst_v4l2_object_new), (gst_v4l2_object_set_property_helper), (gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_create): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_contains_channel), (gst_v4l2_tuner_list_channels), (gst_v4l2_tuner_set_channel_and_notify), (gst_v4l2_tuner_get_channel), (gst_v4l2_tuner_contains_norm), (gst_v4l2_tuner_list_norms), (gst_v4l2_tuner_set_norm_and_notify), (gst_v4l2_tuner_get_norm): * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities), (gst_v4l2_fill_lists), (gst_v4l2_empty_lists): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_get_fps): Renamed some properties to match the tuner interface naming. 2006-09-27 16:14:18 +0000 Wim Taymans Small cleanups. Original commit message from CVS: * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_set_property_helper), (gst_v4l2_set_defaults): * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read), (gst_v4l2src_create): * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open): * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities), (gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_set_norm), (gst_v4l2_get_frequency), (gst_v4l2_set_frequency), (gst_v4l2_signal_strength), (gst_v4l2_get_attribute), (gst_v4l2_set_attribute), (gst_v4l2_get_input), (gst_v4l2_set_input): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame), (gst_v4l2src_get_capture), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init), (gst_v4l2src_capture_start), (gst_v4l2src_capture_stop), (gst_v4l2src_buffer_new): * tests/icles/v4l2src-test.c: (my_bus_callback), (main): Small cleanups. Fix error messages. Use locks when getting timestamps. Fix leaks in test. Add licensing header to tests. 2006-09-27 15:14:07 +0000 Edgard Lima * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: * tests/icles/v4l2src-test.c: Some cleanups and comments. Original commit message from CVS: Some cleanups and comments. 2006-09-27 13:41:35 +0000 Christian Schaller * gst-plugins-good.spec.in: add audiofx plugin Original commit message from CVS: add audiofx plugin 2006-09-26 14:17:54 +0000 Wim Taymans docs/plugins/: Add v4l2 plugin to the docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: Add v4l2 plugin to the docs. * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read), (gst_v4l2src_get_mmap), (gst_v4l2src_create): * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2vidorient.c: Fix docs. Remove some more externs. 2006-09-26 13:18:06 +0000 Wim Taymans sys/v4l2/Makefile.am: Fix makefile, list libs in stack order. Original commit message from CVS: * sys/v4l2/Makefile.am: Fix makefile, list libs in stack order. * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2object.c: (gst_v4l2_device_get_type), (gst_v4l2_object_install_properties_helper): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read), (gst_v4l2src_get_mmap), (gst_v4l2src_create): * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2vidorient.h: * sys/v4l2/gstv4l2xoverlay.h: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.h: Fix coding style: - Remove extern from functions. - Fix header indentation. Fix Flags, add defaults for properties. Remove unused enums. Fix TOO_LAZY in error messages. 2006-09-26 11:06:17 +0000 Wim Taymans sys/v4l2/: Fix pass at code cleanups, move errors cases out of the normal flow for additional code clarity. Original commit message from CVS: * sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices), (gst_v4l2_probe_needs_probe), (gst_v4l2_object_install_properties_helper), (gst_v4l2_object_new), (gst_v4l2_object_destroy), (gst_v4l2_object_set_property_helper), (gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults), (gst_v4l2_object_start), (gst_v4l2_object_stop): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_init), (gst_v4l2src_dispose), (gst_v4l2src_set_property), (gst_v4l2src_get_property), (gst_v4l2src_fixate), (gst_v4l2src_get_caps), (gst_v4l2src_set_caps), (gst_v4l2src_get_read), (gst_v4l2src_get_mmap), (gst_v4l2src_create): * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities), (gst_v4l2_open), (gst_v4l2_close), (gst_v4l2_get_norm), (gst_v4l2_set_norm), (gst_v4l2_get_frequency), (gst_v4l2_set_frequency), (gst_v4l2_signal_strength), (gst_v4l2_get_attribute), (gst_v4l2_set_attribute), (gst_v4l2_get_input), (gst_v4l2_set_input): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame), (gst_v4l2src_get_capture), (gst_v4l2src_set_capture), (gst_v4l2src_capture_init), (gst_v4l2src_capture_start), (gst_v4l2src_capture_stop), (gst_v4l2src_capture_deinit), (gst_v4l2src_get_size_limits), (gst_v4l2src_set_fps), (gst_v4l2src_get_fps), (gst_v4l2src_buffer_finalize), (gst_v4l2src_buffer_new): Fix pass at code cleanups, move errors cases out of the normal flow for additional code clarity. 2006-09-25 13:55:44 +0000 Wim Taymans gst/autodetect/: Small cleanups. don't try to set "sync" property when it is not available. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_find_best): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect): Small cleanups. don't try to set "sync" property when it is not available. 2006-09-25 11:47:42 +0000 Peter Kjellerstedt gst/: Include stdlib.h in some more places, makes things compile with uClibc and -Werror (#357592). Original commit message from CVS: Patch by: Peter Kjellerstedt * gst/alpha/gstalpha.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstudpsrc.c: * gst/videomixer/videomixer.c: Include stdlib.h in some more places, makes things compile with uClibc and -Werror (#357592). 2006-09-25 09:15:10 +0000 Tim-Philipp Müller ext/jpeg/gstjpegdec.c: our code should handle that fine. Some of the buttons on the apple trailer site are apparently... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: Set minimum height to 8 (from 16), our code should handle that fine. Some of the buttons on the apple trailer site are apparently only 15 pixels high (see #357470). 2006-09-23 15:31:56 +0000 Wim Taymans gst/rtsp/: Improve error reporting. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_open): * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Improve error reporting. 2006-09-23 15:30:40 +0000 Wim Taymans gst/rtp/: Fix klass typos. Original commit message from CVS: * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init): * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init): * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init): * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init): * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_plugin_init): * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init): * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init): * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init): Fix klass typos. Mark RANK_MARGINAL, decodebin can handle the depayloaders fine. 2006-09-22 17:53:48 +0000 Tim-Philipp Müller configure.ac: Need -base CVS for gst_base_rtp_depayload_push_ts(). Original commit message from CVS: * configure.ac: Need -base CVS for gst_base_rtp_depayload_push_ts(). 2006-09-22 17:22:34 +0000 Wim Taymans gst/avi/gstavidemux.c: Don't check for a tag that is never there and check if we read the correct tag. Fixes seeking ... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index): Don't check for a tag that is never there and check if we read the correct tag. Fixes seeking again. We must post an error when all pads are unlinked. 2006-09-22 15:15:13 +0000 Wim Taymans gst/rtp/: More fixage, set endoder-params correctly in the payloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_reset_packet), (gst_rtp_vorbis_pay_init_packet), (gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id), (gst_rtp_vorbis_pay_handle_buffer): More fixage, set endoder-params correctly in the payloader. 2006-09-22 12:12:10 +0000 Tim-Philipp Müller gst/autodetect/: Make static pad templates static to appease valgrind's leak detector. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): Make static pad templates static to appease valgrind's leak detector. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/autodetect.c: (GST_START_TEST), (autodetect_suite): Add simple test for the ghostpad lockup on shutdown fixed in core CVS (audio bit disabled because it would need dozens of alsa suppressions and I'm too lazy to add those now). 2006-09-22 12:08:14 +0000 Wim Taymans gst/rtp/: Small cleanups. Original commit message from CVS: * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state): * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init): Small cleanups. * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init), (gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init), (gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps), (gst_rtp_vorbis_depay_process), (gst_rtp_vorbis_depay_set_property), (gst_rtp_vorbis_depay_get_property), (gst_rtp_vorbis_depay_change_state), (gst_rtp_vorbis_depay_plugin_init): * gst/rtp/gstrtpvorbisdepay.h: * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init), (gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init), (gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet), (gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_append_buffer), (gst_rtp_vorbis_pay_handle_buffer), (gst_rtp_vorbis_pay_plugin_init): * gst/rtp/gstrtpvorbispay.h: Add experimental vorbis pay and depayloaders. 2006-09-21 13:33:16 +0000 Wim Taymans gst/rtp/gstrtpmp4gpay.c: Fix profile-level-id parsing and setup. Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config): Fix profile-level-id parsing and setup. 2006-09-21 09:50:41 +0000 Wim Taymans gst/udp/: Update README, simple cleanup. Original commit message from CVS: * gst/udp/README: * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property): Update README, simple cleanup. 2006-09-21 09:35:13 +0000 Wim Taymans gst/rtp/README: Update README with some examples. Original commit message from CVS: * gst/rtp/README: Update README with some examples. * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps): * gst/rtp/gstrtpmp4gpay.h: Make optional RTP parameters of type STRING, as required by the application/x-rtp caps specification. 2006-09-20 19:37:45 +0000 Philippe Kalaf gst/rtp/: Correctly calculate size of each H263+ RTP buffer taking into account MTU and Original commit message from CVS: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: Correctly calculate size of each H263+ RTP buffer taking into account MTU and RTP header. 2006-09-20 16:41:48 +0000 Wim Taymans gst/rtp/Makefile.am: And makefile too. Original commit message from CVS: * gst/rtp/Makefile.am: And makefile too. 2006-09-20 16:09:03 +0000 Wim Taymans gst/rtp/: Added preliminary ASF depayloader. Original commit message from CVS: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init), (gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init), (decode_base64), (gst_rtp_asf_depay_setcaps), (gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property), (gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state), (gst_rtp_asf_depay_plugin_init): * gst/rtp/gstrtpasfdepay.h: Added preliminary ASF depayloader. * gst/rtp/gstrtph264depay.c: (decode_base64): Fix base64 decoding. 2006-09-20 16:06:27 +0000 Wim Taymans gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now. 2006-09-19 17:25:15 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (find_stream_by_pt), (gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel), (gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: Reorganize stream parsing and creation. Detect container formats in interleaved mode. Keep more state about the streams. Assume a server also supports PLAY if it does not say. Add unicast and interleaved properties to TCP transport requests to make some servers happy (WMServer). * gst/rtsp/sdpmessage.h: Add some defines for the standard Bandwidth types. 2006-09-19 16:24:10 +0000 Edgard Lima * tests/icles/v4l2src-test.c: Just a small fix to the app options. Original commit message from CVS: Just a small fix to the app options. 2006-09-19 13:08:35 +0000 Edgard Lima * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2vidorient.c: * sys/v4l2/gstv4l2vidorient.h: * tests/icles/v4l2src-test.c: Add Video Orientation interface support to v4l2src. Original commit message from CVS: Add Video Orientation interface support to v4l2src. 2006-09-19 10:53:56 +0000 Wim Taymans gst/rtsp/test.c: Fix build. Original commit message from CVS: * gst/rtsp/test.c: (main): Fix build. 2006-09-19 10:14:52 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Add ms-gsm to the src template. Original commit message from CVS: * gst/wavparse/gstwavparse.c: Add ms-gsm to the src template. 2006-09-18 17:37:46 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state), (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: Small cleanups, added documentation. Try to clean up the requests and responses. Refactor parsing the supported methods. * gst/rtsp/rtspconnection.c: (rtsp_connection_open), (rtsp_connection_create), (rtsp_connection_send), (parse_response_status), (parse_request_line), (rtsp_connection_receive), (rtsp_connection_close), (rtsp_connection_free): * gst/rtsp/rtsptransport.c: (rtsp_transport_new), (rtsp_transport_init), (rtsp_transport_parse), (rtsp_transport_free): * gst/rtsp/rtspurl.c: (rtsp_url_parse): * gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init), (sdp_message_clean), (sdp_message_free), (sdp_media_new), (sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump): Use g_return_val some more. * gst/rtsp/rtspdefs.h: Add more enum values to track initial states. * gst/rtsp/rtspmessage.c: (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_remove_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_take_body), (rtsp_message_get_body), (rtsp_message_steal_body), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Reorder arguments, object goes as the first one. Use g_return_val some more. 2006-09-18 15:36:14 +0000 Edgard Lima * sys/v4l2/v4l2src_calls.c: Fix GST_BUFFER_DURATION. Original commit message from CVS: Fix GST_BUFFER_DURATION. 2006-09-18 14:00:41 +0000 Wim Taymans gst/rtsp/gstrtspsrc.*: Export sometimes source pad with correct caps on the template, create the ghostpad from the te... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: Export sometimes source pad with correct caps on the template, create the ghostpad from the template. Remove RTCP template as we never expose RTCP. Protect against invalid body size. Avoid memcpy when creating the output buffer. Properly post an error and send EOS when the loop function is shut down. 2006-09-18 11:29:12 +0000 Lutz Mueller gst/rtsp/gstrtspsrc.*: Make sure we can never set an invalid location. Original commit message from CVS: Based on patch by: Lutz Mueller * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open), (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: Make sure we can never set an invalid location. * gst/rtsp/rtspmessage.c: (rtsp_message_steal_body): * gst/rtsp/rtspmessage.h: Added _steal_body method for future use. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free): Make freeing of NULL url return immediatly. 2006-09-18 10:42:52 +0000 Lutz Mueller gst/rtsp/gstrtspsrc.*: Use boilerplate. Original commit message from CVS: Based on patch by: Lutz Mueller * gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Use boilerplate. Make rtspsrc subclass GstBin to make state changes easier. Add Range header field on the PLAY request. 2006-09-18 08:59:17 +0000 Thijs Vermeir gst/rtsp/: Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multica... Original commit message from CVS: Based on patch by: Thijs Vermeir * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/rtspconnection.c: (inet_aton): Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multicast group. Move parsing and setting of caps to a common place. Fixes #349894. 2006-09-16 22:14:35 +0000 Stefan Kost More G_OBJECT macro fixing. Original commit message from CVS: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/wavpack/gstwavpackenc.h: * ext/xine/xineaudiodec.c: * ext/xine/xineaudiosink.c: * ext/xine/xineinput.c: * gst/chart/gstchart.c: * gst/equalizer/gstiirequalizer.c: * gst/games/gstpuzzle.c: * gst/librfb/gstrfbsrc.c: * gst/mixmatrix/mixmatrix.c: * gst/nsf/gstnsf.h: * gst/vbidec/gstvbidec.c: * gst/virtualdub/gstxsharpen.c: More G_OBJECT macro fixing. 2006-09-16 21:57:29 +0000 Stefan Kost More G_OBJECT macro fixing. Original commit message from CVS: * ext/flac/gstflactag.c: * gst/alpha/gstalpha.c: * gst/debug/breakmydata.c: * gst/debug/negotiation.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideotemplate.c: * gst/videomixer/videomixer.c: * sys/sunaudio/gstsunaudiosrc.h: More G_OBJECT macro fixing. 2006-09-16 14:30:59 +0000 Yves Lefebvre gst/avi/gstavimux.c: Correctly set the dwLength in strh. Original commit message from CVS: Patch by: Yves Lefebvre * gst/avi/gstavimux.c: (gst_avi_mux_stop_file): Correctly set the dwLength in strh. With this patch, the file duration is now displayed correctly in window media player and the AVI plays completely. Fixes #356147 2006-09-15 19:11:00 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: * tests/icles/v4l2src-test.c: The test application and the plgind error messages has been improved. Original commit message from CVS: The test application and the plgind error messages has been improved. 2006-09-15 17:10:22 +0000 Darren Kenny sys/sunaudio/gstsunaudiomixerctrl.c: Set the output track as the MASTER so that the gnome-settings-daemon keybindings... Original commit message from CVS: Patch by: Darren Kenny * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_build_list): Set the output track as the MASTER so that the gnome-settings-daemon keybindings for changing the volume using the keyboard works. Fixes #356142. 2006-09-15 16:01:48 +0000 Wim Taymans gst/multipart/multipartdemux.c: Fix documentation, it is not possible to control the framerate of jpegdec using filte... Original commit message from CVS: * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain): Fix documentation, it is not possible to control the framerate of jpegdec using filtered caps yet. Fixes #355210. Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we stop when there is an error. 2006-09-14 11:05:35 +0000 Tim-Philipp Müller gst/: Don't interpret a first buffer with an offset of NONE as 'from the middle of the stream', but only a first buff... Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag): * gst/id3demux/gstid3demux.c: (gst_id3demux_chain): Don't interpret a first buffer with an offset of NONE as 'from the middle of the stream', but only a first buffer that has a valid buffer offset that's non-zero (see #345449). 2006-09-14 10:38:42 +0000 Tim-Philipp Müller gst/icydemux/gsticydemux.*: When we merge/collect multiple incoming buffers for typefinding purposes, keep an initial... Original commit message from CVS: * gst/icydemux/gsticydemux.c: (gst_icydemux_reset), (gst_icydemux_typefind_or_forward): * gst/icydemux/gsticydemux.h: When we merge/collect multiple incoming buffers for typefinding purposes, keep an initial 0 offset on the first outgoing buffer as well (otherwise id3demux won't work right). Fixes #345449. Also Make buffer metadata writable before setting buffer caps. * tests/check/elements/icydemux.c: (typefind_succeed), (cleanup_icydemux), (push_data), (GST_START_TEST), (icydemux_suite): Small test case for the above. 2006-09-13 13:26:15 +0000 Stefan Kost gst/avi/gstavidemux.c: More code reuse and better logging in _peek_chunk(). Reintroduce check for chunk sizes before ... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_header_push), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): More code reuse and better logging in _peek_chunk(). Reintroduce check for chunk sizes before reading them (avoid oom). Better handling for invalid chunksizes when streaming. 2006-09-12 20:18:55 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Implements stop() to clear the adapter and event() to clear the adapter on FLUSH_STOP and... Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init), (gst_spectrum_start), (gst_spectrum_stop), (gst_spectrum_event): Implements stop() to clear the adapter and event() to clear the adapter on FLUSH_STOP and EOS. 2006-09-11 20:38:41 +0000 Stefan Kost gst/level/gstlevel.*: Fix type mixup in level->interval (gdouble<->guint64). Spotted by Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_set_property): * gst/level/gstlevel.h: Fix type mixup in level->interval (gdouble<->guint64). Spotted by René Stadler 2006-09-11 18:23:59 +0000 Stefan Kost gst/spectrum/gstspectrum.*: Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_init), (gst_spectrum_set_property): * gst/spectrum/gstspectrum.h: Fix type mixup in spectrum->interval (gdouble<->guint64). Spotted by René Stadler 2006-09-11 18:02:39 +0000 Stefan Kost gst/spectrum/demo-osssrc.c: Use more defines Original commit message from CVS: * gst/spectrum/demo-osssrc.c: (draw_spectrum), (main): Use more defines * gst/spectrum/gstspectrum.c: (gst_spectrum_init), (gst_spectrum_dispose), (gst_spectrum_set_caps), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: Apply some of the spectrum cleanup changes suggested in #348085. 2006-09-08 16:47:46 +0000 Tim-Philipp Müller configure.ac: Bump requirements of -base (videocrop test case needs this). Original commit message from CVS: * configure.ac: Bump requirements of -base (videocrop test case needs this). * gst/videocrop/gstvideocrop.c: Document sloppy handling of subsampled chroma planes if left/top cropping is an odd number. * tests/check/elements/videocrop.c: (handoff_cb), (videocrop_test_cropping_init_context), (videocrop_test_cropping_deinit_context), (videocrop_test_cropping), (check_1x1_buffer), (GST_START_TEST), (videocrop_suite), (main): Add another unit test that crops the input to 1x1 (and checks that that pixel has the expected values in a number of formats). 2006-09-08 11:04:24 +0000 Tim-Philipp Müller gst/videocrop/: Some quick tests indicate that it doesn't make a great deal of sense to use liboil here, at least not... Original commit message from CVS: * gst/videocrop/Makefile.am: * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init), (gst_video_crop_transform_packed), (gst_video_crop_transform_planar): Some quick tests indicate that it doesn't make a great deal of sense to use liboil here, at least not for the memcpy()s we do, so remove liboil usage until there is clear evidence it actually makes a positive difference somewhere. 2006-09-06 09:05:33 +0000 Stefan Kost gst/avi/gstavidemux.c: Revert one change to fix streaming avi (adapter size != data size). Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_sync), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data): Revert one change to fix streaming avi (adapter size != data size). 2006-09-04 16:21:17 +0000 Frédéric Riss gst/matroska/: Add support for VOBSUB subtitle tracks and zlib-compressed tracks. Make sure we start on a keyframe af... Original commit message from CVS: Patch by: Frédéric Riss * gst/matroska/matroska-demux.c: (gst_matroska_track_free), (gst_matroska_demux_reset), (gst_matroska_demux_read_track_encodings), (gst_matroska_demux_add_stream), (gst_matroska_decode_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_subtitle_caps): * gst/matroska/matroska-ids.h: Add support for VOBSUB subtitle tracks and zlib-compressed tracks. Make sure we start on a keyframe after a seek. (#343348) 2006-09-04 15:06:25 +0000 Tim-Philipp Müller gst/matroska/: not perfect yet though, needs some tweaking in flacdec; also, seeking could be better. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf), (gst_matroska_demux_push_flac_codec_priv_data), (gst_matroska_demux_push_xiph_codec_priv_data), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps): * gst/matroska/matroska-ids.h: Add basic FLAC support (#311586), not perfect yet though, needs some tweaking in flacdec; also, seeking could be better. Do better bounds checking when deserialising vorbis stream headers to make sure we don't read beyond the end of the buffer on bad input. 2006-09-04 09:34:25 +0000 Alessandro Decina ext/annodex/gstcmmldec.c: Seeking back in a file containing a CMML stream errors out if the seek goes back up to the ... Original commit message from CVS: Patch by: Alessandro Decina * ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain): Seeking back in a file containing a CMML stream errors out if the seek goes back up to the CMML headers. This is because after the seek the xml processing instruction is submitted to the xml parser again, which results in an error. The attached patch fixes the problem. Fixes #353908. * ext/annodex/gstcmmlenc.h: Fix authors name. 2006-09-03 10:46:17 +0000 Tim-Philipp Müller tests/check/elements/videocrop.c: More tests: check passthrough mode and caps transform in both directions with fixed... Original commit message from CVS: * tests/check/elements/videocrop.c: (handoff_cb), (buffer_probe_cb), (test_caps_transform), (test_passthrough), (notgst_value_list_get_nth_int), (videocrop_suite): More tests: check passthrough mode and caps transform in both directions with fixed values, ranges and lists. 2006-09-02 18:49:01 +0000 Tim-Philipp Müller docs/plugins/: Add videocrop to docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: Add videocrop to docs. * gst/videocrop/Makefile.am: * gst/videocrop/gstvideocrop.c: * gst/videocrop/gstvideocrop.h: Move boilerplate stuff and structures into a header file. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/videocrop.c: (video_crop_get_test_caps), (test_unit_sizes), (videocrop_test_cropping_init_context), (videocrop_test_cropping_deinit_context), (videocrop_test_cropping), (test_cropping), (videocrop_suite): Add unit tests for videocrop. 2006-09-02 15:30:45 +0000 Tim-Philipp Müller Port/rewrite videocrop from scratch for GStreamer-0.10, and make it support all formats videoscale supports (#345653). Original commit message from CVS: * configure.ac: * gst/videocrop/Makefile.am: * gst/videocrop/gstvideocrop.c: (gst_video_crop_base_init), (gst_video_crop_class_init), (gst_video_crop_init), (gst_video_crop_get_image_details_from_caps), (gst_video_crop_get_unit_size), (gst_video_crop_transform_packed), (gst_video_crop_transform_planar), (gst_video_crop_transform), (gst_video_crop_transform_dimension), (gst_video_crop_transform_dimension_value), (gst_video_crop_transform_caps), (gst_video_crop_set_caps), (gst_video_crop_set_property), (gst_video_crop_get_property), (plugin_init): Port/rewrite videocrop from scratch for GStreamer-0.10, and make it support all formats videoscale supports (#345653). 2006-09-02 14:45:04 +0000 Stefan Kost sys/v4l2/: Whitespace cleanups, dashify property-names. Original commit message from CVS: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_install_properties_helper): * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init): * sys/v4l2/gstv4l2src.h: Whitespace cleanups, dashify property-names. 2006-09-02 14:28:55 +0000 Stefan Kost sys/v4l2/: Cleanup error messages and unify header comments Original commit message from CVS: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open): * sys/v4l2/gstv4l2xoverlay.h: * sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities), (gst_v4l2_open): * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_queue_frame), (gst_v4l2src_capture_init): * sys/v4l2/v4l2src_calls.h: Cleanup error messages and unify header comments 2006-08-31 13:04:31 +0000 Jan Schmidt Add missing GST_LIBS to the link flags Original commit message from CVS: * ext/lame/Makefile.am: * ext/mpeg2dec/Makefile.am: * gst/dvdlpcmdec/Makefile.am: * gst/dvdsub/Makefile.am: * gst/mpegaudioparse/Makefile.am: Add missing GST_LIBS to the link flags 2006-08-30 18:01:52 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: Another small fix to set_caps function. Original commit message from CVS: Another small fix to set_caps function. 2006-08-30 13:30:13 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: Send new_segment in GST_FORMAT_TIME instead of in GST_FORMAT_BYTES. Original commit message from CVS: Send new_segment in GST_FORMAT_TIME instead of in GST_FORMAT_BYTES. 2006-08-30 11:36:06 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: A small fix to set_caps function. Original commit message from CVS: A small fix to set_caps function. 2006-08-30 11:27:40 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Reset each streams last_flow to GST_FLOW_OK. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_do_seek): Reset each streams last_flow to GST_FLOW_OK. (gst_qtdemux_activate_segment): Removing mystic modifications for good. 2006-08-30 11:07:37 +0000 Stefan Kost gst/qtdemux/qtdemux.c: put back 'segment start<=stop' change that was mystically reverted by the last commit Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment), (qtdemux_parse_tree): put back 'segment start<=stop' change that was mystically reverted by the last commit 2006-08-30 10:43:53 +0000 Stefan Kost gst/qtdemux/qtdemux.c: Fix the build for disabled debug Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment), (qtdemux_parse_tree): Fix the build for disabled debug 2006-08-29 20:59:47 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: Fixed framerate negotiation. Original commit message from CVS: Fixed framerate negotiation. 2006-08-28 17:47:29 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Make sure segment start<=stop in weird quicktime files. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment), (gst_qtdemux_add_stream), (qtdemux_parse_trak), (qtdemux_video_caps): Make sure segment start<=stop in weird quicktime files. 2006-08-28 16:59:13 +0000 Andy Wingo ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle): New helper function to lessen the ifdefs. Original commit message from CVS: 2006-08-28 Andy Wingo * ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle): New helper function to lessen the ifdefs. (GST_INFO_OBJECT): (gst_dv1394src_iso_receive): Use it. (gst_dv1394src_create): Also use the control sockets in iec61883 mode. (gst_dv1394src_start, gst_dv1394src_stop): Always use a separate handle for AVC operations; fixes #348233. 2006-08-28 14:59:05 +0000 Stefan Kost sys/v4l2/v4l2_calls.c: add comments and more debug logging Original commit message from CVS: * sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists): add comments and more debug logging 2006-08-27 17:14:06 +0000 Stefan Kost Rename again (audiofxgood -> audiofx). Original commit message from CVS: * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audiofxgood.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: * gst/audiofxgood/.cvsignore: * gst/audiofxgood/Makefile.am: * gst/audiofxgood/audiofx.c: * gst/audiofxgood/audiopanorama.c: * gst/audiofxgood/audiopanorama.h: Rename again (audiofxgood -> audiofx). 2006-08-27 13:12:52 +0000 Stefan Kost gst/avi/gstavidemux.c: Initialze variables. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan): Initialze variables. 2006-08-25 16:21:37 +0000 Wim Taymans gst/avi/gstavidemux.*: More attempts to turn this into readable code. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_init), (gst_avi_demux_finalize), (gst_avi_demux_reset), (gst_avi_demux_index_last), (gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_peek_tag), (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_loop), (gst_avi_demux_chain), (gst_avi_demux_sink_activate), (gst_avi_demux_change_state): * gst/avi/gstavidemux.h: More attempts to turn this into readable code. Don't leak adapters. Calculate duration according to index more efficiently. Don't try to act like we drive the pipeline in chain mode. 2006-08-25 09:53:18 +0000 Wim Taymans ext/annodex/gstcmmlutils.c: Fix build. Original commit message from CVS: * ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt): Fix build. 2006-08-25 09:42:43 +0000 Alessandro Decina ext/annodex/gstannodex.c: Do some extra sanity checks. Original commit message from CVS: Patch by: Alessandro Decina * ext/annodex/gstannodex.c: (gst_annodex_granule_to_time): Do some extra sanity checks. Fixes #350340. * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state), (gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip), (gst_cmml_enc_push_clip), (gst_cmml_enc_push): Check if clip->start_time is valid before adding the clip to the track list. Reset enc->preamble going from PAUSED to READY. Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is only used for EOS. Only post an error message if we were the one that created the fatal GstFlowReturn value. * ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt), (gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip): Parse the seconds field of the npt-sec time format using %llu rather than %d and check that the value scaled by GST_SECOND doesn't overflow. Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos. Lookup a clip's track with clip->track rather than clip->id which makes no sense. Identify a clip by its track and start time and not its xml id. do some more input checking and make sure we don't do undefined shifts. * tests/check/elements/cmmldec.c: (setup_cmmldec), (teardown_cmmldec), (check_output_buffer_is_equal), (push_data), (cmml_tag_message_pop), (check_headers), (push_clip_full), (push_clip), (push_empty_clip), (check_output_clip), (GST_START_TEST), (cmmldec_suite): * tests/check/elements/cmmlenc.c: (setup_cmmlenc), (teardown_cmmlenc), (check_output_buffer_is_equal), (push_data), (check_headers), (push_clip), (check_clip_times), (check_clip), (check_empty_clip), (GST_START_TEST), (cmmlenc_suite): Added some more checks. 2006-08-24 19:00:22 +0000 Stefan Kost Make also the pan-property float (saves scaling and yields better resolution) Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init), (gst_audio_panorama_set_property), (gst_audio_panorama_get_property), (gst_audio_panorama_transform_m2s_int), (gst_audio_panorama_transform_s2s_int), (gst_audio_panorama_transform_m2s_float), (gst_audio_panorama_transform_s2s_float): * gst/audiofxgood/audiopanorama.h: * tests/check/elements/audiopanorama.c: (GST_START_TEST): Make also the pan-property float (saves scaling and yields better resolution) 2006-08-24 18:23:14 +0000 Stefan Kost gst/audiofxgood/audiopanorama.c: ChangeLog surgery to add cymax's real name Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s_float), (gst_audio_panorama_transform_s2s_float): ChangeLog surgery to add cymax's real name 2006-08-24 18:17:20 +0000 Stefan Kost gst/audiofxgood/audiopanorama.*: Added float support (thanks cymax) Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s_int), (gst_audio_panorama_transform_s2s_int), (gst_audio_panorama_transform_m2s_float), (gst_audio_panorama_transform_s2s_float), (gst_audio_panorama_transform): * gst/audiofxgood/audiopanorama.h: Added float support (thanks cymax) 2006-08-24 14:16:55 +0000 Stefan Kost gst/audiofxgood/audiopanorama.c: Fix docs & debug category. Add Fixme for volume pan levels. Original commit message from CVS: * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_transform_m2s): Fix docs & debug category. Add Fixme for volume pan levels. 2006-08-24 13:51:15 +0000 Stefan Kost gst/avi/gstavidemux.c: unbreak AVI index handling, some more debug, remove an obsolete adapter_flush that caused stre... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_sync), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_chain): unbreak AVI index handling, some more debug, remove an obsolete adapter_flush that caused streaming to wander off in the wild 2006-08-24 11:21:06 +0000 Wim Taymans gst/avi/gstavidemux.*: Some more cleanups. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull): * gst/avi/gstavidemux.h: Some more cleanups. Fix totalFrames parsing in ODML. Disable use of index for length calculation in case of ODML as this is broken now. 2006-08-24 10:03:03 +0000 Tim-Philipp Müller ext/flac/gstflacdec.c: Use libgsttag helper function here too. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_update_metadata): Use libgsttag helper function here too. 2006-08-24 09:24:11 +0000 Sebastian Dröge ext/wavpack/gstwavpackdec.c: Post audio codec and average bitrate tags on bus (#344472). Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain): Post audio codec and average bitrate tags on bus (#344472). * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init), (gst_wavpack_parse_src_query): Forward queries in other formats (BYTE format in particular) upstream; add Sebastian to authors. 2006-08-24 00:40:07 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: Fix set_caps to set width and height to the values the driver is really working with. Original commit message from CVS: Fix set_caps to set width and height to the values the driver is really working with. 2006-08-23 15:33:47 +0000 Stefan Kost gst/avi/gstavidemux.*: Initial streaming support for avidemux (fixes #336465) Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_init), (gst_avi_demux_dispose), (gst_avi_demux_reset), (gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk), (gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull), (gst_avi_demux_parse_subindex), (gst_avi_demux_read_subindexes_push), (gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream), (sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_peek_tag), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push), (gst_avi_demux_stream_header_pull), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain), (gst_avi_demux_sink_activate), (gst_avi_demux_activate_push), (gst_avi_demux_change_state): * gst/avi/gstavidemux.h: Initial streaming support for avidemux (fixes #336465) 2006-08-23 10:30:31 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackenc.c: Fix mem leak, send newsegment event on correction pad as well (#352476). Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block): Fix mem leak, send newsegment event on correction pad as well (#352476). * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): Restore original author (on Sebastian's request). * tests/check/Makefile.am: * tests/check/gst-plugins-bad.supp: Add (so far empty) suppression file for -bad. Remove wavpackenc test from VALGRIND_TO_FIX now that the leak is fixed. 2006-08-23 09:22:07 +0000 Sebastian Dröge tests/check/: Add unit tests for wavpack elements (#352476). Original commit message from CVS: Patch by: Sebastian Dröge * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/wavpackdec.c: (setup_wavpackdec), (cleanup_wavpackdec), (GST_START_TEST), (wavpackdec_suite), (main): * tests/check/elements/wavpackenc.c: (setup_wavpackenc), (cleanup_wavpackenc), (GST_START_TEST), (wavpackenc_suite), (main): * tests/check/elements/wavpackparse.c: (wavpackparse_found_pad), (setup_wavpackparse), (cleanup_wavpackparse), (GST_START_TEST), (wavpackparse_suite), (main): Add unit tests for wavpack elements (#352476). 2006-08-23 08:52:50 +0000 Sebastian Dröge Add docs for wavpack elements (#352476). Original commit message from CVS: Patch by: Sebastian Dröge * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/inspect/plugin-wavpack.xml: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: * ext/wavpack/gstwavpackparse.h: Add docs for wavpack elements (#352476). 2006-08-22 20:39:26 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: Fixed query size to work with drivers that uses intermediate step like "width * height" to find closest size. Original commit message from CVS: Fixed query size to work with drivers that uses intermediate step like "width * height" to find closest size. 2006-08-22 17:20:41 +0000 Tim-Philipp Müller docs/plugins/gst-plugins-good-plugins-docs.sgml: There is no taglibmux element ... Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: There is no taglibmux element ... * gst/rtsp/gstrtspsrc.c: Use '%' rather than '&perc;' in gtk-doc blurb, docs build was complaining about unknown entity here. 2006-08-22 17:02:39 +0000 Wim Taymans gst/avi/gstavidemux.*: Mark DISCONT. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_do_seek), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry): * gst/avi/gstavidemux.h: Mark DISCONT. Remove old unused fields and reorder the struct a bit. 2006-08-22 16:45:37 +0000 Wim Taymans Small documentation updates. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: * sys/oss/gstosssink.c: (gst_oss_sink_open), (gst_oss_sink_prepare), (gst_oss_sink_unprepare): Small documentation updates. 2006-08-22 16:42:22 +0000 Wim Taymans gst/avi/gstavidemux.*: Precalc most of the duration query for each stream. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_init), (gst_avi_demux_parse_stream), (gst_avi_demux_stream_index), (gst_avi_demux_peek_tag), (gst_avi_demux_next_data_buffer), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header), (gst_avi_demux_do_seek), (gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow), (gst_avi_demux_process_next_entry), (gst_avi_demux_loop), (gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state): * gst/avi/gstavidemux.h: Precalc most of the duration query for each stream. Make seeking more correct. Use GstSegment to track position and duration. Code cleanups and leak fixes. Calculate correct total duration based on index length. 2006-08-22 13:53:34 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: If strings in text fields are marked ISO8859-1, but contain valid UTF-8 already, then han... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_text_identification_frame), (parse_insert_string_field): If strings in text fields are marked ISO8859-1, but contain valid UTF-8 already, then handle them as UTF-8 and ignore the encoding. (#351794) 2006-08-22 12:28:24 +0000 Tim-Philipp Müller ext/flac/gstflacdec.*: Make flac-in-ogg work (#352100). Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame), (gst_flac_dec_write), (gst_flac_dec_loop), (gst_flac_dec_sink_event), (gst_flac_dec_chain), (gst_flac_dec_src_query): * ext/flac/gstflacdec.h: Make flac-in-ogg work (#352100). 2006-08-22 12:10:32 +0000 Tim-Philipp Müller gst/monoscope/gstmonoscope.c: Don't unref buffers of which we've already given away ownership to the adapter. Original commit message from CVS: * gst/monoscope/gstmonoscope.c: (gst_monoscope_chain): Don't unref buffers of which we've already given away ownership to the adapter. 2006-08-22 10:32:34 +0000 Tim-Philipp Müller ext/speex/gstspeexdec.c: Make metadata extraction actually work. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments): Make metadata extraction actually work. * ext/speex/gstspeexenc.c: (gst_speexenc_base_init), (gst_speexenc_init), (gst_speexenc_create_metadata_buffer), (gst_speexenc_chain): Fix metadata writing: replace old code which wrote completely broken tags with libgsttag-based code. Plus miscellaneous code cleanups (use static pad templates etc.) and a bunch of leak fixes. 2006-08-21 19:34:03 +0000 Stefan Kost gst/audiopanorama/: die! die! die! you should never have been there Original commit message from CVS: * gst/audiopanorama/.cvsignore: * gst/audiopanorama/Makefile.am: * gst/audiopanorama/audiofx.c: * gst/audiopanorama/audiopanorama.c: * gst/audiopanorama/audiopanorama.h: die! die! die! you should never have been there 2006-08-21 16:24:28 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Some more constification. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_node_dump_foreach), (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps): Some more constification. Fix some paletted data formats again. Fix ulaw/alaw in qt. Set correct caps for raw RGB. Add support for yuv2, which is like Yuv2. Add support for raw audio with the NONE fourcc, which is like raw. 2006-08-21 13:59:52 +0000 Tim-Philipp Müller ext/wavpack/: More clean-ups: use shorter variable names to make code easier to read; prefix structures we define wit... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_init), (gst_wavpack_enc_finalize), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_format_samples), (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block), (gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state), (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset), (gst_wavpack_parse_src_query), (gst_wavpack_parse_src_event), (gst_wavpack_parse_init), (gst_wavpack_parse_get_upstream_length), (gst_wavpack_parse_loop): More clean-ups: use shorter variable names to make code easier to read; prefix structures we define with 'Gst' to make it clearer where they come from. 2006-08-21 13:26:37 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackenc.c: Fix caps set on buffers and template caps (output is framed) and make them match (#35166... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block), (gst_wavpack_enc_sink_event): Fix caps set on buffers and template caps (output is framed) and make them match (#351663); use GST_WARNING_OBJECT instead of GST_ELEMENT_WARNING; simplify push_block(); do some small clean-ups here and there; fix memleak (#351663). 2006-08-21 13:12:47 +0000 Jan Schmidt tests/check/elements/audiopanorama.c: Fix invalid memory access in audiopanorama test suite. Original commit message from CVS: * tests/check/elements/audiopanorama.c: (GST_START_TEST): Fix invalid memory access in audiopanorama test suite. 2006-08-21 11:34:41 +0000 Edward Hervey tests/check/elements/.cvsignore: ignore built file Original commit message from CVS: * tests/check/elements/.cvsignore: ignore built file 2006-08-21 10:46:21 +0000 Wim Taymans gst/rtp/Makefile.am: Fix the build again. Original commit message from CVS: * gst/rtp/Makefile.am: Fix the build again. 2006-08-21 09:21:27 +0000 Stefan Kost gst/audiofxgood/: resubmit with the desired name *again* Original commit message from CVS: * gst/audiofxgood/.cvsignore: * gst/audiofxgood/Makefile.am: * gst/audiofxgood/audiofx.c: (plugin_init): * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init), (gst_audio_panorama_class_init), (gst_audio_panorama_init), (gst_audio_panorama_set_property), (gst_audio_panorama_get_property), (gst_audio_panorama_get_unit_size), (gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s), (gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform): * gst/audiofxgood/audiopanorama.h: resubmit with the desired name *again* 2006-08-20 13:09:51 +0000 Stefan Kost use g_assert in _get_unit_size Original commit message from CVS: * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size): * gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size): use g_assert in _get_unit_size 2006-08-20 13:06:44 +0000 Stefan Kost docs/plugins/: cleanup -unused.txt to make it useful, add previously missing docs Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-audiofxgood.xml: cleanup -unused.txt to make it useful, add previously missing docs * ext/Makefile.am: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/esd/gstesd.c: (plugin_init): reflow to get rid of two external symbols * gst/audiofxgood/audiofx.c: (plugin_init): re-add 2006-08-20 12:09:16 +0000 Stefan Kost gst/audiofxgood/audiofx.c Original commit message from CVS: * configure.ac: * gst/audiofxgood/.cvsignore: * gst/audiofxgood/Makefile.am: * gst/audiofxgood/audiofx.c * gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init), (gst_audio_panorama_class_init), (gst_audio_panorama_init), (gst_audio_panorama_set_property), (gst_audio_panorama_get_property), (gst_audio_panorama_get_unit_size), (gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps), (gst_audio_panorama_transform_m2s), (gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform): * gst/audiofxgood/audiopanorama.h: * tests/check/Makefile.am: * tests/check/elements/audiopanorama.c: (setup_panorama_m), (setup_panorama_s), (cleanup_panorama), (GST_START_TEST), (panorama_suite), (main): Add audiofxgood plugin with audiopanorama element 2006-08-18 21:39:00 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.c: Fix resyncing in push mode not stopping re-syncing at embedded zeroes; skip garbage be... Original commit message from CVS: Based on patch by: Sebastian Dröge * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_sink_event), (gst_wavpack_parse_get_upstream_length), (gst_wavpack_parse_find_marker), (gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop), (gst_wavpack_parse_resync_adapter): Fix resyncing in push mode not stopping re-syncing at embedded zeroes; skip garbage between frames in pull mode as well if necessary; use gst_pad_query_peer_duration(); push EOS and NEWSEGMENT event in right direction (#351659). 2006-08-18 17:00:53 +0000 Wim Taymans docs/plugins/Makefile.am: More Oss docs fixage. Original commit message from CVS: * docs/plugins/Makefile.am: More Oss docs fixage. 2006-08-18 16:52:21 +0000 Wim Taymans gst/rtp/: Added experimental SVQ3 depayloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_base_init), (gst_rtp_sv3v_depay_class_init), (gst_rtp_sv3v_depay_init), (gst_rtp_sv3v_depay_finalize), (gst_rtp_sv3v_depay_setcaps), (gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_set_property), (gst_rtp_sv3v_depay_get_property), (gst_rtp_sv3v_depay_change_state), (gst_rtp_sv3v_depay_plugin_init): * gst/rtp/gstrtpsv3vdepay.h: Added experimental SVQ3 depayloader. 2006-08-18 13:25:06 +0000 Edward Hervey ext/dv/gstdvdemux.*: When handling seek requests, don't send the newsegment event from the calling thread. Instead sa... Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek), (gst_dvdemux_loop), (gst_dvdemux_change_state): * ext/dv/gstdvdemux.h: When handling seek requests, don't send the newsegment event from the calling thread. Instead save it so it can be sent from the streaming thread. 2006-08-17 15:51:50 +0000 Sjoerd Simons gst/multipart/multipartdemux.c: Accept leading whitespace before the boundary Original commit message from CVS: Patch by: Sjoerd Simons * gst/multipart/multipartdemux.c: (multipart_parse_header): Accept leading whitespace before the boundary This patch makes the demuxer allow some whitespace before the actual boundary. This makes the demuxer work with the ``old'' gstreamer multipartmuxer again (which placed an extra \n before the start of the stream) Fixes #349068. 2006-08-17 15:47:28 +0000 Wim Taymans gst/rtp/gstrtph264depay.c: Error out on non-implemented stuff. Original commit message from CVS: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process): Error out on non-implemented stuff. 2006-08-16 16:50:00 +0000 Andy Wingo ext/ladspa/gstsignalprocessor.c: Make ladspa elements reusable. Fixes #350006. Original commit message from CVS: Patch by: Andy Wingo * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setup), (gst_signal_processor_start), (gst_signal_processor_stop), (gst_signal_processor_cleanup), (gst_signal_processor_setcaps), (gst_signal_processor_pen_buffer), (gst_signal_processor_flush), (gst_signal_processor_do_pulls), (gst_signal_processor_do_pushes), (gst_signal_processor_change_state): Make ladspa elements reusable. Fixes #350006. 2006-08-16 15:33:12 +0000 Wim Taymans ext/ladspa/gstladspa.c: Convert ' ' into '_'. Try to keep as many characters in the padtemplate names as possible. Original commit message from CVS: * ext/ladspa/gstladspa.c: (gst_ladspa_base_init): Convert ' ' into '_'. Try to keep as many characters in the padtemplate names as possible. 2006-08-16 14:47:50 +0000 Wim Taymans ext/ladspa/gstsignalprocessor.c: A push() gives away our refcount so we should not use the buffer on the pen anymore. Original commit message from CVS: * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush), (gst_signal_processor_do_pushes): A push() gives away our refcount so we should not use the buffer on the pen anymore. 2006-08-16 13:48:00 +0000 Tim-Philipp Müller sys/oss/gstossmixerelement.c: Don't leak device string. Original commit message from CVS: * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init), (gst_oss_mixer_element_finalize): Don't leak device string. 2006-08-16 13:01:32 +0000 Tim-Philipp Müller configure.ac: Require CVS of GStreamer core and -base (for Original commit message from CVS: * configure.ac: Require CVS of GStreamer core and -base (for GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()). * ext/taglib/gstid3v2mux.cc: Write extended comment tags properly (#348762). * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_comment_frame): Extract COMM frames into extended comments, which makes it easier to properly retain the description bit of the tag and maintain this information when re-tagging (#348762). 2006-08-16 12:02:48 +0000 Tim-Philipp Müller tests/check/Makefile.am: Don't try to run annodex unit tests if the annodex plugin has not been built (Fixes #351116). Original commit message from CVS: * tests/check/Makefile.am: Don't try to run annodex unit tests if the annodex plugin has not been built (Fixes #351116). 2006-08-16 10:53:32 +0000 Tim-Philipp Müller gst/autodetect/gstautoaudiosink.c: When we can't find a usable audiosink, don't error out, but use a fake sink instea... Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best): When we can't find a usable audiosink, don't error out, but use a fake sink instead and post a warning message on the bus (#341278). 2006-08-16 10:40:04 +0000 Sebastian Dröge ext/wavpack/: In push mode, re-sync to next wavpack header if sync is lost (#351557). Also use hyphens instead of und... Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_resync_adapter), (gst_wavpack_parse_chain): In push mode, re-sync to next wavpack header if sync is lost (#351557). Also use hyphens instead of underscores in GObject property names. 2006-08-16 10:22:32 +0000 Tim-Philipp Müller sys/oss/: Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for ossmixer's new device property. Original commit message from CVS: * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init): * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for ossmixer's new device property. * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: Add docs for OSS elements. * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Update to CVS version. 2006-08-16 10:05:00 +0000 Wim Taymans gst/rtp/: Caps extra properties must be defined as strings for depayloaders because they are generated from an SDP. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpmp4gdepay.c: Caps extra properties must be defined as strings for depayloaders because they are generated from an SDP. * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init), (gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init), (gst_rtp_h264_depay_finalize), (decode_base64), (gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process), (gst_rtp_h264_depay_set_property), (gst_rtp_h264_depay_get_property), (gst_rtp_h264_depay_change_state), (gst_rtp_h264_depay_plugin_init): * gst/rtp/gstrtph264depay.h: Added basic, not completely functional RFC 3984 H264 depayloader. 2006-08-16 09:48:26 +0000 Wim Taymans gst/rtsp/gstrtpdec.c: Add pads after setting them up. Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps): Add pads after setting them up. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_combine_flows), (gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: Fix interleaved mode. - Protect streaming with lock. - Combine flows - set caps on outgoing buffers. - strip trailing \0 from data packets. - Configure RTP/RTCP in stream. Use DEBUG_OBJECT more. 2006-08-16 09:29:20 +0000 Wim Taymans gst/udp/gstmultiudpsink.c: Turn a g_print into a DEBUG line. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add): Turn a g_print into a DEBUG line. 2006-08-16 09:25:17 +0000 Wim Taymans sys/oss/: Small cleanups. Better error reporting. Original commit message from CVS: * sys/oss/gstossmixer.c: (gst_ossmixer_open), (gst_ossmixer_new): * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init), (gst_oss_mixer_element_init), (gst_oss_mixer_element_set_property), (gst_oss_mixer_element_get_property), (gst_oss_mixer_element_change_state): * sys/oss/gstossmixerelement.h: Small cleanups. Better error reporting. Add device property for the mixer instead of the hardcoded /dev/mixer. Fixes #350785. API: GstOssMixerElement::device property 2006-08-15 22:44:27 +0000 Jens Granseuer gconf/Makefile.am: Make --disable-schemas work right (they still need to be copied to the installation directory, jus... Original commit message from CVS: Patch by: Jens Granseuer * gconf/Makefile.am: Make --disable-schemas work right (they still need to be copied to the installation directory, just not applied). Fixes #351347 (also #344100). 2006-08-15 20:29:45 +0000 Sebastian Dröge ext/wavpack/gstwavpackparse.*: Make wavpackparse also work in push-mode (not seekable yet though); some small clean-u... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_class_init), (gst_wavpack_parse_reset), (gst_wavpack_parse_get_src_query_types), (gst_wavpack_parse_src_query), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_sink_event), (gst_wavpack_parse_init), (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_loop), (gst_wavpack_parse_chain), (gst_wavpack_parse_sink_activate), (gst_wavpack_parse_sink_activate_pull): * ext/wavpack/gstwavpackparse.h: Patch by: Sebastian Dröge Make wavpackparse also work in push-mode (not seekable yet though); some small clean-ups along the way; add support for SEEKING query and query types function. (#351495). 2006-08-14 11:37:10 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: * win32/common/config.h: back to HEAD Original commit message from CVS: back to HEAD 2006-08-14 11:14:43 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * win32/common/config.h: releasing 0.10.4 Original commit message from CVS: releasing 0.10.4 2006-08-14 10:06:55 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Extract all references/redirections if there is more than one and sort them; also extract mini... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_redirects_sort_func), (qtdemux_process_redirects), (qtdemux_parse_tree): Extract all references/redirections if there is more than one and sort them; also extract minimum required bitrate information if available. (#350399) 2006-08-10 14:10:28 +0000 Edward Hervey Send the newsegment event in the streaming thread. Original commit message from CVS: Patch by: Edward Hervey * configure.ac: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_data): Send the newsegment event in the streaming thread. Fixes #347529 2006-08-10 14:02:45 +0000 Thomas Vander Stichele * win32/common/config.h: bumped for prerel Original commit message from CVS: bumped for prerel 2006-08-10 13:10:38 +0000 Thomas Vander Stichele * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: update translations Original commit message from CVS: update translations 2006-08-08 14:55:53 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix silly typo. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_tree): Fix silly typo. 2006-08-08 14:46:00 +0000 Tim-Philipp Müller * ChangeLog: ChangeLog surgery: mention bug number Original commit message from CVS: ChangeLog surgery: mention bug number 2006-08-08 14:40:47 +0000 Tim-Philipp Müller ext/jpeg/: Refuse sink caps in the encoder if width or height is not a multiple of 16, the encoder does not support t... Original commit message from CVS: * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps), (gst_smokeenc_resync), (gst_smokeenc_chain): Refuse sink caps in the encoder if width or height is not a multiple of 16, the encoder does not support that yet; along the same lines, check the return value of the encoder setup function; also remove some debug log clutter. 2006-08-04 11:38:54 +0000 Andy Wingo ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing whether a processor can work in place or not, and for... Original commit message from CVS: 2006-08-04 Andy Wingo * ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing whether a processor can work in place or not, and for keeping track of its state. Change the FlowReturn instance variable from "state" to "flow_state", all callers changed. * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup) (gst_signal_processor_start, gst_signal_processor_stop) (gst_signal_processor_cleanup): New functions to manage the processor's state. (gst_signal_processor_setcaps): start() as well as setup() here. (gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE. (gst_signal_processor_change_state): Stop and cleanup the processor as we go to NULL. * ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if INPLACE_BROKEN is not set. * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare): Do the alloc_buffer in bytes, not frames. 2006-08-04 10:21:26 +0000 Zaheer Abbas Merali sys/ximage/ximageutil.c: Fix rgb masks when recording in < 24bpp. Original commit message from CVS: 2006-08-04 Zaheer Abbas Merali * sys/ximage/ximageutil.c: (ximageutil_xcontext_get): Fix rgb masks when recording in < 24bpp. 2006-08-04 09:20:26 +0000 Andy Wingo * ChangeLog: * ext/ladspa/gstsignalprocessor.c: BPB Original commit message from CVS: (gst_signal_processor_src_activate_pull): BPB 2006-08-04 09:05:53 +0000 Andy Wingo * ChangeLog: * ext/ladspa/gstsignalprocessor.c: ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps) (gst_signal_processor_prepare) (gst_signal_processor_u... Original commit message from CVS: 2006-08-04 Andy Wingo * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps) (gst_signal_processor_prepare) (gst_signal_processor_update_inputs) (gst_signal_processor_process, gst_signal_processor_pen_buffer) (gst_signal_processor_flush) (gst_signal_processor_sink_activate_push) (gst_signal_processor_src_activate_pull) (gst_signal_processor_change_state): Remove the last of the code that assumes that we process whole buffers at a time. Fix some debugging. Seems to work now in some cases. 2006-07-31 22:27:22 +0000 Andy Wingo ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process): Fix nframes-choosing. Original commit message from CVS: 2006-08-01 Andy Wingo * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process): Fix nframes-choosing. (gst_signal_processor_init): Init pending_in and pending_out. 2006-07-31 22:03:09 +0000 Andy Wingo ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No more default sample rate, although we never check tha... Original commit message from CVS: 2006-08-01 Andy Wingo * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No more default sample rate, although we never check that the sample rate actually gets set. Something for the future. (gst_signal_processor_setcaps): Some refcount fixes, flow fixes. (gst_signal_processor_event): Refcount fixen. (gst_signal_processor_process): Pull the number of frames to process from the sizes of the buffers in the input pens. (gst_signal_processor_pen_buffer): Remove an incorrect FIXME :) (gst_signal_processor_do_pulls): Add an nframes argument, and use it instead of buffer_frames. (gst_signal_processor_getrange): Refcount fixen, pass nframes on to do_pulls. (gst_signal_processor_chain) (gst_signal_processor_sink_activate_push) (gst_signal_processor_src_activate_pull): Refcount fixen. * ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay. 2006-07-31 19:44:18 +0000 Stefan Kost ext/ladspa/gstsignalprocessor.c: don't query buffer-frames from caps, add lots of debug-log, try fix for assert (#349... Original commit message from CVS: * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps), (gst_signal_processor_process): don't query buffer-frames from caps, add lots of debug-log, try fix for assert (#349189) 2006-07-31 15:58:43 +0000 Wim Taymans gst/udp/gstudpsrc.c: Fix docs. Original commit message from CVS: * gst/udp/gstudpsrc.c: Fix docs. 2006-07-29 16:32:26 +0000 Stefan Kost ext/ladspa/gstsignalprocessor.c: Add debugs logs here and there, add more error handling, add some Original commit message from CVS: * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_add_pad_from_template), (gst_signal_processor_init), (gst_signal_processor_setcaps), (gst_signal_processor_process), (gst_signal_processor_pen_buffer), (gst_signal_processor_do_pulls), (gst_signal_processor_getrange), (gst_signal_processor_sink_activate_push), (gst_signal_processor_src_activate_pull), (gst_signal_processor_change_state): Add debugs logs here and there, add more error handling, add some FIXME comments, filed #349189 2006-07-29 11:22:47 +0000 Zaheer Abbas Merali ext/jpeg/gstsmokeenc.c: Set caps on buffer correctly. Fixes bug #349155. Original commit message from CVS: 2006-07-29 Zaheer Abbas Merali * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps), (gst_smokeenc_chain): Set caps on buffer correctly. Fixes bug #349155. 2006-07-28 16:17:17 +0000 Sjoerd Simons gst/multipart/multipartdemux.c: Uses GstAdapter instead of own buffering. Original commit message from CVS: Patch by: Sjoerd Simons * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init), (gst_multipart_demux_class_init), (gst_multipart_demux_init), (gst_multipart_demux_finalize), (get_line_end), (multipart_parse_header), (multipart_find_boundary), (gst_multipart_demux_chain), (gst_multipart_demux_change_state), (gst_multipart_set_property), (gst_multipart_get_property): Uses GstAdapter instead of own buffering. Actually parses the mime-type correctly (In tests the mime-type was always "" with the old version). Uses the Content-length header if available to speed up things. Reliably autoscans the boundary name by default. Fixes #349068. * gst/multipart/multipartmux.c: (gst_multipart_mux_collected): Don't start the stream with a \n. 2006-07-28 08:32:47 +0000 Brian Cameron sys/sunaudio/gstsunaudiosrc.c: Open source with O_NONBLOCK (#349015). Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open): Open source with O_NONBLOCK (#349015). 2006-07-28 08:21:27 +0000 Stefan Kost gst/avi/gstavidemux.*: Whitespace fixes and more debug Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_index), (gst_avi_demux_massage_index): * gst/avi/gstavidemux.h: Whitespace fixes and more debug 2006-07-27 11:21:53 +0000 Tim-Philipp Müller gst/autodetect/gstautoaudiosink.c: Get rid of old and unused magic sound-server properties stuff. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_create_element_with_pretty_name), (gst_auto_audio_sink_find_best), (gst_auto_audio_sink_change_state): Get rid of old and unused magic sound-server properties stuff. Add suffix to child sink's name that makes it easy to see from the name alone which type it actually is (alsa, oss, esd, etc.). 2006-07-27 10:05:27 +0000 Wim Taymans gst/udp/gstudpsrc.*: Rename "buffer" to "buffer-size" to make clear it is a size we set and not some sort of feature ... Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_set_property), (gst_udpsrc_get_property), (gst_udpsrc_start): * gst/udp/gstudpsrc.h: Rename "buffer" to "buffer-size" to make clear it is a size we set and not some sort of feature we enable. 2006-07-27 10:01:49 +0000 Tim-Philipp Müller gst/udp/gstudpsrc.c: Use CLOSE_SOCKET() here instead of close() to maintain win32 workiness. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_start): Use CLOSE_SOCKET() here instead of close() to maintain win32 workiness. 2006-07-27 09:04:51 +0000 Thijs Vermeir gst/udp/gstudpsrc.*: Added "buffer" property to control the kernel receive buffer size. Original commit message from CVS: Patch by: Thijs Vermeir * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property), (gst_udpsrc_start): * gst/udp/gstudpsrc.h: Added "buffer" property to control the kernel receive buffer size. Update documentation. Small cleanups. Fixes #348752. API: buffer property 2006-07-26 17:09:04 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: Fix lame putting lots of 0's at start of mp3. Fixes bug #348786. Original commit message from CVS: 2006-07-26 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_setup): Fix lame putting lots of 0's at start of mp3. Fixes bug #348786. 2006-07-26 16:36:59 +0000 Kai Vehmanen gst/rtp/: Fix timestamp calculation on outgoing RTP packets. Original commit message from CVS: Patch by: Kai Vehmanen * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush), (gst_rtp_pcma_pay_handle_buffer): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush), (gst_rtp_pcmu_pay_handle_buffer): Fix timestamp calculation on outgoing RTP packets. Fixes #348675. 2006-07-26 10:07:29 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: is still sub-optimal though, since we don't retain or extract the comment descriptions pro... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Fix writing of comment frames (should be COMM not TCOM), is still sub-optimal though, since we don't retain or extract the comment descriptions properly (#334375, also see #334375). 2006-07-26 09:02:56 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: #define 'fact' RIFF chunk if we are not compiling against Original commit message from CVS: * gst/wavparse/gstwavparse.c: #define 'fact' RIFF chunk if we are not compiling against -base CVS (we don't want to depend on -base CVS for this one define only, and also not for release order reasons). 2006-07-26 08:17:45 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: Handle multiple tags of the same type properly. Re-inject unparsed ID3v2 frames that we ge... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Handle multiple tags of the same type properly. Re-inject unparsed ID3v2 frames that we get as binary blobs from id3demux into the tag again so we don't lose information when retagging (#334375). 2006-07-25 17:54:25 +0000 Tim-Philipp Müller sys/ximage/gstximagesrc.c: Document newly-added properties properly, so that there is a 'Since: 0.10.4' in the plugin... Original commit message from CVS: * sys/ximage/gstximagesrc.c: (gst_ximage_src_class_init): Document newly-added properties properly, so that there is a 'Since: 0.10.4' in the plugin docs. Convert some property names into canonical GObject style (GObject will do that internally anyway). 2006-07-25 16:47:04 +0000 Tim-Philipp Müller gst/id3demux/id3tags.c: Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as well, and add the version to... Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_add_id3v2_frame_blob_to_taglist): Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as well, and add the version to the blob's buffer caps, since that information will be needed for deserialisation later on (#348644). 2006-07-25 13:14:05 +0000 Stefan Kost gst/avi/gstavidemux.c: Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed indentation and spacing. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream): Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed indentation and spacing. 2006-07-24 21:43:06 +0000 Sébastien Moutte sys/directsound/gstdirectsoundsink.*: Add an attenuation property that will directly attenuate the directsound buffer. Original commit message from CVS: * sys/directsound/gstdirectsoundsink.h: * sys/directsound/gstdirectsoundsink.c: Add an attenuation property that will directly attenuate the directsound buffer. Change the size of the directsound secondary buffer to a half second. Add more debug logs. Add a lock to protect dsound buffer write access. Fix a bad implementation of reset. * sys/directsound/gstdirectdrawsink.c: * sys/directsound/gstdirectdrawsink.h: Add a keep_aspect_ratio property. Do not use overlay if not supported. Add more debug logs. Remove overwrite of WM_ERASEBKGND message handling. It was not redrawing border when keep_aspect_ratio was enabled. * win32/common/config.h: update version waiting an auto-generated config.h 2006-07-24 15:25:49 +0000 Tim-Philipp Müller docs/plugins/: Update files to CVS/Prerelease version, add esdsink docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: Update files to CVS/Prerelease version, add esdsink docs. * ext/esd/esdsink.c: Add gtk-doc blurb. * gst/rtp/gstrtpmp4vpay.c: Fix typo in element description. 2006-07-24 14:54:04 +0000 Tim-Philipp Müller * ChangeLog: ChangeLog surgery: fix Stefan's e-mail address Original commit message from CVS: ChangeLog surgery: fix Stefan's e-mail address 2006-07-24 14:49:19 +0000 Tim-Philipp Müller ext/esd/esdsink.c: Prevent libesd from auto-spawning a sound daemon if it is not already running. Now that we don't d... Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_open), (gst_esdsink_factory_init): Prevent libesd from auto-spawning a sound daemon if it is not already running. Now that we don't do evil stuff like that any longer we can give esdsink a rank so that autoaudiosink will try it as well if all other audio sinks fail (#343051). 2006-07-24 14:42:11 +0000 Tim-Philipp Müller ext/esd/Makefile.am: Oops, need to remove README from EXTRA_DIST as well. Original commit message from CVS: * ext/esd/Makefile.am: Oops, need to remove README from EXTRA_DIST as well. 2006-07-24 14:37:36 +0000 Tim-Philipp Müller ext/esd/README: Remove, it contains nothing useful anyway. Original commit message from CVS: * ext/esd/README: Remove, it contains nothing useful anyway. * ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare), (gst_esdsink_delay): Some small clean-ups; use GST_BOILERPLATE etc. 2006-07-24 14:16:06 +0000 Wim Taymans gst/law/: Fix negotiation to deal with ANY/EMPTY caps instead of leaking. Original commit message from CVS: * gst/law/alaw-decode.c: (alawdec_getcaps): * gst/law/alaw-encode.c: (alawenc_getcaps), (gst_alawenc_chain): * gst/law/mulaw-decode.c: (mulawdec_getcaps): * gst/law/mulaw-encode.c: (mulawenc_getcaps): Fix negotiation to deal with ANY/EMPTY caps instead of leaking. 2006-07-24 13:40:56 +0000 Stefan Kost gst/wavparse/gstwavparse.*: Use information from 'fact' chunk for length calculation of compressed samples. Calculate... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_reset), (gst_wavparse_other), (gst_wavparse_perform_seek), (gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_pad_query): * gst/wavparse/gstwavparse.h: Use information from 'fact' chunk for length calculation of compressed samples. Calculate bps if bogus value is found in wav header (embeded mp2/mp3). 2006-07-24 11:48:03 +0000 Joni Valtanen Port udp plugin to win32 (#345288). Original commit message from CVS: Based on patch by: Joni Valtanen * configure.ac: * gst/udp/Makefile.am: * gst/udp/gstdynudpsink.c: (gst_dynudpsink_init), (gst_dynudpsink_finalize), (gst_dynudpsink_close): * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init), (gst_multiudpsink_finalize), (gst_multiudpsink_close): * gst/udp/gstmultiudpsink.h: * gst/udp/gstudp.c: (plugin_init): * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create), (gst_udpsrc_start), (gst_udpsrc_stop): * gst/udp/gstudpsrc.h: * gst/udp/gstudpnetutils.c: (gst_udp_net_utils_win32_inet_aton), (gst_udp_net_utils_win32_wsa_startup): * gst/udp/gstudpnetutils.h: Port udp plugin to win32 (#345288). 2006-07-24 11:00:34 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Remove unwanted DEBUG line. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (rtsp_connection_send): Remove unwanted DEBUG line. 2006-07-23 11:33:54 +0000 Tim-Philipp Müller gst/id3demux/: On second thought, it might be wiser and more efficient not to do tag registration from a streaming th... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (plugin_init): * gst/id3demux/id3tags.c: (id3demux_add_id3v2_frame_blob_to_taglist): * gst/id3demux/id3tags.h: On second thought, it might be wiser and more efficient not to do tag registration from a streaming thread. 2006-07-23 10:56:27 +0000 Tim-Philipp Müller gst/id3demux/id3tags.c: Put ID3v2 frames we can't parse as binary blobs into private tags, so that they are not lost ... Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_add_id3v2_frame_blob_to_taglist), (id3demux_id3v2_frames_to_tag_list): Put ID3v2 frames we can't parse as binary blobs into private tags, so that they are not lost when retagging, at least once id3v2mux has been taught to re-inject those frames again. See bug #334375. 2006-07-21 10:57:00 +0000 Wim Taymans gst/avi/gstavidemux.c: Fix some leaks. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_process_next_entry): Fix some leaks. * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list): Don't use \n in debug lines. 2006-07-20 18:48:32 +0000 Stefan Kost docs/plugins/: Add annodex and icydemux, cleanup the sections a bit Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: Add annodex and icydemux, cleanup the sections a bit 2006-07-19 14:36:00 +0000 Martin Szulecki sys/v4l2/gstv4l2object.c: If "device-name" is requested and the device is not open, try to temporarily open it to obt... Original commit message from CVS: Patch by: Martin Szulecki * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_get_property_helper): If "device-name" is requested and the device is not open, try to temporarily open it to obtain this information (#342494). 2006-07-19 11:52:53 +0000 Alex Lancaster ext/taglib/gstid3v2mux.cc: Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as Original commit message from CVS: Patch by: Alex Lancaster * ext/taglib/gstid3v2mux.cc: Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as ID3v2 TSSE frames (#347898). 2006-07-19 07:40:52 +0000 Tim-Philipp Müller * ChangeLog: ChangeLog surgery: mention fixed bug Original commit message from CVS: ChangeLog surgery: mention fixed bug 2006-07-18 19:59:01 +0000 Stefan Kost gst/avi/gstavimux.c: Respect mpegversion for "video/mpeg" and give message in case of unhandled versions. Original commit message from CVS: * gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps): Respect mpegversion for "video/mpeg" and give message in case of unhandled versions. 2006-07-18 18:05:15 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackdec.c: Fix caps after previous change to byte order endianness. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): Fix caps after previous change to byte order endianness. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset), (gst_wavpack_parse_sink_event), (gst_wavpack_parse_init), (gst_wavpack_parse_loop): * ext/wavpack/gstwavpackparse.h: Queue incoming events if there's no source pad yet and send them downstream later when the pad is there. 2006-07-18 16:47:25 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackdec.*: Output audio in native byte order (which is also how we get samples from wavpack); outpu... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init), (gst_wavpack_dec_format_samples), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain), (gst_wavpack_dec_change_state): * ext/wavpack/gstwavpackdec.h: Output audio in native byte order (which is also how we get samples from wavpack); output samples with 21-24 bit depth with 32 bit width (makes things easier for us). 2006-07-18 15:53:35 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackdec.*: More clean-ups: remove most of the disfunctional correction pad stuff for now, if it eve... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init), (gst_wavpack_dec_class_init), (gst_wavpack_dec_init), (gst_wavpack_dec_finalize), (gst_wavpack_dec_format_samples), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event), (gst_wavpack_dec_change_state): * ext/wavpack/gstwavpackdec.h: More clean-ups: remove most of the disfunctional correction pad stuff for now, if it ever gets implemented a lot of stuff will have to be rewritten anyway; redo chain function, move errors to end, error out instead of g_assert()ing. Also rename overly long variable 'wavpackdec' to just 'dec'; miscellaneous other small stuff. 2006-07-18 14:08:06 +0000 Sebastian Dröge configure.ac: Check for wavpack version and define WAVPACK_OLD_API if necessary. Original commit message from CVS: Patch by: Sebastian Dröge * configure.ac: Check for wavpack version and define WAVPACK_OLD_API if necessary. * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpackcommon.c: (gst_wavpack_read_header), (gst_wavpack_read_metadata): * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init), (gst_wavpack_dec_class_init), (gst_wavpack_dec_init), (gst_wavpack_dec_finalize), (gst_wavpack_dec_format_samples), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event), (gst_wavpack_dec_change_state), (gst_wavpack_dec_request_new_pad), (gst_wavpack_dec_plugin_init): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init), (gst_wavpack_enc_init), (gst_wavpack_enc_finalize), (gst_wavpack_enc_set_wp_config): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init), (gst_wavpack_parse_finalize), (gst_wavpack_parse_class_init), (gst_wavpack_parse_index_get_entry_from_sample), (gst_wavpack_parse_scan_to_find_sample), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_create_src_pad): * ext/wavpack/gstwavpackstreamreader.c: * ext/wavpack/gstwavpackstreamreader.h: Port to new/official wavpack API, don't use API that was exported in wavpack header files and in the lib but meant to be private, at least not for recent wavpack versions; misc. 'cleanups' (#347443). 2006-07-17 10:25:57 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Store duration in uint64 too instead of clipping. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek), (gst_qtdemux_prepare_current_sample), (gst_qtdemux_loop_state_movie): Store duration in uint64 too instead of clipping. When we do a keyframe seek and the requested time is at the keyframe, don't seek back to the beginning of the keyframe. Fixes #347439. 2006-07-17 10:22:54 +0000 Wim Taymans ext/libpng/gstpngdec.*: Use statically allocated segment instead of leaking. Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_init), (buffer_clip), (gst_pngdec_caps_create_and_set), (gst_pngdec_task), (gst_pngdec_chain), (gst_pngdec_sink_event), (gst_pngdec_libpng_init), (gst_pngdec_change_state), (gst_pngdec_sink_activate_push): * ext/libpng/gstpngdec.h: Use statically allocated segment instead of leaking. Various cleanups. Fix flush and seek handling. 2006-07-16 14:31:48 +0000 Wim Taymans gst/rtp/: Added simple generic mpeg4 depayloader. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_base_init), (gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_set_property), (gst_rtp_mp4g_depay_get_property), (gst_rtp_mp4g_depay_change_state), (gst_rtp_mp4g_depay_plugin_init): * gst/rtp/gstrtpmp4gdepay.h: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init), (gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush): Added simple generic mpeg4 depayloader. Fix generic mpeg4 payloader. 2006-07-15 15:25:05 +0000 Tim-Philipp Müller gst/rtsp/gstrtspsrc.c: Don't try doing state changes on a NULL pointer. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state): Don't try doing state changes on a NULL pointer. 2006-07-15 11:50:25 +0000 Stefan Kost gst/spectrum/: Do not use deprecated gtk functions. Original commit message from CVS: * gst/spectrum/demo-audiotest.c: (main): * gst/spectrum/demo-osssrc.c: (main): Do not use deprecated gtk functions. 2006-07-14 13:33:54 +0000 Sebastien Cote gst/rtp/gstrtpamrdepay.*: rtpamrdec isn't a subclass of GstBaseRtpDepayload. Original commit message from CVS: Patch by: Sebastien Cote * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_base_init), (gst_rtp_amr_depay_class_init), (gst_rtp_amr_depay_init), (gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process): * gst/rtp/gstrtpamrdepay.h: rtpamrdec isn't a subclass of GstBaseRtpDepayload. Fixes #321191 2006-07-14 12:01:05 +0000 Zaheer Abbas Merali sys/ximage/gstximagesrc.c: Fix segfault when moving mouse pointer to the bottom right corner. Original commit message from CVS: 2006-07-14 Zaheer Abbas Merali * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get), (gst_ximage_src_get_caps), (gst_ximage_src_class_init): Fix segfault when moving mouse pointer to the bottom right corner. 2006-07-13 15:22:20 +0000 Thomas Vander Stichele * common: * docs/plugins/inspect/plugin-qtdemux.xml: remove sdlvideosink plugin and update the rest Original commit message from CVS: remove sdlvideosink plugin and update the rest 2006-07-12 09:34:15 +0000 Wim Taymans gst/rtp/: Added mpeg2 TS depayloader. Closing #347234. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_base_init), (gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process), (gst_rtp_mp2t_depay_set_property), (gst_rtp_mp2t_depay_get_property), (gst_rtp_mp2t_depay_change_state), (gst_rtp_mp2t_depay_plugin_init): * gst/rtp/gstrtpmp2tdepay.h: Added mpeg2 TS depayloader. Closing #347234. 2006-07-12 09:28:46 +0000 Tim-Philipp Müller gst/spectrum/gstspectrum.c: Fix typo in property nick. Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): Fix typo in property nick. 2006-07-11 22:46:47 +0000 Tim-Philipp Müller ext/cdio/gstcdiocddasrc.c: Remove g_assert that shouldn't be there. Original commit message from CVS: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_close): Remove g_assert that shouldn't be there. 2006-07-10 20:11:34 +0000 Edward Hervey gst/avi/gstavidemux.*: Don't push tag events found by gst_riff_parse_info() before outputting Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_stream_header), (push_tag_lists): * gst/avi/gstavidemux.h: Don't push tag events found by gst_riff_parse_info() before outputting GST_EVENT_NEWSEGMENT. 2006-07-10 16:41:57 +0000 Wim Taymans gst/rtsp/: replaced closesocket and close in code with one CLOSE_SOCKET. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/rtspconnection.c: (rtsp_connection_send), (rtsp_connection_close): * gst/rtsp/rtspdefs.h: replaced closesocket and close in code with one CLOSE_SOCKET. Some more cleanups. Fixes #345301. 2006-07-10 15:26:39 +0000 Tim-Philipp Müller gst/autodetect/gstautoaudiosink.c: Fix example pipeline in docs. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: Fix example pipeline in docs. 2006-07-10 14:49:46 +0000 Wim Taymans gst/filter/: Don't forget new files. Original commit message from CVS: * gst/filter/gstbpwsinc.h: * gst/filter/gstiir.h: * gst/filter/gstlpwsinc.h: Don't forget new files. 2006-07-10 14:42:15 +0000 Mathis Hofer Ported the gstfilter plugin to GStreamer 0.10. Original commit message from CVS: Patch by: Mathis Hofer * configure.ac: * gst/filter/Makefile.am: * gst/filter/gstbpwsinc.c: (gst_bpwsinc_dispose), (gst_bpwsinc_base_init), (gst_bpwsinc_class_init), (gst_bpwsinc_init), (bpwsinc_set_caps), (bpwsinc_transform_ip), (bpwsinc_set_property), (bpwsinc_get_property): * gst/filter/gstfilter.c: (plugin_init): * gst/filter/gstfilter.h: * gst/filter/gstiir.c: (gst_iir_dispose), (gst_iir_base_init), (gst_iir_class_init), (gst_iir_init), (iir_set_caps), (iir_transform_ip), (iir_set_property), (iir_get_property): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_dispose), (gst_lpwsinc_base_init), (gst_lpwsinc_class_init), (gst_lpwsinc_init), (lpwsinc_set_caps), (lpwsinc_transform_ip), (lpwsinc_set_property), (lpwsinc_get_property): Ported the gstfilter plugin to GStreamer 0.10. 2006-07-10 10:21:57 +0000 Rob Taylor gst/udp/gstmultiudpsink.c: If a destination is added before the stream is set to PAUSED, the multicast group is not j... Original commit message from CVS: Patch by: Rob Taylor * gst/udp/gstmultiudpsink.c: (join_multicast), (gst_multiudpsink_init_send), (gst_multiudpsink_add): If a destination is added before the stream is set to PAUSED, the multicast group is not joined as the socket is not created yet. Also TTL and LOOP should also be set. Fixes #346921. 2006-07-10 09:57:26 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Extract comment information!! Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta): Extract comment information!! 2006-07-10 09:46:25 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Extract year/date information (fixes #347079). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta), (qtdemux_tag_add_date): Extract year/date information (fixes #347079). 2006-07-08 22:41:25 +0000 Zaheer Abbas Merali sys/ximage/gstximagesrc.*: Fix use-damage property to actually work :) Original commit message from CVS: 2006-07-09 Zaheer Abbas Merali * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get), (gst_ximage_src_set_property), (gst_ximage_src_get_property), (gst_ximage_src_get_caps), (gst_ximage_src_class_init), (gst_ximage_src_init): * sys/ximage/gstximagesrc.h: Fix use-damage property to actually work :) Add startx, starty, endx, endy properties so screencasts other than full screen ones can work. 2006-07-08 19:03:54 +0000 Zaheer Abbas Merali sys/ximage/gstximagesrc.*: Add use_damage property to offer ability to choose whether to use Original commit message from CVS: 2006-07-08 Zaheer Abbas Merali * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get), (gst_ximage_src_set_property), (gst_ximage_src_get_property), (gst_ximage_src_class_init), (gst_ximage_src_init): * sys/ximage/gstximagesrc.h: Add use_damage property to offer ability to choose whether to use XDamage or not. 2006-07-07 15:04:29 +0000 Wim Taymans gst/goom/filters.c: Avoid goom coredumping by clearing memory. Original commit message from CVS: * gst/goom/filters.c: (zoomFilterSetResolution): Avoid goom coredumping by clearing memory. Fixes 345679. 2006-07-07 14:30:26 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Don't crash on twos/sowt/raw audio. #345830. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Don't crash on twos/sowt/raw audio. #345830. 2006-07-05 20:21:02 +0000 Sébastien Moutte win32/vs6/libgstid3demux.dsp: Add a link to libgsttag-0.10.lib. Original commit message from CVS: * win32/vs6/libgstid3demux.dsp: Add a link to libgsttag-0.10.lib. 2006-07-05 14:52:13 +0000 Tim-Philipp Müller gst/: Don't return FLOW_UNEXPECTED when a buffer is before the start of the stream (which might happen with large ID3... Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer), (gst_tag_demux_read_range): * gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer), (gst_id3demux_read_range): Don't return FLOW_UNEXPECTED when a buffer is before the start of the stream (which might happen with large ID3v2 tags if the tag reading was done pullrange based and we then switched to push mode later on). Fixes regression introduced by commit from June 29th. 2006-07-05 10:14:16 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: Make UTF-8 the default encoding when writing string tags (before, our UTF-8 strings would ... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Make UTF-8 the default encoding when writing string tags (before, our UTF-8 strings would automatically be converted to ISO-8859-1 by taglib and written as ISO-8859-1 fields if that was possible). * tests/check/elements/id3v2mux.c: (utf8_string_in_buf), (test_taglib_id3mux_check_tag_buffer), (identity_cb), (test_taglib_id3mux_with_tags): Add test case that makes sure our UTF-8 strings have actually been written into the tag as UTF-8. 2006-07-04 16:00:26 +0000 Tim-Philipp Müller configure.ac: Let's try that again. Original commit message from CVS: * configure.ac: Let's try that again. 2006-07-04 15:40:47 +0000 Tim-Philipp Müller configure.ac: Disable monoscope plugin for now until it fulfills all the requirements. Original commit message from CVS: * configure.ac: Disable monoscope plugin for now until it fulfills all the requirements. 2006-07-03 20:35:45 +0000 Tim-Philipp Müller Port monoscope visualisation to 0.10. Original commit message from CVS: * configure.ac: * gst/monoscope/Makefile.am: * gst/monoscope/gstmonoscope.c: (gst_monoscope_base_init), (gst_monoscope_class_init), (gst_monoscope_init), (gst_monoscope_finalize), (gst_monoscope_reset), (gst_monoscope_sink_setcaps), (gst_monoscope_src_setcaps), (gst_monoscope_src_negotiate), (get_buffer), (gst_monoscope_chain), (gst_monoscope_sink_event), (gst_monoscope_src_event), (gst_monoscope_change_state), (plugin_init): * gst/monoscope/gstmonoscope.h: Port monoscope visualisation to 0.10. 2006-07-03 20:02:56 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix silly crasher in state change function; add Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state), (gst_qtdemux_loop_state_header), (qtdemux_video_caps): Fix silly crasher in state change function; add IV41 fourcc (see bug #171111); don't output confusing debug message when skipping atoms. 2006-07-03 16:43:10 +0000 Tim-Philipp Müller gst/: Return FLOW_UNEXPECTED when at the end of the file, not Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain): * gst/id3demux/gstid3demux.c: (gst_id3demux_chain): Return FLOW_UNEXPECTED when at the end of the file, not FLOW_ERROR. Fixes 'internal stream error' errors that would sometimes occur in totem when scrubbing to the end of an ID3v1 tagged mp3 file. 2006-07-03 15:31:22 +0000 Edward Hervey ext/libpng/gstpngdec.*: Implement buffer clipping/dropping using GstSegment. Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_info_callback), (buffer_clip), (user_end_callback), (gst_pngdec_chain), (gst_pngdec_sink_event), (gst_pngdec_change_state): * ext/libpng/gstpngdec.h: Implement buffer clipping/dropping using GstSegment. This provides accurate seeking. 2006-07-03 15:28:48 +0000 Edward Hervey gst/avi/gstavidemux.*: Proper aggregation of each stream's GstFlowReturn in order to figure out whether the task shou... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream), (gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow), (gst_avi_demux_process_next_entry), (push_tag_lists), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Proper aggregation of each stream's GstFlowReturn in order to figure out whether the task should stop or not. Don't send inline events before pushing out a NEW_SEGMENT, more specifically for GST_TAG_EVENT. Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading sub-indexes. 2006-06-30 07:11:24 +0000 Brian Cameron sys/sunaudio/gstsunaudiomixerctrl.c: Move "Monitor" slider to input tab so it works more like sdtaudiocontrol, which ... Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_build_list): Move "Monitor" slider to input tab so it works more like sdtaudiocontrol, which is what people on Solaris are used to using for their mixer program (#346259). 2006-06-29 14:50:18 +0000 Thomas Vander Stichele tests/check/elements/level.c: fix a leak, clean up at the end Original commit message from CVS: * tests/check/elements/level.c: (GST_START_TEST): fix a leak, clean up at the end 2006-06-29 11:41:55 +0000 Tim-Philipp Müller gst/matroska/: Send tag event after newsegment event. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_send_event), (gst_matroska_demux_loop_stream_parse_id): * gst/matroska/matroska-ids.h: Send tag event after newsegment event. 2006-06-29 11:11:50 +0000 Tim-Philipp Müller gst/id3demux/gstid3demux.c: Make sure we don't return GST_FLOW_OK with a NULL buffer in certain cases where a read be... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer), (gst_id3demux_read_range): Make sure we don't return GST_FLOW_OK with a NULL buffer in certain cases where a read beyond the end of the file is requested. Fixes #345930. * gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer), (gst_tag_demux_read_range): Fix same issue here as well. 2006-06-29 11:05:14 +0000 Zaheer Abbas Merali sys/ximage/gstximagesrc.c: Fix hypothetical crash. Original commit message from CVS: 2006-06-29 Zaheer Abbas Merali * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get): Fix hypothetical crash. 2006-06-28 08:36:30 +0000 Brian Cameron sys/sunaudio/gstsunaudiosink.c: Do not modify the ports value. If the user has turned off the built-in speakers, then... Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare): Do not modify the ports value. If the user has turned off the built-in speakers, then we should not reset it in the prepare function, since this causes the built-in speakers to turn back on anytime the user changes a track in totem, rhythmbox, etc. (#346066). 2006-06-23 09:35:45 +0000 Wim Taymans gst/goom/gstgoom.c: Fix double caps unref when negotiation fails. Original commit message from CVS: * gst/goom/gstgoom.c: (gst_goom_src_negotiate): Fix double caps unref when negotiation fails. 2006-06-22 19:31:04 +0000 Tim-Philipp Müller Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) plus two minor macro fixes. Original commit message from CVS: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/annodex/gstcmmlparser.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalphacolor.c: * gst/cutter/gstcutter.c: * gst/debug/gstnavigationtest.c: * gst/icydemux/gsticydemux.c: * gst/level/gstlevel.c: * gst/multipart/multipart.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstvideoflip.c: Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) plus two minor macro fixes. 2006-06-22 16:27:03 +0000 Tim-Philipp Müller gst/matroska/: Try to fix up broken matroska files containing subtitle streams with non-UTF8 character encodings (cou... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_check_subtitle_buffer), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_subtitle_caps): * gst/matroska/matroska-ids.c: (gst_matroska_track_init_subtitle_context): * gst/matroska/matroska-ids.h: Try to fix up broken matroska files containing subtitle streams with non-UTF8 character encodings (courtesy of mkvmerge) using either the encoding specified in the GST_SUBTITLE_ENCODING environment variable or the current locale's character set if it is non-UTF8. Fixes #337076. 2006-06-22 12:17:13 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Set image type from APIC frame as "image-type" field of GST_TAG_IMAGE buffer caps (#344605). Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_picture_frame): Set image type from APIC frame as "image-type" field of GST_TAG_IMAGE buffer caps (#344605). 2006-06-20 19:40:29 +0000 Tim-Philipp Müller ext/flac/: Support chain-based operation, should make flac-over-DAAP work (#340492). Original commit message from CVS: * ext/flac/Makefile.am: * ext/flac/gstflacdec.c: (gst_flac_dec_init), (gst_flac_dec_reset_decoders), (gst_flac_dec_setup_seekable_decoder), (gst_flac_dec_setup_stream_decoder), (gst_flac_dec_finalize), (gst_flac_dec_metadata_callback), (gst_flac_dec_metadata_callback_seekable), (gst_flac_dec_metadata_callback_stream), (gst_flac_dec_error_callback), (gst_flac_dec_error_callback_seekable), (gst_flac_dec_error_callback_stream), (gst_flac_dec_read_seekable), (gst_flac_dec_read_stream), (gst_flac_dec_write), (gst_flac_dec_write_seekable), (gst_flac_dec_write_stream), (gst_flac_dec_loop), (gst_flac_dec_sink_event), (gst_flac_dec_chain), (gst_flac_dec_convert_sink), (gst_flac_dec_get_sink_query_types), (gst_flac_dec_sink_query), (gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query), (gst_flac_dec_handle_seek_event), (gst_flac_dec_sink_activate), (gst_flac_dec_sink_activate_push), (gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state): * ext/flac/gstflacdec.h: Support chain-based operation, should make flac-over-DAAP work (#340492). 2006-06-20 15:35:05 +0000 Wim Taymans docs/plugins/gst-plugins-good-plugins-sections.txt: Doc updates, merge some unused symbols. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins-sections.txt: Doc updates, merge some unused symbols. 2006-06-20 14:57:09 +0000 Wim Taymans Added documentation for the rtsp plugin. Fixes #345393. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init): * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: Added documentation for the rtsp plugin. Fixes #345393. 2006-06-20 12:10:29 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Use better G_OS_* macros. Fixes #345301 some more. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send), (rtsp_connection_close), (rtsp_connection_free): Use better G_OS_* macros. Fixes #345301 some more. 2006-06-20 10:35:48 +0000 Brian Cameron sys/sunaudio/: Add a SunAudio source plugin. Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/Makefile.am: * sys/sunaudio/gstsunaudio.c: (plugin_init): * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new), (gst_sunaudiomixer_ctrl_list_tracks), (gst_sunaudiomixer_ctrl_get_volume), (gst_sunaudiomixer_ctrl_set_volume), (gst_sunaudiomixer_ctrl_set_mute), (gst_sunaudiomixer_ctrl_set_record): * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixertrack.c: (gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new): * sys/sunaudio/gstsunaudiomixertrack.h: * sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose), (gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init), (gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property), (gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps), (gst_sunaudiosrc_open), (gst_sunaudiosrc_close), (gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare), (gst_sunaudiosrc_read), (gst_sunaudiosrc_delay), (gst_sunaudiosrc_reset): * sys/sunaudio/gstsunaudiosrc.h: Add a SunAudio source plugin. Support stereo and right/left channel gain in the mixer plugin. Support the RECORD flag so that you can switch between line-input and microphone in gnome-volume-control. Code cleanups like using an enumerator for track number instead of an integer. Fixes #344923. 2006-06-20 10:31:41 +0000 Joni Valtanen gst/rtsp/rtspconnection.c: Make RTSP plugin compile on windows. Fixes #345301. Original commit message from CVS: Patch by: Joni Valtanen * gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send), (rtsp_connection_close): Make RTSP plugin compile on windows. Fixes #345301. Some changes to original patch to catch errors better. use ifdef WIN32 instead of ifndef. 2006-06-19 10:00:18 +0000 Zaheer Abbas Merali configure.ac: If we have libraw1394 >= 1.2.1, then we need libiec61883. Original commit message from CVS: 2006-06-19 Zaheer Abbas Merali * configure.ac: If we have libraw1394 >= 1.2.1, then we need libiec61883. 2006-06-18 14:00:19 +0000 Edward Hervey ext/jpeg/gstjpegdec.c: After a failed buffer alloc, we need to abort the jpeg decoding (it started when parsing heade... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): After a failed buffer alloc, we need to abort the jpeg decoding (it started when parsing headers to figure out how many bytes we need to request downstream). 2006-06-18 12:37:12 +0000 Mark Nauwelaerts gst/wavparse/gstwavparse.c: Make sure we don't read beyond the end of the file (#345232). Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek): Make sure we don't read beyond the end of the file (#345232). 2006-06-17 14:35:37 +0000 Tim-Philipp Müller configure.ac: Fix --disable-external (can't set conditionals conditionally, #343602). Original commit message from CVS: * configure.ac: Fix --disable-external (can't set conditionals conditionally, #343602). 2006-06-16 12:35:08 +0000 Zaheer Abbas Merali gst/spectrum/Makefile.am: Fix build. Original commit message from CVS: 2006-06-16 Zaheer Abbas Merali * gst/spectrum/Makefile.am: Fix build. 2006-06-16 10:56:24 +0000 Tim-Philipp Müller Use GST_PLUGIN_DOCS, --enable-plugin-docs etc. Original commit message from CVS: * autogen.sh: * configure.ac: * docs/Makefile.am: Use GST_PLUGIN_DOCS, --enable-plugin-docs etc. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect/plugin-taglib.xml: Add/fix apev2mux docs. 2006-06-16 09:49:07 +0000 Stefan Kost gst/spectrum/: port to use message to get results, cleanly exit when closing the window Original commit message from CVS: * gst/spectrum/demo-audiotest.c: (on_window_destroy), (draw_spectrum), (message_handler), (main): * gst/spectrum/demo-osssrc.c: (on_window_destroy), (draw_spectrum), (message_handler), (main): port to use message to get results, cleanly exit when closing the window * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_dispose), (gst_spectrum_set_property), (gst_spectrum_get_property), (gst_spectrum_set_caps), (gst_spectrum_start), (gst_spectrum_message_new), (gst_spectrum_transform_ip): * gst/spectrum/gstspectrum.h: port to derive from basetransform and send results via messages (like level element) 2006-06-15 15:58:09 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Combine return values from src pad pushes. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek), (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_parse_trak): Combine return values from src pad pushes. 2006-06-15 08:50:09 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Don't crash on files with 0 samples, EOS immediatly instead. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header), (gst_qtdemux_prepare_current_sample), (gst_qtdemux_advance_sample), (gst_qtdemux_add_stream): Don't crash on files with 0 samples, EOS immediatly instead. Fixes #344944. 2006-06-14 15:59:56 +0000 Wim Taymans ext/dv/gstdvdec.c: Reset segment info on flush. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_class_init), (gst_dvdec_init), (gst_dvdec_finalize), (gst_dvdec_sink_event), (gst_dvdec_change_state): Reset segment info on flush. Alloc segment in _init, free in _finalize. * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek): Don't send segments twice. 2006-06-14 15:07:22 +0000 Wim Taymans ext/dv/gstdvdemux.c: Respect segment.stop. Fixes #342592. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame): Respect segment.stop. Fixes #342592. 2006-06-14 11:28:41 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: No language specified means the implied language is English according to the matroska ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream): No language specified means the implied language is English according to the matroska spec (partially fixes #344708); add some more debug output. 2006-06-14 09:32:27 +0000 Sebastian Dröge ext/wavpack/gstwavpackenc.*: Use bitrate property solely for bitrates and add new bits-per-sample property for the ot... Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_base_init), (gst_wavpack_enc_class_init), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_chain), (gst_wavpack_enc_sink_event), (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property): * ext/wavpack/gstwavpackenc.h: Use bitrate property solely for bitrates and add new bits-per-sample property for the other stuff. Set duration to 'unknown' in initial header and resend header with proper duration on EOS; update Sebastian's e-mail address. 2006-06-14 08:06:43 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: When operating chain-based, don't make any assumptions about the chunking of the incoming... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_chain): When operating chain-based, don't make any assumptions about the chunking of the incoming data and make streaming work on days other than the second Thursday after a full moon. Also fix up debug messages here and there and make use of the most excellent new gst_pad_query_peer_duration() utility function. Skip any 'bext' chunks in front of the 'fmt ' chunk. Fixes #343837. * gst/wavparse/gstwavparse.h: Remove trailing comma after last enum value, some compilers don't like that. 2006-06-13 17:05:25 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Handle premature EOS gracefully. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_data): Handle premature EOS gracefully. 2006-06-13 09:54:26 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Prevent out of bounds array access when scrubbing towards the end of the file between the last... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek): Prevent out of bounds array access when scrubbing towards the end of the file between the last index entry and the end. Fixes occasional 'start <= stop' newsegment event assertions when scrubbing in MJPEG files. 2006-06-12 11:13:39 +0000 Tim-Philipp Müller tests/check/elements/.cvsignore: And another one. Original commit message from CVS: * tests/check/elements/.cvsignore: And another one. 2006-06-12 11:04:59 +0000 Tim-Philipp Müller gst/spectrum/.cvsignore: Ignore more. Original commit message from CVS: * gst/spectrum/.cvsignore: Ignore more. 2006-06-12 10:53:26 +0000 Tim-Philipp Müller ext/libmms/gstmms.c: Set caps on outgoing buffers. Original commit message from CVS: * ext/libmms/gstmms.c: (gst_mms_create): Set caps on outgoing buffers. * sys/directdraw/gstdirectdrawsink.c: (gst_directdrawsink_init): Comment out unused global instance variable. 2006-06-11 19:31:10 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Extract images from ID3v2 tags (APIC frames). Fixes #339704. Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (scan_encoded_string), (parse_picture_frame): Extract images from ID3v2 tags (APIC frames). Fixes #339704. * configure.ac: Require core >= 0.10.8 (for GST_TAG_IMAGE and GST_TAG_PPEVIEW_IMAGE used in the patch above). 2006-06-11 18:56:24 +0000 Thomas Vander Stichele * ext/raw1394/.gitignore: * ext/taglib/.gitignore: * tests/check/elements/.gitignore: * tests/examples/level/.gitignore: moap ignore Original commit message from CVS: moap ignore 2006-06-11 18:52:19 +0000 Thomas Vander Stichele ext/raw1394/gstdv1394src.c: gratuitous comment changes Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (gst_dv1394src_discover_avc_node): gratuitous comment changes * tests/check/elements/level.c: (GST_START_TEST): fix level test leaks 2006-06-11 18:44:54 +0000 Thomas Vander Stichele * .gitignore: ignore more Original commit message from CVS: ignore more 2006-06-11 18:20:39 +0000 Tim-Philipp Müller gst/: Use gst_pad_query_peer_duration() utility function here. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size): * gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size): Use gst_pad_query_peer_duration() utility function here. 2006-06-11 17:08:11 +0000 Thomas Vander Stichele update build files Original commit message from CVS: * autogen.sh: * configure.ac: * ext/a52dec/Makefile.am: * ext/dvdnav/Makefile.am: * ext/dvdread/Makefile.am: * ext/lame/Makefile.am: * ext/mad/Makefile.am: * ext/mpeg2dec/Makefile.am: * ext/sidplay/Makefile.am: update build files 2006-06-11 13:57:19 +0000 Thomas Vander Stichele autogen.sh: require am17 Original commit message from CVS: * autogen.sh: require am17 * configure.ac: * ext/annodex/Makefile.am: * ext/cdio/Makefile.am: * ext/dv/Makefile.am: * ext/esd/Makefile.am: * ext/flac/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/ladspa/Makefile.am: * ext/libcaca/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/Makefile.am: * sys/oss/Makefile.am: * sys/sunaudio/Makefile.am: * sys/ximage/Makefile.am: clean up build further 2006-06-11 13:55:34 +0000 Thomas Vander Stichele * common: * win32/common/config.h: update Original commit message from CVS: update 2006-06-10 15:33:18 +0000 Sebastian Dröge ext/wavpack/: Add wavpack encoder element (#343131). Original commit message from CVS: Patch by: Sebastian Dröge * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpack.c: (plugin_init): * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type), (gst_wavpack_enc_correction_mode_get_type), (gst_wavpack_enc_joint_stereo_mode_get_type), (gst_wavpack_enc_base_init), (gst_wavpack_enc_class_init), (gst_wavpack_enc_init), (gst_wavpack_enc_dispose), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_format_samples), (gst_wavpack_enc_push_block), (gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block), (gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state), (gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property), (gst_wavpack_enc_plugin_init): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/md5.c: * ext/wavpack/md5.h: Add wavpack encoder element (#343131). 2006-06-09 20:36:00 +0000 Tim-Philipp Müller gconf/Makefile.am: Honour --disable-schemas-install configure option. Fixes #344100. Original commit message from CVS: * gconf/Makefile.am: Honour --disable-schemas-install configure option. Fixes #344100. 2006-06-09 18:33:01 +0000 Tim-Philipp Müller tests/examples/level/Makefile.am: Add -lm to LIBS for pow() function, don't assume one of our dependencies (such as l... Original commit message from CVS: * tests/examples/level/Makefile.am: Add -lm to LIBS for pow() function, don't assume one of our dependencies (such as libxml-2.0) drags it in automatically (#343603). 2006-06-09 18:17:23 +0000 Peter Kjellerstedt configure.ac: We should use $SED and not $(SED) in configure.ac (#343678). Original commit message from CVS: Patch by: Peter Kjellerstedt * configure.ac: We should use $SED and not $(SED) in configure.ac (#343678). 2006-06-09 17:38:19 +0000 Tim-Philipp Müller configure.ac: Check for X before using X_CFLAGS in the check for opengl (#343866). Original commit message from CVS: * configure.ac: Check for X before using X_CFLAGS in the check for opengl (#343866). * ext/musepack/Makefile.am: * ext/wavpack/Makefile.am: * gst/speed/Makefile.am: Add missing GST_LIBS, fixes build on cygwin (#343866). 2006-06-09 17:29:08 +0000 Brian Cameron sys/sunaudio/: Attached find a patch that fixes a number of bugs with the SunAudio mixer plugin and fixes #344101: 1.... Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_open), (gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new), (gst_sunaudiomixer_ctrl_set_volume), (gst_sunaudiomixer_ctrl_set_mute): * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init), (gst_sunaudiosink_init), (gst_sunaudiosink_prepare), (gst_sunaudiosink_write): Attached find a patch that fixes a number of bugs with the SunAudio mixer plugin and fixes #344101: 1. The gst_sunaudiomixer_ctrl_build_list kept appending the same 3 tracks onto the tracklist causing gnome-volume-control's preferences dialog to be messed up and would core dump if you checked/unchecked any item. 2. We weren't previously setting the MUTE flag properly. Fixing this makes gnome-volume-control work better. 3. Now we properly define the input track to be GST_MIXER_TRACK_INPUT and the monitor to be GST_MIXER_TRACK_OUTPUT, so that makes gnome-volume-control look better. Also some minor cleanup in gstsunaudiosink.c. 2006-06-09 17:12:52 +0000 Wim Taymans ext/jpeg/gstjpegdec.*: API: Added IDCT method property Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_idct_method_get_type), (gst_jpeg_dec_class_init), (gst_jpeg_dec_init), (gst_jpeg_dec_decode_indirect), (gst_jpeg_dec_decode_direct), (gst_jpeg_dec_chain), (gst_jpeg_dec_sink_event), (gst_jpeg_dec_set_property), (gst_jpeg_dec_get_property): * ext/jpeg/gstjpegdec.h: API: Added IDCT method property Small cleanups. Avoid dynamic allocation of trivial fixed structure. Allocate enough space for temp 4:4:4 YUV buffers. Fixes #343661. 2006-06-07 09:25:16 +0000 Zaheer Abbas Merali configure.ac: We now require libraw1394 >= 1.1.0 and that version onwards all have .pc files. Original commit message from CVS: 2006-06-07 Zaheer Abbas Merali * configure.ac: We now require libraw1394 >= 1.1.0 and that version onwards all have .pc files. 2006-06-02 15:02:54 +0000 Edward Hervey gst/law/alaw-decode.c: Trying to get items from an ANY or EMPTY caps is ... stupid. Original commit message from CVS: * gst/law/alaw-decode.c: (alawdec_getcaps): Trying to get items from an ANY or EMPTY caps is ... stupid. 2006-06-02 11:33:18 +0000 Edward Hervey ext/dv/gstdvdec.*: Added GstSegment handling, now implements dropping/clipping. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_event), (gst_dvdec_chain), (gst_dvdec_change_state): * ext/dv/gstdvdec.h: Added GstSegment handling, now implements dropping/clipping. 2006-06-01 22:00:26 +0000 Stefan Kost Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass Original commit message from CVS: * ext/alsaspdif/alsaspdifsink.h: * ext/amrwb/gstamrwbdec.h: * ext/amrwb/gstamrwbenc.h: * ext/amrwb/gstamrwbparse.h: * ext/arts/gst_arts.h: * ext/artsd/gstartsdsink.h: * ext/audiofile/gstafparse.h: * ext/audiofile/gstafsink.h: * ext/audiofile/gstafsrc.h: * ext/audioresample/gstaudioresample.h: * ext/bz2/gstbz2dec.h: * ext/bz2/gstbz2enc.h: * ext/dirac/gstdiracdec.h: * ext/directfb/dfbvideosink.h: * ext/divx/gstdivxdec.h: * ext/divx/gstdivxenc.h: * ext/dts/gstdtsdec.h: * ext/faac/gstfaac.h: * ext/gsm/gstgsmdec.h: * ext/gsm/gstgsmenc.h: * ext/ivorbis/vorbisenc.h: * ext/libfame/gstlibfame.h: * ext/nas/nassink.h: * ext/neon/gstneonhttpsrc.h: * ext/polyp/polypsink.h: * ext/sdl/sdlaudiosink.h: * ext/sdl/sdlvideosink.h: * ext/shout/gstshout.h: * ext/snapshot/gstsnapshot.h: * ext/sndfile/gstsf.h: * ext/swfdec/gstswfdec.h: * ext/tarkin/gsttarkindec.h: * ext/tarkin/gsttarkinenc.h: * ext/theora/theoradec.h: * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackparse.h: * ext/xine/gstxine.h: * ext/xvid/gstxviddec.h: * ext/xvid/gstxvidenc.h: * gst/cdxaparse/gstcdxaparse.h: * gst/cdxaparse/gstcdxastrip.h: * gst/colorspace/gstcolorspace.h: * gst/festival/gstfestival.h: * gst/freeze/gstfreeze.h: * gst/gdp/gstgdpdepay.h: * gst/gdp/gstgdppay.h: * gst/modplug/gstmodplug.h: * gst/mpeg1sys/gstmpeg1systemencode.h: * gst/mpeg1videoparse/gstmp1videoparse.h: * gst/mpeg2sub/gstmpeg2subt.h: * gst/mpegaudioparse/gstmpegaudioparse.h: * gst/multifilesink/gstmultifilesink.h: * gst/overlay/gstoverlay.h: * gst/playondemand/gstplayondemand.h: * gst/qtdemux/qtdemux.h: * gst/rtjpeg/gstrtjpegdec.h: * gst/rtjpeg/gstrtjpegenc.h: * gst/smooth/gstsmooth.h: * gst/smoothwave/gstsmoothwave.h: * gst/spectrum/gstspectrum.h: * gst/speed/gstspeed.h: * gst/stereo/gststereo.h: * gst/switch/gstswitch.h: * gst/tta/gstttadec.h: * gst/tta/gstttaparse.h: * gst/videodrop/gstvideodrop.h: * gst/xingheader/gstxingmux.h: * sys/directdraw/gstdirectdrawsink.h: * sys/directsound/gstdirectsoundsink.h: * sys/dxr3/dxr3audiosink.h: * sys/dxr3/dxr3spusink.h: * sys/dxr3/dxr3videosink.h: * sys/qcam/gstqcamsrc.h: * sys/vcd/vcdsrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass 2006-06-01 22:00:26 +0000 Stefan Kost Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass Original commit message from CVS: * ext/alsaspdif/alsaspdifsink.h: * ext/amrwb/gstamrwbdec.h: * ext/amrwb/gstamrwbenc.h: * ext/amrwb/gstamrwbparse.h: * ext/arts/gst_arts.h: * ext/artsd/gstartsdsink.h: * ext/audiofile/gstafparse.h: * ext/audiofile/gstafsink.h: * ext/audiofile/gstafsrc.h: * ext/audioresample/gstaudioresample.h: * ext/bz2/gstbz2dec.h: * ext/bz2/gstbz2enc.h: * ext/dirac/gstdiracdec.h: * ext/directfb/dfbvideosink.h: * ext/divx/gstdivxdec.h: * ext/divx/gstdivxenc.h: * ext/dts/gstdtsdec.h: * ext/faac/gstfaac.h: * ext/gsm/gstgsmdec.h: * ext/gsm/gstgsmenc.h: * ext/ivorbis/vorbisenc.h: * ext/libfame/gstlibfame.h: * ext/nas/nassink.h: * ext/neon/gstneonhttpsrc.h: * ext/polyp/polypsink.h: * ext/sdl/sdlaudiosink.h: * ext/sdl/sdlvideosink.h: * ext/shout/gstshout.h: * ext/snapshot/gstsnapshot.h: * ext/sndfile/gstsf.h: * ext/swfdec/gstswfdec.h: * ext/tarkin/gsttarkindec.h: * ext/tarkin/gsttarkinenc.h: * ext/theora/theoradec.h: * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackparse.h: * ext/xine/gstxine.h: * ext/xvid/gstxviddec.h: * ext/xvid/gstxvidenc.h: * gst/cdxaparse/gstcdxaparse.h: * gst/cdxaparse/gstcdxastrip.h: * gst/colorspace/gstcolorspace.h: * gst/festival/gstfestival.h: * gst/freeze/gstfreeze.h: * gst/gdp/gstgdpdepay.h: * gst/gdp/gstgdppay.h: * gst/modplug/gstmodplug.h: * gst/mpeg1sys/gstmpeg1systemencode.h: * gst/mpeg1videoparse/gstmp1videoparse.h: * gst/mpeg2sub/gstmpeg2subt.h: * gst/mpegaudioparse/gstmpegaudioparse.h: * gst/multifilesink/gstmultifilesink.h: * gst/overlay/gstoverlay.h: * gst/playondemand/gstplayondemand.h: * gst/qtdemux/qtdemux.h: * gst/rtjpeg/gstrtjpegdec.h: * gst/rtjpeg/gstrtjpegenc.h: * gst/smooth/gstsmooth.h: * gst/smoothwave/gstsmoothwave.h: * gst/spectrum/gstspectrum.h: * gst/speed/gstspeed.h: * gst/stereo/gststereo.h: * gst/switch/gstswitch.h: * gst/tta/gstttadec.h: * gst/tta/gstttaparse.h: * gst/videodrop/gstvideodrop.h: * gst/xingheader/gstxingmux.h: * sys/directdraw/gstdirectdrawsink.h: * sys/directsound/gstdirectsoundsink.h: * sys/dxr3/dxr3audiosink.h: * sys/dxr3/dxr3spusink.h: * sys/dxr3/dxr3videosink.h: * sys/qcam/gstqcamsrc.h: * sys/vcd/vcdsrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass 2006-06-01 21:07:26 +0000 Stefan Kost Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass Original commit message from CVS: * ext/aalib/gstaasink.h: * ext/annodex/gstcmmldec.h: * ext/cairo/gsttimeoverlay.h: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.h: * ext/esd/esdmon.h: * ext/esd/esdsink.h: * ext/flac/gstflacenc.h: * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.h: * ext/gdk_pixbuf/gstgdkanimation.h: * ext/gdk_pixbuf/pixbufscale.h: * ext/hal/gsthalaudiosink.h: * ext/hal/gsthalaudiosrc.h: * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokedec.h: * ext/jpeg/gstsmokeenc.h: * ext/libcaca/gstcacasink.h: * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.h: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.h: * ext/raw1394/gstdv1394src.h: * ext/speex/gstspeexenc.h: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautovideosink.h: * gst/avi/gstavidemux.h: * gst/cutter/gstcutter.h: * gst/debug/efence.h: * gst/debug/gstnavigationtest.h: * gst/debug/gstnavseek.h: * gst/flx/gstflxdec.h: * gst/goom/gstgoom.h: * gst/icydemux/gsticydemux.h: * gst/id3demux/gstid3demux.h: * gst/law/alaw-decode.h: * gst/law/alaw-encode.h: * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.h: * gst/matroska/matroska-mux.h: * gst/median/gstmedian.h: * gst/oldcore/gstaggregator.h: * gst/oldcore/gstfdsink.h: * gst/oldcore/gstmd5sink.h: * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.h: * gst/oldcore/gstshaper.h: * gst/oldcore/gststatistics.h: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtspsrc.h: * gst/smpte/gstsmpte.h: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.h: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.h: * sys/oss/gstossdmabuffer.h: * sys/oss/gstossmixerelement.h: * sys/oss/gstosssink.h: * sys/oss/gstosssrc.h: * sys/osxvideo/osxvideosink.h: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiosink.h: * sys/ximage/gstximagesrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass 2006-05-31 16:23:54 +0000 Wim Taymans gst/goom/gstgoom.*: Handle QoS. Original commit message from CVS: * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init), (gst_goom_finalize), (gst_goom_reset), (gst_goom_sink_setcaps), (gst_goom_src_setcaps), (gst_goom_src_event), (gst_goom_sink_event), (get_buffer), (gst_goom_chain), (gst_goom_change_state): * gst/goom/gstgoom.h: Handle QoS. Handle flushing, discont and events. Fix timestamps and various other cleanups. 2006-05-31 15:37:16 +0000 Zaheer Abbas Merali ext/raw1394/gstdv1394src.c: Fix bus reset when using libiec61883 Original commit message from CVS: 2006-05-31 Zaheer Abbas Merali * ext/raw1394/gstdv1394src.c: (gst_dv1394src_bus_reset): Fix bus reset when using libiec61883 2006-05-31 10:31:23 +0000 Zaheer Abbas Merali configure.ac: Detect libiec61883 and set necessary CFLAGS and LIBS for dv1394. Original commit message from CVS: 2006-05-31 Zaheer Abbas Merali * configure.ac: Detect libiec61883 and set necessary CFLAGS and LIBS for dv1394. * ext/raw1394/Makefile.am: Add CFLAGS. * ext/raw1394/gstdv1394src.c: (gst_dv1394src_iec61883_receive), New method, to receive using libiec61883. (gst_dv1394src_iso_receive), #ifdef'd out if libiec61883 is present. (gst_dv1394src_bus_reset), Get userdata correctly if using libiec61883. (gst_dv1394src_create), When using libiec61883, only poll one fd and no need to read. (gst_dv1394src_discover_avc_node), Replace g_warnings. (gst_dv1394src_start), Create new handle when we know which dv port. More reliable than setting port on an existing handle. Initialise libiec61883. (gst_dv1394src_stop): If using libiec61883, then cleanup its handle properly. * ext/raw1394/gstdv1394src.h: Add libiec61883 handle. 2006-05-30 21:07:38 +0000 Sébastien Moutte gst/avi/gstavidemux.c: add an explicit dll imported declaration for GST_CAT_EVENT+WIN32 Original commit message from CVS: * gst/avi/gstavidemux.c: add an explicit dll imported declaration for GST_CAT_EVENT+WIN32 * win32/MANIFEST: sort file listing * win32/vs6/libgstavi.dsp: add gstavimux.c to the project * win32/vs6/libgstid3demux.dsp: add link to zlib library * win32/vs6/libgstmatroska.dsp: add matroska-ids.c to the project 2006-05-30 14:35:18 +0000 Sebastian Dröge Add apev2mux element (#343122). Original commit message from CVS: Patch by: Sebastian Dröge * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/taglib/Makefile.am: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstapev2mux.h: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gsttaglibmux.c: (plugin_init): * ext/taglib/gsttaglibmux.h: Add apev2mux element (#343122). * tests/check/Makefile.am: * tests/check/elements/apev2mux.c: (test_taglib_apev2mux_create_tags), (test_taglib_apev2mux_check_tags), (fill_mp3_buffer), (got_buffer), (demux_pad_added), (test_taglib_apev2mux_check_output_buffer), (test_taglib_apev2mux_with_tags), (GST_START_TEST), (apev2mux_suite), (main): Add unit test for apev2mux element. 2006-05-28 17:33:13 +0000 Tim-Philipp Müller gst/: GST_PTR_FORMAT should be used to print caps in debug statements. Original commit message from CVS: * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps): * gst/debug/negotiation.c: (gst_negotiation_update_caps): * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps): GST_PTR_FORMAT should be used to print caps in debug statements. 2006-05-28 14:38:11 +0000 Sebastian Dröge gst/apetag/gstapedemux.c: Some clean-ups and additions: map APE 'file' tag to Original commit message from CVS: Patch by: Sebastian Dröge * gst/apetag/gstapedemux.c: (ape_demux_get_gst_tag_from_tag), (ape_demux_parse_tags): Some clean-ups and additions: map APE 'file' tag to GST_TAG_LOCATION (#343123); add support for extracting the track count and clean up parsing a bit (#343127). 2006-05-28 13:49:12 +0000 Edward Hervey ext/jpeg/gstjpegdec.c: Initialize segment to GST_FORMAT_UNDEFINED in READY->PAUSED. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_change_state): Initialize segment to GST_FORMAT_UNDEFINED in READY->PAUSED. 2006-05-28 13:30:13 +0000 Edward Hervey ext/jpeg/gstjpegdec.*: Clip outgoing buffers according to currently configured segment. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_finalize), (gst_jpeg_dec_init), (gst_jpeg_dec_chain), (gst_jpeg_dec_sink_event), (gst_jpeg_dec_change_state): * ext/jpeg/gstjpegdec.h: Clip outgoing buffers according to currently configured segment. 2006-05-28 10:39:00 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: Handle writing of track-count or album-volume-count without track-number or albume-volume... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Handle writing of track-count or album-volume-count without track-number or albume-volume-number (in this case the number will just be set to 0). * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_check_tags): It would be nice if we actually checked the values received for track/album-volume number/count in _check_tags(), rather than setting them again ... 2006-05-28 10:05:47 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: A track/volume number or count of 0 does not make sense, just ignore it along with negati... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist): A track/volume number or count of 0 does not make sense, just ignore it along with negative numbers (a tag might only contain a track count without a track number). 2006-05-27 13:11:37 +0000 Edward Hervey ext/jpeg/gstjpegdec.c: Abort decompression when receiving FLUSH_STOP. This should avoid issues when interrupting deco... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_sink_event): Abort decompression when receiving FLUSH_STOP. This should avoid issues when interrupting decoding with flushes. 2006-05-27 12:10:50 +0000 Tim-Philipp Müller ext/flac/gstflac.c: Don't #include file we don't dist any longer. Original commit message from CVS: * ext/flac/gstflac.c: Don't #include file we don't dist any longer. 2006-05-27 11:27:59 +0000 Tim-Philipp Müller README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from... Original commit message from CVS: * README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from the core. 2006-05-26 22:35:00 +0000 Stefan Kost gst/spectrum/: added another example Original commit message from CVS: * gst/spectrum/Makefile.am: * gst/spectrum/demo-audiotest.c: (on_frequency_changed), (spectrum_chain), (main): * gst/spectrum/demo-osssrc.c: added another example * sys/v4l2/gstv4l2src.c: fix typo 2006-05-26 13:16:54 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Clip the outputed NEWSEGMENT stop time to the configured segment stop time. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment): Clip the outputed NEWSEGMENT stop time to the configured segment stop time. 2006-05-26 11:48:44 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Don't clear the running variable in the seek code. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_do_seek): Don't clear the running variable in the seek code. 2006-05-24 16:03:40 +0000 Wim Taymans ext/dv/gstdvdemux.c: Implement EOS correctly by either posting Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_loop): Implement EOS correctly by either posting SEGMENT_DONE or pushing an EOS message depending on the seek type. Fixes #342592 2006-05-24 11:56:43 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Detect QCELP in mp4a descriptors. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_esds): Detect QCELP in mp4a descriptors. 2006-05-24 10:00:50 +0000 Wim Taymans gst/law/: Some cleanups in the chain functions. Original commit message from CVS: * gst/law/alaw-decode.c: (gst_alawdec_chain): * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: (gst_alawenc_chain): * gst/law/alaw-encode.h: * gst/law/mulaw-decode.c: (gst_mulawdec_chain): * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.c: (gst_mulawenc_chain): * gst/law/mulaw-encode.h: Some cleanups in the chain functions. Remove some GStreamer 0.0.2 bits. 2006-05-23 20:15:04 +0000 Mark Nauwelaerts gst/matroska/matroska-mux.c: gst_collect_pads_stop() needs to be called before chaining up to the parent class (#3427... Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/matroska/matroska-mux.c: (gst_matroska_mux_change_state): gst_collect_pads_stop() needs to be called before chaining up to the parent class (#342734). 2006-05-23 16:45:22 +0000 Tim-Philipp Müller ext/flac/: Remove backwards compatibility cruft for dealing with FLAC API changes in the 1.0.x series - we require 1.... Original commit message from CVS: * ext/flac/Makefile.am: * ext/flac/flac_compat.h: * ext/flac/gstflac.c: * ext/flac/gstflacdec.c: (gst_flac_dec_init): * ext/flac/gstflacenc.c: Remove backwards compatibility cruft for dealing with FLAC API changes in the 1.0.x series - we require 1.1.1 or newer these days. 2006-05-23 13:44:11 +0000 Tim-Philipp Müller gst/matroska/: Add support for muxing/demuxing theora video (#342448; too bad none of the usual linux players can act... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_push_xiph_codec_priv_data), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps): * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init), (gst_matroska_mux_video_pad_setcaps), (xiph3_streamheader_to_codecdata), (vorbis_streamheader_to_codecdata), (theora_streamheader_to_codecdata), (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_write_data): Add support for muxing/demuxing theora video (#342448; too bad none of the usual linux players can actually play this). Playback in GStreamer will require additional changes to theoradec in -base. Refactor streamheaders <=> CodecPrivateData code a bit; some small cleanups. 2006-05-22 18:00:52 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: po/POTFILES.in: Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (plugin_init): po/POTFILES.in: Throw an error when the file is encrypted. Move plugin_init stuff to the end of the file, add stuff for i18n, make debug category static. 2006-05-22 15:23:05 +0000 Tim-Philipp Müller ext/jpeg/gstjpegdec.c: Fix crashes when the horizontal subsampling is 1. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (hresamplecpy1), (gst_jpeg_dec_decode_indirect), (gst_jpeg_dec_chain): Fix crashes when the horizontal subsampling is 1. Fixes #342097. 2006-05-22 14:56:29 +0000 Thomas Vander Stichele * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: cover up the dirty truth Original commit message from CVS: cover up the dirty truth 2006-05-22 13:53:18 +0000 Mark Nauwelaerts gst/avi/gstavimux.*: - add odml (large file) index support Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/avi/gstavimux.c: (gst_avi_mux_finalize), (gst_avi_mux_init), (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps), (gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header), (gst_avi_mux_riff_get_avix_header), (gst_avi_mux_write_avix_index), (gst_avi_mux_add_index), (gst_avi_mux_bigfile), (gst_avi_mux_start_file), (gst_avi_mux_stop_file), (gst_avi_mux_handle_event), (gst_avi_mux_do_audio_buffer), (gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer), (gst_avi_mux_change_state): * gst/avi/gstavimux.h: Some enhancements for avimux (#342526): - add odml (large file) index support - store codec init data (e.g. huffyuv) - miscellaneous other fixes/cleanups 2006-05-22 13:51:30 +0000 Thomas Vander Stichele * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: fix descriptions and license blocks cut and paste anyone ? Original commit message from CVS: fix descriptions and license blocks cut and paste anyone ? 2006-05-21 16:41:44 +0000 Stefan Kost gst/spectrum/gstspectrum.c: Use boilerplate macro, fix strings to match plugin-moval-requirements Original commit message from CVS: * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_set_sink_caps), (gst_spectrum_get_sink_caps), (gst_spectrum_chain): Use boilerplate macro, fix strings to match plugin-moval-requirements 2006-05-21 16:23:23 +0000 Stefan Kost gst/spectrum/Makefile.am: Link to base libraries Original commit message from CVS: * gst/spectrum/Makefile.am: Link to base libraries * gst/spectrum/demo-osssrc.c: (main): use new threshhold property * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_dispose), (gst_spectrum_set_property), (gst_spectrum_set_sink_caps), (gst_spectrum_get_sink_caps), (gst_spectrum_chain), (gst_spectrum_change_state): * gst/spectrum/gstspectrum.h: Use gst_adapter, support multiple-channels, add threshold property for result, add docs, fix resulting spectrum range (was including mirrored results) 2006-05-20 22:42:15 +0000 Stefan Kost Initial port of the spectrum element Original commit message from CVS: * configure.ac: * gst/spectrum/demo-osssrc.c: (spectrum_chain), (main): * gst/spectrum/fix_fft.c: (gst_spectrum_fix_dot): * gst/spectrum/gstspectrum.c: (gst_spectrum_get_type), (gst_spectrum_base_init), (gst_spectrum_class_init), (gst_spectrum_init), (gst_spectrum_dispose), (gst_spectrum_set_property), (gst_spectrum_chain): * gst/spectrum/gstspectrum.h: Initial port of the spectrum element 2006-05-19 18:58:05 +0000 Edgard Lima * sys/v4l2/gstv4l2xoverlay.c: I forget to add sys/v4l2/gstv4l2xoverlay.c in las commit Original commit message from CVS: I forget to add sys/v4l2/gstv4l2xoverlay.c in las commit 2006-05-19 18:31:25 +0000 Edgard Lima * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: Some clean-ups requested by wingo in bug #338818. Original commit message from CVS: Some clean-ups requested by wingo in bug #338818. 2006-05-19 14:05:53 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Don't output any tag when we encounter a negative track number - the tag type is uint, so... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist): Don't output any tag when we encounter a negative track number - the tag type is uint, so we end up outputting huge positive numbers instead. (Fixes: #342029) 2006-05-18 23:04:59 +0000 Thomas Vander Stichele configure.ac: update for new GSTPB_PLUGINS_DIR Original commit message from CVS: * configure.ac: update for new GSTPB_PLUGINS_DIR 2006-05-18 19:34:47 +0000 Stefan Kost configure.ac: Check for X11 Original commit message from CVS: * configure.ac: Check for X11 * sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_iface_supported): * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open): * sys/v4l2/gstv4l2xoverlay.h: Code cleanups, fix debug macros 2006-05-18 14:45:33 +0000 Philippe Kalaf rtp/gst/gstrtph263pay.c: Properly set static caps for H263 at 34. Original commit message from CVS: 2006-05-18 Philippe Kalaf * rtp/gst/gstrtph263pay.c: Properly set static caps for H263 at 34. 2006-05-18 12:46:08 +0000 James Doc Livingston ext/taglib/gsttaglibmux.c: Merge event tags and tag setter tags correctly (#339918). Also, don't leak taglist in case... Original commit message from CVS: Patch by: James "Doc" Livingston * ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag): Merge event tags and tag setter tags correctly (#339918). Also, don't leak taglist in case of an error. 2006-05-17 18:09:06 +0000 Philippe Kalaf * common: * gst/rtp/gstrtph263pay.c: Fixed caps for H263 (not the same as H263+) Original commit message from CVS: Fixed caps for H263 (not the same as H263+) 2006-05-17 12:36:26 +0000 Edward Hervey gst/law/mulaw-decode.c: We can only do caps intersection if the othercaps are non-empty and not Original commit message from CVS: * gst/law/mulaw-decode.c: (mulawdec_getcaps): We can only do caps intersection if the othercaps are non-empty and not ANY. Else we return the pad template (base_caps). 2006-05-17 11:20:44 +0000 Tim-Philipp Müller ext/jpeg/gstjpegdec.c: Fix crash when outputting debugging information for certain pictures (always good to use the r... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): Fix crash when outputting debugging information for certain pictures (always good to use the right struct member for the number of records in an array). 2006-05-17 08:10:31 +0000 Jindrich Makovicka gst/matroska/ebml-read.c: Don't create unnecessary sub-buffers all the time. Dramatically improves performance with m... Original commit message from CVS: Patch by: Jindrich Makovicka * gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes), (gst_ebml_read_pull_bytes), (gst_ebml_read_element_id), (gst_ebml_read_element_length), (gst_ebml_read_buffer), (gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint), (gst_ebml_read_float), (gst_ebml_read_ascii), (gst_ebml_read_binary): Don't create unnecessary sub-buffers all the time. Dramatically improves performance with multiple concurrently running matroskademux instances (#341818) (and avoids doing unnecessarily inefficient things in the general case). 2006-05-16 17:20:04 +0000 Edward Hervey ext/libpng/gstpngenc.c: In snapshot mode, we always return GST_FLOW_UNEXPECTED whatever the return value of gst_pad_p... Original commit message from CVS: * ext/libpng/gstpngenc.c: (gst_pngenc_chain): In snapshot mode, we always return GST_FLOW_UNEXPECTED whatever the return value of gst_pad_push_event(). 2006-05-16 14:07:29 +0000 Jan Schmidt gst/autodetect/: Make the name of the child element be based on the name of the parent, so that debug output is more ... Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_find_best): Make the name of the child element be based on the name of the parent, so that debug output is more useful. * gst/id3demux/id3v2frames.c: (find_utf16_bom), (parse_insert_string_field), (parse_split_strings): Rework string parsing to always walk over BOM markers in UTF16 strings, using the endianness indicated by the innermost one, then trying the opposite endianness if that fails to convert to valid UTF-8. Fixes #341774 2006-05-16 13:31:02 +0000 Zaheer Abbas Merali ext/libpng/Makefile.am: Add LIBPNG_CFLAGS. Original commit message from CVS: 2006-05-16 Zaheer Abbas Merali Patch from: Matthieu * ext/libpng/Makefile.am: Add LIBPNG_CFLAGS. 2006-05-15 11:20:21 +0000 Christian Schaller * gst-plugins-good.spec.in: update with latest changes Original commit message from CVS: update with latest changes 2006-05-15 09:00:42 +0000 Tim-Philipp Müller ext/taglib/gstid3v2mux.cc: Add support for writing images (APIC frames) into ID3v2 tags (picture type always set to '... Original commit message from CVS: * ext/taglib/gstid3v2mux.cc: Add support for writing images (APIC frames) into ID3v2 tags (picture type always set to 'other' for now though). 2006-05-14 12:50:07 +0000 Michael Smith gst/wavparse/gstwavparse.c: Update docs; wavparse implements push and pull modes. Original commit message from CVS: * gst/wavparse/gstwavparse.c: Update docs; wavparse implements push and pull modes. 2006-05-12 18:10:36 +0000 Wim Taymans gst/avi/gstavidemux.c: Ooops, bitten by the copy-and-paste design paradigm, fixes seek again. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_index_next), (gst_avi_demux_parse_index), (gst_avi_demux_massage_index), (gst_avi_demux_handle_seek), (gst_avi_demux_loop): Ooops, bitten by the copy-and-paste design paradigm, fixes seek again. 2006-05-12 18:04:22 +0000 Wim Taymans gst/avi/gstavidemux.*: Some cleanups, prepare to use GstSegment. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_index_next), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_push_event), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Some cleanups, prepare to use GstSegment. Fix error in entry walking code. Fix VBR detection. Smarter timestamp calculation code. Uniform error/eos handling. 2006-05-12 17:44:15 +0000 Michael Smith gst/wavparse/gstwavparse.c: Fix use of uninitialised values if we're NOT seeking in ready. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers): Fix use of uninitialised values if we're NOT seeking in ready. Fix typos. 2006-05-12 08:23:18 +0000 Tim-Philipp Müller gst/wavparse/Makefile.am: Add CFLAGS and LIBS for libgstbase, fixes build on Original commit message from CVS: * gst/wavparse/Makefile.am: Add CFLAGS and LIBS for libgstbase, fixes build on Cygwin (#341489). 2006-05-12 08:21:37 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Some more debug info. No need to check whether the string returned by g_convert() is real... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_insert_string_field): Some more debug info. No need to check whether the string returned by g_convert() is really UTF-8 - either it is or we get NULL returned. 2006-05-11 17:59:59 +0000 Edgard Lima * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2element.c: * sys/v4l2/gstv4l2element.h: * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/gstv4l2xoverlay.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: Changes proposed by Wingo in bug #338818. Original commit message from CVS: Changes proposed by Wingo in bug #338818. 2006-05-11 09:09:49 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Figure out the real audio type in mp4a boxes by parsing the optional descriptors in the option... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak), (gst_qtdemux_handle_esds): Figure out the real audio type in mp4a boxes by parsing the optional descriptors in the optional esds box. Promote the default AAC to mp3 when indicated. Fixes #330632. 2006-05-10 17:44:50 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Parse version 2 sample descriptions. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_dump_unknown), (qtdemux_parse_trak), (gst_qtdemux_handle_esds): Parse version 2 sample descriptions. Don't #define gst_util_dump_mem(), use something more specific instead to avoid confusion. 2006-05-10 13:51:01 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Fix parsing of numeric genre strings some more, by ensuring that we only try and parse st... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist): Fix parsing of numeric genre strings some more, by ensuring that we only try and parse strings that a) Start with '(' and b) Consist only of digits. Also, when finding an escaping '((' sequence, bust it back to '(' by swallowing the first parenthesis 2006-05-10 11:17:31 +0000 Tim-Philipp Müller ext/esd/esdsink.*: Move the esd_get_server_info() into gst_esdsink_open() and fail with a decent error message on err... Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_finalize), (gst_esdsink_getcaps), (gst_esdsink_open), (gst_esdsink_close): * ext/esd/esdsink.h: Move the esd_get_server_info() into gst_esdsink_open() and fail with a decent error message on errors. 2006-05-10 10:29:54 +0000 Tim-Philipp Müller Const-ify GEnumValue arrays. Original commit message from CVS: * ext/esd/esdmon.c: (gst_esdmon_depths_get_type), (gst_esdmon_channels_get_type): * ext/gconf/gstgconfaudiosink.c: (gst_gconf_profile_get_type): * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type): * ext/libcaca/gstcacasink.c: (gst_cacasink_dither_get_type): * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type): * gst/alpha/gstalpha.c: (gst_alpha_method_get_type): * gst/rtp/gstrtpilbcdepay.c: (gst_ilbc_mode_get_type): * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type): * gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type): * gst/videofilter/gstvideoflip.c: (gst_video_flip_method_get_type): * gst/videomixer/videomixer.c: (gst_video_mixer_background_get_type): Const-ify GEnumValue arrays. 2006-05-09 14:08:15 +0000 Mark Nauwelaerts gst/avi/gstavimux.c: Work around gst_buffer_make_metadata_writable() bug that results in avimux marking all frames in... Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/avi/gstavimux.c: (gst_avi_mux_do_audio_buffer), (gst_avi_mux_do_video_buffer): Work around gst_buffer_make_metadata_writable() bug that results in avimux marking all frames in the index as keyframes (#340859). 2006-05-08 19:21:18 +0000 Martin Rubli * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: Fix fourcc name printed out. Patch from Martin Rubli. Original commit message from CVS: Fix fourcc name printed out. Patch from Martin Rubli. 2006-05-08 15:20:10 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Don't cause side effects in a debugging function. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query), (qtdemux_dump_mvhd): Don't cause side effects in a debugging function. Also report duration in push mode since we can. 2006-05-08 14:35:20 +0000 Wim Taymans gst/rtsp/rtspurl.c: Make parsing of urls suck slightly less. Original commit message from CVS: * gst/rtsp/rtspurl.c: (rtsp_url_parse): Make parsing of urls suck slightly less. 2006-05-08 11:53:03 +0000 Edward Hervey autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize. Original commit message from CVS: * autogen.sh: (CONFIGURE_DEF_OPT): libtoolize on Darwin/MacOSX is called glibtoolize. 2006-05-08 10:59:05 +0000 Jens Granseuer C89 compliance fixes. Fixes #340980 Original commit message from CVS: Patch by: Jens Granseuer * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_init): * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose): C89 compliance fixes. Fixes #340980 2006-05-06 11:38:30 +0000 Tim-Philipp Müller ext/lame/gstlame.*: Remove tag writing from lame (which was completely broken anyway, #329184). Leaving GstTagSetter ... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_get_type), (gst_lame_release_memory), (gst_lame_init), (gst_lame_sink_event), (gst_lame_setup), (gst_lame_change_state): * ext/lame/gstlame.h: Remove tag writing from lame (which was completely broken anyway, #329184). Leaving GstTagSetter interface around for now, albeit non-functional. Should be removed completely in 0.11. Use the 'id3v2mux' plugin from -good for writing tags. 2006-05-06 09:01:34 +0000 Tim-Philipp Müller ext/flac/gstflacdec.*: Handle segment seeks that include the end of the file as stop point properly: when the decoder... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_loop): * ext/flac/gstflacdec.h: Handle segment seeks that include the end of the file as stop point properly: when the decoder hits EOS we want to send a SEGMENT_DONE message instead of an EOS event in case we're in segment seek mode (fixes #340699). 2006-05-06 00:14:09 +0000 Maciej Katafiasz * ChangeLog: * ext/cairo/gsttextoverlay.c: * ext/flac/gstflacdec.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/wavpack/gstwavpackdec.c: * gst/apetag/gstapedemux.c: * gst/debug/breakmydata.c: * gst/debug/testplugin.c: * gst/matroska/ebml-write.c: * gst/multipart/multipartdemux.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: Add semicolons after GST_BOILERPLATE[_FULL] so that indent doesn't mess up following lines. Original commit message from CVS: Add semicolons after GST_BOILERPLATE[_FULL] so that indent doesn't mess up following lines. 2006-05-05 20:12:59 +0000 Martin Rubli * sys/v4l2/gstv4l2element.c: * sys/v4l2/gstv4l2element.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: * tests/icles/v4l2src-test.c: Some changes proposed by wingo in bug #338818 (but not everything yet). Patch from Martin Rubli to fix framerate dete... Original commit message from CVS: Some changes proposed by wingo in bug #338818 (but not everything yet). Patch from Martin Rubli to fix framerate detection. 2006-05-05 08:23:39 +0000 Andres Salomon ext/lame/gstlame.c: Fix typo (comma vs. semicolon) (#340710). Original commit message from CVS: Patch by: Andres Salomon * ext/lame/gstlame.c: (gst_lame_sink_event): Fix typo (comma vs. semicolon) (#340710). 2006-05-04 17:27:27 +0000 Michal Benes gst/matroska/matroska-demux.c: Don't leak caps when freeing the stream context (#340623). Original commit message from CVS: Patch by: Michal Benes * gst/matroska/matroska-demux.c: (gst_matroska_demux_reset): Don't leak caps when freeing the stream context (#340623). 2006-05-04 15:40:18 +0000 Jan Schmidt configure.ac: Back to CVS Original commit message from CVS: * configure.ac: Back to CVS === release 0.10.3 === 2006-05-04 15:36:02 +0000 Jan Schmidt * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * win32/common/config.h: Really release 0.10.3 Original commit message from CVS: Really release 0.10.3 2006-05-04 15:28:53 +0000 Jan Schmidt * docs/plugins/inspect/plugin-qtdemux.xml: Really release 0.10.3 this time Original commit message from CVS: Really release 0.10.3 this time 2006-05-04 15:05:00 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * win32/common/config.h: Release 0.10.3 Original commit message from CVS: Release 0.10.3 2006-05-03 18:44:38 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2006-05-03 18:41:47 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: Don't strcmp() NULL strings. Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_stream_is_vorbis_header), (gst_matroska_mux_write_data): Don't strcmp() NULL strings. Only start new clusters on video keyframes, not on any random audio buffer that doesn't have the DELTA_UNIT flag set (fixes 'make check' again). 2006-05-03 14:51:50 +0000 Mark Nauwelaerts gst/matroska/matroska-mux.c: Don't misinterpret GST_CLOCK_TIME_NONE as very high timestamp value and then dead-lock w... Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad), (gst_matroska_mux_stream_is_vorbis_header), (gst_matroska_mux_write_data): Don't misinterpret GST_CLOCK_TIME_NONE as very high timestamp value and then dead-lock when muxing vorbis audio streams (the three vorbis header buffers carry no timestamp, and it would try to mux these after all video buffers). Fixes #340346. Improve clustering: start a new cluster also whenever we get a keyframe. 2006-05-03 14:30:21 +0000 Jan Schmidt gst/qtdemux/qtdemux.c: Clean up one piece of logic slightly and remove a dead code block. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Clean up one piece of logic slightly and remove a dead code block. 2006-05-03 14:28:57 +0000 Thomas Vander Stichele add win32 stuff Original commit message from CVS: * Makefile.am: * configure.ac: * win32/common/config.h.in: add win32 stuff 2006-05-03 14:26:51 +0000 Thomas Vander Stichele add win32 stuff Original commit message from CVS: * Makefile.am: * configure.ac: * win32/common/config.h.in: add win32 stuff 2006-05-02 22:34:52 +0000 Michael Smith ext/cairo/gsttimeoverlay.c: Fix timeoverlay for non-multiple-of-4 widths. This fourcc crap Original commit message from CVS: * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform): Fix timeoverlay for non-multiple-of-4 widths. This fourcc crap SUCKS. 2006-05-02 21:52:48 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: Fix get_caps func to work when no framerate is available and the caps isn't simple. Original commit message from CVS: Fix get_caps func to work when no framerate is available and the caps isn't simple. 2006-05-02 18:50:23 +0000 Stefan Kost gst/: don't leak caps-string Original commit message from CVS: * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps): * gst/debug/negotiation.c: (gst_negotiation_update_caps): * gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps): don't leak caps-string 2006-05-02 15:46:02 +0000 Tim-Philipp Müller gst/id3demux/gstid3demux.c: Let core insert default error message for TYPE_NOT_FOUND errors, it's just as good as our... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_chain), (gst_id3demux_sink_activate): Let core insert default error message for TYPE_NOT_FOUND errors, it's just as good as our own and has the added bonus of being translated. 2006-05-02 15:40:15 +0000 Tim-Philipp Müller gst/: Post an error message when we get an EOS event and were not able to find out the type of stream. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_init), (gst_tag_demux_sink_event): * gst/id3demux/gstid3demux.c: (gst_id3demux_init), (gst_id3demux_sink_event): Post an error message when we get an EOS event and were not able to find out the type of stream. * tests/check/elements/id3v2mux.c: (fill_mp3_buffer), (got_buffer), (test_taglib_id3mux_with_tags): Decrease num-buffers to 16 per iteration again, otherwise the many memcpy()s and reallocations in the test will hammer slow CPUs completely and make the test timeout. 2006-05-02 13:24:38 +0000 Thomas Vander Stichele configure.ac: figure out where plugins-base plugins are Original commit message from CVS: * configure.ac: figure out where plugins-base plugins are * tests/check/Makefile.am: use plugins-base plugins, so we have typefind functions * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags): increase num-buffers, this makes sure the test errors out instead of timing out when no typefind functions are present 2006-05-02 13:01:50 +0000 Thomas Vander Stichele * gst/wavparse/gstwavparse.c: fix docs for wavparse Original commit message from CVS: fix docs for wavparse 2006-05-01 21:37:51 +0000 Edgard Lima * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/v4l2_calls.c: * tests/icles/v4l2src-test.c: Few improvements to move to good. Original commit message from CVS: Few improvements to move to good. 2006-05-01 11:46:33 +0000 Thomas Vander Stichele docs/plugins/Makefile.am: also check .cc files for gtk-doc markup Original commit message from CVS: * docs/plugins/Makefile.am: also check .cc files for gtk-doc markup * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * tests/check/Makefile.am: * tests/check/elements/id3v2mux.c: (id3v2mux_suite), (main): * ext/Makefile.am: * ext/taglib/Makefile.am: * ext/taglib/gstid3v2mux.h: * ext/taglib/gsttaglibmux.c: * ext/taglib/gsttaglibmux.h: move taglib-based id3v2muxer to -good. Fixes #336110. 2006-05-01 11:45:15 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-icydemux.xml: add icydemux inspection Original commit message from CVS: add icydemux inspection 2006-05-01 11:43:31 +0000 Thomas Vander Stichele * po/POTFILES.in: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: add ximagesrc for translation Original commit message from CVS: add ximagesrc for translation 2006-04-30 16:16:59 +0000 Thomas Vander Stichele * ext/taglib/gstid3v2mux.cc: * ext/taglib/gsttaglibmux.c: small cleanups Original commit message from CVS: small cleanups 2006-04-30 15:32:13 +0000 Thomas Vander Stichele * ext/taglib/gstid3v2mux.cc: fix docs Original commit message from CVS: fix docs 2006-04-30 14:55:15 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-taglib.xml: update to latest version Original commit message from CVS: update to latest version 2006-04-29 18:46:36 +0000 Tim-Philipp Müller ext/taglib/gsttaglib.cc: Post an error message on the bus in the (extremely unlikely) case of an error. Original commit message from CVS: * ext/taglib/gsttaglib.cc: Post an error message on the bus in the (extremely unlikely) case of an error. 2006-04-29 18:18:24 +0000 Tim-Philipp Müller ext/taglib/: Split the actual ID3v2 tag rendering code into its own subclass. Original commit message from CVS: * ext/taglib/Makefile.am: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gstid3v2mux.h: * ext/taglib/gsttaglib.cc: * ext/taglib/gsttaglib.h: Split the actual ID3v2 tag rendering code into its own subclass. 2006-04-29 16:14:20 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: ... and fix multichannel/WAVFORMATEX support again. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): ... and fix multichannel/WAVFORMATEX support again. 2006-04-28 23:09:17 +0000 Stefan Kost gst/wavparse/gstwavparse.*: Add push (streaming) mode to wavparse (fixes #337625) Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl), (gst_wavparse_parse_cues), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state), (plugin_init): * gst/wavparse/gstwavparse.h: Add push (streaming) mode to wavparse (fixes #337625) 2006-04-28 21:43:07 +0000 Thomas Vander Stichele * tests/check/elements/id3v2mux.c: element renamed Original commit message from CVS: element renamed 2006-04-28 19:22:46 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-ximagesrc.xml: add plugin docs for ximagesrc Original commit message from CVS: add plugin docs for ximagesrc 2006-04-28 19:15:08 +0000 Thomas Vander Stichele add ximagesrc icles test Original commit message from CVS: * configure.ac: * tests/Makefile.am: add ximagesrc icles test 2006-04-28 18:57:09 +0000 Thomas Vander Stichele Move ximagesrc plug-in to good after review. Fixes #336756. Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_class_init), (gst_cmml_enc_push_clip): * sys/Makefile.am: * sys/ximage/Makefile.am: * sys/ximage/gstximagesrc.c: Move ximagesrc plug-in to good after review. Fixes #336756. 2006-04-28 16:51:33 +0000 Thomas Vander Stichele * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: borgify naming Original commit message from CVS: borgify naming 2006-04-28 16:46:52 +0000 Thomas Vander Stichele * sys/ximage/gstximagesrc.c: doc tweaks Original commit message from CVS: doc tweaks 2006-04-28 16:15:20 +0000 Thomas Vander Stichele * sys/ximage/Makefile.am: * sys/ximage/gstximagesrc.c: clean up Makefile.am Original commit message from CVS: clean up Makefile.am 2006-04-28 15:33:09 +0000 Thomas Vander Stichele * ext/taglib/gsttaglibmux.c: * ext/taglib/gsttaglibmux.h: pedantic cleanups Original commit message from CVS: pedantic cleanups 2006-04-28 14:57:57 +0000 Michael Smith gst/icydemux/gsticydemux.*: Fix event handling: cache events when typefinding and forward later. Original commit message from CVS: * gst/icydemux/gsticydemux.c: (gst_icydemux_reset), (gst_icydemux_init), (gst_icydemux_sink_setcaps), (gst_icydemux_add_srcpad), (gst_icydemux_parse_and_send_tags), (gst_icydemux_handle_event), (gst_icydemux_send_cached_events), (gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta), (gst_icydemux_chain), (gst_icydemux_send_tag_event): * gst/icydemux/gsticydemux.h: Fix event handling: cache events when typefinding and forward later. 2006-04-28 14:55:20 +0000 Zaheer Abbas Merali sys/osxaudio/gstosxaudiosink.c: Register osxaudiosrc to the plugin. Original commit message from CVS: 2006-04-28 Zaheer Abbas Merali * sys/osxaudio/gstosxaudiosink.c: (plugin_init): Register osxaudiosrc to the plugin. * sys/osxaudio/gstosxaudiosrc.c: (gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_base_init), (gst_osx_audio_src_class_init), (gst_osx_audio_src_init), (gst_osx_audio_src_set_property), (gst_osx_audio_src_get_property), (gst_osx_audio_src_create_ringbuffer), (gst_osx_audio_src_io_proc), (gst_osx_audio_src_osxelement_init): * sys/osxaudio/gstosxaudiosrc.h: Port of osxaudiosrc to 0.10. * sys/osxaudio/Makefile.am: Add osxaudiosrc 2006-04-28 12:00:39 +0000 Zaheer Abbas Merali * ChangeLog: commit Changelog for previous commit Original commit message from CVS: commit Changelog for previous commit 2006-04-28 11:57:39 +0000 Zaheer Abbas Merali * sys/osxaudio/gstosxringbuffer.c: * sys/osxaudio/gstosxringbuffer.h: Forgot to commit, quick commit be4 apple dies Original commit message from CVS: Forgot to commit, quick commit be4 apple dies 2006-04-28 11:37:22 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Recognise and skip any byte order marker (BOM) in Original commit message from CVS: * gst/id3demux/id3v2frames.c: (has_utf16_bom), (parse_split_strings): Recognise and skip any byte order marker (BOM) in UTF-16 strings. 2006-04-27 16:05:54 +0000 Tim-Philipp Müller Add docs for both avidemux and avimux. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-avi.xml: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: Add docs for both avidemux and avimux. 2006-04-27 14:51:06 +0000 Mark Nauwelaerts gst/avi/: Port AVI muxer to GStreamer-0.10 (#332031). Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/avi/Makefile.am: * gst/avi/gstavi.c: (plugin_init): * gst/avi/gstavimux.c: (gst_avi_mux_get_type), (gst_avi_mux_base_init), (gst_avi_mux_finalize), (gst_avi_mux_class_init), (gst_avi_mux_init), (gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps), (gst_avi_mux_pad_link), (gst_avi_mux_pad_unlink), (gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad), (gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header), (gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_video_header), (gst_avi_mux_riff_get_audio_header), (gst_avi_mux_add_index), (gst_avi_mux_write_index), (gst_avi_mux_bigfile), (gst_avi_mux_start_file), (gst_avi_mux_stop_file), (gst_avi_mux_restart_file), (gst_avi_mux_handle_event), (gst_avi_mux_fill_queue), (gst_avi_mux_send_pad_data), (gst_avi_mux_strip_buffer), (gst_avi_mux_do_audio_buffer), (gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer), (gst_avi_mux_loop), (gst_avi_mux_collect_pads), (gst_avi_mux_get_property), (gst_avi_mux_set_property), (gst_avi_mux_change_state): * gst/avi/gstavimux.h: Port AVI muxer to GStreamer-0.10 (#332031). * tests/check/Makefile.am: * tests/check/elements/avimux.c: * tests/check/elements/.cvsignore: Add unit test for AVI muxer. 2006-04-26 21:29:45 +0000 Stefan Kost gst/wavparse/gstwavparse.*: reverted patch #337625 for the price of 1 hour sleep Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state), (plugin_init): * gst/wavparse/gstwavparse.h: reverted patch #337625 for the price of 1 hour sleep 2006-04-26 20:11:18 +0000 Stefan Kost gst/wavparse/gstwavparse.*: correct partial implementation of push mode (from my last commit) Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl), (gst_wavparse_parse_cues), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_chain), (plugin_init): * gst/wavparse/gstwavparse.h: correct partial implementation of push mode (from my last commit) 2006-04-26 17:37:10 +0000 Wim Taymans ext/esd/esdsink.c: Fix compile problem by defining ESD_MAX_WRITE_SIZE if it is not in esd.h Original commit message from CVS: * ext/esd/esdsink.c: Fix compile problem by defining ESD_MAX_WRITE_SIZE if it is not in esd.h 2006-04-26 17:08:24 +0000 Tim-Philipp Müller gst/auparse/gstauparse.*: Rewrite auparse to suck a little bit less: make source pad dynamic, so decodebin/playbin wo... Original commit message from CVS: * gst/auparse/gstauparse.c: (gst_au_parse_base_init), (gst_au_parse_class_init), (gst_au_parse_init), (gst_au_parse_reset), (gst_au_parse_add_srcpad), (gst_au_parse_remove_srcpad), (gst_au_parse_parse_header), (gst_au_parse_chain), (gst_au_parse_src_convert), (gst_au_parse_src_query), (gst_au_parse_handle_seek), (gst_au_parse_sink_event), (gst_au_parse_src_event), (gst_au_parse_change_state): * gst/auparse/gstauparse.h: Rewrite auparse to suck a little bit less: make source pad dynamic, so decodebin/playbin work with non-raw formats like alaw/mulaw; add query function for duration/position queries; check whether we have enough data before attempting to parse the header (instead of crashing when that is not the case); work around audioconvert sucking by swapping endianness to the native endianness ourselves for float formats; send initial newsegment event. Fixes #161712. 2006-04-26 16:29:38 +0000 Zaheer Abbas Merali sys/osxaudio/: Port of osxaudiosink to 0.10 Original commit message from CVS: 2006-04-26 Zaheer Abbas Merali * sys/osxaudio/Makefile.am: * sys/osxaudio/gstosxaudioelement.c: (gst_osx_audio_element_get_type), (gst_osx_audio_element_class_init): * sys/osxaudio/gstosxaudioelement.h: * sys/osxaudio/gstosxaudiosink.c: (gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_base_init), (gst_osx_audio_sink_class_init), (gst_osx_audio_sink_init), (gst_osx_audio_sink_set_property), (gst_osx_audio_sink_get_property), (gst_osx_audio_sink_getcaps), (gst_osx_audio_sink_create_ringbuffer), (gst_osx_audio_sink_io_proc), (gst_osx_audio_sink_osxelement_init), (plugin_init): * sys/osxaudio/gstosxaudiosink.h: Port of osxaudiosink to 0.10 2006-04-26 08:55:27 +0000 Wim Taymans ext/esd/esdsink.c: Always write ESD_BUF_SIZE bytes and use ESD_MAX_WRITE_SIZE as the size of the ringbuffer. This sho... Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_prepare), (gst_esdsink_delay): Always write ESD_BUF_SIZE bytes and use ESD_MAX_WRITE_SIZE as the size of the ringbuffer. This should fix hangs with older esd sound servers. 2006-04-25 21:56:38 +0000 Stefan Kost Define GstElementDetails as const and also static (when defined as global) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/arts/gst_arts.c: * ext/artsd/gstartsdsink.c: * ext/audiofile/gstafparse.c: * ext/audiofile/gstafsink.c: * ext/audiofile/gstafsrc.c: * ext/audioresample/gstaudioresample.c: * ext/bz2/gstbz2dec.c: * ext/bz2/gstbz2enc.c: * ext/cdaudio/gstcdaudio.c: * ext/directfb/dfbvideosink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/libfame/gstlibfame.c: * ext/libmms/gstmms.c: (gst_mms_base_init): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/nas/nassink.c: (gst_nassink_base_init): * ext/neon/gstneonhttpsrc.c: * ext/sdl/sdlaudiosink.c: * ext/sdl/sdlvideosink.c: * ext/shout/gstshout.c: * ext/snapshot/gstsnapshot.c: * ext/sndfile/gstsf.c: * ext/swfdec/gstswfdec.c: * ext/tarkin/gsttarkindec.c: * ext/tarkin/gsttarkinenc.c: * ext/theora/theoradec.c: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init): * gst/chart/gstchart.c: * gst/colorspace/gstcolorspace.c: * gst/deinterlace/gstdeinterlace.c: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init): * gst/festival/gstfestival.c: * gst/filter/gstbpwsinc.c: * gst/filter/gstiir.c: * gst/filter/gstlpwsinc.c: * gst/freeze/gstfreeze.c: * gst/games/gstpuzzle.c: (gst_puzzle_base_init): * gst/librfb/gstrfbsrc.c: * gst/mixmatrix/mixmatrix.c: * gst/mpeg1sys/gstmpeg1systemencode.c: * gst/mpeg1videoparse/gstmp1videoparse.c: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/multifilesink/gstmultifilesink.c: * gst/overlay/gstoverlay.c: * gst/passthrough/gstpassthrough.c: * gst/playondemand/gstplayondemand.c: * gst/qtdemux/qtdemux.c: * gst/rtjpeg/gstrtjpegdec.c: * gst/rtjpeg/gstrtjpegenc.c: * gst/smooth/gstsmooth.c: * gst/smoothwave/gstsmoothwave.c: * gst/spectrum/gstspectrum.c: * gst/speed/gstspeed.c: * gst/stereo/gststereo.c: * gst/switch/gstswitch.c: * gst/tta/gstttadec.c: (gst_tta_dec_base_init): * gst/tta/gstttaparse.c: (gst_tta_parse_base_init): * gst/vbidec/gstvbidec.c: * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/virtualdub/gstxsharpen.c: * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init): * gst/y4m/gsty4mencode.c: * sys/cdrom/gstcdplayer.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: * sys/glsink/glimagesink.c: * sys/qcam/gstqcamsrc.c: * sys/v4l2/gstv4l2src.c: * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init): * sys/ximagesrc/ximagesrc.c: Define GstElementDetails as const and also static (when defined as global) 2006-04-25 21:56:38 +0000 Stefan Kost Define GstElementDetails as const and also static (when defined as global) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/arts/gst_arts.c: * ext/artsd/gstartsdsink.c: * ext/audiofile/gstafparse.c: * ext/audiofile/gstafsink.c: * ext/audiofile/gstafsrc.c: * ext/audioresample/gstaudioresample.c: * ext/bz2/gstbz2dec.c: * ext/bz2/gstbz2enc.c: * ext/cdaudio/gstcdaudio.c: * ext/directfb/dfbvideosink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/libfame/gstlibfame.c: * ext/libmms/gstmms.c: (gst_mms_base_init): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/nas/nassink.c: (gst_nassink_base_init): * ext/neon/gstneonhttpsrc.c: * ext/sdl/sdlaudiosink.c: * ext/sdl/sdlvideosink.c: * ext/shout/gstshout.c: * ext/snapshot/gstsnapshot.c: * ext/sndfile/gstsf.c: * ext/swfdec/gstswfdec.c: * ext/tarkin/gsttarkindec.c: * ext/tarkin/gsttarkinenc.c: * ext/theora/theoradec.c: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init): * gst/chart/gstchart.c: * gst/colorspace/gstcolorspace.c: * gst/deinterlace/gstdeinterlace.c: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init): * gst/festival/gstfestival.c: * gst/filter/gstbpwsinc.c: * gst/filter/gstiir.c: * gst/filter/gstlpwsinc.c: * gst/freeze/gstfreeze.c: * gst/games/gstpuzzle.c: (gst_puzzle_base_init): * gst/librfb/gstrfbsrc.c: * gst/mixmatrix/mixmatrix.c: * gst/mpeg1sys/gstmpeg1systemencode.c: * gst/mpeg1videoparse/gstmp1videoparse.c: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/multifilesink/gstmultifilesink.c: * gst/overlay/gstoverlay.c: * gst/passthrough/gstpassthrough.c: * gst/playondemand/gstplayondemand.c: * gst/qtdemux/qtdemux.c: * gst/rtjpeg/gstrtjpegdec.c: * gst/rtjpeg/gstrtjpegenc.c: * gst/smooth/gstsmooth.c: * gst/smoothwave/gstsmoothwave.c: * gst/spectrum/gstspectrum.c: * gst/speed/gstspeed.c: * gst/stereo/gststereo.c: * gst/switch/gstswitch.c: * gst/tta/gstttadec.c: (gst_tta_dec_base_init): * gst/tta/gstttaparse.c: (gst_tta_parse_base_init): * gst/vbidec/gstvbidec.c: * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/virtualdub/gstxsharpen.c: * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init): * gst/y4m/gsty4mencode.c: * sys/cdrom/gstcdplayer.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: * sys/glsink/glimagesink.c: * sys/qcam/gstqcamsrc.c: * sys/v4l2/gstv4l2src.c: * sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init): * sys/ximagesrc/ximagesrc.c: Define GstElementDetails as const and also static (when defined as global) 2006-04-25 21:39:46 +0000 Stefan Kost Define GstElementDetails as const and also static (when defined as global) Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: (gst_shout2send_init): * ext/shout2/gstshout2.h: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavidemux.c: (gst_avi_demux_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/gstnavseek.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstfdsink.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: Define GstElementDetails as const and also static (when defined as global) 2006-04-25 17:57:23 +0000 Tim-Philipp Müller ext/jpeg/gstjpegdec.c: Source pad has fixed caps. If we don't set this, bad things happen when the window is resized. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): Source pad has fixed caps. If we don't set this, bad things happen when the window is resized. 2006-04-25 16:38:50 +0000 Tim-Philipp Müller gst/matroska/: Handle case where the TrackType ebml chunk does not come before the Original commit message from CVS: * gst/matroska/Makefile.am: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_handle_src_event): * gst/matroska/matroska-ids.c: (gst_matroska_track_init_video_context), (gst_matroska_track_init_audio_context), (gst_matroska_track_init_subtitle_context), (gst_matroska_track_init_complex_context): * gst/matroska/matroska-ids.h: Handle case where the TrackType ebml chunk does not come before the TrackInfoAudio or TrackInfoVideo ebml chunk (#339446). Ignore QoS events. 2006-04-25 16:09:55 +0000 Wim Taymans gst/rtp/: It's codec_data, not codec_info. Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps): It's codec_data, not codec_info. 2006-04-25 11:45:00 +0000 Mark Nauwelaerts gst/matroska/matroska-demux.c: Handle codec_data for VfW compatibility codec IDs (#339451) Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps): Handle codec_data for VfW compatibility codec IDs (#339451) * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): Same here, handle codec_data and add additional caps we can handle now to the pad template (huffyuv, dv and h263 video) (#339451) 2006-04-25 11:09:24 +0000 Josef Zlomek gst/matroska/matroska-mux.c: Fix timestamping of B-frames, use signed integers, do some rounding (#339678). Original commit message from CVS: Patch by: Josef Zlomek * gst/matroska/matroska-mux.c: (gst_matroska_mux_create_buffer_header), (gst_matroska_mux_write_data): Fix timestamping of B-frames, use signed integers, do some rounding (#339678). 2006-04-24 18:30:55 +0000 Edgard Lima * ChangeLog: * ext/annodex/gstcmmlparser.c: just make it compile with --disable-gst-debug. Original commit message from CVS: just make it compile with --disable-gst-debug. 2006-04-23 15:55:30 +0000 Sébastien Moutte gst/matroska/matroska-demux.c: Fix a bad conversion using gst_guint64_to_gdouble. fabs ((gdouble) demux->index[entry]... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek): Fix a bad conversion using gst_guint64_to_gdouble. fabs ((gdouble) demux->index[entry].time - (gdouble) seek_pos) can not be replaced by fabs (gst_guint64_to_gdouble (demux->index[entry].time - seek_pos)) as the difference could be negative. fabs (gst_guint64_to_gdouble (demux->index[entry].time) - gst_guint64_to_gdouble (seek_pos)) is the good solution. Thanks to Tim who has seen my mistake. 2006-04-22 15:32:48 +0000 Sébastien Moutte gst/matroska/matroska-demux.c: Use gst_guint64_to_gdouble for conversions Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek): Use gst_guint64_to_gdouble for conversions * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgsticydemux.dsp: Add a project file for icydemux 2006-04-21 18:07:10 +0000 Fabrizio Gennari gst/avi/gstavidemux.c: When splitting audio chunks, the block alignment is not taken in consideration, so the smaller... Original commit message from CVS: Patch by: Fabrizio Gennari * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_massage_index): When splitting audio chunks, the block alignment is not taken in consideration, so the smaller chunks could be of size which is not a multiple of the block alignment. Fixes #336904 2006-04-21 17:59:03 +0000 Wim Taymans ext/raw1394/gstdv1394src.c: Use scale functions Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (gst_dv1394src_convert): Use scale functions 2006-04-21 17:27:40 +0000 Tim-Philipp Müller ext/dv/gstdv.c: Fix build. Original commit message from CVS: * ext/dv/gstdv.c: (plugin_init): Fix build. 2006-04-21 17:15:29 +0000 Tim-Philipp Müller gst/debug/progressreport.c: Add 'format' property to force querying to a particular format. Original commit message from CVS: * gst/debug/progressreport.c: (gst_progress_report_finalize), (gst_progress_report_class_init), (gst_progress_report_init), (gst_progress_report_do_query), (gst_progress_report_report), (gst_progress_report_set_property), (gst_progress_report_get_property): Add 'format' property to force querying to a particular format. 2006-04-21 15:50:28 +0000 Andy Wingo ext/dv/gstdv.c (plugin_init): libdv is a marginal decoder, at best, on big endian systems. Drop its rank in that case... Original commit message from CVS: 2006-04-21 Andy Wingo * ext/dv/gstdv.c (plugin_init): libdv is a marginal decoder, at best, on big endian systems. Drop its rank in that case. OTOH on x86 it's quite fine. See changes from today in gst-ffmpeg as well. 2006-04-21 12:40:41 +0000 Ed Catmur ext/lame/gstlame.c: Don't crash if we get an EOS event before the encoder has been set up (#339287). Original commit message from CVS: Patch by: Ed Catmur * ext/lame/gstlame.c: (gst_lame_sink_event): Don't crash if we get an EOS event before the encoder has been set up (#339287). 2006-04-21 09:27:11 +0000 Michael Smith Add icydemux, and tests. Original commit message from CVS: * configure.ac: * gst/icydemux/Makefile.am: * gst/icydemux/gsticydemux.c: (gst_icydemux_get_type), (gst_icydemux_base_init), (gst_icydemux_class_init), (gst_icydemux_reset), (gst_icydemux_init), (gst_icydemux_sink_setcaps), (gst_icydemux_dispose), (gst_icydemux_add_srcpad), (gst_icydemux_remove_srcpad), (unicodify), (gst_icydemux_unicodify), (gst_icydemux_parse_and_send_tags), (gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta), (gst_icydemux_chain), (gst_icydemux_change_state), (gst_icydemux_send_tag_event), (plugin_init): * gst/icydemux/gsticydemux.h: * tests/check/Makefile.am: * tests/check/elements/icydemux.c: (typefind_succeed), (plugin_init), (icydemux_found_pad), (create_icydemux), (cleanup_icydemux), (push_data), (GST_START_TEST), (icydemux_suite), (main): Add icydemux, and tests. 2006-04-20 17:48:29 +0000 Tim-Philipp Müller ext/flac/gstflacdec.c: Post SEGMENT_DONE message in TIME format. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_loop): Post SEGMENT_DONE message in TIME format. 2006-04-20 17:29:56 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: Added a couple of ifdefs to make it compile with other kernels. Original commit message from CVS: Added a couple of ifdefs to make it compile with other kernels. 2006-04-20 16:33:55 +0000 Fabrizio Gennari gst/avi/gstavidemux.c: Fix index creation when we have to scan the file to create an index. There may be other types ... Original commit message from CVS: Patch by: Fabrizio Gennari * gst/avi/gstavidemux.c: (gst_avi_demux_peek_tag), (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan): Fix index creation when we have to scan the file to create an index. There may be other types of RIFF 'LIST' chunks than 'movi' and we need to skip them properly as well or we'll end up reading garbage (#336889). Some other cosmetic changes. 2006-04-20 14:21:42 +0000 Tim-Philipp Müller ext/flac/gstflacdec.c: Add support for segment seeks (fixes #338290). Also demote some recurring debug message from D... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_loop), (gst_flac_dec_handle_seek_event): Add support for segment seeks (fixes #338290). Also demote some recurring debug message from DEBUG to LOG level. 2006-04-20 13:23:40 +0000 Tim-Philipp Müller gst/matroska/: Set DISCONT flag on first buffer after a discontinuity. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroskademux_do_index_seek), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_parse_blockgroup_or_simpleblock): * gst/matroska/matroska-ids.h: Set DISCONT flag on first buffer after a discontinuity. Fix newsegment events sent when seeking and honour KEY_UNIT seek flag. Create pad with bogus caps if we don't recognise the stream codec id. * gst/matroska/matroska-demux.h: Fix GObject macros. 2006-04-20 11:00:16 +0000 Mark Nauwelaerts gst/matroska/matroska-demux.c: Handle end of segment properly when set; don't dead-lock when posting start of segment... Original commit message from CVS: Patch by: Mark Nauwelaerts * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop): Handle end of segment properly when set; don't dead-lock when posting start of segment message when doing a segment seek. Fixes #338810. 2006-04-20 09:48:05 +0000 j^ gst/qtdemux/qtdemux.c: Never treat video streams as an audio stream. Original commit message from CVS: Patch by: j^ * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (qtdemux_video_caps): Never treat video streams as an audio stream. Add qtdrw mime type. Fixes #339041 2006-04-20 09:11:22 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Make mpeg2 aac audio work: create artificial private codec data chunk which faad2 seem... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps), (gst_matroska_demux_plugin_init): Make mpeg2 aac audio work: create artificial private codec data chunk which faad2 seems to require, just as we do for mpeg4 aac. Also call gst_riff_init(). Partially fixes #338767. 2006-04-19 15:16:33 +0000 Tim-Philipp Müller gst/wavenc/gstwavenc.*: Set caps on first outgoing buffer, so that it doesn't error out immediately with a non-negoti... Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_base_init), (gst_wavenc_class_init), (gst_wavenc_init), (gst_wavenc_create_header_buf), (gst_wavenc_push_header), (gst_wavenc_sink_setcaps), (get_id_from_name), (gst_wavenc_event), (gst_wavenc_chain), (gst_wavenc_change_state): * gst/wavenc/gstwavenc.h: Set caps on first outgoing buffer, so that it doesn't error out immediately with a non-negotiated error (#338716). Rewrite and clean up a bit; fix setcaps function to parse things properly; fix sink caps (8bit audio is unsigned and doesn't have depth); use boilerplate macros; remove unused properties stuff. 2006-04-19 09:27:00 +0000 Wim Taymans gst/qtdemux/qtdemux.c: For VBR audio, don't try to calculate the samples_per_frame. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): For VBR audio, don't try to calculate the samples_per_frame. Fixes #338935. 2006-04-18 18:14:34 +0000 Tim-Philipp Müller ext/gdk_pixbuf/gstgdkpixbuf.c: Leave JPEG decoding to our jpegdec plugin. gdkpixbufdec cannot handle MJPEG streams an... Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: Leave JPEG decoding to our jpegdec plugin. gdkpixbufdec cannot handle MJPEG streams and might be autoplugged for those if the user doesn't have jpegdec installed (resulting in a cryptic error message about huffman tables). Better to disable JPEG decoding here and let the user figure out that she needs to install jpegdec. 2006-04-18 18:04:48 +0000 Tim-Philipp Müller ext/gdk_pixbuf/gstgdkpixbuf.*: Make work with packetised/framed input (e.g. png-in-quicktime). Use Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_init), (gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_chain): * ext/gdk_pixbuf/gstgdkpixbuf.h: Make work with packetised/framed input (e.g. png-in-quicktime). Use GST_ELEMENT_ERROR when we return GST_FLOW_ERROR. Add some GST_DEBUG_FUNCPTR here and there. Use GST_LOG for recurring debug messages. Fix boilerplate macros. 2006-04-18 17:29:42 +0000 Tim-Philipp Müller ext/gdk_pixbuf/gstgdkpixbuf.c: No need to special-case for Gdk-2.0 any longer, we require Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_get_capslist), (gst_gdk_pixbuf_set_property), (gst_gdk_pixbuf_get_property): No need to special-case for Gdk-2.0 any longer, we require Gdk 2.2 or newer; minor clean-ups. 2006-04-18 17:17:55 +0000 Tim-Philipp Müller Rewrite a bit: use GstBaseSink::start and stop instead of a state change function; use GST_ELEMENT_ERROR for error re... Original commit message from CVS: * ext/shout2/gstshout2.c: (gst_shout2send_base_init), (gst_shout2send_class_init), (gst_shout2send_init), (set_shout_metadata), (gst_shout2send_set_metadata), (gst_shout2send_event), (gst_shout2send_start), (gst_shout2send_connect), (gst_shout2send_stop), (gst_shout2send_render), (gst_shout2send_set_property), (gst_shout2send_get_property), (gst_shout2send_setcaps), (plugin_init): * ext/shout2/gstshout2.h: * po/POTFILES.in: Rewrite a bit: use GstBaseSink::start and stop instead of a state change function; use GST_ELEMENT_ERROR for error reporting, not g_error() or GST_ERROR(); don't unref caps in setcaps function, will cause crashes or assertion failures; remove (unused) "sync" property, basesink already has such a property; misc. other minor fixes and cleanups. 2006-04-18 14:15:33 +0000 Tim-Philipp Müller Add translatable error message for when we cannot connect to the sound server, as "Cannot open resource for writing" ... Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_open), (gst_esdsink_prepare): * ext/esd/gstesd.c: (plugin_init): * po/POTFILES.in: Add translatable error message for when we cannot connect to the sound server, as "Cannot open resource for writing" isn't really an acceptable message to show to the user in this case. 2006-04-18 13:32:29 +0000 Tim-Philipp Müller sys/oss/gst-i18n-plugin.h: Remove bogus file that doesn't belong here. Original commit message from CVS: * sys/oss/gst-i18n-plugin.h: Remove bogus file that doesn't belong here. 2006-04-17 19:57:10 +0000 Philippe Valembois ext/shout2/gstshout2.*: Handle tags being received before the connection to the server is established properly (see #... Original commit message from CVS: Patch by: Philippe Valembois * ext/shout2/gstshout2.c: (gst_shout2send_init), (gst_shout2send_set_metadata), (gst_shout2send_event), (gst_shout2send_render), (gst_shout2send_change_state): * ext/shout2/gstshout2.h: Handle tags being received before the connection to the server is established properly (see #338636). 2006-04-17 19:43:32 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: Just added a gtk-doc comment. Original commit message from CVS: Just added a gtk-doc comment. 2006-04-17 19:12:36 +0000 Tim-Philipp Müller ext/shout2/gstshout2.c: Don't crash in case the connection to the server fails: don't set pointer to NULL by assignin... Original commit message from CVS: * ext/shout2/gstshout2.c: (gst_shout2send_render): Don't crash in case the connection to the server fails: don't set pointer to NULL by assigning FALSE; error out properly by using GST_ELEMENT_ERROR and returning GST_FLOW_ERROR (fixes #338636). Lastly, free connection before resetting the pointer. 2006-04-17 10:01:51 +0000 Alex Lancaster gst/id3demux/id3tags.c: (Fixes #338713) Original commit message from CVS: * gst/id3demux/id3tags.c: Recognise TCO (Genre) tags in ID3v2.2. Patch by Alex Lancaster (Fixes #338713) 2006-04-13 21:45:57 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: * sys/v4l2/v4l2src_calls.c: Fixed some memory leaks. Original commit message from CVS: Fixed some memory leaks. 2006-04-13 09:15:31 +0000 Thomas Vander Stichele * ChangeLog: * gst/rtp/Makefile.am: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: reverting rtp patches to fix freeze break on -base as explained on the list Original commit message from CVS: reverting rtp patches to fix freeze break on -base as explained on the list 2006-04-13 09:01:17 +0000 Tim-Philipp Müller gst/rtp/: Fix GObject macros. Original commit message from CVS: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpilbcdepay.h: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: Fix GObject macros. 2006-04-13 03:42:51 +0000 Philippe Kalaf gst/rtp/: Ported mulaw and alaw payloaders to use new base class Original commit message from CVS: 2006-04-12 Philippe Kalaf * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: Ported mulaw and alaw payloaders to use new base class * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpilbcpay.h: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcdepay.h: Added new iLBC payloader/depayloader. Payloader uses new audio payload base class. 2006-04-12 21:57:02 +0000 Edgard Lima * sys/v4l2/gstv4l2src.c: Fix to work in read mode. Original commit message from CVS: Fix to work in read mode. 2006-04-12 09:42:10 +0000 Wim Taymans ext/gdk_pixbuf/gstgdkpixbuf.c: Some cleanups. Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_get_capslist), (gst_gdk_pixbuf_sink_getcaps), (gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_init), (gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_chain): Some cleanups. Added RGBA as a possible output format. Correctly free the supported mimetypes. deprecate silent arg, it's not used. Return result from _alloc_buffer to peer. 2006-04-11 18:03:36 +0000 Tim-Philipp Müller gst/rtp/gstrtpmp4vdepay.c: Don't leak memory allocated by gst_buffer_new_and_alloc() by overwriting GST_BUFFER_MALLOC... Original commit message from CVS: * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_process): Don't leak memory allocated by gst_buffer_new_and_alloc() by overwriting GST_BUFFER_MALLOCDATA. 2006-04-11 15:27:31 +0000 Christian Schaller * gst-plugins-good.spec.in: fix version number macro Original commit message from CVS: fix version number macro 2006-04-11 09:35:45 +0000 Tim-Philipp Müller ext/libpng/gstpngdec.*: Handle more than one frame if the content is framed, like with png-in-quicktime (#331917). Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_endrow_callback), (user_end_callback), (gst_pngdec_caps_create_and_set), (gst_pngdec_chain), (gst_pngdec_sink_setcaps), (gst_pngdec_sink_event), (gst_pngdec_libpng_clear), (gst_pngdec_change_state): * ext/libpng/gstpngdec.h: Handle more than one frame if the content is framed, like with png-in-quicktime (#331917). 2006-04-10 19:55:31 +0000 Thomas Vander Stichele sys/oss/: - the user-visible error strings were in the wrong category Original commit message from CVS: * sys/oss/Makefile.am: * sys/oss/common.h: * sys/oss/gstosssink.c: (gst_oss_sink_init), (gst_oss_sink_open), (gst_oss_sink_prepare), (gst_oss_sink_unprepare): * sys/oss/gstosssrc.c: (gst_oss_src_prepare), (gst_oss_src_unprepare): - the user-visible error strings were in the wrong category - and the messages were not marked for translation - which is actually a good thing, because they were exactly the kind of message you would never want anyone to see - the macros were using variables that didn't exist in the macro arguments - and they were obviously copied from each other and then modified - so a common header makes sense 2006-04-10 17:16:09 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Fix parsing of newer stsd chunks again. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Fix parsing of newer stsd chunks again. 2006-04-10 16:09:03 +0000 Tim-Philipp Müller gst/matroska/ebml-read.c: Don't try to modify read-only data. Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_sint): Don't try to modify read-only data. * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_blockgroup_or_simpleblock): Fix comment (won't crash any longer now). 2006-04-10 15:48:55 +0000 Michael Smith ext/annodex/gstcmmlenc.c: Use copies of header buffers for caps to avoid circular refcounting problems (as in theorad... Original commit message from CVS: * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_set_header_on_caps): Use copies of header buffers for caps to avoid circular refcounting problems (as in theoradec, vorbisdec). * tests/check/elements/cmmldec.c: (GST_START_TEST): Fix a typo in test that meant it was testing the wrong thing. * tests/check/elements/cmmlenc.c: (check_headers): Fix refcount checks now that we use buffer-copies for caps. 2006-04-10 15:43:54 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Use static pad templates with ANY caps for audio and video source pads and get rid of ... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps), (gst_matroska_demux_subtitle_caps), (gst_matroska_demux_plugin_init): Use static pad templates with ANY caps for audio and video source pads and get rid of a lot of unnecessary (and partially broken) code for the template caps. Clean up caps finding functions. Fixes playback of audio files/streams that do not contain the sample rate and/or number of channels in the audio context (happens a lot with vorbis/mp3 .mka files it seems). Fixes #337183. Also add myself to copyright holders. 2006-04-10 15:29:21 +0000 Michael Smith ext/annodex/gstcmmlutils.c: Use g_list_delete_link () instead of g_list_remove_link () so that we free the link as we... Original commit message from CVS: * ext/annodex/gstcmmlutils.c: (gst_cmml_track_list_del_clip): Use g_list_delete_link () instead of g_list_remove_link () so that we free the link as well as the contained data. 2006-04-10 14:20:41 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Fix framerate calculation. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse_trak): Fix framerate calculation. 2006-04-10 10:10:55 +0000 Ryan Lortie (desrt) gst/avi/gstavidemux.c: Fix some crashers with empty chunks. (Fixes #337749) Original commit message from CVS: Patch by: Ryan Lortie (desrt) * gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_index), (gst_avi_demux_stream_header): Fix some crashers with empty chunks. (Fixes #337749) 2006-04-10 08:31:40 +0000 Wim Taymans gst/qtdemux/qtdemux.c: force mono 8000 Hz on AMR samples. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): force mono 8000 Hz on AMR samples. 2006-04-09 18:30:51 +0000 Sébastien Moutte ext/neon/gstneonhttpsrc.c: remove atoll by using g_ascii_strtoull (atoll is not supported on WIN32) Original commit message from CVS: * ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_start): remove atoll by using g_ascii_strtoull (atoll is not supported on WIN32) * sys/directdraw/gstdirectdrawsink.c: * sys/directsound/gstdirectsoundsink.c: done some cleans in sources * win32/vs6: add project files for neon, qtdemux 2006-04-09 17:31:37 +0000 Sébastien Moutte gst/level/gstlevel.c: use G_GINT64_CONSTANT for INT64 constants Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_set_caps),(gst_level_transform_ip): use G_GINT64_CONSTANT for INT64 constants * gst/videofilter/gstvideobalance.c: define rint for WIN32 #define rint(x) (floor((x)+0.5)) * win32/vs6/libgstavi.dsp: add missing libraries for the link and remove avimux.c from the project as it isn't ported to 0.10 yet 2006-04-09 14:00:32 +0000 Tim-Philipp Müller gst/matroska/ebml-read.c: Even better would be if we actually did the right thing here (also, G_GUINT64_CONSTANT only... Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_sint): Even better would be if we actually did the right thing here (also, G_GUINT64_CONSTANT only exists since GLib-2.10). 2006-04-09 13:52:03 +0000 Tim-Philipp Müller gst/matroska/ebml-read.c: Can't just replace 1LL with 1L here just because MSVC doesn't support it, as it might lead ... Original commit message from CVS: * gst/matroska/ebml-read.c: (gst_ebml_read_sint): Can't just replace 1LL with 1L here just because MSVC doesn't support it, as it might lead to incorrect results when doing the bitshifting here. Using GLib's G_GUINT64_CONSTANT() macro to force a 64-bit constant in a way that all compilers are happy with. 2006-04-08 21:48:01 +0000 Stefan Kost Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): * ext/arts/gst_arts.c: (gst_arts_class_init): * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): * ext/audiofile/gstafsink.c: (gst_afsink_class_init): * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): * ext/audioresample/gstaudioresample.c: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_class_init): * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): * ext/jack/gstjack.c: (gst_jack_class_init): * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): * ext/nas/nassink.c: (gst_nassink_class_init): * ext/shout/gstshout.c: (gst_icecastsend_class_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): * ext/sndfile/gstsf.c: (gst_sf_class_init): * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), (gst_swfdec_class_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): * gst/chart/gstchart.c: (gst_chart_class_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): * gst/festival/gstfestival.c: (gst_festival_class_init): * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstiir.c: (gst_iir_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_class_init): * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_class_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init): * gst/overlay/gstoverlay.c: (gst_overlay_class_init): * gst/passthrough/gstpassthrough.c: (passthrough_class_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): * gst/smooth/gstsmooth.c: (gst_smooth_class_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): * gst/stereo/gststereo.c: (gst_stereo_class_init): * gst/switch/gstswitch.c: (gst_switch_class_init): * gst/tta/gstttadec.c: (gst_tta_dec_class_init): * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_class_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): * sys/v4l2/gstv4l2colorbalance.c: (gst_v4l2_color_balance_channel_class_init): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), (gst_v4l2_tuner_norm_class_init): * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 2006-04-08 21:48:01 +0000 Stefan Kost Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): * ext/arts/gst_arts.c: (gst_arts_class_init): * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): * ext/audiofile/gstafsink.c: (gst_afsink_class_init): * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): * ext/audioresample/gstaudioresample.c: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_class_init): * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): * ext/jack/gstjack.c: (gst_jack_class_init): * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): * ext/nas/nassink.c: (gst_nassink_class_init): * ext/shout/gstshout.c: (gst_icecastsend_class_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): * ext/sndfile/gstsf.c: (gst_sf_class_init): * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), (gst_swfdec_class_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): * gst/chart/gstchart.c: (gst_chart_class_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): * gst/festival/gstfestival.c: (gst_festival_class_init): * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstiir.c: (gst_iir_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_class_init): * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_class_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init): * gst/overlay/gstoverlay.c: (gst_overlay_class_init): * gst/passthrough/gstpassthrough.c: (passthrough_class_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): * gst/smooth/gstsmooth.c: (gst_smooth_class_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): * gst/stereo/gststereo.c: (gst_stereo_class_init): * gst/switch/gstswitch.c: (gst_switch_class_init): * gst/tta/gstttadec.c: (gst_tta_dec_class_init): * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_class_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): * sys/v4l2/gstv4l2colorbalance.c: (gst_v4l2_color_balance_channel_class_init): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), (gst_v4l2_tuner_norm_class_init): * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 2006-04-08 21:21:45 +0000 Stefan Kost Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_class_init): * ext/esd/esdsink.c: (gst_esdsink_class_init): * ext/flac/gstflactag.c: (gst_flac_tag_class_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init): * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init): * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init): * ext/libmng/gstmngdec.c: (gst_mngdec_class_init): * ext/libmng/gstmngenc.c: (gst_mngenc_class_init): * ext/libpng/gstpngdec.c: (gst_pngdec_class_init): * ext/libpng/gstpngenc.c: (gst_pngenc_class_init): * ext/mikmod/gstmikmod.c: (gst_mikmod_class_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init): * ext/speex/gstspeexenc.c: (gst_speexenc_class_init): * gst/alpha/gstalpha.c: (gst_alpha_class_init): * gst/avi/gstavimux.c: (gst_avimux_class_init): * gst/debug/efence.c: (gst_efence_class_init): * gst/debug/negotiation.c: (gst_negotiation_class_init): * gst/flx/gstflxdec.c: (gst_flxdec_class_init): * gst/goom/gstgoom.c: (gst_goom_class_init): * gst/id3demux/gstid3demux.c: (gst_id3demux_class_init): * gst/interleave/deinterleave.c: (deinterleave_class_init): * gst/interleave/interleave.c: (interleave_class_init): * gst/law/alaw-decode.c: (gst_alawdec_class_init): * gst/law/alaw-encode.c: (gst_alawenc_class_init): * gst/law/mulaw-encode.c: (gst_mulawenc_class_init): * gst/median/gstmedian.c: (gst_median_class_init): * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init): * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init): * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init): * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init): * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init): * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init): * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init): * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init): * gst/smpte/gstsmpte.c: (gst_smpte_class_init): * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init): * gst/udp/gstudpsink.c: (gst_udpsink_class_init): * gst/videomixer/videomixer.c: (gst_videomixer_class_init): * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init): * sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init): * sys/oss/gstosssink.c: (gst_oss_sink_class_init): * sys/osxaudio/gstosxaudioelement.c: (gst_osxaudioelement_class_init): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init): * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 2006-04-08 19:06:25 +0000 Stefan Kost Fix more broken GObject macros Original commit message from CVS: * ext/mikmod/gstmikmod.h: * gst/level/gstlevel.h: Fix more broken GObject macros 2006-04-08 18:41:07 +0000 Stefan Kost Fix broken GObject macros Original commit message from CVS: * ext/xine/gstxine.h: * gst-libs/gst/play/play.h: * sys/v4l2/gstv4l2element.h: * sys/ximagesrc/ximageutil.h: Fix broken GObject macros 2006-04-08 18:25:55 +0000 Stefan Kost Fix broken GObject macros Original commit message from CVS: * ext/annodex/gstcmmldec.h: * ext/annodex/gstcmmlenc.h: * ext/annodex/gstcmmltag.h: * ext/cairo/gsttextoverlay.h: * ext/ladspa/gstsignalprocessor.h: * gst/matroska/ebml-read.h: * gst/matroska/ebml-write.h: * sys/osxaudio/gstosxaudioelement.h: Fix broken GObject macros 2006-04-08 18:23:04 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Don't make rounding errors in timestamp/duration calculations. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample), (gst_qtdemux_chain), (gst_qtdemux_add_stream), (qtdemux_dump_stsz), (qtdemux_dump_stco), (qtdemux_parse_trak): Don't make rounding errors in timestamp/duration calculations. Fix timestamps for AMR and IMA4. Fixes (#337436). Create a dummy segment even when there is no edit list. 2006-04-08 13:09:50 +0000 Tim-Philipp Müller ext/flac/gstflacdec.c: Don't try to seek beyond the end of the file (would occasionally display error dialogs in tote... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_handle_seek_event): Don't try to seek beyond the end of the file (would occasionally display error dialogs in totem when seeking to the end) (#335869). Will still throw an error though if the file is truncated and the total_samples value in the stream header is wrong. 2006-04-07 18:15:08 +0000 Tim-Philipp Müller ext/flac/gstflacdec.*: If the stream header doesn't contain the total number of samples, search for the last flac fra... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_calculate_crc8), (gst_flac_dec_scan_got_frame), (gst_flac_dec_scan_for_last_block), (gst_flac_dec_metadata_callback): * ext/flac/gstflacdec.h: If the stream header doesn't contain the total number of samples, search for the last flac frame at the end of the file and calculate the total duration from that frame's offset (fixes #337609). 2006-04-07 15:53:43 +0000 Zaheer Abbas Merali Typo fix, s/XFree86/X11 and added doc blurb saying that it fixates to 25fps Original commit message from CVS: 2006-04-07 Zaheer Abbas Merali * ext/amrwb/amrwb-code/Makefile.am: * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_recalc), (gst_ximagesrc_create), (gst_ximagesrc_set_property): Typo fix, s/XFree86/X11 and added doc blurb saying that it fixates to 25fps 2006-04-07 15:47:27 +0000 Zaheer Abbas Merali tests/icles/ximagesrc-test.c: Actually assert that pipeline goes to playing Original commit message from CVS: 2006-04-07 Zaheer Abbas Merali * tests/icles/ximagesrc-test.c: (main): Actually assert that pipeline goes to playing 2006-04-07 15:27:40 +0000 Zaheer Abbas Merali sys/ximagesrc/ximagesrc.c: Fix typo, C++ style comments and other small cleanups Original commit message from CVS: 2006-04-07 Zaheer Abbas Merali * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_recalc), (composite_pixel), (gst_ximagesrc_ximage_get), (gst_ximagesrc_create), (gst_ximagesrc_set_property): Fix typo, C++ style comments and other small cleanups 2006-04-07 10:48:19 +0000 Edward Hervey gst/avi/gstavidemux.c: Don't unref the GstPadTemplate returned by gst_element_class_get_pad_template(). Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream): Don't unref the GstPadTemplate returned by gst_element_class_get_pad_template(). 2006-04-06 19:16:02 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Added full edit list support. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_init), (gst_qtdemux_handle_src_query), (gst_qtdemux_find_index), (gst_qtdemux_find_keyframe), (gst_qtdemux_find_segment), (gst_qtdemux_move_stream), (gst_qtdemux_perform_seek), (gst_qtdemux_do_seek), (gst_qtdemux_change_state), (gst_qtdemux_activate_segment), (gst_qtdemux_prepare_current_sample), (gst_qtdemux_advance_sample), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (qtdemux_parse_trak): Added full edit list support. Avoid overflows in prologue image detection code. Avoid roundoff errors in timestamp calculations. 2006-04-06 11:35:26 +0000 j^ Unify the long descriptions in the plugin details (#337263). Original commit message from CVS: Patch by: j^ * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/arts/gst_arts.c: * ext/artsd/gstartsdsink.c: * ext/audiofile/gstafparse.c: * ext/audiofile/gstafsink.c: * ext/audiofile/gstafsrc.c: * ext/cdaudio/gstcdaudio.c: * ext/directfb/dfbvideosink.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: * ext/gsm/gstgsmdec.c: * ext/gsm/gstgsmenc.c: * ext/hermes/gsthermescolorspace.c: * ext/ivorbis/vorbisfile.c: * ext/lcs/gstcolorspace.c: * ext/libfame/gstlibfame.c: * ext/libmms/gstmms.c: (gst_mms_base_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/nas/nassink.c: (gst_nassink_base_init): * ext/neon/gstneonhttpsrc.c: * ext/polyp/polypsink.c: (gst_polypsink_base_init): * ext/sdl/sdlaudiosink.c: * ext/sdl/sdlvideosink.c: * ext/shout/gstshout.c: * ext/snapshot/gstsnapshot.c: * ext/sndfile/gstsf.c: * ext/tarkin/gsttarkindec.c: * ext/tarkin/gsttarkinenc.c: * ext/theora/theoradec.c: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init): * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: * gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init): * gst/chart/gstchart.c: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init): * gst/festival/gstfestival.c: * gst/filter/gstiir.c: * gst/filter/gstlpwsinc.c: * gst/freeze/gstfreeze.c: * gst/games/gstpuzzle.c: (gst_puzzle_base_init): * gst/mixmatrix/mixmatrix.c: * gst/mpeg1sys/gstmpeg1systemencode.c: * gst/mpeg1videoparse/gstmp1videoparse.c: * gst/mpeg2sub/gstmpeg2subt.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/multifilesink/gstmultifilesink.c: * gst/overlay/gstoverlay.c: * gst/passthrough/gstpassthrough.c: * gst/playondemand/gstplayondemand.c: * gst/qtdemux/qtdemux.c: * gst/rtjpeg/gstrtjpegdec.c: * gst/rtjpeg/gstrtjpegenc.c: * gst/smooth/gstsmooth.c: * gst/tta/gstttadec.c: (gst_tta_dec_base_init): * gst/tta/gstttaparse.c: (gst_tta_parse_base_init): * gst/videocrop/gstvideocrop.c: * gst/videodrop/gstvideodrop.c: * gst/virtualdub/gstxsharpen.c: * gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init): * gst/y4m/gsty4mencode.c: Unify the long descriptions in the plugin details (#337263). 2006-04-06 09:14:30 +0000 Brian Cameron sys/sunaudio/gstsunaudiosink.*: Use spec->segsize and spec->segtotal in the prepare function to initialise the ring b... Original commit message from CVS: Patch by: Brian Cameron * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_init), (gst_sunaudiosink_prepare), (gst_sunaudiosink_write): * sys/sunaudio/gstsunaudiosink.h: Use spec->segsize and spec->segtotal in the prepare function to initialise the ring buffer instead of using the buffer-time property (#337421). 2006-04-06 08:52:51 +0000 Tim-Philipp Müller configure.ac: Bump core requirements to CVS for gst_pad_query_peer_duration() which is used by speexdec. Original commit message from CVS: * configure.ac: Bump core requirements to CVS for gst_pad_query_peer_duration() which is used by speexdec. 2006-04-05 18:27:22 +0000 Tim-Philipp Müller ext/speex/: Fix seeking and duration queries (#337033); clean up and refactor a bit. Original commit message from CVS: * ext/speex/gstspeex.c: (plugin_init): * ext/speex/gstspeexdec.c: (gst_speex_dec_class_init), (gst_speex_dec_reset), (gst_speex_dec_init), (speex_dec_convert), (speex_get_sink_query_types), (speex_dec_sink_query), (speex_get_src_query_types), (speex_dec_src_query), (speex_dec_src_event), (speex_dec_sink_event), (speex_dec_chain_parse_header), (speex_dec_chain_parse_comments), (speex_dec_chain_parse_data), (speex_dec_chain), (gst_speex_dec_get_property), (gst_speex_dec_set_property), (speex_dec_change_state): * ext/speex/gstspeexdec.h: Fix seeking and duration queries (#337033); clean up and refactor a bit. 2006-04-05 12:41:14 +0000 Thomas Vander Stichele ext/raw1394/gstdv1394src.c: distinguish between device not found and could not open for reading Original commit message from CVS: * ext/raw1394/gstdv1394src.c: distinguish between device not found and could not open for reading 2006-04-05 08:36:55 +0000 Wim Taymans gst/qtdemux/qtdemux.c: Use duration as segment stop position if none is explicitly configured. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_perform_seek), (gst_qtdemux_do_seek), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop): Use duration as segment stop position if none is explicitly configured. Also perform EOS when we run past the segment stop. 2006-04-04 11:20:58 +0000 Wim Taymans gst/qtdemux/qtdemux.c: More cleanups, added comments. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_go_back), (gst_qtdemux_perform_seek), (gst_qtdemux_do_seek), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_parse_tree), (qtdemux_parse_trak): More cleanups, added comments. Mark discontinuities on outgoing buffers. Post better errors when something goes wrong. Handle EOS and segment end properly. 2006-04-04 08:31:10 +0000 Wim Taymans gst/qtdemux/qtdemux.*: Handle stss boxes so we can mark and find keyframes. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_init), (gst_qtdemux_push_event), (gst_qtdemux_go_back), (gst_qtdemux_perform_seek), (gst_qtdemux_do_seek), (gst_qtdemux_handle_src_event), (plugin_init), (gst_qtdemux_change_state), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop), (gst_qtdemux_chain), (qtdemux_sink_activate_pull), (gst_qtdemux_add_stream), (qtdemux_parse), (qtdemux_parse_tree), (qtdemux_parse_trak), (qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num), (qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds): * gst/qtdemux/qtdemux.h: Handle stss boxes so we can mark and find keyframes. Implement correct accurate and keyframe seeking. Use _DEBUG_OBJECT when possible. 2006-04-03 13:29:20 +0000 Thomas Vander Stichele * tests/check/elements/.gitignore: ignore more Original commit message from CVS: ignore more 2006-04-03 13:28:55 +0000 Thomas Vander Stichele * pkgconfig/Makefile.am: fix dist Original commit message from CVS: fix dist 2006-04-03 09:02:29 +0000 Thomas Vander Stichele add a .pc file so other modules can use good plugins in tests Original commit message from CVS: * Makefile.am: * configure.ac: * pkgconfig/.cvsignore: * pkgconfig/Makefile.am: * pkgconfig/gstreamer-plugins-good-uninstalled.pc.in: add a .pc file so other modules can use good plugins in tests 2006-04-01 16:50:49 +0000 Thomas Vander Stichele * common: * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-taglib.xml: * ext/taglib/gsttaglibmux.c: * tests/check/elements/id3v2mux.c: add taglib checks and docs Original commit message from CVS: add taglib checks and docs 2006-04-01 15:30:51 +0000 Thomas Vander Stichele * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/upload.mak: disable use of AS_LIBTOOL_TAGS, it doesn't work correctly Original commit message from CVS: disable use of AS_LIBTOOL_TAGS, it doesn't work correctly 2006-04-01 14:03:03 +0000 Thomas Vander Stichele * common: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-videobalance.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: adding inspect files Original commit message from CVS: adding inspect files 2006-04-01 10:15:33 +0000 Thomas Vander Stichele * tests/icles/ximagesrc-test.c: 5 second timeout Original commit message from CVS: 5 second timeout 2006-04-01 10:14:26 +0000 Thomas Vander Stichele * tests/icles/.gitignore: * tests/icles/Makefile.am: * tests/icles/ximagesrc-test.c: rename test Original commit message from CVS: rename test 2006-04-01 10:09:11 +0000 Thomas Vander Stichele * gst/audiofx/gststereo.c: rework build; add translations for v4l2 Original commit message from CVS: rework build; add translations for v4l2 2006-04-01 10:09:11 +0000 Thomas Vander Stichele * gst/equalizer/gstiirequalizer.c: * gst/qtdemux/qtdemux.c: * gst/spectrum/gstspectrum.c: * gst/videocrop/gstvideocrop.c: * sys/directdraw/gstdirectdrawplugin.c: * sys/directsound/gstdirectsoundplugin.c: * sys/v4l2/gstv4l2.c: * sys/ximage/gstximagesrc.c: rework build; add translations for v4l2 Original commit message from CVS: rework build; add translations for v4l2 2006-04-01 09:56:45 +0000 Thomas Vander Stichele configure.ac: clean up, use AS_VERSION and AS_NANO Original commit message from CVS: * configure.ac: clean up, use AS_VERSION and AS_NANO * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): use PACKAGE_VERSION define * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: updated 2006-04-01 09:54:39 +0000 Thomas Vander Stichele configure.ac: rework similarly to other modules Original commit message from CVS: * configure.ac: rework similarly to other modules * ext/a52dec/gsta52dec.c: * ext/amrnb/amrnb.c: * ext/dvdnav/dvdnavsrc.c: * ext/dvdread/dvdreadsrc.c: * ext/lame/gstlame.c: * ext/mad/gstid3tag.c: * ext/mpeg2dec/gstmpeg2dec.c: * ext/sidplay/gstsiddec.cc: * gst/asfdemux/gstasf.c: * gst/dvdlpcmdec/gstdvdlpcmdec.c: * gst/dvdsub/gstdvdsubdec.c: * gst/iec958/ac3iec.c: * gst/mpegaudioparse/gstmpegaudioparse.c: * gst/mpegstream/gstmpegstream.c: * gst/realmedia/rmdemux.c: (plugin_init): use the correct defines 2006-03-31 17:52:36 +0000 Zaheer Abbas Merali Add tests and fix PAR caps issue to ximagesrc Original commit message from CVS: 2006-03-31 Zaheer Abbas Merali * Makefile.am: * configure.ac: * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_ximage_get), (gst_ximagesrc_get_caps), (gst_ximagesrc_class_init): * sys/ximagesrc/ximageutil.c: * tests/Makefile.am: * tests/icles/Makefile.am: * tests/icles/ximagesrc-test.c: (terminate_playback), (main): Add tests and fix PAR caps issue to ximagesrc 2006-03-31 16:32:47 +0000 Zaheer Abbas Merali sys/ximagesrc/ximagesrc.c: Add docs to ximagesrc Original commit message from CVS: 2006-03-31 Zaheer Abbas Merali * sys/ximagesrc/ximagesrc.c: Add docs to ximagesrc 2006-03-31 15:21:35 +0000 Zaheer Abbas Merali sys/ximagesrc/: Fix ximagesrc so a) the cursor doesnt trail and b) there are no yellow rectangles with the cursor Original commit message from CVS: 2006-03-31 Zaheer Abbas Merali * sys/ximagesrc/ximagesrc.c: (composite_pixel), (gst_ximagesrc_ximage_get), (gst_ximagesrc_set_property), (gst_ximagesrc_get_caps), (gst_ximagesrc_class_init): * sys/ximagesrc/ximagesrc.h: * sys/ximagesrc/ximageutil.c: (ximageutil_xcontext_get): * sys/ximagesrc/ximageutil.h: Fix ximagesrc so a) the cursor doesnt trail and b) there are no yellow rectangles with the cursor 2006-03-30 23:46:42 +0000 Sébastien Moutte * win32/vs6/gst_plugins_good.dsw: * win32/vs6/libgstalaw.dsp: * win32/vs6/libgstalpha.dsp: * win32/vs6/libgstalphacolor.dsp: * win32/vs6/libgstapetag.dsp: * win32/vs6/libgstauparse.dsp: * win32/vs6/libgstautodetect.dsp: * win32/vs6/libgstavi.dsp: * win32/vs6/libgstcutter.dsp: * win32/vs6/libgsteffectv.dsp: * win32/vs6/libgstflx.dsp: * win32/vs6/libgstgoom.dsp: * win32/vs6/libgstid3demux.dsp: * win32/vs6/libgstinterleave.dsp: * win32/vs6/libgstjpeg.dsp: * win32/vs6/libgstlevel.dsp: * win32/vs6/libgstmatroska.dsp: * win32/vs6/libgstmedian.dsp: * win32/vs6/libgstmonoscope.dsp: * win32/vs6/libgstmulaw.dsp: * win32/vs6/libgstmultipart.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstsmpte.dsp: * win32/vs6/libgstspeex.dsp: * win32/vs6/libgstvideobalance.dsp: * win32/vs6/libgstvideobox.dsp: * win32/vs6/libgstvideoflip.dsp: * win32/vs6/libgstvideomixer.dsp: * win32/vs6/libgstwavenc.dsp: * win32/vs6/libgstwavparse.dsp: I'm too lazy to comment this Original commit message from CVS: *** empty log message *** 2006-03-30 23:37:16 +0000 Sébastien Moutte ext\jpeg\smokecodec.c: use of GST_DEBUG instead of DEBUG(a...) for WIN32 Original commit message from CVS: * ext\jpeg\smokecodec.c: use of GST_DEBUG instead of DEBUG(a...) for WIN32 * ext\speex\gstspeexenc.c: (gst_speexenc_set_header_on_caps): move first instruction after all variables declarations * gst\alpha\gstalpha.c: * gst\effectv\gstshagadelic.c: * gst\smpte\paint.c: * gst\videofilter\gstvideobalance.c: define M_PI if it's not defined (it's not defined on WIN32) * gst\cutter\gstcutter.c: (gst_cutter_chain): * gst\id3demux\id3v2frames.c: (parse_relative_volume_adjustment_two): * gst\level\gstlevel.c: (gst_level_set_property), (gst_level_transform_ip): * gst\matroska\matroska-demux.c: (gst_matroska_demux_parse_info), (gst_matroska_demux_video_caps): * gst\matroska\matroska-mux.c: (gst_matroska_mux_start), (gst_matroska_mux_finish): * gst\wavparse\gstwavparse.c: (gst_wavparse_stream_data): use gst_guint64_to_gdouble for conversions * gst\goom\filters.c: (setPixelRGB_): fix a debug which was using undefined variable * gst\level\gstlevel.c: (gst_level_set_caps), (gst_level_transform_ip): * gst\matroska\ebml-read.c: (gst_ebml_read_sint): replace LL suffix with L suffix (LL isn't supported by MSVC6.0) * win32/vs6: add vs6 projects files for most of plugins-good 2006-03-30 15:37:05 +0000 Wim Taymans better/unified long descriptions Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/smpte/gstsmpte.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init): better/unified long descriptions Fixed #336602 Some cleanups to auparse, don't send multiple newsegments. 2006-03-29 16:06:50 +0000 Michael Dominic K ext/dv/gstdvdemux.*: Seek in READY patch. Only works for pull based mode. Original commit message from CVS: From a patch by: Michael Dominic K. * ext/dv/gstdvdemux.c: (gst_dvdemux_class_init), (gst_dvdemux_reset), (gst_dvdemux_src_convert), (gst_dvdemux_send_event), (gst_dvdemux_flush), (gst_dvdemux_loop), (gst_dvdemux_sink_activate_pull), (gst_dvdemux_change_state): * ext/dv/gstdvdemux.h: Seek in READY patch. Only works for pull based mode. Fixes #323880 2006-03-28 16:06:05 +0000 Tim-Philipp Müller ext/lame/gstlame.*: Make xingheader property non-functional, it's broken anyway after all (use xingmux instead). Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup): * ext/lame/gstlame.h: Make xingheader property non-functional, it's broken anyway after all (use xingmux instead). 2006-03-28 15:10:18 +0000 Tim-Philipp Müller ext/lame/gstlame.c: On EOS, flush encoder and send remaining data. Fix return value handling in sink event function. Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_event): On EOS, flush encoder and send remaining data. Fix return value handling in sink event function. 2006-03-27 17:06:45 +0000 Edgard Lima * sys/v4l2/v4l2src_calls.c: Small fix, now pwc driver can tell about its buffers. Original commit message from CVS: Small fix, now pwc driver can tell about its buffers. 2006-03-27 14:09:18 +0000 Tim-Philipp Müller ext/gdk_pixbuf/gstgdkpixbuf.c: Fix two crashers: don't unref the same caps twice, and set pixbuf loader to NULL after... Original commit message from CVS: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_event): Fix two crashers: don't unref the same caps twice, and set pixbuf loader to NULL after freeing it. 2006-03-27 14:00:02 +0000 Wim Taymans ext/speex/gstspeexenc.*: Don't leak adapter. Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speexenc_class_init), (gst_speexenc_finalize), (gst_speexenc_sink_setcaps), (gst_speexenc_chain): * ext/speex/gstspeexenc.h: Don't leak adapter. A push *always* takes ownership of the buffer, even on errors. Small cleanups. 2006-03-26 19:56:37 +0000 Tim-Philipp Müller ext/taglib/gsttaglib.*: Fix newsegment event handling a bit. We need to cache the first newsegment event, because we ... Original commit message from CVS: * ext/taglib/gsttaglib.cc: * ext/taglib/gsttaglib.h: Fix newsegment event handling a bit. We need to cache the first newsegment event, because we can't adjust offsets yet when we get it, as we don't know the size of the tag yet for sure at that point. Also do some minor cleaning up here and there and add some debug statements. 2006-03-26 12:24:56 +0000 Tim-Philipp Müller gst/id3demux/gstid3demux.c: Create source pad without leaking. Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad): Create source pad without leaking. 2006-03-25 21:57:24 +0000 Tim-Philipp Müller ext/taglib/gsttaglib.cc: We do not want to proxy the caps on the sink pad; our source pad should have application/x-i... Original commit message from CVS: * ext/taglib/gsttaglib.cc: We do not want to proxy the caps on the sink pad; our source pad should have application/x-id3 caps; also, don't use already-freed strings in debug messages; finally, adjust buffer offsets on buffers sent out. 2006-03-25 13:02:55 +0000 Tim-Philipp Müller sys/v4l2/gstv4l2src.c: Older kernels don't seem to have this particular v4l2 format, so comment out until this gets f... Original commit message from CVS: * sys/v4l2/gstv4l2src.c: Older kernels don't seem to have this particular v4l2 format, so comment out until this gets fixed properly (and make buildbots happy). 2006-03-25 05:31:28 +0000 Edgard Lima * common: * sys/v4l2/gstv4l2colorbalance.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2element.c: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: Just make few things more robust and also some identation. Original commit message from CVS: Just make few things more robust and also some identation. 2006-03-24 19:41:03 +0000 Wim Taymans ext/flac/: Spifify a bit. Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_handle_seek_event): * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.h: Spifify a bit. Fix deadly lock order error in seeking code, STREAM_LOCK cannot be taken within LOCK and the streaming variables are protected with the STREAM_LOCK anyway. 2006-03-24 18:56:16 +0000 Wim Taymans gst/avi/gstavidemux.c: this patch combines the global init_frames with the stream init_frames. Rationale being that t... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_index), (gst_avi_demux_stream_index), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_handle_seek): this patch combines the global init_frames with the stream init_frames. Rationale being that the global delay should be subtracted from any stream delay. Fixes #335858. 2006-03-24 17:11:56 +0000 Stefan Kost gst/: use DEBUG_FUNCPTR for collectpads Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_init): * gst/smpte/gstsmpte.c: (gst_smpte_init): * gst/videomixer/videomixer.c: (gst_videomixer_init): use DEBUG_FUNCPTR for collectpads 2006-03-24 09:54:00 +0000 Tim-Philipp Müller ext/jpeg/gstjpegenc.c: Don't crash when encoding images where the number of rows isn't a multiple of 2*DCTSIZE. Add s... Original commit message from CVS: * ext/jpeg/gstjpegenc.c: (gst_jpegenc_init), (gst_jpegenc_chain): Don't crash when encoding images where the number of rows isn't a multiple of 2*DCTSIZE. Add some GST_DEBUG_FUNCPTR. 2006-03-23 21:28:06 +0000 Tim-Philipp Müller More state change function fixes. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_change_state): * gst/interleave/deinterleave.c: (deinterleave_change_state): * gst/interleave/interleave.c: (interleave_change_state): * gst/wavenc/gstwavenc.c: (gst_wavenc_change_state): More state change function fixes. 2006-03-23 20:12:47 +0000 Wim Taymans ext/esd/esdsink.*: Fix esd choppy playback by configuring audiosink correctly. Fixes #325191 Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_class_init), (gst_esdsink_getcaps), (gst_esdsink_open), (gst_esdsink_close), (gst_esdsink_prepare), (gst_esdsink_unprepare), (gst_esdsink_delay), (gst_esdsink_reset): * ext/esd/esdsink.h: Fix esd choppy playback by configuring audiosink correctly. Fixes #325191 2006-03-23 19:57:34 +0000 Tim-Philipp Müller ext/libpng/gstpngdec.c: Make state change function thread-safe. Original commit message from CVS: * ext/libpng/gstpngdec.c: (gst_pngdec_change_state): Make state change function thread-safe. 2006-03-23 16:50:32 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: Don't try to read beyond the end of the file just because the header claims a bigger size... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Don't try to read beyond the end of the file just because the header claims a bigger size (like with truncated files). 2006-03-23 15:36:27 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.*: Delay source pad creation until we have the first chunk of media data, so the we can exam... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_loop): * gst/wavparse/gstwavparse.h: Delay source pad creation until we have the first chunk of media data, so the we can examine the data and adjust the caps accordingly if required. This makes playback of .wav files with DTS-declared-as-PCM content work (#313266). 2006-03-22 19:50:56 +0000 Christian Schaller * gst-plugins-good.spec.in: add videobalance plugn Original commit message from CVS: add videobalance plugn 2006-03-22 13:02:11 +0000 Jan Schmidt * ChangeLog: mention fixed bug number in the changelog Original commit message from CVS: mention fixed bug number in the changelog 2006-03-22 13:00:34 +0000 Jan Schmidt gst/: Don't attempt typefinding on too-short buffers that have been completely trimmed away. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain): * gst/id3demux/gstid3demux.c: (gst_id3demux_chain): Don't attempt typefinding on too-short buffers that have been completely trimmed away. * gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag): Improve the debug output 2006-03-21 18:12:59 +0000 Wim Taymans ext/esd/esdsink.c: Some cleanups. Original commit message from CVS: * ext/esd/esdsink.c: (gst_esdsink_class_init), (gst_esdsink_init), (gst_esdsink_finalize), (gst_esdsink_getcaps), (gst_esdsink_open), (gst_esdsink_close), (gst_esdsink_prepare), (gst_esdsink_write), (gst_esdsink_set_property), (gst_esdsink_get_property): Some cleanups. Reset fd to -1 when we close them. 2006-03-21 16:19:37 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: the OPTIONS request result is optional so don't fail on it. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): the OPTIONS request result is optional so don't fail on it. 2006-03-21 14:53:36 +0000 Edward Hervey gst/: gcc 4.1 unreferenced pointer fixes. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_reset): * gst/id3demux/gstid3demux.c: (gst_id3demux_reset): * gst/wavparse/gstwavparse.c: (gst_wavparse_create_sourcepad), (gst_wavparse_stream_headers), (gst_wavparse_send_event), (gst_wavparse_change_state): gcc 4.1 unreferenced pointer fixes. 2006-03-21 13:07:31 +0000 Tommi Myöhänen gst/wavparse/gstwavparse.c: Fix block alignment calculation. Alignment should be done before adding the byte offset w... Original commit message from CVS: Patch by: Tommi Myöhänen * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek): Fix block alignment calculation. Alignment should be done before adding the byte offset where the data starts (#335231). 2006-03-20 18:34:21 +0000 Jan Schmidt gst/matroska/ebml-write.c: Ensure that we set correct caps on buffers that are transferred direct from the input. Original commit message from CVS: * gst/matroska/ebml-write.c: (gst_ebml_write_element_push): Ensure that we set correct caps on buffers that are transferred direct from the input. 2006-03-20 17:38:48 +0000 Jan Schmidt gst/goom/: Free filter data when cleaning up. (Fixes: #334995) Original commit message from CVS: * gst/goom/filters.c: (zoomFilterDestroy): * gst/goom/goom_core.c: (goom_close): Free filter data when cleaning up. (Fixes: #334995) 2006-03-20 08:59:29 +0000 Tim-Philipp Müller ext/taglib/gsttaglib.h: Fix left-over gst_my_filter_get_type. Original commit message from CVS: * ext/taglib/gsttaglib.h: Fix left-over gst_my_filter_get_type. 2006-03-17 16:34:36 +0000 Zaheer Abbas Merali * sys/ximage/gstximagesrc.c: Have a show mouse pointer property and use it if we can Original commit message from CVS: Have a show mouse pointer property and use it if we can 2006-03-17 15:33:08 +0000 Tim-Philipp Müller configure.ac: Don't compile udp and rtsp plugins on win32 (mingw) or other systems that don't have for... Original commit message from CVS: * configure.ac: Don't compile udp and rtsp plugins on win32 (mingw) or other systems that don't have for some reason (#316203). 2006-03-16 17:28:07 +0000 Zaheer Abbas Merali * ChangeLog: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gstdv1394src.h: Change bus reset handler so it reports useful information such as whether the device being used connected or disconne... Original commit message from CVS: Change bus reset handler so it reports useful information such as whether the device being used connected or disconnected 2006-03-16 16:06:22 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: We only care about gain and peak data for the master volume. Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_relative_volume_adjustment_two): We only care about gain and peak data for the master volume. 2006-03-16 13:22:28 +0000 Tim-Philipp Müller gst/id3demux/id3v2frames.c: Read replay gain tags (#323721). Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_id_string), (parse_unique_file_identifier), (parse_relative_volume_adjustment_two), (id3v2_tag_to_taglist): Read replay gain tags (#323721). 2006-03-15 23:19:30 +0000 Tim-Philipp Müller configure.ac: Bump requirements to gst-plugins-base CVS because of buggy gst_tag_from_id3_user_tag() in 0.10.5. Original commit message from CVS: * configure.ac: Bump requirements to gst-plugins-base CVS because of buggy gst_tag_from_id3_user_tag() in 0.10.5. 2006-03-15 22:30:24 +0000 Philippe Kalaf * ChangeLog: * gst/rtp/gstrtppcmadepay.c: Fixed one of the caps in the code from mulaw to alaw. Original commit message from CVS: Fixed one of the caps in the code from mulaw to alaw. 2006-03-15 16:21:38 +0000 Jan Schmidt gst/apetag/gsttagdemux.c: Ensure that we set caps on the buffers we pass. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain): Ensure that we set caps on the buffers we pass. * gst/id3demux/gstid3demux.c: (gst_id3demux_chain), (gst_id3demux_sink_activate): Ensure that we set caps on the buffers we pass. Use STREAM, TYPE_NOT_FOUND as the error class when typefinding fails. 2006-03-15 16:17:12 +0000 Edward Hervey Fix memleak with gst_static_pad_template_get(). Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_init): * ext/dv/gstdvdemux.c: (gst_dvdemux_init), (gst_dvdemux_add_pads): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_setcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_init): * ext/jpeg/gstsmokedec.c: (gst_smokedec_init): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init): * ext/libmng/gstmngdec.c: (gst_mngdec_init), (gst_mngdec_src_getcaps): * ext/libpng/gstpngdec.c: (gst_pngdec_init), (gst_pngdec_caps_create_and_set): * ext/libpng/gstpngenc.c: (gst_pngenc_init): * ext/mikmod/gstmikmod.c: (gst_mikmod_init): * ext/speex/gstspeexdec.c: (gst_speex_dec_init): * gst/alpha/gstalpha.c: (gst_alpha_init): * gst/auparse/gstauparse.c: (gst_au_parse_init): * gst/avi/gstavidemux.c: (gst_avi_demux_init), (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream): * gst/cutter/gstcutter.c: (gst_cutter_init): * gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange), (gst_efence_checkgetrange): * gst/debug/negotiation.c: (gst_negotiation_init): * gst/flx/gstflxdec.c: (gst_flxdec_init): * gst/goom/gstgoom.c: (gst_goom_init): * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_init): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init): * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_init): * gst/rtsp/gstrtpdec.c: (gst_rtpdec_init): * gst/smpte/gstsmpte.c: (gst_smpte_init): * gst/wavparse/gstwavparse.c: (gst_wavparse_init), (gst_wavparse_create_sourcepad): Fix memleak with gst_static_pad_template_get(). This uses gst_pad_new_from_static_template() instead. Fixes #333512 2006-03-15 15:08:20 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Let's not forget to chain up to the parent dispose. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_dispose): Let's not forget to chain up to the parent dispose. 2006-03-15 14:39:25 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Series of memleak fixes: Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init), (gst_qtdemux_init), (gst_qtdemux_dispose), (gst_qtdemux_add_stream), (qtdemux_parse_trak): Series of memleak fixes: - Unref the GstAdapter in finalize. - Use gst_pad_new_from_static_template(), shorter and safer. - Free unused QtDemuxStream when not used. 2006-03-15 13:43:42 +0000 Christophe Fergeau ext/lame/gstlame.c: use GST_DEBUG_FUNCPTR more often. Original commit message from CVS: Patch by: Christophe Fergeau * ext/lame/gstlame.c: (gst_lame_release_memory), (gst_lame_finalize), (gst_lame_class_init), (gst_lame_sink_setcaps), (gst_lame_init), (gst_lame_sink_event), (gst_lame_change_state): Fix some memory leaks (#333345), use GST_DEBUG_FUNCPTR more often. 2006-03-14 17:56:02 +0000 Tim-Philipp Müller configure.ac: Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(), used by id3demux. Original commit message from CVS: * configure.ac: Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(), used by id3demux. * gst/id3demux/gstid3demux.c: (plugin_init): * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_user_text_identification_frame), (parse_unique_file_identifier): Add support for UFID and TXXX frames and extract musicbrainz tags. 2006-03-14 17:24:03 +0000 Edward Hervey sys/v4l2/gstv4l2src.c: Initialization of the debugging category should be as early as possible, moving it from _class... Original commit message from CVS: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_base_init), (gst_v4l2src_class_init): Initialization of the debugging category should be as early as possible, moving it from _class_init() to beginning of _base_init(). 2006-03-14 15:28:00 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Catch short reads, like they might happen with truncated files (see #305279); remove unnecessa... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry): Catch short reads, like they might happen with truncated files (see #305279); remove unnecessary indentation. 2006-03-14 14:18:16 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Fix DIB image inversion for pictures with a depth != 8 (#305279). Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_invert): Fix DIB image inversion for pictures with a depth != 8 (#305279). 2006-03-14 09:23:09 +0000 Tim-Philipp Müller ext/jpeg/gstjpegdec.*: Fix durations on outgoing buffers after seeking in MJPEG files (#334083); some minor clean-ups. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_class_init), (gst_jpeg_dec_chain), (gst_jpeg_dec_change_state): * ext/jpeg/gstjpegdec.h: Fix durations on outgoing buffers after seeking in MJPEG files (#334083); some minor clean-ups. 2006-03-13 18:28:18 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Implement seek in READY (re-fixes #327658) Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_reset), (gst_wavparse_change_state): Implement seek in READY (re-fixes #327658) 2006-03-13 17:22:19 +0000 Tim-Philipp Müller ext/taglib/gsttaglib.cc: Add gtk-doc blurb (unused for the time being); match registered plugin name to the filename ... Original commit message from CVS: * ext/taglib/gsttaglib.cc: Add gtk-doc blurb (unused for the time being); match registered plugin name to the filename of the plugin (taglibmux => taglib) 2006-03-13 15:49:08 +0000 Wim Taymans close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. 2006-03-13 10:05:09 +0000 Julien Moutte Fix build of v4l2 (sigh) Original commit message from CVS: 2006-03-13 Julien MOUTTE * docs/plugins/gst-plugins-bad-plugins-decl-list.txt: * sys/v4l2/Makefile.am: Fix build of v4l2 (sigh) 2006-03-12 15:33:00 +0000 Edward Hervey sys/v4l2/v4l2src_calls.c: g_atomic_int_set is only available in glib-0.10, use gst_atomic_int_et instead. Original commit message from CVS: * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_init), (gst_v4l2src_buffer_pool_free): g_atomic_int_set is only available in glib-0.10, use gst_atomic_int_et instead. 2006-03-12 15:25:51 +0000 Edward Hervey sys/v4l2/gstv4l2element.h: Remove tim's addition of "_stdint.h" since it doesn't make the PPC buildbot happy. Original commit message from CVS: * sys/v4l2/gstv4l2element.h: Remove tim's addition of "_stdint.h" since it doesn't make the PPC buildbot happy. I will just use the same comment Ronald used when he added these lines: Yet Another Hack (tm) for kernel header borkedness. 2006-03-12 15:02:02 +0000 Tim-Philipp Müller ext/taglib/: Add support for writing MusicBrainz IDs. Original commit message from CVS: * ext/taglib/Makefile.am: * ext/taglib/gsttaglib.cc: * ext/taglib/gsttaglib.h: Add support for writing MusicBrainz IDs. 2006-03-12 14:43:57 +0000 Tim-Philipp Müller sys/v4l2/gstv4l2element.h: Include "_stdint.h" in an attempt to make the Original commit message from CVS: * sys/v4l2/gstv4l2element.h: Include "_stdint.h" in an attempt to make the PPC-buildbot happy. 2006-03-12 11:00:33 +0000 Christophe Fergeau ext/lame/gstlame.c: mark the xing-header property as BROKEN (see http://bugzilla.gnome.org/show_bug.cgi?id=330317#c19... Original commit message from CVS: 2006-03-12 Christophe Fergeau Reviewed by: Tim-Philipp Müller * ext/lame/gstlame.c: (gst_lame_class_init): mark the xing-header property as BROKEN (see http://bugzilla.gnome.org/show_bug.cgi?id=330317#c19 for an explanation why it's broken). 2006-03-11 22:50:03 +0000 Edgard Lima * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2.c: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2element.c: * sys/v4l2/gstv4l2element.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2xoverlay.c: * sys/v4l2/gstv4l2xoverlay.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: V4L2 ported to 0.10. Original commit message from CVS: V4L2 ported to 0.10. 2006-03-11 10:58:08 +0000 Alex Lancaster ext/taglib/gsttaglib.cc: and add support for TCOP (copyright) Original commit message from CVS: 2006-03-11 Christophe Fergeau Patch by: Alex Lancaster * ext/taglib/gsttaglib.cc: fix writing of TPOS tags (album number), and add support for TCOP (copyright) 2006-03-09 20:02:44 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix build with gcc-4.1 (#327355). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_send_event): Fix build with gcc-4.1 (#327355). 2006-03-09 17:44:17 +0000 Christophe Fergeau new id3v2 muxer based on TagLib Original commit message from CVS: 2006-03-09 Christophe Fergeau reviewed by: Tim-Philipp Müller * configure.ac: * ext/Makefile.am: * ext/taglib/Makefile.am: * ext/taglib/gsttaglib.cc: * ext/taglib/gsttaglib.h: new id3v2 muxer based on TagLib 2006-03-09 11:47:32 +0000 Wim Taymans ext/dv/gstdvdemux.c: Handle events in push mode better, can now do non-flushing seeks in push mode as well. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event), (gst_dvdemux_convert_segment), (gst_dvdemux_demux_frame): Handle events in push mode better, can now do non-flushing seeks in push mode as well. 2006-03-08 12:16:14 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Extract disc number and count from files that use 'disk' instead of 'disc' as node identifier ... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta): Extract disc number and count from files that use 'disk' instead of 'disc' as node identifier for that (fixes #332066). 2006-03-07 17:31:03 +0000 Wim Taymans gst/udp/gstdynudpsink.c: Applied patch from Kai Vehmanen, fixes #333624. Original commit message from CVS: * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init): Applied patch from Kai Vehmanen, fixes #333624. 2006-03-06 22:22:45 +0000 Julien Moutte ext/libpng/gstpngdec.c: Implement paletted and grayscale png files handling. (#150363). Original commit message from CVS: 2006-03-06 Julien MOUTTE * ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set): Implement paletted and grayscale png files handling. (#150363). 2006-03-06 00:10:29 +0000 Thomas Vander Stichele ext/speex/gstspeexenc.c: fix a tag list assert follow gst-plugins-base/ext/ogg/README; set OFFSET and OFFSET_END. Mu... Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speexenc_set_header_on_caps), (gst_speexenc_chain): fix a tag list assert follow gst-plugins-base/ext/ogg/README; set OFFSET and OFFSET_END. Muxes correctly with gst-plugins-base > 0.9.3 2006-03-05 13:03:40 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Add support for '3IVD' fourcc (#333403). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add support for '3IVD' fourcc (#333403). 2006-03-04 20:11:35 +0000 Tim-Philipp Müller gst/id3demux/: Use new typefind helper functions here as well, and do typefinding in pull-mode if upstream supports t... Original commit message from CVS: * gst/id3demux/Makefile.am: * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad), (gst_id3demux_chain), (gst_id3demux_sink_activate): Use new typefind helper functions here as well, and do typefinding in pull-mode if upstream supports that. 2006-03-04 18:57:37 +0000 Benjamin Pineau sys/sunaudio/: Remove unused variables, breaks build from CVS Original commit message from CVS: * sys/sunaudio/gstsunaudiomixerctrl.c: (gst_sunaudiomixer_ctrl_get_volume), (gst_sunaudiomixer_ctrl_set_volume): * sys/sunaudio/gstsunaudiomixertrack.c: (gst_sunaudiomixer_track_new): Remove unused variables, breaks build from CVS with -Werror (#333392, patch by: Benjamin Pineau) 2006-03-03 23:45:23 +0000 Sébastien Moutte sys/: sinks are now using GST_RANK_PRIMARY to be used with autodectection Original commit message from CVS: * sys/directdraw: * sys/directsound: sinks are now using GST_RANK_PRIMARY to be used with autodectection * win32/vs6: project files updated to fix some bugs * win32/vs7: * win32/vs8: vs7 and vs8 project files added 2006-03-03 18:36:53 +0000 Wim Taymans docs/plugins/: Added wavparse docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: Added wavparse docs. * gst/wavparse/gstwavparse.c: (gst_wavparse_class_init), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_send_event), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: Implement seek in READY (fixes #327658) Added docs and did some cleanups. 2006-03-03 17:51:16 +0000 Tim-Philipp Müller gst/avi/gstavidemux.*: If we have an index, use a duration based on the index instead of blindly trusting the informa... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream), (gst_avi_demux_calculate_durations_from_index), (gst_avi_demux_stream_header): * gst/avi/gstavidemux.h: If we have an index, use a duration based on the index instead of blindly trusting the information in the stream headers (fixes #331817). 2006-03-03 15:50:40 +0000 Wim Taymans docs/plugins/: Added smoke and jpeg to the docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: Added smoke and jpeg to the docs. * ext/jpeg/Makefile.am: * ext/jpeg/gstjpeg.c: (plugin_init): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokedec.c: (gst_smokedec_init), (gst_smokedec_chain): * ext/jpeg/gstsmokedec.h: * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain): * ext/jpeg/gstsmokeenc.h: * ext/jpeg/smokecodec.h: Port smokedec (fixes #331905). Added some docs. Some cleanups. 2006-03-03 14:39:55 +0000 Wim Taymans docs/plugins/: Added videobalance and videoflip to the docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: Added videobalance and videoflip to the docs. * gst/videofilter/Makefile.am: * gst/videofilter/gstvideobalance.c: (gst_video_balance_update_tables_planar411), (gst_video_balance_is_passthrough), (gst_video_balance_update_properties), (oil_tablelookup_u8), (gst_video_balance_planar411_ip), (gst_video_balance_set_caps), (gst_video_balance_transform_ip), (gst_video_balance_base_init), (gst_video_balance_finalize), (gst_video_balance_class_init), (gst_video_balance_init), (gst_video_balance_interface_supported), (gst_video_balance_interface_init), (gst_video_balance_colorbalance_list_channels), (gst_video_balance_colorbalance_set_value), (gst_video_balance_colorbalance_get_value), (gst_video_balance_colorbalance_init), (gst_video_balance_set_property), (gst_video_balance_get_property), (gst_video_balance_get_type), (plugin_init): * gst/videofilter/gstvideobalance.h: Ported to 0.10. (Fixes #326160) Added docs. * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: Added docs. 2006-03-03 11:07:41 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Use GST_WARNING instead of GST_ERROR for all the too short/long atoms when parsing. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak): Use GST_WARNING instead of GST_ERROR for all the too short/long atoms when parsing. Also let's be a bit less vulgar in our warning messages :) 2006-03-02 15:14:22 +0000 Tim-Philipp Müller configure.ac: Bump requirements to current core and -base CVS (core for new typefind helper API, and -base for the Original commit message from CVS: * configure.ac: Bump requirements to current core and -base CVS (core for new typefind helper API, and -base for the WAVFORMATEX support that was added to libgstriff and is needed by wavparse). * gst/apetag/Makefile.am: * gst/apetag/gsttagdemux.c: (gst_tag_demux_chain), (gst_tag_demux_sink_activate): Use new typefind helpers for typefinding instead of our home-grown stuff; also, do typefinding in pull-mode if upstream supports that. 2006-02-28 11:59:49 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Can't divide through zero (suppress warning in case of stream with one single still picture) (... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Can't divide through zero (suppress warning in case of stream with one single still picture) (see #327083) 2006-02-28 10:40:01 +0000 Christian Schaller * ChangeLog: remove conflict indicator Original commit message from CVS: remove conflict indicator 2006-02-28 10:39:08 +0000 Christian Schaller * ChangeLog: add missing entry Original commit message from CVS: add missing entry 2006-02-28 10:29:16 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Use DEBUG_OBJECT more. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull): Use DEBUG_OBJECT more. 2006-02-28 10:22:11 +0000 Wim Taymans docs/plugins/: Added dvdec and dvdemux to docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: Added dvdec and dvdemux to docs. * ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_chain): Added docs. Check frame sizes so we don't crash when don't have enough data. Send nice error messages on error. * ext/dv/gstdvdemux.c: (gst_dvdemux_base_init), (gst_dvdemux_class_init), (gst_dvdemux_init), (gst_dvdemux_finalize), (gst_dvdemux_reset), (gst_dvdemux_src_convert), (gst_dvdemux_sink_convert), (gst_dvdemux_src_query), (gst_dvdemux_sink_query), (gst_dvdemux_push_event), (gst_dvdemux_handle_sink_event), (gst_dvdemux_convert_src_pair), (gst_dvdemux_convert_sink_pair), (gst_dvdemux_convert_src_to_sink), (gst_dvdemux_handle_push_seek), (gst_dvdemux_do_seek), (gst_dvdemux_handle_pull_seek), (gst_dvdemux_handle_src_event), (gst_dvdemux_demux_audio), (gst_dvdemux_demux_video), (gst_dvdemux_demux_frame), (gst_dvdemux_flush), (gst_dvdemux_chain), (gst_dvdemux_loop), (gst_dvdemux_sink_activate_push), (gst_dvdemux_sink_activate_pull), (gst_dvdemux_sink_activate), (gst_dvdemux_change_state): * ext/dv/gstdvdemux.h: Added docs. Implement pull mode. Fix memleaks. Reduce memcpy for the video demuxing. 2006-02-28 09:21:27 +0000 Jan Schmidt ext/annodex/: Add a little extra debug. Make the decoder not return NOT_LINKED, as we want to continue decoding all C... Original commit message from CVS: * ext/annodex/gstcmmldec.c: (gst_cmml_dec_sink_event), (gst_cmml_dec_new_buffer), (gst_cmml_dec_parse_preamble), (gst_cmml_dec_parse_head), (gst_cmml_dec_push_clip): * ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk): Add a little extra debug. Make the decoder not return NOT_LINKED, as we want to continue decoding all CMML and emitting tags. 2006-02-27 14:37:29 +0000 Christian Schaller * gst-plugins-good.spec.in: add annodex plugin Original commit message from CVS: add annodex plugin 2006-02-27 14:00:18 +0000 Michael Smith ext/annodex/gstskeltag.*: Deleted; these files aren't used any more either. Original commit message from CVS: * ext/annodex/gstskeltag.c: * ext/annodex/gstskeltag.h: Deleted; these files aren't used any more either. 2006-02-25 20:37:29 +0000 Julien Moutte ext/Makefile.am: Fix dist-check. Original commit message from CVS: 2006-02-25 Julien MOUTTE * ext/Makefile.am: Fix dist-check. 2006-02-25 19:36:24 +0000 Julien Moutte ext/annodex/gstcmmlenc.c: Fix another memleak. Original commit message from CVS: 2006-02-25 Julien MOUTTE * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_push_clip): Fix another memleak. 2006-02-25 19:07:41 +0000 Julien Moutte Fix a memleak in gst_cmml_track_list_add_clip. Original commit message from CVS: 2006-02-25 Alessandro Decina * ext/annodex/Makefile.am: * ext/annodex/gstannodex.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/annodex/gstcmmlparser.c: * ext/annodex/gstcmmlparser.h: * ext/annodex/gstcmmlutils.c: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: Fix a memleak in gst_cmml_track_list_add_clip. Handle overflows in clip's start and end times. Add the "encoded" parameter to cmmldec and cmmlenc caps. Do not parse junk at the end of a CMML preamble buffer. Register a libxml error handler to not print stuff on stderr. Check for bad clip start and end times in the testsuites. 2006-02-25 11:37:10 +0000 Julien Moutte ext/annodex/: Fix possible memleaks. Original commit message from CVS: 2006-02-25 Julien MOUTTE * ext/annodex/gstcmmldec.c: (gst_cmml_dec_class_init), (gst_cmml_dec_finalize), (gst_cmml_dec_change_state): * ext/annodex/gstcmmlenc.c: (gst_cmml_enc_class_init), (gst_cmml_enc_finalize), (gst_cmml_enc_change_state): * ext/annodex/gstcmmlutils.c: (gst_cmml_track_list_destroy): Fix possible memleaks. 2006-02-24 23:52:28 +0000 Julien Moutte tests/check/: Fix tests so that they use the plugins-base tags. Original commit message from CVS: 2006-02-25 Julien MOUTTE * tests/check/Makefile.am: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: Fix tests so that they use the plugins-base tags. 2006-02-24 23:36:58 +0000 Julien Moutte ext/Makefile.am: Re-enable module. Original commit message from CVS: 2006-02-25 Julien MOUTTE * ext/Makefile.am: Re-enable module. 2006-02-24 23:32:14 +0000 Julien Moutte tests/check/Makefile.am: Forgot to remove that test. Original commit message from CVS: 2006-02-25 Julien MOUTTE * tests/check/Makefile.am: Forgot to remove that test. 2006-02-24 23:31:08 +0000 Julien Moutte Try to fix Annodex plugin. Original commit message from CVS: 2006-02-25 Julien MOUTTE * ext/annodex/Makefile.am: * ext/annodex/gstannodex.c: (plugin_init): * ext/annodex/gstcmmldec.c: * ext/annodex/gstskeldec.c: * ext/annodex/gstskeldec.h: * tests/check/Makefile.am: * tests/check/elements/skeldec.c: Try to fix Annodex plugin. 2006-02-24 23:06:27 +0000 Julien Moutte tests/check/Makefile.am: Disable those checks as well. Original commit message from CVS: 2006-02-25 Julien MOUTTE * tests/check/Makefile.am: Disable those checks as well. 2006-02-24 22:49:29 +0000 Julien Moutte ext/Makefile.am: Disable annodex for now until we figure out how to make it build. Original commit message from CVS: 2006-02-24 Julien MOUTTE * ext/Makefile.am: Disable annodex for now until we figure out how to make it build. * ext/gdk_pixbuf/Makefile.am: Note for Thomas : Add a rule to your checklist : "please try to at least build what you are going to commit into -good, or if you are too lazy to do that, please check that the buildbots are not crying because of your commit." 2006-02-24 19:51:29 +0000 Edgard Lima * ChangeLog: * configure.ac: * ext/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbuf.h: * ext/gdk_pixbuf/pixbufscale.c: * ext/gdk_pixbuf/pixbufscale.h: I'm too lazy to comment this Original commit message from CVS: Gdkpixbuf ported from 0.8 to 0.10 by Renato Filho . gst_loader and gdkpixbufanimation still need port. 2006-02-24 19:49:32 +0000 Fabrizio Gennari gst/qtdemux/qtdemux.c: Add support for palettised Apple SMC videos (#327075, based on Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream), (qtdemux_parse_trak), (qtdemux_video_caps): Add support for palettised Apple SMC videos (#327075, based on patch by: Fabrizio Gennari ). 2006-02-24 19:07:10 +0000 Michael Smith Add Annodex elements from Alessendro Decina: skeleton and CMML. Original commit message from CVS: * configure.ac: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/Makefile.am: * ext/annodex/Makefile.am: * ext/annodex/gstannodex.c: * ext/annodex/gstannodex.h: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmldec.h: * ext/annodex/gstcmmlenc.c: * ext/annodex/gstcmmlenc.h: * ext/annodex/gstcmmlparser.c: * ext/annodex/gstcmmlparser.h: * ext/annodex/gstcmmltag.c: * ext/annodex/gstcmmltag.h: * ext/annodex/gstcmmlutils.c: * ext/annodex/gstcmmlutils.h: * ext/annodex/gstskeldec.c: * ext/annodex/gstskeldec.h: * ext/annodex/gstskeltag.c: * ext/annodex/gstskeltag.h: * tests/check/Makefile.am: * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: * tests/check/elements/skeldec.c: Add Annodex elements from Alessendro Decina: skeleton and CMML. Includes tests & docs, oh my! Passes Thomas's -good checklist entirely. Wow. 2006-02-24 17:09:56 +0000 Michael Smith autogen.sh: Check for automake 1.9 as well. Original commit message from CVS: * autogen.sh: Check for automake 1.9 as well. 2006-02-24 14:49:48 +0000 Tim-Philipp Müller ext/flac/gstflacenc.c: Change min. sample rate to 8kHz to match flacdec's. Original commit message from CVS: * ext/flac/gstflacenc.c: Change min. sample rate to 8kHz to match flacdec's. 2006-02-23 20:08:58 +0000 Tim-Philipp Müller ext/cdio/Makefile.am: Add GST_BASE_CFLAGS and GST_BASE_LIBS (seems to be required for Cygwin, see #317048) Original commit message from CVS: * ext/cdio/Makefile.am: Add GST_BASE_CFLAGS and GST_BASE_LIBS (seems to be required for Cygwin, see #317048) * gst/rtp/gstasteriskh263.c: Cygwin has includes for both the unix network socket API and the windows API, but only one can be included, so fix includes to only use one or the other, prefering the unxi one (#317048). 2006-02-23 12:21:25 +0000 Philippe Kalaf rtp/gst/: Separated the G711 payloaders/depayloaders into separate elements for mulaw/alaw. Also removed the old g711... Original commit message from CVS: 2006-02-23 Philippe Kalaf * rtp/gst/gstrtppcmadepay.c: * rtp/gst/gstrtppcmadepay.h: * rtp/gst/gstgstrtppcmapay.c: * rtp/gst/gstgstrtppcmapay.h: * rtp/gst/gstrtppcmudepay.c: * rtp/gst/gstrtppcmudepay.h: * rtp/gst/gstrtppcmupay.c: * rtp/gst/gstrtppcmupay.h: * rtp/gst/Makefile.am: * rtp/gst/gstrtp.c: * rtp/gst/README: Separated the G711 payloaders/depayloaders into separate elements for mulaw/alaw. Also removed the old g711 payloaders/depayloaders. 2006-02-22 20:22:25 +0000 Wim Taymans ext/dv/: Ueber spiffify some more, added debug category. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_init), (gst_dvdec_change_state): * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.c: (gst_dvdemux_base_init), (gst_dvdemux_init), (gst_dvdemux_src_convert), (gst_dvdemux_sink_convert), (gst_dvdemux_src_query), (gst_dvdemux_sink_query), (gst_dvdemux_handle_sink_event), (gst_dvdemux_demux_frame), (gst_dvdemux_flush), (gst_dvdemux_chain), (gst_dvdemux_change_state): * ext/dv/gstdvdemux.h: Ueber spiffify some more, added debug category. Use _scale. Use segments, respect playback rate from newsegment. Fix refcount issue. 2006-02-22 09:33:25 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Add 'dvsd' and 'dv25' to list of possible fourcc values for DV Video. Original commit message from CVS: Reviewed by : Edward Hervey * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add 'dvsd' and 'dv25' to list of possible fourcc values for DV Video. Add image/png for fourcc 'png ' 2006-02-20 21:19:59 +0000 Jan Schmidt Port ximagesrc to 0.10 (Closes #304795) Original commit message from CVS: * configure.ac: * sys/Makefile.am: * sys/ximagesrc/Makefile.am: * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_return_buf), (gst_ximagesrc_open_display), (gst_ximagesrc_start), (gst_ximagesrc_stop), (gst_ximagesrc_unlock), (gst_ximagesrc_recalc), (composite_pixel), (gst_ximagesrc_ximage_get), (gst_ximagesrc_create), (gst_ximagesrc_set_property), (gst_ximagesrc_get_property), (gst_ximagesrc_clear_bufpool), (gst_ximagesrc_base_init), (gst_ximagesrc_dispose), (gst_ximagesrc_finalize), (gst_ximagesrc_get_caps), (gst_ximagesrc_set_caps), (gst_ximagesrc_fixate), (gst_ximagesrc_class_init), (gst_ximagesrc_init), (plugin_init): * sys/ximagesrc/ximagesrc.h: * sys/ximagesrc/ximageutil.c: (ximageutil_handle_xerror), (ximageutil_check_xshm_calls), (ximageutil_xcontext_get), (ximageutil_xcontext_clear), (ximageutil_calculate_pixel_aspect_ratio), (gst_ximagesrc_buffer_finalize), (gst_ximage_buffer_free), (gst_ximagesrc_buffer_init), (gst_ximagesrc_buffer_class_init), (gst_ximagesrc_buffer_get_type), (gst_ximageutil_ximage_new), (gst_ximageutil_ximage_destroy): * sys/ximagesrc/ximageutil.h: Port ximagesrc to 0.10 (Closes #304795) === release 0.10.1 === 2006-02-20 19:12:10 +0000 Jan Schmidt configure.ac: releasing 0.10.1, "Slimy - yet satisfying" Original commit message from CVS: 2006-02-20 Jan Schmidt * configure.ac: releasing 0.10.1, "Slimy - yet satisfying" 2006-02-20 13:08:50 +0000 Jan Schmidt ext/ladspa/gstsignalprocessor.c: Fix compilation of LADPSA. It doesn't seem to work, and isn't enabled for the build,... Original commit message from CVS: * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event), (gst_signal_processor_process): Fix compilation of LADPSA. It doesn't seem to work, and isn't enabled for the build, but it helps me win the feature-count competitions ooh yeah. 2006-02-19 16:02:25 +0000 Wim Taymans gst/avi/gstavidemux.c: Use scaling code for added precission and more correct stop position in case scale==0. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_file_header), (gst_avi_demux_stream_init), (gst_avi_demux_parse_avih), (gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream), (gst_avi_demux_stream_header), (gst_avi_demux_change_state): Use scaling code for added precission and more correct stop position in case scale==0. 2006-02-19 12:09:19 +0000 Wim Taymans gst/flx/gstflxdec.*: Implement DURATION query. Original commit message from CVS: * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/flx/gstflxdec.h: Implement DURATION query. 2006-02-19 11:57:58 +0000 Wim Taymans gst/flx/: Set MALLOCDATA for the temp buffers so we don't leak. Original commit message from CVS: * gst/flx/flx_color.h: * gst/flx/flx_fmt.h: * gst/flx/gstflxdec.c: (gst_flxdec_init), (gst_flxdec_src_query_handler), (flx_decode_color), (gst_flxdec_chain): * gst/flx/gstflxdec.h: Set MALLOCDATA for the temp buffers so we don't leak. Some debug cleanups. Consume all data in the adapter before leaving the chain function. Fixes #330678. 2006-02-18 20:48:09 +0000 Jan Schmidt gst/id3demux/: Handle 0 data size in otherwise valid frames. Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list): * gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist): Handle 0 data size in otherwise valid frames. Handle numeric strings in 2.4.0 even when not in parentheses 2006-02-18 17:20:48 +0000 Tim-Philipp Müller gst/matroska/: Recognise SSA/ASS and USF subtitle formats and set proper caps when they are found. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_subtitle_caps), (gst_matroska_demux_plugin_init): * gst/matroska/matroska-ids.h: Recognise SSA/ASS and USF subtitle formats and set proper caps when they are found. 2006-02-17 18:25:42 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Don't GST_LOG timestamps from nonexistent index entries (#331582). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie): Don't GST_LOG timestamps from nonexistent index entries (#331582). 2006-02-17 17:54:05 +0000 Tim-Philipp Müller ext/jpeg/gstjpegdec.c: Fix invalid memory access for some odd-sized images (see image contained in quicktime stream i... Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_decode_direct), (gst_jpeg_dec_chain): Fix invalid memory access for some odd-sized images (see image contained in quicktime stream in #327083); use g_malloc() instead of g_alloca(). 2006-02-17 16:28:29 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Check that the size of the returned buffer is of the correct size because the parser assumes t... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_header): Check that the size of the returned buffer is of the correct size because the parser assumes that. Fixes #331543. 2006-02-17 15:37:38 +0000 Wim Taymans gst/rtp/gstrtpamrdepay.c: Patch from Sebastien Cote, fixes #319884 Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain): Patch from Sebastien Cote, fixes #319884 2006-02-17 11:19:34 +0000 Tim-Philipp Müller ext/cdio/gstcdio.c: Init debug category (#331253). Original commit message from CVS: * ext/cdio/gstcdio.c: (plugin_init): Init debug category (#331253). 2006-02-17 10:53:38 +0000 Christian Schaller * ext/gconf/gconf.c: * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfaudiosink.h: * gconf/gstreamer.schemas.in: * gst-plugins-good.spec.in: add Jurg's patch for multidevice support Original commit message from CVS: add Jurg's patch for multidevice support 2006-02-16 20:30:13 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: Pass extra_data to gst_riff_create_audio_caps(), so that Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Pass extra_data to gst_riff_create_audio_caps(), so that WAVEFORMATEX stuff works. Post audio codec name and post it as taglist on the bus. Allow up to 8 channesl for raw PCM in the source pad template caps. 2006-02-16 17:16:06 +0000 Wim Taymans ext/lame/gstlame.c: Fix up lame a bit. Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain), (gst_lame_change_state): Fix up lame a bit. Apply patch #319782 by Gautier Portet. 2006-02-16 16:53:52 +0000 Wim Taymans gst/multipart/multipartdemux.c: Applied #318663. Gives quite a few false positives in autoscan mode, but it's better ... Original commit message from CVS: * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init), (gst_multipart_demux_class_init), (gst_multipart_demux_init), (gst_multipart_demux_finalize), (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain), (gst_multipart_demux_change_state), (gst_multipart_set_property), (gst_multipart_get_property): Applied #318663. Gives quite a few false positives in autoscan mode, but it's better than nothing. Not closing yet. 2006-02-16 14:13:48 +0000 Wim Taymans Update documentation. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-udp.xml: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init), (gst_udpsrc_start): Update documentation. Fix args. 2006-02-16 14:02:57 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Don't stop the task if the pad isn't linked. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event), (gst_qtdemux_loop), (qtdemux_sink_activate_pull): Don't stop the task if the pad isn't linked. 2006-02-16 10:58:18 +0000 Jan Schmidt gst/id3demux/id3tags.c: ID3 2.3.0 used synch-safe integers for the tag size, but not for the frame size. (Fixes #331368) Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list): ID3 2.3.0 used synch-safe integers for the tag size, but not for the frame size. (Fixes #331368) 2006-02-16 10:42:25 +0000 Wim Taymans gst/rtsp/README: Updated README. Original commit message from CVS: * gst/rtsp/README: Updated README. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp): * gst/rtsp/gstrtspsrc.h: Make sure the RTP port is an even port an try to allocate another if not. Added retry property to control max retries for port allocation. Make sure RTCP port is RTP port+1. Cleanup when port allocation fails. Fixes #319183. 2006-02-16 09:17:58 +0000 Wouter Paesen gst/alpha/gstalpha.c: Don't ignore return value of the parent class's state Original commit message from CVS: * gst/alpha/gstalpha.c: (gst_alpha_change_state): Don't ignore return value of the parent class's state change function (#331385, patch by: Wouter Paesen). 2006-02-15 12:17:28 +0000 Wim Taymans Add HAL sound device wrapper plugins. Closes #329106 Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * ext/Makefile.am: * ext/hal/Makefile.am: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init), (gst_hal_audio_sink_class_init), (gst_hal_audio_sink_reset), (gst_hal_audio_sink_init), (gst_hal_audio_sink_dispose), (do_toggle_element), (gst_hal_audio_sink_set_property), (gst_hal_audio_sink_get_property), (gst_hal_audio_sink_change_state): * ext/hal/gsthalaudiosink.h: * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init), (gst_hal_audio_src_class_init), (gst_hal_audio_src_reset), (gst_hal_audio_src_init), (gst_hal_audio_src_dispose), (do_toggle_element), (gst_hal_audio_src_set_property), (gst_hal_audio_src_get_property), (gst_hal_audio_src_change_state): * ext/hal/gsthalaudiosrc.h: * ext/hal/gsthalelements.c: (plugin_init): * ext/hal/gsthalelements.h: * ext/hal/hal.c: (gst_hal_get_string), (gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink), (gst_hal_get_audio_src): * ext/hal/hal.h: Add HAL sound device wrapper plugins. Closes #329106 2006-02-15 12:13:47 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: It appears 100% equals 1/1 and not 100/1 ... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_chain): It appears 100% equals 1/1 and not 100/1 ... 2006-02-15 10:15:47 +0000 Wim Taymans gst/avi/gstavidemux.c: Add comment in a fultile attempt to stop the copy-and-paste paradigm leading to duplication of... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event): Add comment in a fultile attempt to stop the copy-and-paste paradigm leading to duplication of bad code. * gst/rtsp/rtsptransport.c: (rtsp_transport_parse): Mime parameters have to be checked case insensitive 2006-02-15 09:45:27 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: When buffering MDAT data, show the user something is happening by posting 'buffering' messages... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_post_buffering), (gst_qtdemux_chain): When buffering MDAT data, show the user something is happening by posting 'buffering' messages on the bus. 2006-02-14 23:23:08 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Advance stream time for lagging subtitle streams by sending newsegment events with the... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams): Advance stream time for lagging subtitle streams by sending newsegment events with the update flag set. 2006-02-14 18:50:13 +0000 Edward Hervey gst/qtdemux/qtdemux.*: Make push-based work if mdat atom is before moov atom. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_init), (gst_qtdemux_handle_src_query), (gst_qtdemux_change_state), (next_entry_size), (gst_qtdemux_chain): * gst/qtdemux/qtdemux.h: Make push-based work if mdat atom is before moov atom. Don't answer duration query. This should be transformed into replying FALSE to seek events. 2006-02-14 16:58:30 +0000 Edward Hervey gst/avi/gstavidemux.c: There can be bogus data before the hdrl LIST tag in the RIFF header. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header): There can be bogus data before the hdrl LIST tag in the RIFF header. It's hard to say if it's not respecting the AVI specifications or not, but since Google Video is producing AVIs like that and the other player don't seem to complain, I guess we should do the same. 2006-02-14 11:24:53 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Handle the case where data atoms are before moov atoms in push-based mode. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (next_entry_size), (gst_qtdemux_chain): Handle the case where data atoms are before moov atoms in push-based mode. Errors out gracefully. 2006-02-13 22:04:42 +0000 Edward Hervey gst/qtdemux/: QtDemux can now work push-based. Original commit message from CVS: * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: (gst_qtdemux_init), (gst_qtdemux_handle_sink_event), (gst_qtdemux_change_state), (extract_initial_length_and_fourcc), (gst_qtdemux_loop_state_header), (gst_qtdemux_loop_state_movie), (gst_qtdemux_loop_header), (next_entry_size), (gst_qtdemux_chain), (qtdemux_sink_activate), (qtdemux_sink_activate_pull), (qtdemux_sink_activate_push), (qtdemux_parse_trak): * gst/qtdemux/qtdemux.h: QtDemux can now work push-based. It still needs some love for seeking. 2006-02-13 12:00:51 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Add more validation to ensure that a char encoding conversion produced a valid UTF-8 string. Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_insert_string_field), (parse_split_strings): Add more validation to ensure that a char encoding conversion produced a valid UTF-8 string. 2006-02-13 10:43:15 +0000 Edward Hervey gst/avi/gstavidemux.c: Properly handle end of segment. Closes #330885. Original commit message from CVS: Reviewed by: Edward Hervey * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry): Properly handle end of segment. Closes #330885. 2006-02-13 10:36:23 +0000 Wim Taymans gst/rtp/gstrtpmp4gpay.h: For got to commit this one. Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.h: For got to commit this one. 2006-02-12 18:59:36 +0000 Wim Taymans gst/rtp/gstrtpmp4gpay.*: Make more things work. Original commit message from CVS: * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init), (gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush): * gst/rtp/gstrtpmp4gpay.h: Make more things work. Handle ACC config strings. 2006-02-12 13:10:20 +0000 Thomas Vander Stichele gst/rtp/gstrtpamrpay.c: set timestamps if no incoming timestamps set Original commit message from CVS: * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer): set timestamps if no incoming timestamps set 2006-02-11 13:54:26 +0000 Tim-Philipp Müller gst/apetag/gsttagdemux.c: ... and fix the very same leaks in GstTagDemux. Original commit message from CVS: * gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size), (gst_tag_demux_do_typefind): ... and fix the very same leaks in GstTagDemux. 2006-02-11 13:35:13 +0000 Jon Trowbridge gst/id3demux/gstid3demux.c: Original commit message from CVS: * gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size), (gst_id3demux_do_typefind): Fix a couple of mem leaks. (Patch by Jonathan Matthew ) 2006-02-10 17:37:39 +0000 Wim Taymans gst/rtp/gstrtpmp4vpay.c: First set options, then set caps or else the baseclass will not know about the options, duh. Original commit message from CVS: * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps): First set options, then set caps or else the baseclass will not know about the options, duh. 2006-02-10 17:16:55 +0000 Wim Taymans gst/rtp/gstrtpmp4vpay.c: Don't waste time looking for a config string if we have codec_info on the incomming caps. Original commit message from CVS: * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init), (gst_rtp_mp4v_pay_setcaps): Don't waste time looking for a config string if we have codec_info on the incomming caps. 2006-02-10 16:40:58 +0000 Wim Taymans gst/rtp/README: Say something about case-sensitivity of caps vs mime-attributes. Original commit message from CVS: * gst/rtp/README: Say something about case-sensitivity of caps vs mime-attributes. * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_handle_buffer): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_get_type), (gst_rtp_mp4g_pay_base_init), (gst_rtp_mp4g_pay_class_init), (gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush), (gst_rtp_mp4g_pay_handle_buffer), (gst_rtp_mp4g_pay_set_property), (gst_rtp_mp4g_pay_get_property), (gst_rtp_mp4g_pay_plugin_init): * gst/rtp/gstrtpmp4gpay.h: Added beginnings of mpeg4-generic payloader (RFC 3640) 2006-02-09 14:20:14 +0000 Wim Taymans gst/rtsp/: Resurected rtpdec to make rtspsrc happy again. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type), (gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps), (gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp), (gst_rtpdec_set_property), (gst_rtpdec_get_property), (gst_rtpdec_change_state): * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport): * gst/rtsp/rtspconnection.c: (read_body), (rtsp_connection_receive): * gst/rtsp/rtspmessage.c: (rtsp_message_dump): Resurected rtpdec to make rtspsrc happy again. Skip attributes from the session id. Don't crash when dumping a message with an empty body. 2006-02-09 14:14:07 +0000 Wim Taymans gst/rtp/gstrtpamrdepay.c: Added more meaningfull warnings when something goes wrong. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain): Added more meaningfull warnings when something goes wrong. Clear F bit on outgoing AMR packets. * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_handle_buffer): Added debugging category Support payloading of multiple AMR frames. * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_depay_data): Added some debugging. 2006-02-09 11:25:42 +0000 Jan Schmidt configure.ac: Back to CVS Original commit message from CVS: * configure.ac: Back to CVS === release 0.10.2 === 2006-02-09 11:22:38 +0000 Jan Schmidt * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cdio.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: Releasing 0.10.2 Original commit message from CVS: Releasing 0.10.2 2006-02-08 17:35:05 +0000 Jan Schmidt * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2006-02-08 17:18:20 +0000 Jan Schmidt * ChangeLog: Oops, jumping the gun with the ChangeLog entry Original commit message from CVS: Oops, jumping the gun with the ChangeLog entry 2006-02-08 17:16:46 +0000 Jan Schmidt configure.ac: Bump core and plugins-base requirement to 0.10.2.2 for API additions (and 1 migration of gst_bin_find_u... Original commit message from CVS: * configure.ac: Bump core and plugins-base requirement to 0.10.2.2 for API additions (and 1 migration of gst_bin_find_unconnected_pad) 2006-02-08 17:12:40 +0000 Tim-Philipp Müller ext/: Register musicbrainz tags. Original commit message from CVS: * ext/flac/gstflac.c: (plugin_init): * ext/speex/gstspeex.c: (plugin_init): Register musicbrainz tags. 2006-02-07 18:31:31 +0000 Thomas Vander Stichele * gst/qtdemux/qtdemux.c: remove unused var Original commit message from CVS: remove unused var 2006-02-07 18:01:17 +0000 Thomas Vander Stichele gst/qtdemux/qtdemux.c: use the correct variable to check if we can calculate the last chunk. Looks like an obvious b... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (qtdemux_parse_trak): use the correct variable to check if we can calculate the last chunk. Looks like an obvious bug, and makes the dump of offsets comparable to other tools 2006-02-07 17:54:42 +0000 Thomas Vander Stichele gst/qtdemux/qtdemux.c: clean up some debugging, using _OBJECT, moving recurring messages to LOG level Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (qtdemux_parse_trak): clean up some debugging, using _OBJECT, moving recurring messages to LOG level 2006-02-07 16:23:33 +0000 Tim-Philipp Müller ext/gconf/gconf.h: Remove declaration of function that no longer exists. Original commit message from CVS: * ext/gconf/gconf.h: Remove declaration of function that no longer exists. 2006-02-07 13:39:08 +0000 Zaheer Abbas Merali ext/shout2/gstshout2.c: Make shout2 work for non ogg streams Original commit message from CVS: 2006-02-07 Zaheer Abbas Merali * ext/shout2/gstshout2.c: (gst_shout2send_render), (gst_shout2send_setcaps), (gst_shout2send_change_state): Make shout2 work for non ogg streams 2006-02-06 17:26:43 +0000 Wim Taymans gst/udp/gstmultiudpsink.*: Updated docs. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_render), (gst_multiudpsink_get_property), (gst_multiudpsink_init_send), (gst_multiudpsink_add), (gst_multiudpsink_remove), (gst_multiudpsink_clear), (gst_multiudpsink_get_stats), (gst_multiudpsink_change_state): * gst/udp/gstmultiudpsink.h: Updated docs. Added properties bytes-served, bytes_to_serve. Post proper error messages, Emit client added signal too. 2006-02-06 15:41:25 +0000 Wim Taymans gst/qtdemux/qtdemux.*: Some QT demux loving. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event), (gst_qtdemux_loop_header), (qtdemux_inflate), (qtdemux_parse), (qtdemux_parse_trak), (qtdemux_parse_udta), (qtdemux_tag_add_str), (qtdemux_tag_add_num), (qtdemux_tag_add_gnre), (gst_qtdemux_handle_esds), (qtdemux_video_caps), (qtdemux_audio_caps): * gst/qtdemux/qtdemux.h: Some QT demux loving. Handle seeking in a less broken way. Fix AMR caps to match the AMR decoder. Set first timestamp on AMR samples to 0 for now. Remove some \n in DEBUG strings. Use _scale_int for maximum precision. 2006-02-06 15:31:16 +0000 Thomas Vander Stichele * ChangeLog: * common: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/udp/gstmultiudpsink.c: adding docs for multiudpsink Original commit message from CVS: adding docs for multiudpsink 2006-02-06 15:28:56 +0000 Thomas Vander Stichele gst/level/gstlevel.c: peak below decay is not necessarily an error, so don't ERROR log Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_transform_ip): peak below decay is not necessarily an error, so don't ERROR log 2006-02-06 15:27:06 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: cvs versions Original commit message from CVS: cvs versions 2006-02-06 14:25:34 +0000 Tim-Philipp Müller gst/matroska/ebml-write.*: Make sure we send a newsegment event in BYTES format before sending buffers (#328531). Original commit message from CVS: * gst/matroska/ebml-write.c: (gst_ebml_write_reset), (gst_ebml_write_flush_cache), (gst_ebml_write_element_push), (gst_ebml_write_seek): * gst/matroska/ebml-write.h: Make sure we send a newsegment event in BYTES format before sending buffers (#328531). 2006-02-06 12:18:45 +0000 Tim-Philipp Müller Pass unhandled queries upstream instead of just dropping them (#326446). Update query type arrays here and there. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_src_query), (gst_dvdemux_sink_query): * ext/flac/gstflacdec.c: (gst_flac_dec_src_query): * ext/speex/gstspeexdec.c: (speex_get_query_types), (speex_dec_src_query): * ext/speex/gstspeexenc.c: (gst_speexenc_src_query), (gst_speexenc_sink_query): * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query): * gst/matroska/matroska-demux.c: (gst_matroska_demux_get_src_query_types), (gst_matroska_demux_handle_src_query): * gst/wavparse/gstwavparse.c: (gst_wavparse_get_query_types), (gst_wavparse_pad_query): Pass unhandled queries upstream instead of just dropping them (#326446). Update query type arrays here and there. 2006-02-06 11:57:52 +0000 Tim-Philipp Müller tests/check/elements/matroskamux.c: Collectpads in core got changed and now also holds a reference to any pad that is... Original commit message from CVS: * tests/check/elements/matroskamux.c: (setup_src_pad): Collectpads in core got changed and now also holds a reference to any pad that is part of it. Fix refcount checks in test case accordingly. 2006-02-06 11:41:43 +0000 Tim-Philipp Müller gst/apetag/gstapedemux.h: Fix include, for now GstTagDemux is in the apetag dir. Original commit message from CVS: * gst/apetag/gstapedemux.h: Fix include, for now GstTagDemux is in the apetag dir. 2006-02-06 11:34:23 +0000 Tim-Philipp Müller docs/plugins/: Add cdio plugin to docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-cdio.xml: Add cdio plugin to docs. * ext/cdio/gstcdiocddasrc.c: Add gtk-doc blurb. * ext/cdio/gstcdio.c: The plugin is called 'cdio' not 'cddio'. 2006-02-06 10:56:07 +0000 Tim-Philipp Müller Add APE tag demuxer (#325649). Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-apetag.xml: * gst/apetag/Makefile.am: * gst/apetag/gstapedemux.c: * gst/apetag/gstapedemux.h: * gst/apetag/gsttagdemux.c: * gst/apetag/gsttagdemux.h: Add APE tag demuxer (#325649). 2006-02-05 22:22:56 +0000 Jan Schmidt ext/gconf/: Ignore changing the GConf key to "". Ignore GConf key updates that don't actually change the string. Original commit message from CVS: * ext/gconf/gconf.c: (gst_gconf_get_default_audio_sink), (gst_gconf_get_default_video_sink), (gst_gconf_get_default_audio_src), (gst_gconf_get_default_video_src): * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init), (gst_gconf_audio_sink_dispose), (do_toggle_element): * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset), (gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose), (do_toggle_element): * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset), (gst_gconf_video_sink_init), (gst_gconf_video_sink_dispose), (do_toggle_element): * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset), (gst_gconf_video_src_init), (gst_gconf_video_src_dispose), (do_toggle_element): * ext/gconf/gstgconfvideosrc.h: Ignore changing the GConf key to "". Ignore GConf key updates that don't actually change the string. For now, ignore the GConf key when the state is > READY, as it breaks streaming. Sometime it will be nice to bring the new sink online even mid-stream, by sending NEWSEGMENT info and possibly prerolling. (Fixes #326736) 2006-02-05 20:43:49 +0000 Jan Schmidt gst/goom/: Make goom reentrant by moving all important static variables into instance structures. Original commit message from CVS: * gst/goom/filters.c: (zoomFilterNew), (calculatePXandPY), (setPixelRGB), (setPixelRGB_), (getPixelRGB), (getPixelRGB_), (zoomFilterSetResolution), (zoomFilterDestroy), (zoomFilterFastRGB), (pointFilter): * gst/goom/filters.h: * gst/goom/goom_core.c: (goom_init), (goom_set_resolution), (goom_update), (goom_close): * gst/goom/goom_core.h: * gst/goom/goom_tools.h: * gst/goom/graphic.c: * gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init), (gst_goom_dispose), (gst_goom_src_setcaps), (gst_goom_chain): * gst/goom/gstgoom.h: * gst/goom/lines.c: (goom_lines): * gst/goom/lines.h: Make goom reentrant by moving all important static variables into instance structures. (Fixes #329181) 2006-02-04 15:41:43 +0000 Tim-Philipp Müller gst/avi/gstavidemux.*: Third attempt, use gst_pad_is_linked() this time. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_all_source_pads_unlinked), (gst_avi_demux_process_next_entry): * gst/avi/gstavidemux.h: Third attempt, use gst_pad_is_linked() this time. 2006-02-04 13:30:12 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Adjust for data length indicators when parsing (Fixes #329810) Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_split_strings): Adjust for data length indicators when parsing (Fixes #329810) Fix stupid bug parsing UTF-8 tag text. Output tag strings with multiple fields as multiple tags, so the app gets all the data. 2006-02-03 20:05:20 +0000 Edgard Lima * ChangeLog: * ext/flac/gstflacenc.c: Fixed a bug add in last commit, where no event is send. Thanks Tim to show me. Original commit message from CVS: Fixed a bug add in last commit, where no event is send. Thanks Tim to show me. 2006-02-03 18:07:35 +0000 Edgard Lima * ChangeLog: * ext/flac/gstflacenc.c: * gst/matroska/ebml-read.c: Just make it compile with --disable-gst-debug. Original commit message from CVS: Just make it compile with --disable-gst-debug. 2006-02-03 16:55:42 +0000 Christian Schaller * gst-plugins-good.spec.in: update spec file Original commit message from CVS: update spec file 2006-02-03 13:06:24 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Never output a tag with a null contents string. Original commit message from CVS: * gst/id3demux/id3v2frames.c: (parse_text_identification_frame), (id3v2_tag_to_taglist), (id3v2_genre_string_to_taglist), (id3v2_genre_fields_to_taglist): Never output a tag with a null contents string. 2006-02-02 21:00:16 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Only pause if all pads are unlinked AND we've tried to send data on all of them at least once. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked): Only pause if all pads are unlinked AND we've tried to send data on all of them at least once. 2006-02-02 12:29:24 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Make loop function/task pause itself when all source pads are unlinked. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked), (gst_avi_demux_process_next_entry), (gst_avi_demux_loop): Make loop function/task pause itself when all source pads are unlinked. 2006-02-02 10:47:15 +0000 Tim-Philipp Müller Use new functions from core to render a bin from a string. Fixes build. Up requirements to core CVS. Original commit message from CVS: * configure.ac: * ext/gconf/gconf.c: (gst_gconf_render_bin_from_key): Use new functions from core to render a bin from a string. Fixes build. Up requirements to core CVS. 2006-02-01 11:01:04 +0000 Tim-Philipp Müller gst/auparse/gstauparse.c: Don't push buffers into the adapter that we are going to push downstream again without fram... Original commit message from CVS: * gst/auparse/gstauparse.c: (gst_au_parse_chain): Don't push buffers into the adapter that we are going to push downstream again without framing anyway. Also, the adaptor takes ownership of buffers put into it (fixes auparse pushing invalid buffers for .au files with ADPCM contents). Finally, set caps on all outgoing buffers. 2006-01-30 23:13:05 +0000 Jan Schmidt gst/id3demux/: Someone should kick my butt. Remove ID3v1 tags from the end of the file. Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_chain), (gst_id3demux_read_id3v1), (gst_id3demux_sink_activate), (gst_id3demux_send_tag_event): * gst/id3demux/id3tags.c: (id3demux_read_id3v1_tag): Someone should kick my butt. Remove ID3v1 tags from the end of the file. Improve error messages. Send the TAG message as soon as we complete typefinding, instead of waiting until we send the first buffer. Downstream tag event is still sent before the first buffer. 2006-01-29 20:07:49 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackdec.c: Add debug category, use boilerplate macros, fix handling of widths of 32 bits. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_setcaps), (gst_wavpack_dec_base_init), (gst_wavpack_dec_dispose), (gst_wavpack_dec_class_init), (gst_wavpack_dec_sink_event), (gst_wavpack_dec_init), (gst_wavpack_dec_format_samples), (gst_wavpack_dec_chain), (gst_wavpack_dec_plugin_init): Add debug category, use boilerplate macros, fix handling of widths of 32 bits. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init), (gst_wavpack_parse_dispose), (gst_wavpack_parse_class_init), (gst_wavpack_parse_index_get_last_entry), (gst_wavpack_parse_index_get_entry_from_sample), (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset), (gst_wavpack_parse_src_query), (gst_wavpack_parse_scan_to_find_sample), (gst_wavpack_parse_send_newsegment), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_src_event), (gst_wavpack_parse_init), (gst_wavpack_parse_get_upstream_length), (gst_wavpack_parse_pull_buffer), (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_loop), (gst_wavpack_parse_change_state), (gst_wavepack_parse_sink_activate), (gst_wavepack_parse_sink_activate_pull), (gst_wavpack_parse_plugin_init): * ext/wavpack/gstwavpackparse.h: Rewrite a bit, mostly to fix flow logic and to make seeking work. Fix buffer/event refcounting. Add some debug statements. Add width of 32 to source pad template caps. Use boilerplate macros. 2006-01-27 12:17:56 +0000 Andy Wingo ext/dv/: Call dv_set_error_log (dv_decoder_t *, NULL); after dv_decoder_new to not have warings flooding stderr. this... Original commit message from CVS: 2006-01-27 Jan Gerber Reviewed by: Andy Wingo * ext/dv/gstdvdec.c (gst_dvdec_change_state): * ext/dv/gstdvdemux.c (gst_dvdemux_change_state): Call dv_set_error_log (dv_decoder_t *, NULL); after dv_decoder_new to not have warings flooding stderr. this is the suggested way also used in dvgrab and kino. (#328336) 2006-01-27 01:43:07 +0000 Jan Schmidt sys/oss/gstosssink.c: Free the device name string when finalised. Original commit message from CVS: * sys/oss/gstosssink.c: (gst_oss_sink_class_init), (gst_oss_sink_init), (gst_oss_sink_finalise): Free the device name string when finalised. 2006-01-26 16:23:42 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Fix wrong memcpy source pointer. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Fix wrong memcpy source pointer. 2006-01-25 22:05:28 +0000 Tim-Philipp Müller gst/id3demux/gstid3demux.c: Don't put function calls in g_return_if_fail() statements, or they'll be replaced with NO... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_remove_srcpad): Don't put function calls in g_return_if_fail() statements, or they'll be replaced with NOOPs if someone compiles with G_DISABLE_CHECKS defined. 2006-01-25 20:33:05 +0000 Jan Schmidt * ChangeLog: changelog surgery Original commit message from CVS: changelog surgery 2006-01-25 18:23:05 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Never trust ANY information encoded in a media file, especially when it's giving you size... Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame): Never trust ANY information encoded in a media file, especially when it's giving you sizes. (Fixes #328452) 2006-01-24 18:03:46 +0000 Edgard Lima * ChangeLog: * gst/rtp/gstrtpg711pay.c: I'm too lazy to comment this Original commit message from CVS: Patch written by Kai Vehmanen applied. See bug #325148. 2006-01-24 11:58:53 +0000 Edward Hervey gst/qtdemux/qtdemux.c: More coherent framerate setting on caps. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (gst_qtdemux_add_stream), (qtdemux_parse_trak): More coherent framerate setting on caps. If sample_size is available, use that for the samples' duration in the index. This enables single frame streams to work (and I imagine fixes some other cases). Tested on testsuite, no regression. 2006-01-23 18:39:31 +0000 Edward Hervey gst/matroska/: Added recognition of Real Audio and Video streams in matroska demuxer. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps), (gst_matroska_demux_plugin_init): * gst/matroska/matroska-ids.h: Added recognition of Real Audio and Video streams in matroska demuxer. 2006-01-23 18:37:16 +0000 Tim-Philipp Müller ext/lame/gstlame.*: Contrary to what the const char in the lame API might suggest, lame expects us to keep the string... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_finalize), (gst_lame_class_init), (gst_lame_init), (add_one_tag), (gst_lame_set_metadata): * ext/lame/gstlame.h: Contrary to what the const char in the lame API might suggest, lame expects us to keep the strings we pass to id3tag_set_foo() around; it doesn't free them either though, so we have to store them somewhere and free them later when we can be sure lame doesn't need them any longer. 2006-01-23 15:10:55 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Added codec recognition for: _ VP31 : video/x-vp3 _ AVDJ : image/jpeg _ dvcp, dvc : video/x-d... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps): Added codec recognition for: _ VP31 : video/x-vp3 _ AVDJ : image/jpeg _ dvcp, dvc : video/x-dv, systemstream=(boolean)false _ 0x6d730017 : audio/x-adpcm, layout=(string)quicktime 2006-01-23 15:02:04 +0000 Tim-Philipp Müller ext/lame/gstlame.c: don't pass an uninitialised string pointer to lame if we don't know how to handle the tag type, a... Original commit message from CVS: * ext/lame/gstlame.c: (add_one_tag): Fix handling of GST_TAG_DATE (#311679), don't pass an uninitialised string pointer to lame if we don't know how to handle the tag type, and fix minor memory leak. 2006-01-23 14:32:47 +0000 Jan Schmidt gst/id3demux/id3v2frames.c: Remove errant break statement, and fix compilation with older GCC. Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist): Remove errant break statement, and fix compilation with older GCC. 2006-01-23 12:04:12 +0000 Jan Schmidt * ChangeLog: Mention that my last commit fixes #328241 Original commit message from CVS: Mention that my last commit fixes #328241 2006-01-23 11:06:34 +0000 Tim-Philipp Müller sys/sunaudio/: Export functions that are needed in other parts of the code, makes the mixer actually work; adjust mag... Original commit message from CVS: Reviewed by: Tim-Philipp Müller * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_init): Export functions that are needed in other parts of the code, makes the mixer actually work; adjust magic minimum buffer-time value from 3ms to 5ms to work around stuttering during mp3 playback (#327765). 2006-01-23 10:44:03 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: Fix possible deadlock in matroska muxer (#327825). Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad), (gst_matroska_mux_write_data), (gst_matroska_mux_collected): Fix possible deadlock in matroska muxer (#327825). 2006-01-23 09:59:03 +0000 Jens Granseuer C89 fixes: declare variables at the beginning of a block and Original commit message from CVS: * ext/libpng/gstpngenc.c: (gst_pngenc_chain): * gst/avi/gstavidemux.c: (gst_avi_demux_invert): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps): * gst/rtsp/sdpmessage.h: * gst/udp/gstdynudpsink.c: (gst_dynudpsink_render): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_get_stats): C89 fixes: declare variables at the beginning of a block and make gcc-2.9x happy (#328264; patch by: Jens Granseuer ). 2006-01-23 09:22:17 +0000 Jan Schmidt gst/id3demux/: Rewrite parsing of text tags to handle multiple NULL terminated strings. Parse numeric genre strings a... Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag): * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_comment_frame), (parse_text_identification_frame), (id3v2_tag_to_taglist), (id3v2_are_digits), (id3v2_genre_string_to_taglist), (id3v2_genre_fields_to_taglist), (parse_split_strings), (free_tag_strings): Rewrite parsing of text tags to handle multiple NULL terminated strings. Parse numeric genre strings and ID3v2 type "(3)(6)Alternative" style genre strings. Parse dates that are only YYYY or YYYY-mm format. 2006-01-21 11:43:53 +0000 Fabrizio gst/qtdemux/qtdemux.c: 'twos' and 'sowt' fourcc can be 16bit or 8bit audio. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_parse_trak), (qtdemux_audio_caps): 'twos' and 'sowt' fourcc can be 16bit or 8bit audio. Fix 8bit case (#327133, based on patch by: Fabrizio Gennari ). Also, "G_LITTLE_ENDIAN" and "G_BIG_ENDIAN" are not valid literals for endianness in caps strings, only "LITTLE_ENDIAN" and "BIG_ENDIAN" are valid. 2006-01-20 15:06:28 +0000 Christoph Burghardt gst/videobox/gstvideobox.c: Don't forget to initialize liboil, otherwise our oil functions Original commit message from CVS: * gst/videobox/gstvideobox.c: (gst_video_box_class_init): Don't forget to initialize liboil, otherwise our oil functions will crash (fixes #327871; patch by: Christoph Burghardt ). 2006-01-19 21:46:32 +0000 Tim-Philipp Müller * ChangeLog: ChangeLog surgery (last entry may have been slightly misleading) Original commit message from CVS: ChangeLog surgery (last entry may have been slightly misleading) 2006-01-19 21:00:50 +0000 Brian Cameron configure.ac: just like in the core and gst-plugins-base. Fixes build on Solaris (fixes Original commit message from CVS: * configure.ac: Use plain AS_LIBTOOL_TAGS instead of AS_LIBTOOL_TAGS([CXX]), just like in the core and gst-plugins-base. Fixes build on Solaris (fixes #326683; patch by: Brian Cameron ) 2006-01-19 00:10:51 +0000 Tim-Philipp Müller ext/cdio/: Fix build for libcdio versions >= 76; give slightly lower rank than cdparanoia. Original commit message from CVS: * ext/cdio/gstcdio.c: (gst_cdio_add_cdtext_field), (plugin_init): * ext/cdio/gstcdio.h: * ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_get_cdtext): Fix build for libcdio versions >= 76; give slightly lower rank than cdparanoia. 2006-01-18 19:30:36 +0000 Tim-Philipp Müller Port libcdio cdda source, formerly known as cddasrc, now known as cdiocddasrc (fixes #323327). Should also read CD-TE... Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/cdio/Makefile.am: * ext/cdio/gstcdio.c: * ext/cdio/gstcdio.h: * ext/cdio/gstcdiocddasrc.c: * ext/cdio/gstcdiocddasrc.h: Port libcdio cdda source, formerly known as cddasrc, now known as cdiocddasrc (fixes #323327). Should also read CD-TEXT if available, but that's not tested (fixes #317658). 2006-01-18 19:08:08 +0000 Tommi Myöhänen gst/wavparse/gstwavparse.c: Fix conversion from TIME to BYTES format (fixes #326864; Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_pad_convert): Fix conversion from TIME to BYTES format (fixes #326864; patch by: Tommi Myöhänen ) 2006-01-18 18:54:02 +0000 Edgard Lima * gst/qtdemux/qtdemux.c: Ronald's patch applied. see bug #326318. Original commit message from CVS: Ronald's patch applied. see bug #326318. 2006-01-17 16:45:43 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.*: Fix seeking for quicktime files. Could still use some more love and sophistication. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_init), (gst_qtdemux_send_event), (gst_qtdemux_handle_src_event), (gst_qtdemux_change_state), (gst_qtdemux_loop_header): * gst/qtdemux/qtdemux.h: Fix seeking for quicktime files. Could still use some more love and sophistication. 2006-01-16 10:23:47 +0000 Christian Schaller * gst-plugins-good.spec.in: update with love Original commit message from CVS: update with love 2006-01-15 20:21:48 +0000 Sergey Scobich gst/id3demux/id3v2frames.c: Fix compilation of id3demux when zlib is not present. Original commit message from CVS: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame): Fix compilation of id3demux when zlib is not present. (Fixes #326602; patch by: Sergey Scobich) 2006-01-15 14:12:12 +0000 Tim-Philipp Müller ext/esd/Makefile.am: otherwise build will fail for folks with libesd in a non-standard prefix (#327009). Original commit message from CVS: * ext/esd/Makefile.am: Add $(ESD_CFLAGS), otherwise build will fail for folks with libesd in a non-standard prefix (#327009). 2006-01-13 19:29:27 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: back to head Original commit message from CVS: back to head 2006-01-13 19:25:40 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/upload.mak: releasing 0.10.1 Original commit message from CVS: releasing 0.10.1 2006-01-13 18:37:13 +0000 Wim Taymans ext/jpeg/gstsmokeenc.c: fix memleak. Fixes #326618 Original commit message from CVS: patch by: Wim Taymans * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain): fix memleak. Fixes #326618 2006-01-13 18:35:00 +0000 Mike Smith gst/level/gstlevel.c: Fix memleak. Fixes #326612 Original commit message from CVS: 2006-01-13 Thomas Vander Stichele patch by: Mike Smith * gst/level/gstlevel.c: (gst_level_message_new), (gst_level_message_append_channel): Fix memleak. Fixes #326612 2006-01-11 11:39:10 +0000 Thomas Vander Stichele configure.ac: prereleasing Original commit message from CVS: * configure.ac: prereleasing * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: update translations 2006-01-11 11:04:03 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Add support for Indeo3 video in Quicktime files. Original commit message from CVS: reviewed by: Edward Hervey * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add support for Indeo3 video in Quicktime files. Closes #326524 2006-01-10 12:38:59 +0000 Michael Smith gst/level/gstlevel.c: Don't leak filter arrays. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_dispose): Don't leak filter arrays. 2006-01-09 17:04:52 +0000 Christian Schaller * ChangeLog: * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/upload.mak: * gst-plugins-good.spec.in: * sys/Makefile.am: * sys/sunaudio/Makefile.am: * sys/sunaudio/gstsunaudio.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiomixerctrl.c: * sys/sunaudio/gstsunaudiomixerctrl.h: * sys/sunaudio/gstsunaudiomixertrack.c: * sys/sunaudio/gstsunaudiomixertrack.h: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: add Sun Audio plugin. Verified that nothing breaks and that make check works. Original commit message from CVS: add Sun Audio plugin. Verified that nothing breaks and that make check works. Don't think the docs gets properly built yet, but I don't understand exactly how to enable that. 2006-01-07 20:01:09 +0000 Philippe Kalaf gst-plugins-good/gst/udp/: Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc, overrides the por... Original commit message from CVS: 2005-01-07 Philippe Khalaf * gst-plugins-good/gst/udp/gstdynudpsink.c: * gst-plugins-good/gst/udp/gstudpsrc.c: Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc, overrides the port or multicast parameters. Fixes bugs #323021. 2006-01-06 16:28:30 +0000 Tim-Philipp Müller ext/gconf/: Add new gconfaudiosrc and gconfvideosrc elements (needed for gnome-sound-recorder). Original commit message from CVS: * ext/gconf/Makefile.am: * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init), (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_reset), (gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose), (do_toggle_element), (cb_toggle_element), (gst_gconf_audio_src_change_state): * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfelements.c: (plugin_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init), (gst_gconf_video_src_class_init), (gst_gconf_video_src_reset), (gst_gconf_video_src_init), (gst_gconf_video_src_dispose), (do_toggle_element), (cb_toggle_element), (gst_gconf_video_src_change_state): * ext/gconf/gstgconfvideosrc.h: Add new gconfaudiosrc and gconfvideosrc elements (needed for gnome-sound-recorder). 2006-01-06 11:46:53 +0000 Edward Hervey gst/id3demux/gstid3demux.c: Add gst_element_no_more_pads() for proper decodebin behaviour. Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad): Add gst_element_no_more_pads() for proper decodebin behaviour. * gst/id3demux/id3v2frames.c: (parse_comment_frame), (parse_text_identification_frame), (parse_split_strings): Failure to decode some tags is not a GST_ERROR() but a GST_WARNING() When iterating over a chunk of text, check that we haven't gone too far. 2006-01-05 23:17:44 +0000 Sébastien Moutte * sys/directdraw/gstdirectdrawplugin.c: * sys/directdraw/gstdirectdrawsink.c: * sys/directdraw/gstdirectdrawsink.h: * sys/directsound/gstdirectsoundplugin.c: * sys/directsound/gstdirectsoundsink.c: * sys/directsound/gstdirectsoundsink.h: * win32/vs6/libgstdirectdraw.dsp: * win32/vs6/libgstdirectsound.dsp: added sys/directdraw added sys/directsound added win32/vs6/gst_plugins_bad.dsw added win32/vs6/libgstdirectsound.dsp ... Original commit message from CVS: 2006-01-05 Sebastien Moutte * added sys/directdraw * added sys/directsound * added win32/vs6/gst_plugins_bad.dsw * added win32/vs6/libgstdirectsound.dsp * added win32/vs6/libgstdirectdraw.dsp * added win32/common/config.h 2006-01-05 17:03:45 +0000 Stefan Kost gst/videobox/gstvideobox.c: call oil_init() when using liboil Original commit message from CVS: * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (plugin_init): call oil_init() when using liboil 2006-01-04 17:28:49 +0000 Wim Taymans ext/jpeg/: Fix leaks. Original commit message from CVS: * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain): Fix leaks. 2006-01-02 19:38:32 +0000 Tim-Philipp Müller ext/flac/gstflacdec.c: Don't g_assert() where we should just return FALSE; remove unnecessary g_assert(); initialize ... Original commit message from CVS: Reviewed by: Tim-Philipp Müller * ext/flac/gstflacdec.c: (gst_flac_dec_write), (gst_flac_dec_convert_src), (gst_flac_dec_src_query), (gst_flac_dec_change_state): Don't g_assert() where we should just return FALSE; remove unnecessary g_assert(); initialize some fields properly in state change function (fixes #325504). Also, use GST_DEBUG_OBJECT in two more places. 2005-12-30 15:51:05 +0000 Stefan Kost configure.ac: also remove smoothwave's Makefile.am Original commit message from CVS: * configure.ac: also remove smoothwave's Makefile.am * docs/plugins/Makefile.am: fix plugin docs 2005-12-30 15:39:17 +0000 Thomas Vander Stichele * gst/smoothwave/.gitignore: * gst/smoothwave/Makefile.am: * gst/smoothwave/README: * gst/smoothwave/demo-osssrc.c: * gst/smoothwave/gstsmoothwave.c: * gst/smoothwave/gstsmoothwave.h: remove old plugin that went bad Original commit message from CVS: remove old plugin that went bad 2005-12-30 15:34:18 +0000 Stefan Kost tests/examples/Makefile.am: added missing Makefile.am Original commit message from CVS: * tests/examples/Makefile.am: added missing Makefile.am 2005-12-30 15:28:44 +0000 Stefan Kost moved level-example to tests/examples/level-example Original commit message from CVS: * configure.ac: * gst/level/Makefile.am: * gst/level/level-example.c: * tests/Makefile.am: * tests/examples/level/Makefile.am: * tests/examples/level/level-example.c: (message_handler), (main): moved level-example to tests/examples/level-example * tests/old/examples/level/demo.c: (main): * tests/old/examples/level/plot.c: (main): some initial fixes 2005-12-29 16:36:19 +0000 Michael Smith gst/udp/gstmultiudpsink.*: Track packets sent per client in addition to bytes sent; provide this info through get-sta... Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render), (gst_multiudpsink_remove), (gst_multiudpsink_get_stats): * gst/udp/gstmultiudpsink.h: Track packets sent per client in addition to bytes sent; provide this info through get-stats signal 2005-12-29 11:26:12 +0000 Tim-Philipp Müller gst/auparse/gstauparse.c: Can't use gst_object_unref() on a GstAdapter (#325191). Original commit message from CVS: * gst/auparse/gstauparse.c: (gst_au_parse_dispose): Can't use gst_object_unref() on a GstAdapter (#325191). 2005-12-28 18:55:32 +0000 Jan Schmidt gst/id3demux/id3tags.c: If a broken tag has 0 bytes payload, at least still skip the 10 byte header Original commit message from CVS: * gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag): If a broken tag has 0 bytes payload, at least still skip the 10 byte header 2005-12-22 15:00:41 +0000 Philippe Kalaf gst-plugins-good/gst/rtp/: Making these depayloaders (H263+ and mpeg4 video) inherit from Original commit message from CVS: 2005-12-22 Philippe Khalaf * gst-plugins-good/gst/rtp/gstrtph263pdepay.h: * gst-plugins-good/gst/rtp/gstrtph263pdepay.c: * gst-plugins-good/gst/rtp/gstrtpmp4vdepay.h: * gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c: Making these depayloaders (H263+ and mpeg4 video) inherit from RtpBaseDepayloaderClass. Fixes bugs #323922 and #323908. 2005-12-21 17:15:09 +0000 Jan Schmidt docs/plugins/gst-plugins-good-plugins.*: Regenerate the plugin hiearchy. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: Regenerate the plugin hiearchy. 2005-12-21 15:24:59 +0000 Jan Schmidt Add documentation for id3demux. Original commit message from CVS: 2005-12-21 Jan Schmidt * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * gst/id3demux/gstid3demux.c: (gst_id3demux_get_type), (gst_id3demux_base_init), (gst_id3demux_class_init), (gst_id3demux_chain): * gst/id3demux/gstid3demux.h: Add documentation for id3demux. Don't fail if the first buffer is not at offset 0, just attempt to typefind and do pass through Rename the gst_type function from gst_gst_id3demux.. 2005-12-20 12:44:25 +0000 Michael Smith gst/udp/gstmultiudpsink.*: Collect statistics; return them from get_stats. Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render), (gst_multiudpsink_add), (gst_multiudpsink_remove), (gst_multiudpsink_get_stats): * gst/udp/gstmultiudpsink.h: Collect statistics; return them from get_stats. 2005-12-19 15:43:30 +0000 Edward Hervey gst/avi/gstavidemux.c: Stupid signedness issue... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan): Stupid signedness issue... 2005-12-19 15:19:44 +0000 Edward Hervey ext/swfdec/gstswfdec.c: Add debugging category and return GstFlowReturn in the right places Original commit message from CVS: * ext/swfdec/gstswfdec.c: (gst_swfdec_class_init), (gst_swfdec_chain), (gst_swfdec_render): Add debugging category and return GstFlowReturn in the right places * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link): Get something from the peer pad once we've checked if there is a peer pad. * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state), (qtdemux_tree_get_child_by_type), (qtdemux_parse_trak), (qtdemux_video_caps): Couple of fixes 2005-12-19 15:06:27 +0000 Edward Hervey gst/avi/gstavidemux.c: Construct index for indexless files. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml), (gst_avi_demux_peek_tag), (gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan), (gst_avi_demux_stream_header), (gst_avi_demux_loop): Construct index for indexless files. Make sure pad/buffers are correctly reset to NULL once we don't need them anymore, else we get lovely segfaults/assertions. * gst/wavparse/gstwavparse.c: Yes, you can have 96KHz audio and wma in wav :( 2005-12-18 15:14:44 +0000 Jan Schmidt configure.ac: Check for optional dependency on zlib for id3demux Original commit message from CVS: * configure.ac: Check for optional dependency on zlib for id3demux * gst/id3demux/Makefile.am: * gst/id3demux/gstid3demux.c: (gst_gst_id3demux_get_type), (gst_id3demux_base_init), (gst_id3demux_class_init), (gst_id3demux_reset), (gst_id3demux_init), (gst_id3demux_dispose), (gst_id3demux_add_srcpad), (gst_id3demux_remove_srcpad), (gst_id3demux_trim_buffer), (gst_id3demux_chain), (gst_id3demux_set_property), (gst_id3demux_get_property), (id3demux_get_upstream_size), (gst_id3demux_srcpad_event), (gst_id3demux_read_id3v1), (gst_id3demux_read_id3v2), (gst_id3demux_sink_activate), (gst_id3demux_src_activate_pull), (gst_id3demux_src_checkgetrange), (gst_id3demux_read_range), (gst_id3demux_src_getrange), (gst_id3demux_change_state), (gst_id3demux_pad_query), (gst_id3demux_get_query_types), (simple_find_peek), (simple_find_suggest), (gst_id3demux_do_typefind), (gst_id3demux_send_tag_event), (plugin_init): * gst/id3demux/gstid3demux.h: * gst/id3demux/id3tags.c: (read_synch_uint), (id3demux_read_id3v1_tag), (id3demux_read_id3v2_tag), (id3demux_id3v2_frame_hdr_size), (convert_fid_to_v240), (id3demux_id3v2_frames_to_tag_list): * gst/id3demux/id3tags.h: * gst/id3demux/id3v2.4.0-frames.txt: * gst/id3demux/id3v2.4.0-structure.txt: * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame), (parse_comment_frame), (parse_text_identification_frame), (id3v2_tag_to_taglist), (parse_split_strings): All new LGPL id3 demuxer. Can use zlib for compressed frames, otherwise it discards them. Works on my test files. * gst/wavparse/gstwavparse.c: (gst_wavparse_loop): Don't send EOS to a non-existing srcpad The debug category can be static 2005-12-17 17:48:38 +0000 Julien Moutte docs/plugins/: Updates. Original commit message from CVS: 2005-12-17 Julien MOUTTE * docs/plugins/gst-plugins-bad-plugins-decl.txt: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-undocumented.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/plugins/inspect/plugin-qtdemux.xml: * docs/plugins/inspect/plugin-sdlvideosink.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: Updates. * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_surface_create), (gst_dfbvideosink_event_thread), (gst_dfbvideosink_enum_vmodes), (gst_dfbvideosink_enum_devices), (gst_dfbvideosink_setup), (gst_dfbvideosink_cleanup), (gst_dfbvideosink_can_blit_from_format), (gst_dfbvideosink_get_best_vmode), (gst_dfbvideosink_getcaps), (gst_dfbvideosink_setcaps), (gst_dfbvideosink_show_frame), (gst_dfbvideosink_buffer_alloc), (gst_dfbsurface_finalize), (gst_dfbvideosink_interface_supported), (gst_dfbvideosink_navigation_send_event), (gst_dfbvideosink_update_colorbalance), (gst_dfbvideosink_colorbalance_list_channels), (gst_dfbvideosink_colorbalance_set_value), (gst_dfbvideosink_colorbalance_get_value), (gst_dfbvideosink_colorbalance_init), (gst_dfbvideosink_set_property), (gst_dfbvideosink_get_property), (gst_dfbvideosink_init), (gst_dfbvideosink_class_init): * ext/directfb/dfbvideosink.h: Implement vertical sync and color balance interface. 2005-12-16 21:57:51 +0000 Stefan Kost change some char* into char[] Original commit message from CVS: * ext/esd/esdmon.c: (gst_esdmon_open_audio): * ext/esd/esdsink.c: (gst_esdsink_prepare): * gst/multipart/multipartdemux.c: change some char* into char[] 2005-12-16 19:32:53 +0000 Wim Taymans gst/wavparse/gstwavparse.*: Use GstSegment to implement more seeking features. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_reset), (gst_wavparse_other), (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull): * gst/wavparse/gstwavparse.h: Use GstSegment to implement more seeking features. 2005-12-16 12:25:38 +0000 Tim-Philipp Müller ext/wavpack/gstwavpackdec.c: Oops, remove trailing comma from caps string. Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: Oops, remove trailing comma from caps string. 2005-12-16 10:12:49 +0000 Benjamin Pineau gst/rtsp/rtspconnection.c: Add include and move include to make things work on OpenBSD a... Original commit message from CVS: * gst/rtsp/rtspconnection.c: Add include and move include to make things work on OpenBSD as well (fixes #323717; patch by: Benjamin Pineau) 2005-12-16 09:59:21 +0000 gcocatre@gmail.com ext/wavpack/: Wavpack supports samplerates from 6-192kHz, fix pad template remove buffer-frames from caps, they are g... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link): * ext/wavpack/gstwavpackparse.c: Wavpack supports samplerates from 6-192kHz, fix pad template caps (fixes #322973; patch by: gcocatre@gmail.com). Also remove buffer-frames from caps, they are gone in 0.10. 2005-12-14 20:05:45 +0000 Edgard Lima * ChangeLog: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: Set clock rate to be fixed in 8000. It fixes bug #324012. Original commit message from CVS: Set clock rate to be fixed in 8000. It fixes bug #324012. 2005-12-14 18:07:16 +0000 Philippe Kalaf gst-plugins-good/gst/rtp/: Fixed payload range in payloder caps. Removed payload range completly from depayloaders as... Original commit message from CVS: 2005-12-14 Philippe Khalaf * gst-plugins-good/gst/rtp/gstasteriskh263.c: * gst-plugins-good/gst/rtp/gstrtpamrdepay.c: * gst-plugins-good/gst/rtp/gstrtpamrpay.c: * gst-plugins-good/gst/rtp/gstrtpg711depay.c: * gst-plugins-good/gst/rtp/gstrtpg711depay.c: * gst-plugins-good/gst/rtp/gstrtpgsmdepay.c: * gst-plugins-good/gst/rtp/gstrtph263pay.c: * gst-plugins-good/gst/rtp/gstrtph263pdepay.c: * gst-plugins-good/gst/rtp/gstrtph263ppay.c: * gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c: * gst-plugins-good/gst/rtp/gstrtpmp4vpay.c: * gst-plugins-good/gst/rtp/gstrtpmpadepay.c: * gst-plugins-good/gst/rtp/gstrtpmpapay.c: * gst-plugins-good/gst/rtp/README: Fixed payload range in payloder caps. Removed payload range completly from depayloaders as they don't require payload type in their caps. In effect, there isn't any specific payload type for any given codec, only suggestions. Fixes bug #324011. 2005-12-13 21:58:42 +0000 Julien Moutte gst/videomixer/videomixer.c: Code cleanup and re-enabling queued time validity check for correct EOS handling. Original commit message from CVS: 2005-12-13 Julien MOUTTE * gst/videomixer/videomixer.c: (gst_videomixer_init), (gst_videomixer_fill_queues), (gst_videomixer_blend_buffers), (gst_videomixer_collected): Code cleanup and re-enabling queued time validity check for correct EOS handling. 2005-12-13 17:18:32 +0000 Tim-Philipp Müller sys/oss/gstossmixerelement.c: Add 'device-name' property and fix state change function. Original commit message from CVS: * sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init), (gst_oss_mixer_element_get_property), (gst_oss_mixer_element_change_state): Add 'device-name' property and fix state change function. 2005-12-13 10:45:04 +0000 Edward Hervey gst/flx/gstflxdec.c: If the speed of the file is null in the header, set the frame_time to the default setting of GST... Original commit message from CVS: * gst/flx/gstflxdec.c: (gst_flxdec_chain): If the speed of the file is null in the header, set the frame_time to the default setting of GST_SECOND / 70. Which is the default frame_delay for .fli files as stated in this document : http://www.compuphase.com/flic.htm Would be nice to have the time conversion done properly too (duration = flxh->frames * flxdec->frame_time) 2005-12-12 22:29:34 +0000 Julien Moutte Adding documentation for videomixer on my way with a funny sample pipeline. Original commit message from CVS: 2005-12-12 Julien MOUTTE * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps), (gst_videomixer_fill_queues), (gst_videomixer_update_queues), (gst_videomixer_collected): Adding documentation for videomixer on my way with a funny sample pipeline. 2005-12-12 21:43:00 +0000 Julien Moutte gst/videomixer/videomixer.c: Fix caps negotiation. (#323896) Original commit message from CVS: 2005-12-12 Julien MOUTTE * gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps), (gst_videomixer_fill_queues), (gst_videomixer_update_queues), (gst_videomixer_collected): Fix caps negotiation. (#323896) 2005-12-12 18:14:58 +0000 Arwed v. Merkatz * ChangeLog: * gst/matroska/matroska-demux.c: Set correct timestamps on audio laces, fixes playback of mp3 from matroska. Original commit message from CVS: Set correct timestamps on audio laces, fixes playback of mp3 from matroska. 2005-12-12 10:40:42 +0000 Tim-Philipp Müller ext/: GstObjects must be unref'ed with gst_object_unref() instead of g_object_unref(), otherwise things break for GLi... Original commit message from CVS: * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_loop): * ext/libmms/gstmms.c: (gst_mms_src_query), (gst_mms_create): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_src_query), (gst_musepackdec_loop): * ext/swfdec/gstswfdec.c: (gst_swfdec_video_link), (gst_swfdec_src_query): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query): GstObjects must be unref'ed with gst_object_unref() instead of g_object_unref(), otherwise things break for GLib-2.6 users. 2005-12-12 10:30:20 +0000 Tim-Philipp Müller gst/auparse/gstauparse.*: Use gst_object_unref() for GstObjects instead of g_object_unref() and fix a mem leak in a d... Original commit message from CVS: * gst/auparse/gstauparse.c: (gst_au_parse_base_init), (gst_au_parse_class_init), (gst_au_parse_init), (gst_au_parse_dispose), (gst_au_parse_chain), (gst_au_parse_change_state), (plugin_init): * gst/auparse/gstauparse.h: Use gst_object_unref() for GstObjects instead of g_object_unref() and fix a mem leak in a debug statement; while we're at it, also borgify, use boilerplate macros and clean up a little bit. 2005-12-11 20:27:06 +0000 Edward Hervey gst/debug/efence.c: Added pull mode. Original commit message from CVS: * gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange), (gst_efence_checkgetrange), (gst_efence_activate_src_pull): Added pull mode. 2005-12-11 19:25:41 +0000 Tim-Philipp Müller gst/: Use audiotestsrc instead of sinesrc (#323798). Original commit message from CVS: * gst/goom/gstgoom.c: * gst/level/level-example.c: (main): * gst/smoothwave/demo-osssrc.c: (main): Use audiotestsrc instead of sinesrc (#323798). 2005-12-11 17:50:50 +0000 Stefan Kost sys/oss/gstosssink.c: more debug-func-ptr usage Original commit message from CVS: * sys/oss/gstosssink.c: (gst_oss_sink_class_init): more debug-func-ptr usage 2005-12-11 16:43:42 +0000 Zeeshan Ali * ChangeLog: * gst/flx/flx_color.c: * gst/flx/flx_color.h: * gst/flx/flx_fmt.h: * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: Now flxdec works on big-endian machines as well. Original commit message from CVS: Now flxdec works on big-endian machines as well. 2005-12-11 16:14:22 +0000 Tim-Philipp Müller gst/debug/efence.c: Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix Original commit message from CVS: * gst/debug/efence.c: (gst_efence_init), (gst_efence_chain), (gst_fenced_buffer_copy): Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix GST_DEBUG crasher where GST_TIME_FORMAT was not used in conjunction with GST_TIME_ARGS. Also, don't leak pad templates and use GST_DEBUG_FUNCPTR for pad functions. 2005-12-10 20:26:33 +0000 Tim-Philipp Müller ext/flac/gstflacdec.*: Rewrite flacdec a bit, so that even seeking might work now. Most importantly, don't act upon a... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flac_dec_base_init), (gst_flac_dec_class_init), (gst_flac_dec_init), (gst_flac_dec_metadata_callback), (gst_flac_dec_error_callback), (gst_flac_dec_eof), (gst_flac_dec_write), (gst_flac_dec_loop), (gst_flac_dec_convert_src), (gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query), (gst_flac_dec_send_newsegment), (gst_flac_dec_handle_seek_event), (gst_flac_dec_src_event), (gst_flac_dec_change_state): * ext/flac/gstflacdec.h: Rewrite flacdec a bit, so that even seeking might work now. Most importantly, don't act upon any flow return values we get, just tell the decoder everything's dandy and act on the flow return values later on in the loop function. We don't want to mess up the internal decoder state for non-fatal things like flushing pads etc. Other than that, use GstSegment (segment seeks don't work yet though, but should be easy to add), use boilerplate macros, drop the superfluous 'flacdec:' from debug messages, use gst_util_uint64_scale_int, and lots of other things. 2005-12-10 14:57:48 +0000 Tim-Philipp Müller configure.ac: Update comment in OSS includes check. Original commit message from CVS: * configure.ac: Update comment in OSS includes check. * sys/oss/gstossdmabuffer.c: * sys/oss/gstosshelper.c: * sys/oss/gstossmixer.c: * sys/oss/gstossmixertrack.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss/oss_probe.c: Don't assume the OSS soundcard.h include is always in the sys/ directory. Instead, use the existing defines from config.h to include the right file. Fixes compilation on OpenBSD 3.8 (#323718). 2005-12-09 19:51:03 +0000 Thomas Vander Stichele * ChangeLog: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * ext/flac/gstflac.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: borgify and fix up documentation Original commit message from CVS: borgify and fix up documentation 2005-12-09 15:30:21 +0000 Jan Schmidt ext/faad/gstfaad.c: Assume that an unknown channel mapping with 2 channels is stereo and play it that way instead of ... Original commit message from CVS: * ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst), (gst_faad_update_caps): Assume that an unknown channel mapping with 2 channels is stereo and play it that way instead of erroring. * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (gst_qtdemux_add_stream), (qtdemux_parse_trak): Handle e.g. jpeg streams with 0 duration frames as having 0 framerate. Debug fixes. Some 64 bit variable fixes 2005-12-09 11:12:48 +0000 Michael Smith ext/flac/gstflacdec.c: Accept a wider range of flac files, more closely matching flac sp Original commit message from CVS: * ext/flac/gstflacdec.c: (raw_caps_factory), (gst_flacdec_write): Accept a wider range of flac files, more closely matching flac sp 2005-12-08 16:27:12 +0000 Julien Moutte docs/plugins/Makefile.am: Add multipart elements. Original commit message from CVS: 2005-12-08 Julien MOUTTE * docs/plugins/Makefile.am: Add multipart elements. * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: Fix flac. * docs/plugins/gst-plugins-good-plugins.hierarchy: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: Add docs. 2005-12-07 11:46:15 +0000 Edward Hervey gst/qtdemux/qtdemux.c: Memleak fixes. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (gst_qtdemux_add_stream): Memleak fixes. Send out EOS for valid reasons (couldn't pull_range() from upstream for example). 2005-12-07 11:40:46 +0000 Edward Hervey gst/avi/gstavidemux.c: Memleak and crasher fixes. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream), (gst_avi_demux_stream_header), (gst_avi_demux_invert): Memleak and crasher fixes. * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_create_sourcepad), (gst_wavparse_stream_headers): Memleak fixes 2005-12-06 19:55:58 +0000 Thomas Vander Stichele * gst/equalizer/gstiirequalizer.c: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: * sys/v4l2/gstv4l2colorbalance.h: * sys/v4l2/gstv4l2element.h: * sys/v4l2/gstv4l2src.h: * sys/v4l2/gstv4l2tuner.h: * sys/v4l2/gstv4l2xoverlay.h: * sys/v4l2/v4l2_calls.c: * sys/v4l2/v4l2_calls.h: * sys/v4l2/v4l2src_calls.c: * sys/v4l2/v4l2src_calls.h: expand tabs Original commit message from CVS: expand tabs 2005-12-06 19:48:07 +0000 Thomas Vander Stichele * ext/lame/gstlame.h: expand tabs Original commit message from CVS: expand tabs 2005-12-06 19:44:58 +0000 Thomas Vander Stichele * ChangeLog: * ext/aalib/gstaasink.h: * ext/cairo/gsttextoverlay.h: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: * ext/esd/esdsink.h: * ext/flac/flac_compat.h: * ext/flac/gstflacdec.h: * ext/flac/gstflacenc.h: * ext/gconf/gconf.h: * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfvideosink.h: * ext/gdk_pixbuf/gstgdkanimation.h: * ext/jpeg/gstjpegdec.h: * ext/jpeg/smokecodec.h: * ext/jpeg/smokeformat.h: * ext/ladspa/gstsignalprocessor.h: * ext/ladspa/search.c: * ext/ladspa/utils.h: * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.c: * ext/libmng/gstmngenc.h: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: * ext/shout2/gstshout2.h: * ext/speex/gstspeexdec.h: * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautovideosink.h: * gst/avi/gstavidemux.h: * gst/cutter/gstcutter.h: * gst/debug/tests.c: * gst/debug/tests.h: * gst/effectv/gstwarp.c: * gst/flx/flx_fmt.h: * gst/flx/gstflxdec.h: * gst/goom/filters.c: * gst/goom/filters.h: * gst/goom/goom_tools.h: * gst/law/alaw-encode.c: * gst/level/gstlevel.c: * gst/level/gstlevel.h: * gst/matroska/ebml-write.h: * gst/matroska/matroska-demux.h: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.h: * gst/monoscope/convolve.c: * gst/monoscope/convolve.h: * gst/multipart/multipartmux.c: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstaggregator.h: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmd5sink.h: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.h: * gst/oldcore/gstshaper.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.h: * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.h: * gst/rtsp/rtspurl.c: * gst/rtsp/rtspurl.h: * gst/rtsp/sdpmessage.c: * gst/rtsp/sdpmessage.h: * gst/smpte/barboxwipes.c: * gst/smpte/gstmask.h: * gst/smpte/gstsmpte.h: * gst/smpte/paint.c: * gst/smpte/paint.h: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.c: * gst/videomixer/videomixer.c: * gst/wavenc/riff.h: * gst/wavparse/gstwavparse.h: * sys/oss/gstossdmabuffer.h: * sys/oss/gstossmixer.h: * sys/oss/gstossmixerelement.h: * sys/oss/gstossmixertrack.h: * sys/oss/gstosssink.c: * sys/oss/gstosssink.h: * sys/oss/gstosssrc.c: * sys/oss/gstosssrc.h: * sys/osxaudio/gstosxaudioelement.h: * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.h: expand tabs Original commit message from CVS: expand tabs 2005-12-05 18:12:07 +0000 Thomas Vander Stichele * configure.ac: back to HEAD Original commit message from CVS: back to HEAD === release 0.10.0 === 2005-12-05 18:03:23 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: releasing 0.10.0 Original commit message from CVS: releasing 0.10.0 2005-12-05 18:01:48 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-qtdemux.xml: releasing 0.10.0 Original commit message from CVS: releasing 0.10.0 2005-12-05 16:21:08 +0000 Thomas Vander Stichele * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2005-12-05 15:08:46 +0000 Thomas Vander Stichele * Makefile.am: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/it.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: update translations Original commit message from CVS: update translations 2005-12-05 13:04:22 +0000 Andy Wingo Update for alloc_buffer changes. Original commit message from CVS: 2005-12-05 Andy Wingo * ext/faac/gstfaac.c: (gst_faac_sink_event), (gst_faac_chain): * ext/faad/gstfaad.c: (gst_faad_chain): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_chain): * ext/lcs/gstcolorspace.c: (gst_colorspace_chain): * ext/xine/xineinput.c: (gst_xine_input_get): * gst/colorspace/gstcolorspace.c: (gst_colorspace_chain): * gst/speed/gstspeed.c: (speed_chain): * gst/videocrop/gstvideocrop.c: (gst_video_crop_chain): Update for alloc_buffer changes. 2005-12-05 13:03:00 +0000 Andy Wingo Update for alloc_buffer changes. Original commit message from CVS: 2005-12-05 Andy Wingo * ext/dv/gstdvdec.c: (gst_dvdec_chain): * ext/flac/gstflacdec.c: (gst_flacdec_write): * ext/flac/gstflacenc.c: (gst_flacenc_write_callback): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain): * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_chain): * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain): * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_process): * ext/libpng/gstpngdec.c: (user_info_callback), (gst_pngdec_task): * ext/speex/gstspeexdec.c: (speex_dec_chain): * ext/speex/gstspeexenc.c: (gst_speexenc_chain): * gst/auparse/gstauparse.c: (gst_auparse_chain): * gst/flx/gstflxdec.c: (gst_flxdec_chain): * gst/goom/gstgoom.c: (gst_goom_chain): * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_vorbis_codec_priv_data), (gst_matroska_demux_add_wvpk_header): * gst/multipart/multipartdemux.c: (gst_multipart_demux_chain): * gst/multipart/multipartmux.c: (gst_multipart_mux_collected): * gst/videomixer/videomixer.c: (gst_videomixer_collected): * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Update for alloc_buffer changes. 2005-12-05 12:23:22 +0000 Michael Smith docs/plugins/gst-plugins-good-plugins.args: Remove args for plugins that aren't in -good. Original commit message from CVS: * docs/plugins/gst-plugins-good-plugins.args: Remove args for plugins that aren't in -good. 2005-12-04 22:26:07 +0000 Christian Schaller * gst-plugins-good.spec.in: remove pango plugin as its gone into base Original commit message from CVS: remove pango plugin as its gone into base 2005-12-03 18:51:48 +0000 Thomas Vander Stichele * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpg711pay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpspeexpay.c: fix element descriptions Original commit message from CVS: fix element descriptions 2005-12-03 18:50:12 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-fdsrc.xml: remove fdsrc docs Original commit message from CVS: remove fdsrc docs 2005-12-01 19:18:08 +0000 Thomas Vander Stichele * configure.ac: back to HEAD Original commit message from CVS: back to HEAD === release 0.9.7 === 2005-12-01 19:14:26 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: releasing 0.9.7 Original commit message from CVS: releasing 0.9.7 2005-12-01 19:13:20 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-qtdemux.xml: releasing 0.9.7 Original commit message from CVS: releasing 0.9.7 2005-12-01 17:53:29 +0000 Thomas Vander Stichele * common: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2005-12-01 15:34:13 +0000 Thomas Vander Stichele * ChangeLog: * docs/plugins/.gitignore: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-rtp.xml: add multipart plugin to docs Original commit message from CVS: add multipart plugin to docs 2005-12-01 15:22:25 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: * ext/Makefile.am: * ext/pango/Makefile.am: * ext/pango/gstclockoverlay.c: * ext/pango/gstclockoverlay.h: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextoverlay.h: * ext/pango/gsttextrender.c: * ext/pango/gsttextrender.h: * ext/pango/gsttimeoverlay.c: * ext/pango/gsttimeoverlay.h: move pango to base Original commit message from CVS: move pango to base 2005-12-01 14:39:30 +0000 Thomas Vander Stichele gst/rtp/: parsers are depayers Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16parse.c: * gst/rtp/gstrtpL16parse.h: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmparse.h: parsers are depayers 2005-12-01 14:30:01 +0000 Thomas Vander Stichele * ChangeLog: * common: * gst/rtp/Makefile.am: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16enc.c: * gst/rtp/gstrtpL16enc.h: * gst/rtp/gstrtpL16parse.c: * gst/rtp/gstrtpL16parse.h: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdec.c: * gst/rtp/gstrtpamrdec.h: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrenc.c: * gst/rtp/gstrtpamrenc.h: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdec.c: * gst/rtp/gstrtpdec.h: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711dec.h: * gst/rtp/gstrtpg711depay.c: * gst/rtp/gstrtpg711depay.h: * gst/rtp/gstrtpg711enc.c: * gst/rtp/gstrtpg711enc.h: * gst/rtp/gstrtpg711pay.c: * gst/rtp/gstrtpg711pay.h: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmenc.h: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmparse.h: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263enc.c: * gst/rtp/gstrtph263enc.h: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdec.c: * gst/rtp/gstrtph263pdec.h: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263penc.c: * gst/rtp/gstrtph263penc.h: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4vdec.c: * gst/rtp/gstrtpmp4vdec.h: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4venc.c: * gst/rtp/gstrtpmp4venc.h: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadec.c: * gst/rtp/gstrtpmpadec.h: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpaenc.c: * gst/rtp/gstrtpmpaenc.h: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtpspeexdec.c: * gst/rtp/gstrtpspeexdec.h: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexenc.c: * gst/rtp/gstrtpspeexenc.h: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpspeexpay.h: Do burger's rename for rtp payloaders and depayloaders Original commit message from CVS: Do burger's rename for rtp payloaders and depayloaders 2005-11-30 19:02:35 +0000 Wim Taymans ext/dv/: Fix seeking in dvdemux again, add some more debug info. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_chain): * ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame): * ext/dv/gstdvdemux.h: Fix seeking in dvdemux again, add some more debug info. 2005-11-30 18:48:56 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: fix tests Original commit message from CVS: fix tests 2005-11-30 18:40:19 +0000 Thomas Vander Stichele * Makefile.am: add tests subdir Original commit message from CVS: add tests subdir 2005-11-30 18:36:02 +0000 Thomas Vander Stichele * tests/check/Makefile.am: add Makefile.am Original commit message from CVS: add Makefile.am 2005-11-30 18:28:53 +0000 Thomas Vander Stichele move Original commit message from CVS: * PORTED_09: * docs/random/PORTED_09: move * tests/Makefile.am: add * win32/gst.sln: remove 2005-11-30 18:24:08 +0000 Thomas Vander Stichele * ChangeLog: * Makefile.am: * check/.gitignore: * check/Makefile.am: * check/elements/.gitignore: * check/elements/level.c: * check/elements/matroskamux.c: * configure.ac: * examples/Makefile.am: * examples/capsfilter/Makefile.am: * examples/capsfilter/capsfilter1.c: * examples/gob/Makefile.am: * examples/gob/gst-identity2.gob: * examples/gstplay/.gitignore: * examples/gstplay/Makefile.am: * examples/gstplay/player.c: * examples/indexing/.gitignore: * examples/indexing/Makefile.am: * examples/indexing/indexmpeg.c: * examples/level/Makefile.am: * examples/level/README: * examples/level/demo.c: * examples/level/plot.c: * examples/stats/Makefile.am: * examples/stats/mp2ogg.c: * examples/switch/.gitignore: * examples/switch/Makefile.am: * examples/switch/switcher.c: move under tests Original commit message from CVS: move under tests 2005-11-30 16:57:57 +0000 Christian Schaller * common: * gst-plugins-good.spec.in: update for latest changes Original commit message from CVS: update for latest changes 2005-11-30 14:53:29 +0000 Tim-Philipp Müller ext/pango/gsttextrender.*: Add missing files. Original commit message from CVS: * ext/pango/gsttextrender.c: (gst_text_render_base_init), (gst_text_render_class_init), (resize_bitmap), (gst_text_render_render_text), (gst_text_render_setcaps), (gst_text_render_fixate_caps), (gst_text_renderer_bitmap_to_ayuv), (gst_text_render_chain), (gst_text_render_finalize), (gst_text_render_init), (gst_text_render_set_property): * ext/pango/gsttextrender.h: Add missing files. 2005-11-30 13:20:57 +0000 Tim-Philipp Müller Port pango-based textoverlay, timeoverlay and textrender to 0.9 and add background shading and text wrapping modes. M... Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/pango/Makefile.am: * ext/pango/gstclockoverlay.c: (gst_clock_overlay_base_init), (gst_clock_overlay_render_time), (gst_clock_overlay_get_text), (gst_clock_overlay_class_init), (gst_clock_overlay_init): * ext/pango/gstclockoverlay.h: * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init), (gst_text_overlay_get_text), (gst_text_overlay_class_init), (gst_text_overlay_finalize), (gst_text_overlay_init), (gst_text_overlay_update_wrap_mode), (gst_text_overlay_setcaps), (gst_text_overlay_text_pad_linked), (gst_text_overlay_text_pad_unlinked), (gst_text_overlay_set_property), (gst_text_overlay_getcaps), (gst_text_overlay_shade_y), (gst_text_overlay_blit_yuv420), (gst_text_overlay_resize_bitmap), (gst_text_overlay_render_text), (gst_text_overlay_push_frame), (gst_text_overlay_pop_video), (gst_text_overlay_pop_text), (gst_text_overlay_collected), (gst_text_overlay_change_state), (plugin_init): * ext/pango/gsttextoverlay.h: * ext/pango/gsttimeoverlay.c: (gst_time_overlay_base_init), (gst_time_overlay_render_time), (gst_time_overlay_get_text), (gst_time_overlay_class_init), (gst_time_overlay_init): * ext/pango/gsttimeoverlay.h: Port pango-based textoverlay, timeoverlay and textrender to 0.9 and add background shading and text wrapping modes. Make timoverlay derive from textoverlay. Also add new clockoverlay element. 2005-11-30 11:10:01 +0000 Julien Moutte gst/udp/Makefile.am: Moved to netbuffer. Original commit message from CVS: 2005-11-30 Julien MOUTTE * gst/udp/Makefile.am: Moved to netbuffer. 2005-11-30 10:18:42 +0000 Julien Moutte Ported multipart mux/demux to 0.9. Original commit message from CVS: 2005-11-30 Julien MOUTTE * configure.ac: * PORTED_O9: * gst/multipart/Makefile.am: * gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init), (gst_multipart_demux_class_init), (gst_multipart_demux_init), (gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain), (gst_multipart_demux_change_state), (gst_multipart_demux_plugin_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init), (gst_multipart_mux_init), (gst_multipart_mux_finalize), (gst_multipart_mux_sinkconnect), (gst_multipart_mux_request_new_pad), (gst_multipart_mux_handle_src_event), (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected), (gst_multipart_mux_change_state): Ported multipart mux/demux to 0.9. 2005-11-30 08:26:47 +0000 Thomas Vander Stichele gst/: update for symbols change Original commit message from CVS: * gst/debug/gstnavigationtest.c: (gst_navigationtest_get_type): * gst/debug/gstnavigationtest.h: * gst/effectv/gstaging.c: (gst_agingtv_get_type): * gst/effectv/gstdice.c: (gst_dicetv_get_type): * gst/effectv/gstedge.c: (gst_edgetv_get_type): * gst/effectv/gstquark.c: (gst_quarktv_get_type): * gst/effectv/gstrev.c: (gst_revtv_get_type): * gst/effectv/gstshagadelic.c: (gst_shagadelictv_get_type): * gst/effectv/gstvertigo.c: (gst_vertigotv_get_type): * gst/effectv/gstwarp.c: (gst_warptv_get_type): * gst/videofilter/gstvideoflip.c: (gst_video_flip_set_property), (gst_video_flip_get_type): * gst/videofilter/gstvideoflip.h: update for symbols change 2005-11-29 17:46:04 +0000 Thomas Vander Stichele gst/udp/: the old gstnet lib was renamed gstnetbuffer (#322257) Original commit message from CVS: * gst/udp/gstdynudpsink.c: * gst/udp/gstudpsrc.c: the old gstnet lib was renamed gstnetbuffer (#322257) 2005-11-29 15:42:01 +0000 Tim-Philipp Müller ext/cairo/gsttextoverlay.c: Actually render the text from the text pad. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_render_text), (gst_text_overlay_collected): Actually render the text from the text pad. 2005-11-29 14:49:00 +0000 Edward Hervey gst/debug/: Update for GstBaseTransform event virtual method Original commit message from CVS: * gst/debug/gstnavseek.c: (gst_navseek_event): * gst/debug/progressreport.c: (gst_progress_report_event): Update for GstBaseTransform event virtual method 2005-11-29 10:55:09 +0000 Thomas Vander Stichele ext/cairo/Makefile.am: no need to link to videofilter Original commit message from CVS: 2005-11-29 Thomas Vander Stichele * ext/cairo/Makefile.am: no need to link to videofilter 2005-11-29 10:46:00 +0000 Thomas Vander Stichele * ChangeLog: * gst/debug/Makefile.am: * gst/debug/gstnavigationtest.h: * gst/effectv/Makefile.am: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/videofilter/Makefile.am: * gst/videofilter/gstvideofilter.c: * gst/videofilter/gstvideofilter.h: * gst/videofilter/gstvideoflip.h: remove the videofilter library and link to the one in base Original commit message from CVS: remove the videofilter library and link to the one in base 2005-11-29 01:30:40 +0000 Thomas Vander Stichele * common: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideoflip.h: borgify Original commit message from CVS: borgify 2005-11-28 17:31:44 +0000 Edward Hervey gst/avi/gstavidemux.c: Useless check now we're setting the current entry correctly. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry): Useless check now we're setting the current entry correctly. 2005-11-28 16:54:03 +0000 Tim-Philipp Müller ext/jpeg/gstjpegenc.c: Don't leak input buffer in chain function (fixes #322667); make state change function thread-s... Original commit message from CVS: * ext/jpeg/gstjpegenc.c: (gst_jpegenc_resync), (gst_jpegenc_chain), (gst_jpegenc_set_property), (gst_jpegenc_get_property), (gst_jpegenc_change_state): Don't leak input buffer in chain function (fixes #322667); make state change function thread-safe; don't repeat the current function name in GST_DEBUG statements; use GST_ROUND_UP_* macros; use gst_pad_alloc_buffer(); misc. minor cleanups. 2005-11-28 15:43:29 +0000 Edward Hervey ext/faad/gstfaad.c: Handle gracefully the consequence of "Maximum number of scalefactor bands exceeded", which result... Original commit message from CVS: * ext/faad/gstfaad.c: (gst_faad_srcgetcaps): Handle gracefully the consequence of "Maximum number of scalefactor bands exceeded", which results in 0 channels with samplerates of 0. * gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state): Do upward transitions, then call parent state_change, then do downward transitions. 2005-11-28 15:13:22 +0000 Jan Schmidt gst/matroska/matroska-mux.c: Look for pixel-aspect-ratio in caps, not pixel_width and pixel_height (Fixes: #322645) Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): Look for pixel-aspect-ratio in caps, not pixel_width and pixel_height (Fixes: #322645) 2005-11-28 12:59:05 +0000 Jan Schmidt gst/matroska/matroska-mux.c: From Michal Benes: frame duration should be GST_SECOND / framerate, not Original commit message from CVS: * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): From Michal Benes: frame duration should be GST_SECOND / framerate, not GST_SECOND * framerate. (Fixes: #322643) 2005-11-27 17:02:53 +0000 Thomas Vander Stichele configure.ac: fix up GST_PLUGIN_LDFLAGS Original commit message from CVS: * configure.ac: fix up GST_PLUGIN_LDFLAGS * gst/rtsp/rtspconnection.c: fix includes (see #317043) * gst/videofilter/Makefile.am: stop installing this library 2005-11-27 15:30:25 +0000 Thomas Vander Stichele * configure.ac: no need for an AS_LIBTOOL call Original commit message from CVS: no need for an AS_LIBTOOL call 2005-11-27 14:33:31 +0000 Thomas Vander Stichele * Makefile.am: * common: * gst-plugins-good.spec.in: add ACLOCAL_AMFLAGS; remove old stuff from spec changelog Original commit message from CVS: add ACLOCAL_AMFLAGS; remove old stuff from spec changelog 2005-11-26 12:54:47 +0000 Edward Hervey ext/dv/gstdvdec.c: Handle the case where the incoming Video dv stream doesn't have a pixel aspect ratio set. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_sink_setcaps): Handle the case where the incoming Video dv stream doesn't have a pixel aspect ratio set. 2005-11-25 22:14:47 +0000 Thomas Vander Stichele * ChangeLog: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * ext/flac/gstflacdec.c: document flacdec Original commit message from CVS: document flacdec 2005-11-25 21:36:18 +0000 Thomas Vander Stichele * ChangeLog: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/inspect/plugin-autodetect.xml: * ext/cairo/gstcairo.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: * ext/cairo/gsttimeoverlay.c: * ext/cairo/gsttimeoverlay.h: do some name borgifying document Original commit message from CVS: do some name borgifying document 2005-11-25 21:02:16 +0000 Thomas Vander Stichele documenting auto*sink using strstr for the video sink lookup, class field is not ordered update other plugins Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init), (gst_auto_video_sink_factory_filter): documenting auto*sink using strstr for the video sink lookup, class field is not ordered update other plugins 2005-11-25 19:58:19 +0000 Edgard Lima * ext/wavpack/Makefile.am: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackparse.c: * ext/wavpack/gstwavpackparse.h: Wavpack ported to 0.9. No support for correction file yet. Original commit message from CVS: Wavpack ported to 0.9. No support for correction file yet. 2005-11-25 18:15:51 +0000 Thomas Vander Stichele ext/wavpack/: put back wavpack - still needs porting Original commit message from CVS: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link), (gst_wavpack_dec_wvclink), (gst_wavpack_dec_get_type), (gst_wavpack_dec_base_init), (gst_wavpack_dec_dispose), (gst_wavpack_dec_class_init), (gst_wavpack_dec_src_query), (gst_wavpack_dec_init), (gst_wavpack_dec_setup_context), (gst_wavpack_dec_format_samples), (gst_wavpack_dec_loop), (gst_wavpack_dec_plugin_init): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_get_type), (gst_wavpack_parse_base_init), (gst_wavpack_parse_dispose), (gst_wavpack_parse_class_init), (gst_wavpack_parse_src_query), (gst_wavpack_parse_src_event), (find_header), (find_sample), (gst_wavpack_parse_seek), (gst_wavpack_parse_init), (gst_wavpack_parse_handle_event), (gst_wavpack_parse_loop), (gst_wavpack_parse_change_state), (gst_wavpack_parse_plugin_init): * ext/wavpack/gstwavpackparse.h: put back wavpack - still needs porting 2005-11-25 18:03:24 +0000 Sebastien Cote gst/udp/gstudpsrc.c: Patch from Sebastien Cote to close control sockets in udpsrc. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_stop): Patch from Sebastien Cote to close control sockets in udpsrc. 2005-11-24 15:07:06 +0000 Julien Moutte gst/effectv/gstquark.c: Flush the planes list on reverse caps negotiation. This was crashing because of differently s... Original commit message from CVS: 2005-11-24 Julien MOUTTE * gst/effectv/gstquark.c: (gst_quarktv_set_caps), (gst_quarktv_get_unit_size), (gst_quarktv_transform), (gst_quarktv_planetable_clear), (gst_quarktv_change_state), (gst_quarktv_base_init), (gst_quarktv_class_init), (gst_quarktv_init): Flush the planes list on reverse caps negotiation. This was crashing because of differently sized buffers. 2005-11-24 12:50:28 +0000 Julien Moutte gst/: Handle strides correctly, fix identity flipping, convert navigation event correctly again. Original commit message from CVS: 2005-11-24 Julien MOUTTE * gst/debug/gstnavigationtest.c: (draw_box_planar411): * gst/videofilter/gstvideoflip.c: (gst_videoflip_method_get_type), (gst_videoflip_set_caps), (gst_videoflip_transform_caps), (gst_videoflip_get_unit_size), (gst_videoflip_flip), (gst_videoflip_transform), (gst_videoflip_handle_src_event), (gst_videoflip_set_property), (gst_videoflip_base_init), (gst_videoflip_class_init), (gst_videoflip_init): Handle strides correctly, fix identity flipping, convert navigation event correctly again. 2005-11-24 11:16:53 +0000 Michael Smith * README: Fix #320288: wrong readme in plugins-good Original commit message from CVS: Fix #320288: wrong readme in plugins-good 2005-11-24 11:06:29 +0000 Thomas Vander Stichele * Makefile.am: fix torture target Original commit message from CVS: fix torture target 2005-11-23 21:25:56 +0000 Thomas Vander Stichele * Makefile.am: add a torture target Original commit message from CVS: add a torture target 2005-11-23 20:05:26 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: back to HEAD Original commit message from CVS: back to HEAD === release 0.9.6 === 2005-11-23 19:57:49 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-qtdemux.xml: releasing 0.9.6 Original commit message from CVS: releasing 0.9.6 2005-11-23 19:56:31 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: releasing 0.9.6 Original commit message from CVS: releasing 0.9.6 2005-11-23 19:14:07 +0000 Thomas Vander Stichele * docs/plugins/inspect/plugin-cutter.xml: adding cutter Original commit message from CVS: adding cutter 2005-11-23 19:05:29 +0000 Thomas Vander Stichele * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2005-11-23 16:49:16 +0000 Jan Schmidt gst/debug/gstnavigationtest.c: Oops, initialise the framerate GValue Original commit message from CVS: * gst/debug/gstnavigationtest.c: (gst_navigationtest_init): Oops, initialise the framerate GValue 2005-11-23 15:50:51 +0000 Julien Moutte VideoFilter inherits from Original commit message from CVS: 2005-11-23 Julien MOUTTE * ext/cairo/gsttimeoverlay.c: (gst_timeoverlay_update_font_height), (gst_timeoverlay_set_caps), (gst_timeoverlay_get_unit_size), (gst_timeoverlay_transform), (gst_timeoverlay_base_init), (gst_timeoverlay_class_init), (gst_timeoverlay_init), (gst_timeoverlay_get_type): * ext/cairo/gsttimeoverlay.h: * gst/debug/Makefile.am: * gst/debug/gstnavigationtest.c: (gst_navigationtest_handle_src_event), (gst_navigationtest_get_unit_size), (gst_navigationtest_set_caps), (gst_navigationtest_transform), (gst_navigationtest_change_state), (gst_navigationtest_base_init), (gst_navigationtest_class_init), (gst_navigationtest_init), (gst_navigationtest_get_type), (plugin_init): * gst/debug/gstnavigationtest.h: * gst/effectv/Makefile.am: * gst/effectv/gstaging.c: (gst_agingtv_set_caps), (gst_agingtv_get_unit_size), (gst_agingtv_transform), (gst_agingtv_base_init), (gst_agingtv_class_init), (gst_agingtv_init), (gst_agingtv_get_type): * gst/effectv/gstdice.c: (gst_dicetv_set_caps), (gst_dicetv_get_unit_size), (gst_dicetv_transform), (gst_dicetv_base_init), (gst_dicetv_class_init), (gst_dicetv_init), (gst_dicetv_get_type): * gst/effectv/gstedge.c: (gst_edgetv_set_caps), (gst_edgetv_get_unit_size), (gst_edgetv_transform), (gst_edgetv_base_init), (gst_edgetv_class_init), (gst_edgetv_init), (gst_edgetv_get_type): * gst/effectv/gsteffectv.c: * gst/effectv/gsteffectv.h: * gst/effectv/gstquark.c: (gst_quarktv_set_caps), (gst_quarktv_get_unit_size), (fastrand), (gst_quarktv_transform), (gst_quarktv_change_state), (gst_quarktv_base_init), (gst_quarktv_class_init), (gst_quarktv_init), (gst_quarktv_get_type): * gst/effectv/gstrev.c: (gst_revtv_set_caps), (gst_revtv_get_unit_size), (gst_revtv_transform), (gst_revtv_base_init), (gst_revtv_class_init), (gst_revtv_init), (gst_revtv_get_type): * gst/effectv/gstshagadelic.c: (gst_shagadelictv_set_caps), (gst_shagadelictv_get_unit_size), (gst_shagadelictv_transform), (gst_shagadelictv_base_init), (gst_shagadelictv_class_init), (gst_shagadelictv_init), (gst_shagadelictv_get_type): * gst/effectv/gstvertigo.c: (gst_vertigotv_set_caps), (gst_vertigotv_get_unit_size), (gst_vertigotv_transform), (gst_vertigotv_base_init), (gst_vertigotv_class_init), (gst_vertigotv_init), (gst_vertigotv_get_type): * gst/effectv/gstwarp.c: (gst_warptv_set_caps), (gst_warptv_get_unit_size), (gst_warptv_transform), (gst_warptv_base_init), (gst_warptv_class_init), (gst_warptv_init), (gst_warptv_get_type): * gst/videofilter/Makefile.am: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideofilter.c: (gst_videofilter_get_type), (gst_videofilter_class_init), (gst_videofilter_init): * gst/videofilter/gstvideofilter.h: * gst/videofilter/gstvideoflip.c: (gst_videoflip_set_caps), (gst_videoflip_transform_caps), (gst_videoflip_get_unit_size), (gst_videoflip_flip), (gst_videoflip_transform), (gst_videoflip_handle_src_event), (gst_videoflip_set_property), (gst_videoflip_base_init), (gst_videoflip_class_init), (gst_videoflip_init), (plugin_init), (gst_videoflip_get_type): * gst/videofilter/gstvideoflip.h: VideoFilter inherits from BaseTransform, it's just a place holder for now and every video effect plugin has been ported to use BaseTransform features directly. QuarkTV was fixed too (was broken), navigationtest works and best for the end, videoflip converts navigation events depending on flip method ! Fixes #320953 2005-11-23 14:22:18 +0000 Jan Schmidt Fixes for API changes Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_fixate): * ext/cairo/gsttextoverlay.c: (gst_text_overlay_collected): * gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_setcaps), (gst_goom_src_negotiate), (gst_goom_chain): * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): * sys/osxvideo/osxvideosink.m: Fixes for API changes 2005-11-23 12:19:06 +0000 Christian Schaller * gst-plugins-good.spec.in: add cutter to spec in Original commit message from CVS: add cutter to spec in 2005-11-23 11:57:51 +0000 Jan Schmidt gst/qtdemux/qtdemux.c: Convert to fractional framerates Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (gst_qtdemux_add_stream), (qtdemux_dump_mvhd), (qtdemux_parse_trak): Convert to fractional framerates 2005-11-22 23:58:14 +0000 Michael Smith ext/jpeg/: JPEG fractiony goodness. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_setcaps), (gst_jpeg_dec_chain), (gst_jpeg_dec_change_state): * ext/jpeg/gstjpegdec.h: * ext/jpeg/gstjpegenc.c: (gst_jpegenc_setcaps): * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps), (gst_smokeenc_resync): * ext/jpeg/gstsmokeenc.h: JPEG fractiony goodness. 2005-11-22 22:35:57 +0000 Michael Smith * ChangeLog: * gst/goom/filters.c: * gst/goom/graphic.h: Fix for #321430: unresolved symbols due to incorrect linkage on inline functions in goom. Original commit message from CVS: Fix for #321430: unresolved symbols due to incorrect linkage on inline functions in goom. Does not, however, fix the general crackheadedness of goom (global variables, oh my!); this should be moved to -bad. 2005-11-22 22:21:37 +0000 Jan Schmidt More fractional framerate conversions Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_init), (gst_text_overlay_setcaps), (gst_text_overlay_collected): * ext/cairo/gsttextoverlay.h: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link): * ext/gdk_pixbuf/gstgdkpixbuf.h: * ext/libpng/gstpngdec.c: (gst_pngdec_init), (gst_pngdec_caps_create_and_set): * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps): * gst/avi/gstavimux.c: (gst_avimux_init), (gst_avimux_vidsinkconnect): * gst/flx/gstflxdec.c: (gst_flxdec_chain): * gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_setcaps), (gst_goom_src_negotiate), (gst_goom_chain): * gst/goom/gstgoom.h: * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps): * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps): * sys/osxvideo/osxvideosink.h: * sys/osxvideo/osxvideosink.m: More fractional framerate conversions 2005-11-22 20:07:47 +0000 Jan Schmidt Convert to fractional framerates. Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_fixate): * gst/debug/gstnavigationtest.c: (gst_navigationtest_handle_src_event): * gst/videofilter/gstvideofilter.c: (gst_videofilter_format_get_structure), (gst_videofilter_setcaps), (gst_videofilter_init): * gst/videofilter/gstvideofilter.h: Convert to fractional framerates. 2005-11-22 18:11:58 +0000 Thomas Vander Stichele * ChangeLog: * ext/aalib/gstaasink.c: * ext/dv/gstdvdec.c: * ext/esd/esdmon.c: * ext/flac/gstflacenc.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/libcaca/gstcacasink.c: * ext/shout2/gstshout2.c: * gst/alpha/gstalpha.c: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstshaper.c: * gst/smpte/barboxwipes.c: * gst/smpte/gstsmpte.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/videomixer.c: fix up more enums Original commit message from CVS: fix up more enums 2005-11-22 17:39:11 +0000 Michael Smith gst/videomixer/videomixer.c: Fractional framerates, videomixer. Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps), (gst_videomixer_fill_queues), (gst_videomixer_update_queues): Fractional framerates, videomixer. 2005-11-22 17:15:25 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: doh Original commit message from CVS: doh 2005-11-22 17:09:36 +0000 Michael Smith ext/dv/: Fractional framerates for DV. Original commit message from CVS: * ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps): * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.c: (gst_dvdemux_init), (gst_dvdemux_src_convert), (gst_dvdemux_sink_convert), (gst_dvdemux_demux_video), (gst_dvdemux_demux_frame), (gst_dvdemux_flush): * ext/dv/gstdvdemux.h: Fractional framerates for DV. 2005-11-22 17:04:38 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: fix up GValueEnum Original commit message from CVS: fix up GValueEnum 2005-11-22 14:44:26 +0000 Tim-Philipp Müller gst/autodetect/: Use gst_plugin_feature_list_free() to free feature list and in the case of autovideosink free the li... Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_find_best), (gst_auto_video_sink_detect): Use gst_plugin_feature_list_free() to free feature list and in the case of autovideosink free the list at all. Also miscellaneous cosmetic fixes. 2005-11-22 13:13:21 +0000 Thomas Vander Stichele gst/cutter/gstcutter.c: copy calculation code from level; remove use of some audio functions Original commit message from CVS: * gst/cutter/gstcutter.c: (gst_cutter_chain), (gst_cutter_set_property), (gst_cutter_get_caps): copy calculation code from level; remove use of some audio functions 2005-11-22 13:11:25 +0000 Thomas Vander Stichele * gst/level/gstlevel.c: various cosmetic fixes Original commit message from CVS: various cosmetic fixes 2005-11-22 12:48:10 +0000 Thomas Vander Stichele * gst/level/gstlevel.c: various cosmetic fixes Original commit message from CVS: various cosmetic fixes 2005-11-22 12:41:35 +0000 Thomas Vander Stichele * gst/level/gstlevel.c: various cosmetic fixes Original commit message from CVS: various cosmetic fixes 2005-11-22 12:39:29 +0000 Andy Wingo * ext/lame/gstlame.c: Update for gst_tag_setter API changes. Original commit message from CVS: 2005-11-22 Andy Wingo * Update for gst_tag_setter API changes. 2005-11-22 12:38:33 +0000 Andy Wingo * ChangeLog: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexenc.c: * gst/avi/gstavimux.c: Update for gst_tag_setter API changes. Original commit message from CVS: 2005-11-22 Andy Wingo * Update for gst_tag_setter API changes. 2005-11-22 11:57:51 +0000 Andy Wingo * gst/qtdemux/qtdemux.c: ext/faad/gstfaad.c (gst_faad_event) ext/ivorbis/vorbisfile.c (gst_ivorbisfile_loop) gst/qtdemux/qtdemux.c (gst_qtdemu... Original commit message from CVS: 2005-11-22 Andy Wingo * ext/faad/gstfaad.c (gst_faad_event) * ext/ivorbis/vorbisfile.c (gst_ivorbisfile_loop) * gst/qtdemux/qtdemux.c (gst_qtdemux_loop_header) * gst/speed/gstspeed.c (speed_sink_event) * gst/tta/gstttaparse.c (gst_tta_parse_src_event) (gst_tta_parse_parse_header): Run update-funcnames. 2005-11-22 11:53:34 +0000 Andy Wingo * ChangeLog: * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfvideosink.c: * ext/libpng/gstpngdec.c: * ext/speex/gstspeexdec.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautovideosink.c: * gst/avi/gstavidemux.c: * gst/goom/gstgoom.c: * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: ext/dv/gstdvdemux.c (gst_dvdemux_handle_sink_event) (gst_dvdemux_demux_frame) ext/flac/gstflacdec.c (gst_flacdec_writ... Original commit message from CVS: 2005-11-22 Andy Wingo * ext/dv/gstdvdemux.c (gst_dvdemux_handle_sink_event) (gst_dvdemux_demux_frame) * ext/flac/gstflacdec.c (gst_flacdec_write) * ext/flac/gstflacenc.c (gst_flacenc_seek_callback) (gst_flacenc_sink_event) * ext/gconf/gstgconfaudiosink.c (gst_gconf_audio_sink_init) * ext/gconf/gstgconfvideosink.c (gst_gconf_video_sink_init) * ext/libpng/gstpngdec.c (gst_pngdec_caps_create_and_set) * ext/speex/gstspeexdec.c (speex_dec_event, speex_dec_chain) * gst/auparse/gstauparse.c (gst_auparse_chain) * gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_init) * gst/autodetect/gstautovideosink.c (gst_auto_video_sink_init) * gst/avi/gstavidemux.c (gst_avi_demux_stream_header) (gst_avi_demux_handle_seek) * gst/goom/gstgoom.c (gst_goom_event) * gst/matroska/ebml-write.c (gst_ebml_write_seek) * gst/matroska/matroska-demux.c (gst_matroska_demux_handle_seek_event) (gst_matroska_demux_loop_stream_parse_id) * gst/wavenc/gstwavenc.c (gst_wavenc_stop_file) * gst/wavparse/gstwavparse.c (gst_wavparse_handle_seek) (gst_wavparse_stream_headers): Run update-funcnames. 2005-11-22 11:49:30 +0000 Edward Hervey URIHandler interface and element properties are now properly synchronized for DV1394src and UDPSrc Original commit message from CVS: * ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init), (gst_dv1394src_init), (gst_dv1394src_dispose), (gst_dv1394src_set_property), (gst_dv1394src_discover_avc_node), (gst_dv1394src_uri_set_uri): * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_update_uri), (gst_udpsrc_set_uri), (gst_udpsrc_set_property), (gst_udpsrc_uri_get_uri): URIHandler interface and element properties are now properly synchronized for DV1394src and UDPSrc 2005-11-22 11:36:04 +0000 Tim-Philipp Müller ext/: libgsttagedit has been renamed to libgsttag. Original commit message from CVS: * ext/flac/Makefile.am: * ext/speex/Makefile.am: libgsttagedit has been renamed to libgsttag. 2005-11-21 23:50:02 +0000 Edward Hervey ext/lame/gstlame.c: Don't take the stream lock Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_event): Don't take the stream lock 2005-11-21 20:11:59 +0000 Wim Taymans gst/rtsp/rtspconnection.c: Apply patch from Sebastien Cote to fix #319184. Original commit message from CVS: * gst/rtsp/rtspconnection.c: (read_body): Apply patch from Sebastien Cote to fix #319184. 2005-11-21 19:50:25 +0000 Thomas Vander Stichele port cutter Original commit message from CVS: * configure.ac: * gst/cutter/Makefile.am: * gst/cutter/gstcutter.c: (gst_cutter_class_init), (gst_cutter_init), (gst_cutter_message_new), (gst_cutter_chain), (gst_cutter_set_property), (gst_cutter_get_property), (plugin_init), (gst_cutter_get_caps): port cutter * gst/level/gstlevel.c: fix up plugin details 2005-11-21 18:09:02 +0000 Tim-Philipp Müller Update for stream lock API changes: don't take stream log in sink event handlers any longer and change GST_STREAM_LOC... Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event): * ext/flac/gstflacdec.c: (gst_flacdec_loop), (gst_flacdec_src_event): * ext/flac/gstflacenc.c: (gst_flacenc_sink_event): * ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event), (gst_signal_processor_getrange), (gst_signal_processor_chain): * gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek): * gst/flx/gstflxdec.c: (gst_flxdec_src_event_handler), (gst_flxdec_sink_event_handler): * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_seek_event): * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek): Update for stream lock API changes: don't take stream log in sink event handlers any longer and change GST_STREAM_LOCK to GST_PAD_STREAM_LOCK. Don't leak references in flxdec event functions. 2005-11-21 17:52:15 +0000 Michael Smith * gst/auparse/Makefile.am: * gst/auparse/gstauparse.h: Forgot to commit header file changes, Makefile.am changes. Oops. Original commit message from CVS: Forgot to commit header file changes, Makefile.am changes. Oops. 2005-11-21 17:49:21 +0000 Michael Smith * ChangeLog: * gst/auparse/gstauparse.c: gst_object_unref, not g_object_unref Original commit message from CVS: gst_object_unref, not g_object_unref 2005-11-21 17:37:41 +0000 Wim Taymans Fix for stream lock updates. Original commit message from CVS: * ext/faac/gstfaac.c: (gst_faac_sink_event): * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event): * gst/tta/gstttaparse.c: (gst_tta_parse_src_event): Fix for stream lock updates. 2005-11-21 17:23:46 +0000 Tim-Philipp Müller gst/wavparse/gstwavparse.c: Use GST_DEBUG_FUNCPTR; add debug message in pad activate function. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_init), (gst_wavparse_create_sourcepad), (gst_wavparse_sink_activate): Use GST_DEBUG_FUNCPTR; add debug message in pad activate function. 2005-11-21 17:18:01 +0000 Michael Smith gst/auparse/: Partially fix #161712. playbin still doesn't work on these files, (on the bug report, Andy says we aren... Original commit message from CVS: * gst/auparse/Makefile.am: * gst/auparse/gstauparse.c: (gst_auparse_class_init), (gst_auparse_init), (gst_auparse_dispose), (gst_auparse_chain), (gst_auparse_change_state): * gst/auparse/gstauparse.h: Partially fix #161712. playbin still doesn't work on these files, (on the bug report, Andy says we aren't typefinding it for some reason?) but at least auparse isn't totally busted like it was before. 2005-11-21 16:45:46 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: j@bootlab.org, #321903). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add DX50, DIVX and DIV3 fourccs (patch by j@bootlab.org, #321903). 2005-11-21 16:36:05 +0000 Andy Wingo *.*: Ran scripts/update-macros. Oh yes. Original commit message from CVS: 2005-11-21 Andy Wingo * *.h: * *.c: Ran scripts/update-macros. Oh yes. 2005-11-21 15:06:35 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Filler events are gone for now, comment out section generating them. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams): Filler events are gone for now, comment out section generating them. 2005-11-21 14:39:04 +0000 Tim-Philipp Müller Update for GST_FOURCC_FORMAT API change. Original commit message from CVS: * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_get_format_from_caps): * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_create): * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_type_get), (qtdemux_node_dump_foreach), (qtdemux_dump_hdlr), (qtdemux_dump_dref), (qtdemux_dump_stsd), (qtdemux_dump_dcom), (qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps): * sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps): * sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list), (gst_v4l2src_capture_init), (gst_v4l2src_get_size_limits): Update for GST_FOURCC_FORMAT API change. 2005-11-21 14:33:11 +0000 Jan Schmidt Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) Original commit message from CVS: * ext/audioresample/gstaudioresample.c: * ext/polyp/polypsink.c: (gst_polypsink_sink_fixate): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_fixate): * gst/modplug/gstmodplug.cc: * sys/glsink/glimagesink.c: (gst_glimagesink_fixate): * sys/v4l2/gstv4l2src.c: (gst_v4l2src_fixate): Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) 2005-11-21 14:31:05 +0000 Jan Schmidt Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_fixate): * ext/mikmod/gstmikmod.c: (gst_mikmod_srcfixate): * gst/goom/gstgoom.c: (gst_goom_src_negotiate): * sys/osxvideo/osxvideosink.m: Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027) 2005-11-21 13:38:24 +0000 Tim-Philipp Müller Fixes for GST_FOURCC_FORMAT API change. Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_setcaps): * gst/avi/gstavidemux.c: (gst_avi_demux_parse_file_header), (gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_header), (gst_avi_demux_stream_data): * gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps): * gst/wavenc/gstwavenc.c: (write_metadata): * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_adtl), (gst_wavparse_parse_file_header), (gst_wavparse_stream_headers): Fixes for GST_FOURCC_FORMAT API change. 2005-11-21 12:13:48 +0000 Tim-Philipp Müller Fix for collect pads API change. Also fix textoverlay state change function. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_finalize), (gst_text_overlay_init), (gst_text_overlay_text_pad_linked), (gst_text_overlay_text_pad_unlinked), (gst_text_overlay_pop_video), (gst_text_overlay_pop_text), (gst_text_overlay_collected), (gst_text_overlay_change_state): * gst/matroska/matroska-mux.c: (gst_matroska_mux_init), (gst_matroska_mux_reset), (gst_matroska_mux_request_new_pad), (gst_matroska_mux_best_pad), (gst_matroska_mux_change_state): * gst/smpte/gstsmpte.c: (gst_smpte_init), (gst_smpte_collected): * gst/videomixer/videomixer.c: (gst_videomixer_init), (gst_videomixer_request_new_pad), (gst_videomixer_fill_queues), (gst_videomixer_change_state): Fix for collect pads API change. Also fix textoverlay state change function. 2005-11-20 17:04:55 +0000 Julien Moutte gst/matroska/matroska-mux.c: Replace Original commit message from CVS: 2005-11-20 Julien MOUTTE * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Replace GST_PAD_IS_USABLE by something approaching it. 2005-11-20 16:43:32 +0000 Julien Moutte gst/matroska/matroska-mux.c: Fix for Original commit message from CVS: 2005-11-20 Julien MOUTTE * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Fix for API changes. * gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Fix for API changes, but also fix the code that was not checking return values from pad_push neither using pad_alloc_buffer. 2005-11-18 18:19:21 +0000 Edward Hervey ext/libpng/gstpngenc.c: Added debug category Original commit message from CVS: * ext/libpng/gstpngenc.c: (gst_pngenc_class_init), (gst_pngenc_chain): Added debug category Return GST_FLOW_UNEXPECTED when sending an EOS, so the whole pipeline goes to EOS. 2005-11-17 18:23:23 +0000 Edgard Lima * ChangeLog: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711depay.c: * gst/rtp/gstrtpg711enc.c: * gst/rtp/gstrtpg711enc.h: * gst/rtp/gstrtpg711pay.c: * gst/rtp/gstrtpg711pay.h: * gst/rtp/gstrtpspeexdec.c: * gst/rtp/gstrtpspeexdec.h: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexenc.c: * gst/rtp/gstrtpspeexenc.h: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpspeexpay.h: Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size. Original commit message from CVS: Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size. 2005-11-16 19:08:54 +0000 Wim Taymans check/elements/matroskamux.c: Fix leak in check. Original commit message from CVS: * check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad): Fix leak in check. 2005-11-16 17:00:32 +0000 Wim Taymans gst/flx/gstflxdec.c: Fix state change. Original commit message from CVS: * gst/flx/gstflxdec.c: (gst_flxdec_change_state): Fix state change. 2005-11-16 11:02:24 +0000 Andy Wingo * ChangeLog: * gst/udp/gstudpsrc.c: Move comment. Original commit message from CVS: (gst_udpsrc_create): Move comment. 2005-11-16 10:43:44 +0000 Andy Wingo gst/udp/gstudpsrc.c: Clean up with the boilerplate macro. Original commit message from CVS: 2005-11-16 Andy Wingo * gst/udp/gstudpsrc.c: Clean up with the boilerplate macro. 2005-11-15 19:41:21 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: When seeking, seek to closest index entry at or before the requested seek position, no... Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek): When seeking, seek to closest index entry at or before the requested seek position, not just the closest one (#321001). 2005-11-15 12:16:00 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Invert DIB images again (see #132341). Original commit message from CVS: * gst/avi/gstavidemux.c: (swap_line), (gst_avi_demux_invert), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data): Invert DIB images again (see #132341). 2005-11-14 02:13:35 +0000 Thomas Vander Stichele * ChangeLog: * common: * configure.ac: * ext/aalib/gstaasink.c: * ext/cairo/gstcairo.c: * ext/dv/gstdv.c: * ext/esd/gstesd.c: * ext/flac/gstflac.c: * ext/gconf/gstgconfelements.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/jpeg/gstjpeg.c: * ext/ladspa/gstladspa.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmng.c: * ext/libpng/gstpng.c: * ext/mikmod/gstmikmod.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttimeoverlay.c: * ext/raw1394/gst1394.c: * ext/speex/gstspeex.c: * gst/alpha/Makefile.am: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautodetect.c: * gst/avi/gstavi.c: * gst/cutter/gstcutter.c: * gst/debug/efence.c: * gst/debug/gstdebug.c: * gst/debug/gstnavigationtest.c: * gst/effectv/gsteffectv.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/law/alaw.c: * gst/law/mulaw.c: * gst/level/gstlevel.c: * gst/matroska/matroska.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipart.c: * gst/oldcore/gstelements.c: * gst/rtp/Makefile.am: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtp.c: * gst/rtsp/gstrtsp.c: * gst/smoothwave/gstsmoothwave.c: * gst/smpte/gstsmpte.c: * gst/udp/gstudp.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: * gst/videomixer/videomixer.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * sys/oss/gstossaudio.c: * sys/osxaudio/gstosxaudio.c: rework configure.ac; make asterisk rtp stuff compile on mingw Original commit message from CVS: rework configure.ac; make asterisk rtp stuff compile on mingw 2005-11-12 13:31:56 +0000 Edward Hervey ext/jpeg/gstjpegdec.c: Only GST_DEBUG() information on the valid components. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain): Only GST_DEBUG() information on the valid components. 2005-11-11 19:34:50 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: back to head Original commit message from CVS: back to head === release 0.9.5 === 2005-11-11 19:33:23 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: releasing 0.9.5 Original commit message from CVS: releasing 0.9.5 2005-11-11 18:33:21 +0000 Thomas Vander Stichele * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: Update .po files Original commit message from CVS: Update .po files 2005-11-11 16:48:58 +0000 Edward Hervey gst/avi/gstavidemux.*: Yeah, implement proper seeking. Exact seeking and segment seeking. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Yeah, implement proper seeking. Exact seeking and segment seeking. Still need to do some checks for segment_stop. 2005-11-11 15:17:44 +0000 Christian Schaller * gst-plugins-good.spec.in: fix Cairo entry Original commit message from CVS: fix Cairo entry 2005-11-10 12:34:26 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Add support for custom genre tags. Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/qtdemux/qtdemux.c: (qtdemux_parse_udta): Add support for custom genre tags. 2005-11-10 12:22:30 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: Don't try to ready buffer duration from buffer that we don't own any longer and that mi... Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data): Don't try to ready buffer duration from buffer that we don't own any longer and that might already have been unreffed. (#321136) 2005-11-09 21:35:29 +0000 Zeeshan Ali * ChangeLog: * gst/flx/gstflxdec.c: Attempting to optimize the code for embedded systems. Original commit message from CVS: Attempting to optimize the code for embedded systems. 2005-11-08 08:54:30 +0000 Tim-Philipp Müller sys/oss/gstosssink.c: Don't re-use already closed file descriptor. (#320920) Original commit message from CVS: Reviewed by: Tim-Philipp Müller * sys/oss/gstosssink.c: (gst_oss_sink_close): Don't re-use already closed file descriptor. (#320920) 2005-11-07 17:35:20 +0000 Tim-Philipp Müller sys/oss/gstosssink.*: Cache probed caps; fix debug output for SET_PARAM macros. Original commit message from CVS: * sys/oss/gstosssink.c: (gst_oss_sink_dispose), (gst_oss_sink_set_property), (gst_oss_sink_getcaps), (gst_oss_sink_prepare): * sys/oss/gstosssink.h: Cache probed caps; fix debug output for SET_PARAM macros. 2005-11-07 15:09:54 +0000 Tim-Philipp Müller ext/cairo/: Port cairo textoverlay plugin to 0.9. Add 'shaded-background' property and redo position. Doesn't handle ... Original commit message from CVS: * ext/cairo/Makefile.am: * ext/cairo/gstcairo.c: (plugin_init): * ext/cairo/gsttextoverlay.c: (gst_text_overlay_base_init), (gst_text_overlay_class_init), (gst_text_overlay_finalize), (gst_text_overlay_init), (gst_text_overlay_font_init), (gst_text_overlay_set_property), (gst_text_overlay_render_text), (gst_text_overlay_getcaps), (gst_text_overlay_setcaps), (gst_text_overlay_text_pad_linked), (gst_text_overlay_text_pad_unlinked), (gst_text_overlay_shade_y), (gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2), (gst_text_overlay_push_frame), (gst_text_overlay_pop_video), (gst_text_overlay_pop_text), (gst_text_overlay_collected), (gst_text_overlay_change_state): * ext/cairo/gsttextoverlay.h: Port cairo textoverlay plugin to 0.9. Add 'shaded-background' property and redo position. Doesn't handle upstream renegotiation yet though. 2005-11-07 10:31:32 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: No need to take the STREAM_LOCK in the loop function. Improve some debug messages. Don't leak ... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): No need to take the STREAM_LOCK in the loop function. Improve some debug messages. Don't leak pad names in debug messages. 2005-11-07 10:27:00 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Don't error out when the source pad isn't linked. Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_push_vorbis_codec_priv_data), (gst_matroska_demux_add_wvpk_header): Don't error out when the source pad isn't linked. 2005-11-02 19:42:38 +0000 Tim-Philipp Müller ext/gconf/: Fix state change functions here as well and set kid to NULL state before removing it. Original commit message from CVS: * ext/gconf/gstgconfaudiosink.c: (do_toggle_element), (gst_gconf_audio_sink_change_state): * ext/gconf/gstgconfvideosink.c: (do_toggle_element), (gst_gconf_video_sink_change_state): Fix state change functions here as well and set kid to NULL state before removing it. 2005-11-02 16:48:55 +0000 Thomas Vander Stichele * check/elements/matroskamux.c: * common: * tests/check/elements/matroskamux.c: sigh, static pad templates aren't refcounted properly Original commit message from CVS: sigh, static pad templates aren't refcounted properly 2005-11-01 16:14:25 +0000 Thomas Vander Stichele * check/elements/.gitignore: * gst/level/.gitignore: * tests/check/elements/.gitignore: ignore more Original commit message from CVS: ignore more 2005-11-01 15:15:44 +0000 Edward Hervey gst/wavenc/gstwavenc.c: Added proper event handlind, made downstream newsegment event use GST_FORMAT_BYTES (otherwise... Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file), (gst_wavenc_init), (gst_wavenc_event), (gst_wavenc_chain): Added proper event handlind, made downstream newsegment event use GST_FORMAT_BYTES (otherwise it's ignored), and don't set a duration of 0 for buffers otherwise they are discarded by GstBaseSink. GstWavEnc needs some serious loving, after going through the code I'm really wondering how this can stay in -good ... 2005-11-01 15:11:16 +0000 Thomas Vander Stichele Fix leaks and invalid memory access as reported by valgrind Original commit message from CVS: * check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad), (setup_matroskamux), (check_buffer_data), (GST_START_TEST): * gst/matroska/matroska-mux.c: (gst_matroska_mux_finalize), (gst_matroska_mux_reset), (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start), (gst_matroska_mux_write_data), (gst_matroska_mux_collected): Fix leaks and invalid memory access as reported by valgrind 2005-11-01 14:41:01 +0000 Thomas Vander Stichele * check/elements/matroskamux.c: * tests/check/elements/matroskamux.c: ... and add the missing file Original commit message from CVS: ... and add the missing file 2005-11-01 14:36:02 +0000 Michal Benes add a unit test for matroskamux fix the bugs that the unit test exposed Original commit message from CVS: Patch by: Michal Benes * check/Makefile.am: * gst/matroska/ebml-write.c: (gst_ebml_write_seek): * gst/matroska/matroska-mux.c: (gst_matroska_mux_handle_src_event), (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start): add a unit test for matroskamux fix the bugs that the unit test exposed 2005-11-01 14:34:22 +0000 Thomas Vander Stichele * gst/rtp/Makefile.am: fix Makefile.am Original commit message from CVS: fix Makefile.am 2005-11-01 12:39:16 +0000 Tim-Philipp Müller gst/autodetect/: Fix state change function and use GST_DEBUG_FUNCPTR in class_init. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_class_init), (gst_auto_audio_sink_change_state): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_class_init), (gst_auto_video_sink_change_state): Fix state change function and use GST_DEBUG_FUNCPTR in class_init. 2005-11-01 12:35:39 +0000 Tim-Philipp Müller gst/matroska/: Set timestamps on outgoing ebml headers as well, so that the element after matroskamux can get the tim... Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/matroska/ebml-write.c: (gst_ebml_write_new), (gst_ebml_write_reset), (gst_ebml_write_element_new): * gst/matroska/ebml-write.h: * gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data): Set timestamps on outgoing ebml headers as well, so that the element after matroskamux can get the timestamp already when reading the first ebml element and doesn't have to wait for the actual data buffer for that (#320308). 2005-10-31 22:08:52 +0000 Andy Wingo * ChangeLog: * gst/videomixer/videomixer.c: gst/videomixer/videomixer.c (gst_videomixer_pad_unlink) Original commit message from CVS: 2005-10-31 Andy Wingo * gst/videomixer/videomixer.c (gst_videomixer_pad_unlink) (gst_videomixer_pad_link): Kill some memleaks. (gst_videomixer_pad_get_property): Style fix. (gst_videomixer_pad_set_property): Style fix. (gst_videomixer_pad_init): Style fix. (gst_videomixer_update_queues): Kill memleak. (gst_videomixer_loop): Kill memleak. (gst_videomixer_collected): Kill memleak. 2005-10-31 19:08:27 +0000 Edgard Lima * ChangeLog: * gst/auparse/gstauparse.c: Just some cleanup. Original commit message from CVS: Just some cleanup. 2005-10-31 14:41:31 +0000 Edgard Lima * ChangeLog: * ext/speex/gstspeexenc.c: Add checks to GST_FLOW_NOT_LINKED for values returned from gst_pad_push. Original commit message from CVS: Add checks to GST_FLOW_NOT_LINKED for values returned from gst_pad_push. 2005-10-31 12:00:10 +0000 Zeeshan Ali * ChangeLog: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711depay.c: Payloader now sets some default caps on the srcpad if caps on the sinkpad are never set. This is important for the g7... Original commit message from CVS: Payloader now sets some default caps on the srcpad if caps on the sinkpad are never set. This is important for the g711 to work with burger's rtpbin element. 2005-10-28 19:19:40 +0000 Edgard Lima * ChangeLog: * common: * ext/speex/gstspeexenc.c: Add checks for return values from gst_pad_push and gst_pad_alloc_buffer. Original commit message from CVS: Add checks for return values from gst_pad_push and gst_pad_alloc_buffer. 2005-10-28 15:32:48 +0000 Tim-Philipp Müller gst/matroska/: Add SimpleBlock support to matroska demuxer and muxer (part of Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream), (gst_matroska_demux_parse_info), (gst_matroska_demux_parse_blockgroup_or_simpleblock), (gst_matroska_demux_parse_cluster): * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init), (gst_matroska_mux_init), (gst_matroska_mux_start), (gst_matroska_mux_create_buffer_header), (gst_matroska_mux_write_data), (gst_matroska_mux_set_property), (gst_matroska_mux_get_property): * gst/matroska/matroska-mux.h: Add SimpleBlock support to matroska demuxer and muxer (part of Matroska v2). (#319731) 2005-10-28 13:24:40 +0000 Wim Taymans ext/jpeg/gstjpegdec.*: Cleanups. Don't create caps for every chain. Original commit message from CVS: * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_chain), (gst_jpeg_dec_change_state): * ext/jpeg/gstjpegdec.h: Cleanups. Don't create caps for every chain. 2005-10-27 18:46:32 +0000 Flavio Oliveira * ChangeLog: * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: * gst/law/mulaw-encode.c: * gst/law/mulaw-encode.h: Fix to set timestamp on buffer, it was tested with RTP G711 elements. Original commit message from CVS: Fix to set timestamp on buffer, it was tested with RTP G711 elements. 2005-10-27 11:27:53 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.h: Remove got_redirect from class structure as well. Original commit message from CVS: * gst/qtdemux/qtdemux.h: Remove got_redirect from class structure as well. 2005-10-27 11:25:19 +0000 Tim-Philipp Müller gst/qtdemux/qtdemux.c: Remove 'got-redirect' signal and post element message on the bus instead. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init), (qtdemux_parse_tree): Remove 'got-redirect' signal and post element message on the bus instead. 2005-10-27 11:00:40 +0000 Wim Taymans sys/oss/gstosssrc.c: Set correct format on oss instead of a silly value. Original commit message from CVS: * sys/oss/gstosssrc.c: (gst_oss_src_prepare): Set correct format on oss instead of a silly value. 2005-10-27 09:52:08 +0000 Julien Moutte gst/videobox/gstvideobox.c: Use liboil for Original commit message from CVS: 2005-10-27 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_transform_caps), (gst_video_box_set_caps), (gst_video_box_get_unit_size), (gst_video_box_copy_plane_i420), (gst_video_box_i420), (gst_video_box_ayuv): Use liboil for I420 rendering as well, doesn't bring much for my platform. Might help on some other platforms. 2005-10-26 21:47:36 +0000 Zeeshan Ali * ChangeLog: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmpay.c: Declaring the padtemplate correctly. Original commit message from CVS: Declaring the padtemplate correctly. 2005-10-26 20:28:32 +0000 Zeeshan Ali * ChangeLog: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711depay.c: * gst/rtp/gstrtpg711enc.c: * gst/rtp/gstrtpg711pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmpay.c: Setting the proper copyright notice. Original commit message from CVS: Setting the proper copyright notice. 2005-10-26 17:23:06 +0000 Julien Moutte gst/videobox/Makefile.am: Use liboil. Original commit message from CVS: 2005-10-26 Julien MOUTTE * gst/videobox/Makefile.am: Use liboil. * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_set_property), (gst_video_box_transform_caps), (gst_video_box_set_caps), (gst_video_box_get_unit_size), (gst_video_box_ayuv): Lot of optimization in AYUV rendering using liboil. Will dot the same to I420 border generation tomorrow. 2005-10-26 16:36:01 +0000 Thomas Vander Stichele * gst/rtp/Makefile.am: fix automake warnings Original commit message from CVS: fix automake warnings 2005-10-26 14:50:59 +0000 Zeeshan Ali * ChangeLog: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711dec.h: * gst/rtp/gstrtpg711depay.c: * gst/rtp/gstrtpg711depay.h: * gst/rtp/gstrtpg711enc.c: * gst/rtp/gstrtpg711pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmparse.h: * gst/rtp/gstrtpgsmpay.c: Hacked the G711 (de)payloader to try to make things right. rtpg711dec now inherits from the basertpdepayloader. Original commit message from CVS: Hacked the G711 (de)payloader to try to make things right. rtpg711dec now inherits from the basertpdepayloader. 2005-10-26 14:23:45 +0000 Julien Moutte gst/videobox/gstvideobox.c: Removing this forgotten debug. Original commit message from CVS: 2005-10-26 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_transform_caps), (gst_video_box_get_unit_size), (gst_video_box_ayuv): Removing this forgotten debug. 2005-10-26 14:08:49 +0000 Julien Moutte gst/videobox/gstvideobox.c: Fix the stride issue when boxing to AYUV. Original commit message from CVS: 2005-10-26 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_transform_caps), (gst_video_box_get_unit_size), (gst_video_box_ayuv): Fix the stride issue when boxing to AYUV. 2005-10-26 11:12:34 +0000 Tim-Philipp Müller sys/oss/: Actually use the 'oss' debug category we register. Original commit message from CVS: * sys/oss/gstossaudio.c: * sys/oss/gstossdmabuffer.c: * sys/oss/gstosshelper.c: * sys/oss/gstossmixer.c: * sys/oss/gstossmixerelement.c: * sys/oss/gstossmixertrack.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: Actually use the 'oss' debug category we register. 2005-10-26 10:38:18 +0000 Julien Moutte gst/videomixer/videomixer.c: Use gst_pad_get_parent and drop the ref that was added through that call. Original commit message from CVS: 2005-10-26 Julien MOUTTE * gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property), (gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps): Use gst_pad_get_parent and drop the ref that was added through that call. 2005-10-26 10:03:02 +0000 Thomas Vander Stichele * ChangeLog: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmpay.c: fix compilation Original commit message from CVS: fix compilation 2005-10-25 21:09:36 +0000 Flavio Oliveira * ChangeLog: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711depay.c: Just removed a couple of lines of weird code used during development/test time. Original commit message from CVS: Just removed a couple of lines of weird code used during development/test time. 2005-10-25 19:19:38 +0000 Flavio Oliveira * ChangeLog: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg711dec.c: * gst/rtp/gstrtpg711dec.h: * gst/rtp/gstrtpg711depay.c: * gst/rtp/gstrtpg711depay.h: * gst/rtp/gstrtpg711enc.c: * gst/rtp/gstrtpg711enc.h: * gst/rtp/gstrtpg711pay.c: * gst/rtp/gstrtpg711pay.h: G711 payloader and depayloader created by Edgard Lima (it supports mulaw and alaw (dec)encoders) Original commit message from CVS: G711 payloader and depayloader created by Edgard Lima (it supports mulaw and alaw (dec)encoders) 2005-10-25 17:55:19 +0000 Julien Moutte gst/videobox/gstvideobox.c: Doh ! I introduced wingo's bug again ! Sorry... Original commit message from CVS: 2005-10-25 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_transform_caps), (gst_video_box_get_unit_size): Doh ! I introduced wingo's bug again ! Sorry... 2005-10-25 16:02:38 +0000 Christian Schaller * ChangeLog: * gst/rtp/Makefile.am: add missing header files for disting Original commit message from CVS: add missing header files for disting 2005-10-25 15:07:02 +0000 Zeeshan Ali * ChangeLog: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmenc.h: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmparse.h: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgsmpay.h: Getting the GSM (de)payloader working and compatible with our plans for RTP. Original commit message from CVS: Getting the GSM (de)payloader working and compatible with our plans for RTP. 2005-10-25 13:03:04 +0000 Christian Schaller * gst/rtp/gstrtp.c: fix mistaken claim on GPL, its LGPL Original commit message from CVS: fix mistaken claim on GPL, its LGPL 2005-10-25 10:47:09 +0000 Julien Moutte ext/libpng/gstpngdec.c: Push a newsegment event, move some redundant code in a single place. Original commit message from CVS: 2005-10-25 Julien MOUTTE * ext/libpng/gstpngdec.c: (user_info_callback), (gst_pngdec_caps_create_and_set), (gst_pngdec_task): Push a newsegment event, move some redundant code in a single place. 2005-10-25 10:23:26 +0000 Julien Moutte ext/libpng/gstpngdec.c: Temporary hack to get correct colors order when we have a png image with alpha channel. Original commit message from CVS: 2005-10-25 Julien MOUTTE * ext/libpng/gstpngdec.c: (user_info_callback), (gst_pngdec_caps_create_and_set), (gst_pngdec_task): Temporary hack to get correct colors order when we have a png image with alpha channel. 2005-10-24 17:29:02 +0000 Edward Hervey ext/dv/gstdvdemux.c: Call gst_element_no_more_pads when there will be no more pads. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads): Call gst_element_no_more_pads when there will be no more pads. 2005-10-24 16:39:38 +0000 Wim Taymans gst/rtp/: Added two new payloaders, an RFC 2190 payloader for h263 and a payload convertor for an asterisk server. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_get_type), (gst_asteriskh263_base_init), (gst_asteriskh263_class_init), (gst_asteriskh263_init), (gst_asteriskh263_finalize), (gst_asteriskh263_chain), (gst_asteriskh263_set_property), (gst_asteriskh263_get_property), (gst_asteriskh263_change_state), (gst_asteriskh263_plugin_init): * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtph263enc.c: (gst_rtph263enc_get_type), (gst_rtph263enc_base_init), (gst_rtph263enc_class_init), (gst_rtph263enc_init), (gst_rtph263enc_finalize), (gst_rtph263enc_setcaps), (gst_rtph263enc_gobfiner), (gst_rtph263enc_flush), (gst_rtph263enc_handle_buffer), (gst_rtph263enc_plugin_init): * gst/rtp/gstrtph263enc.h: Added two new payloaders, an RFC 2190 payloader for h263 and a payload convertor for an asterisk server. 2005-10-24 15:57:17 +0000 Tim-Philipp Müller sys/oss/gstosssrc.c: Set bytes_per_sample correctly (is not always 4, but depends on width and number of channels). Original commit message from CVS: * sys/oss/gstosssrc.c: (gst_oss_src_prepare): Set bytes_per_sample correctly (is not always 4, but depends on width and number of channels). 2005-10-24 15:50:06 +0000 Tim-Philipp Müller ext/flac/gstflacenc.*: Fix seeking, so that flacenc can rewrite the header with the correct duration and amount of sa... Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flacenc_base_init), (gst_flacenc_init), (gst_flacenc_sink_setcaps), (gst_flacenc_seek_callback), (gst_flacenc_write_callback), (gst_flacenc_sink_event), (gst_flacenc_chain), (gst_flacenc_set_property), (gst_flacenc_get_property), (gst_flacenc_change_state): * ext/flac/gstflacenc.h: Fix seeking, so that flacenc can rewrite the header with the correct duration and amount of samples and all that at EOS; also set timestamps and granulepos on outgoing buffers; add debug category; fix state change function. 2005-10-24 13:46:09 +0000 Julien Moutte gst/videomixer/videomixer.c: Don't restrict video geometry from 16 to 4096. Original commit message from CVS: 2005-10-24 Julien MOUTTE * gst/videomixer/videomixer.c: Don't restrict video geometry from 16 to 4096. 2005-10-24 13:22:14 +0000 Julien Moutte gst/videobox/gstvideobox.c: Fix caps negotiation correctly, add debugging category. Original commit message from CVS: 2005-10-24 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_transform_caps), (gst_video_box_get_unit_size): Fix caps negotiation correctly, add debugging category. 2005-10-24 13:02:47 +0000 Christian Schaller * ChangeLog: * configure.ac: port over plugin listing from base Original commit message from CVS: port over plugin listing from base 2005-10-24 08:59:24 +0000 Julien Moutte ext/libpng/gstpngdec.c: Don't use fixed caps on a sink pad. Original commit message from CVS: 2005-10-24 Julien MOUTTE * ext/libpng/gstpngdec.c: (gst_pngdec_init): Don't use fixed caps on a sink pad. 2005-10-23 23:05:59 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: * docs/upload.mak: back to HEAD Original commit message from CVS: back to HEAD === release 0.9.4 === 2005-10-23 22:43:08 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * RELEASE: * configure.ac: * docs/Makefile.am: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavparse.xml: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: releasing 0.9.4 Original commit message from CVS: releasing 0.9.4 2005-10-23 11:07:10 +0000 Thomas Vander Stichele * ext/libpng/gstpngdec.c: * gst/wavparse/gstwavparse.c: * po/POTFILES.in: STOPPED->FAILED Original commit message from CVS: STOPPED->FAILED 2005-10-21 17:00:58 +0000 Tim-Philipp Müller ext/speex/gstspeexenc.c: Add position and duration query, fix query type function. Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speexenc_get_query_types), (gst_speexenc_src_query): Add position and duration query, fix query type function. * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream), (gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps): Let's not set non-fixed caps on source pads. 2005-10-21 16:15:57 +0000 Wim Taymans Set correct stream_time in newsegment event. avi can also handle a duration query now. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame): * gst/avi/gstavidemux.c: (gst_avi_demux_get_src_query_types), (gst_avi_demux_handle_seek): Set correct stream_time in newsegment event. avi can also handle a duration query now. 2005-10-21 10:06:40 +0000 Christian Schaller * gst-plugins-good.spec.in: update for latest additions Original commit message from CVS: update for latest additions 2005-10-20 19:14:27 +0000 Tim-Philipp Müller gst/matroska/matroska-demux.c: Fix duration query; fix basetime in newsegment event after seek; fix duration in initi... Original commit message from CVS: * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_src_query), (gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop_stream_parse_id): Fix duration query; fix basetime in newsegment event after seek; fix duration in initial newsegment event. * gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start): Extract number of channels and samplerate from vorbis headers; add some debug messages when querying the durations of the input streams. 2005-10-20 11:50:53 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Set stream time correctly in newsegment. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event): Set stream time correctly in newsegment. 2005-10-20 11:39:40 +0000 Wim Taymans gst/avi/gstavidemux.c: Correctly fill in the stream time. Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek): Correctly fill in the stream time. 2005-10-19 20:48:24 +0000 Thomas Vander Stichele * ChangeLog: * check/elements/level.c: * gst/level/gstlevel.c: * gst/level/level-example.c: * tests/check/elements/level.c: use ELEMENT messages instead Original commit message from CVS: use ELEMENT messages instead 2005-10-19 15:58:00 +0000 Wim Taymans gst/: API change fix. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types), (gst_qtdemux_handle_src_query): * gst/speed/gstspeed.c: (speed_get_query_types), (speed_src_query): * gst/tta/gstttaparse.c: (gst_tta_parse_src_event), (gst_tta_parse_get_query_types), (gst_tta_parse_query): API change fix. 2005-10-19 15:57:04 +0000 Wim Taymans API change fix. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_get_src_query_types), (gst_dvdemux_src_query): * ext/flac/gstflacdec.c: (gst_flacdec_length), (gst_flacdec_src_query): * ext/raw1394/gstdv1394src.c: (gst_dv1394src_query): * ext/speex/gstspeexdec.c: (speex_dec_src_query): * gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query): * gst/debug/gstnavseek.c: (gst_navseek_seek): * gst/debug/progressreport.c: (gst_progress_report_report): * gst/matroska/ebml-read.c: (gst_ebml_read_get_length): * gst/matroska/matroska-demux.c: (gst_matroska_demux_handle_src_query): * gst/matroska/matroska-mux.c: (gst_matroska_mux_start): * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_pad_convert), (gst_wavparse_pad_query), (gst_wavparse_srcpad_event): API change fix. 2005-10-19 10:57:46 +0000 Tim-Philipp Müller gst/goom/: Make inline functions either 'static inline' or 'extern inline', otherwise the Forte compiler apparently w... Original commit message from CVS: * gst/goom/filters.c: * gst/goom/graphic.h: * gst/goom/lines.c: Make inline functions either 'static inline' or 'extern inline', otherwise the Forte compiler apparently won't inline them (#317300). 2005-10-18 22:50:11 +0000 Julien Moutte ext/libpng/gstpngdec.c: forgot the buffer unref in pull. Original commit message from CVS: 2005-10-19 Julien MOUTTE * ext/libpng/gstpngdec.c: forgot the buffer unref in pull. 2005-10-18 22:44:11 +0000 Julien Moutte ext/libpng/gstpngdec.*: Complete rewrite of pngdec. It's now very nice and handle push/pull based model. if you have ... Original commit message from CVS: 2005-10-19 Julien MOUTTE * ext/libpng/gstpngdec.c: (gst_pngdec_class_init), (gst_pngdec_init), (user_error_fn), (user_warning_fn), (user_info_callback), (user_endrow_callback), (user_end_callback), (user_read_data), (gst_pngdec_caps_create_and_set), (gst_pngdec_task), (gst_pngdec_chain), (gst_pngdec_sink_event), (gst_pngdec_libpng_clear), (gst_pngdec_libpng_init), (gst_pngdec_change_state), (gst_pngdec_sink_activate_push), (gst_pngdec_sink_activate_pull), (gst_pngdec_sink_activate): * ext/libpng/gstpngdec.h: Complete rewrite of pngdec. It's now very nice and handle push/pull based model. if you have filesrc connected to it, it will do random access to load the png file. If you have a network source that can't do _getrange, it does progressive loading through the chain function. * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps), (transform_rgb), (transform_bgr): Fix caps negotiation correctly thanks to Master Wim Taymans ;-) 2005-10-18 18:12:31 +0000 Tim-Philipp Müller gst/matroska/: Ported matroska demuxer to 0.9. Original commit message from CVS: * gst/matroska/Makefile.am: * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: * gst/matroska/matroska.c: (plugin_init): Ported matroska demuxer to 0.9. 2005-10-18 18:06:14 +0000 Tim-Philipp Müller gst/matroska/matroska-mux.c: Fix mpeg4 input handling (#318847); also, while we're at it, fix media type for Motion-J... Original commit message from CVS: Reviewed by: Tim-Philipp Müller * gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps), (gst_matroska_mux_audio_pad_setcaps): Fix mpeg4 input handling (#318847); also, while we're at it, fix media type for Motion-JPEG: should be image/jpeg. 2005-10-18 13:21:18 +0000 Wim Taymans gst/wavparse/gstwavparse.c: Fix for segment-start/stop API change. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event): Fix for segment-start/stop API change. 2005-10-17 17:18:56 +0000 Julien Moutte gst/alpha/gstalphacolor.c: Handle caps negotiation in a better way. Original commit message from CVS: 2005-10-17 Julien MOUTTE * gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps), (transform_rgb), (transform_bgr): Handle caps negotiation in a better way. 2005-10-17 16:59:20 +0000 Julien Moutte gst/videobox/gstvideobox.c: Fix caps nego some more to get Original commit message from CVS: 2005-10-17 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_transform_caps), (gst_video_box_get_unit_size): Fix caps nego some more to get AYUV output declared in transform_caps. 2005-10-17 15:23:24 +0000 Julien Moutte ext/libpng/gstpngdec.c: We use fixed caps. Original commit message from CVS: 2005-10-17 Julien MOUTTE * ext/libpng/gstpngdec.c: (gst_pngdec_init): We use fixed caps. 2005-10-17 15:14:29 +0000 Julien Moutte gst/videobox/gstvideobox.c: Fix wrong size calculations and implement get_unit_size correctly. Original commit message from CVS: 2005-10-17 Julien MOUTTE * gst/videobox/gstvideobox.c: (gst_video_box_transform_caps), (gst_video_box_get_unit_size): Fix wrong size calculations and implement get_unit_size correctly. 2005-10-17 14:56:12 +0000 Tim-Philipp Müller configure.ac: Enable flx plugin. Original commit message from CVS: * configure.ac: Enable flx plugin. * gst/flx/gstflxdec.c: (flx_decode_chunks): Fix gcc4 signedness issue. 2005-10-17 08:46:30 +0000 Julien Moutte configure.ac: Adding videomixer. Original commit message from CVS: 2005-10-17 Julien MOUTTE * configure.ac: Adding videomixer. * ext/libpng/gstpngdec.c: (gst_pngdec_class_init), (user_read_data), (gst_pngdec_chain): More debugging. * gst/alpha/Makefile.am: Adding alphacolor * gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init), (gst_alpha_color_class_init), (gst_alpha_color_init), (gst_alpha_color_transform_caps), (gst_alpha_color_set_caps), (transform_rgb), (transform_bgr), (gst_alpha_color_transform_ip), (plugin_init): Ported to 0.9 using in place base tranform. * gst/videomixer/Makefile.am: * gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type), (gst_videomixer_pad_class_init), (gst_videomixer_pad_sink_setcaps), (gst_videomixer_pad_link), (gst_videomixer_pad_unlink), (gst_videomixer_pad_init), (gst_videomixer_class_init), (gst_videomixer_init), (gst_videomixer_getcaps), (gst_videomixer_request_new_pad), (gst_videomixer_fill_queues), (gst_videomixer_blend_buffers), (gst_videomixer_update_queues), (gst_videomixer_collected), (gst_videomixer_change_state): Ported to 0.9 using collectpads. 2005-10-16 21:19:44 +0000 Zeeshan Ali * ChangeLog: * common: * configure.ac: * gst/flx/Makefile.am: * gst/flx/gstflxdec.c: * gst/flx/gstflxdec.h: flx plugin ported to 0.9 Original commit message from CVS: flx plugin ported to 0.9 2005-10-16 14:33:05 +0000 Thomas Vander Stichele * ChangeLog: * ext/shout2/gstshout2.c: use gst_version_string Original commit message from CVS: use gst_version_string 2005-10-16 13:17:11 +0000 Andy Wingo configure.ac: GLIB_CHECK. Original commit message from CVS: 2005-10-16 Andy Wingo * configure.ac: GLIB_CHECK. 2005-10-15 16:48:55 +0000 Julien Moutte ext/libpng/: Ported pngdec to 0.9 Original commit message from CVS: 2005-10-15 Julien MOUTTE * ext/libpng/Makefile.am: * ext/libpng/gstpng.c: (plugin_init): * ext/libpng/gstpngdec.c: (gst_pngdec_class_init), (gst_pngdec_init), (user_read_data), (gst_pngdec_chain): * ext/libpng/gstpngdec.h: Ported pngdec to 0.9 2005-10-14 12:43:30 +0000 Tim-Philipp Müller Port matroska muxer to 0.9 (#318847). Original commit message from CVS: Reviewed by: Tim-Philipp Müller * configure.ac: * gst/matroska/Makefile.am: * gst/matroska/ebml-ids.h: * gst/matroska/ebml-write.c: * gst/matroska/ebml-write.h: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: * gst/matroska/matroska.c: (plugin_init): Port matroska muxer to 0.9 (#318847). 2005-10-13 18:59:35 +0000 Tim-Philipp Müller ext/speex/gstspeexenc.c: Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE; use GST_READ_UINT32_LE() and fr... Original commit message from CVS: * ext/speex/gstspeexenc.c: (gst_speexenc_get_tag_value), (comment_init), (comment_add): Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE; use GST_READ_UINT32_LE() and friends rather than the private implementation of those same macros. 2005-10-13 16:01:35 +0000 Thomas Vander Stichele * ext/cairo/Makefile.am: fix dist Original commit message from CVS: fix dist 2005-10-13 15:28:01 +0000 Stefan Kost examples/stats/mp2ogg.c: more typo fixes Original commit message from CVS: * examples/stats/mp2ogg.c: more typo fixes 2005-10-12 14:30:36 +0000 Stefan Kost renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/a52dec/gsta52dec.c: (gst_a52dec_init): * ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_is_open), (dvdnavsrc_set_property), (dvdnavsrc_open), (dvdnavsrc_close), (dvdnavsrc_event), (dvdnavsrc_convert), (dvdnavsrc_query): * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_set_property), (dvdreadsrc_srcpad_query), (dvdreadsrc_get), (dvdreadsrc_open_file), (dvdreadsrc_close_file): * ext/dvdread/dvdreadsrc.h: * ext/lame/gstlame.h: * gst/asfdemux/gstasfdemux.c: (gst_asf_demux_init): * gst/asfdemux/gstasfmux.c: (gst_asfmux_init): * gst/iec958/ac3iec.h: * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_init): * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_init): * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_init): * gst/mpegstream/gstrfc2250enc.c: (gst_rfc2250_enc_init): * gst/synaesthesia/gstsynaesthesia.c: (gst_synaesthesia_init): renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition 2005-10-12 14:29:55 +0000 Stefan Kost renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio), (gst_artsdsink_close_audio), (gst_artsdsink_change_state): * ext/artsd/gstartsdsink.h: * ext/audiofile/gstafparse.c: (gst_afparse_open_file), (gst_afparse_close_file): * ext/audiofile/gstafparse.h: * ext/audiofile/gstafsink.c: (gst_afsink_open_file), (gst_afsink_close_file), (gst_afsink_chain), (gst_afsink_change_state): * ext/audiofile/gstafsink.h: * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file), (gst_afsrc_close_file), (gst_afsrc_change_state): * ext/audiofile/gstafsrc.h: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init): * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init): * ext/dts/gstdtsdec.c: (gst_dtsdec_init): * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: (gst_jack_bin_init), (gst_jack_bin_change_state): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init): * ext/nas/nassink.c: (gst_nassink_open_audio), (gst_nassink_close_audio), (gst_nassink_change_state): * ext/nas/nassink.h: * ext/polyp/polypsink.c: (gst_polypsink_init): * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state): * ext/sdl/sdlvideosink.h: * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): * ext/sndfile/gstsf.c: (gst_sf_set_property), (gst_sf_change_state), (gst_sf_release_request_pad), (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop): * ext/sndfile/gstsf.h: * ext/swfdec/gstswfdec.c: (gst_swfdec_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init): * gst/apetag/apedemux.c: (gst_ape_demux_init): * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init): * gst/festival/gstfestival.c: (gst_festival_change_state): * gst/festival/gstfestival.h: * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init): * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init), (gst_multifilesink_set_location), (gst_multifilesink_open_file), (gst_multifilesink_close_file), (gst_multifilesink_next_file), (gst_multifilesink_pad_query), (gst_multifilesink_handle_event), (gst_multifilesink_chain), (gst_multifilesink_change_state): * gst/multifilesink/gstmultifilesink.h: * gst/videodrop/gstvideodrop.c: (gst_videodrop_init): * sys/cdrom/gstcdplayer.c: (cdplayer_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init), (dxr3audiosink_open), (dxr3audiosink_close), (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3), (dxr3audiosink_change_state): * sys/dxr3/dxr3audiosink.h: * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open), (dxr3spusink_close), (dxr3spusink_chain), (dxr3spusink_change_state): * sys/dxr3/dxr3spusink.h: * sys/dxr3/dxr3videosink.c: (dxr3videosink_init), (dxr3videosink_open), (dxr3videosink_close), (dxr3videosink_write_data), (dxr3videosink_change_state): * sys/dxr3/dxr3videosink.h: * sys/glsink/glimagesink.c: (gst_glimagesink_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state), (gst_qcamsrc_open), (gst_qcamsrc_close): * sys/qcam/gstqcamsrc.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get), (gst_vcdsrc_open_file), (gst_vcdsrc_close_file), (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate): * sys/vcd/vcdsrc.h: renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition 2005-10-12 14:29:43 +0000 Stefan Kost renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/esd/esdmon.c: (gst_esdmon_open_audio), (gst_esdmon_close_audio), (gst_esdmon_change_state): * ext/esd/esdmon.h: * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init): * ext/pango/gsttextoverlay.c: (gst_textoverlay_init): * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init): * gst/avi/gstavimux.c: (gst_avimux_init): * gst/matroska/matroska-demux.c: (gst_matroska_demux_init): * gst/multipart/multipartdemux.c: (gst_multipart_demux_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_init): * gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init), (gst_multifilesrc_get), (gst_multifilesrc_open_file), (gst_multifilesrc_close_file), (gst_multifilesrc_change_state): * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.c: (gst_pipefilter_init), (gst_pipefilter_open_file), (gst_pipefilter_close_file), (gst_pipefilter_change_state): * gst/oldcore/gstpipefilter.h: * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): * gst/videomixer/videomixer.c: (gst_videomixer_init): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_init): * sys/osxaudio/gstosxaudiosink.h: * sys/osxaudio/gstosxaudiosrc.h: renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition 2005-10-12 03:14:57 +0000 Thomas Vander Stichele * ext/Makefile.am: dist cairo Original commit message from CVS: dist cairo 2005-10-12 03:12:57 +0000 Thomas Vander Stichele ext/: update of cairo-based timeoverlay to 1.0 Cairo API doesn't work yet for resizing of output sink Original commit message from CVS: * ext/Makefile.am: * ext/cairo/Makefile.am: * ext/cairo/gstcairo.c: (plugin_init): * ext/cairo/gsttextoverlay.c: (gst_textoverlay_change_state): * ext/cairo/gsttimeoverlay.c: (gst_timeoverlay_update_font_height), (gst_timeoverlay_setup), (gst_timeoverlay_planar411): * ext/cairo/gsttimeoverlay.h: update of cairo-based timeoverlay to 1.0 Cairo API doesn't work yet for resizing of output sink 2005-10-12 03:07:26 +0000 Thomas Vander Stichele * configure.ac: don't build checks if we don't have check Original commit message from CVS: don't build checks if we don't have check 2005-10-12 03:03:27 +0000 Thomas Vander Stichele * Makefile.am: * common: don't build checks if we don't have gstcheck Original commit message from CVS: don't build checks if we don't have gstcheck 2005-10-11 17:38:29 +0000 Wim Taymans ext/speex/gstspeexdec.c: newsegment API fix. Original commit message from CVS: * ext/speex/gstspeexdec.c: (speex_dec_event), (speex_dec_chain): newsegment API fix. 2005-10-11 16:34:36 +0000 Wim Taymans gst/: newsegment API update. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header): * gst/tta/gstttaparse.c: (gst_tta_parse_src_event), (gst_tta_parse_parse_header): newsegment API update. 2005-10-11 16:33:08 +0000 Wim Taymans newsegment API update. Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event), (gst_dvdemux_demux_frame): * ext/flac/gstflacdec.c: (gst_flacdec_write): * gst/auparse/gstauparse.c: (gst_auparse_chain): * gst/avi/gstavidemux.c: (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek): * gst/goom/gstgoom.c: (gst_goom_event): * gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file): * gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_pad_convert), (gst_wavparse_srcpad_event): newsegment API update. 2005-10-11 10:07:35 +0000 Andy Wingo ext/speex/gstspeexenc.c: Signedness cleanups. Original commit message from CVS: 2005-10-11 Andy Wingo * ext/speex/gstspeexenc.c: Signedness cleanups. 2005-10-10 19:57:40 +0000 Edgard Lima * ChangeLog: * PORTED_09: * ext/speex/Makefile.am: * ext/speex/gstspeex.c: * ext/speex/gstspeexenc.c: Speexenc ported to 0.9. Original commit message from CVS: Speexenc ported to 0.9. 2005-10-10 14:16:21 +0000 Wim Taymans sys/oss/: Cleanups, make device configurable in the sink, handle and report errors. Original commit message from CVS: * sys/oss/gstosssink.c: (gst_oss_sink_class_init), (gst_oss_sink_init), (gst_oss_sink_set_property), (gst_oss_sink_get_property), (gst_oss_sink_open), (gst_oss_sink_prepare), (gst_oss_sink_reset): * sys/oss/gstosssink.h: * sys/oss/gstosssrc.c: (gst_oss_src_class_init), (gst_oss_src_set_property), (gst_oss_src_init), (gst_oss_src_open), (gst_oss_src_prepare): Cleanups, make device configurable in the sink, handle and report errors. 2005-10-10 12:31:07 +0000 Wim Taymans ext/gconf/: Make sure element is NULL before removing from the bin. Original commit message from CVS: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset): Make sure element is NULL before removing from the bin. 2005-10-07 16:28:24 +0000 Andy Wingo * ChangeLog: * ext/raw1394/gstdv1394src.c: Don't unref the message. Original commit message from CVS: (gst_dv1394src_bus_reset): Don't unref the message. 2005-10-07 16:22:59 +0000 Andy Wingo * ChangeLog: * ext/raw1394/gstdv1394src.c: Post a message when the cable is unplugged. Original commit message from CVS: (gst_dv1394src_bus_reset): Post a message when the cable is unplugged. (gst_dv1394src_create, gst_dv1394src_unlock): Remove some prints. 2005-10-07 15:24:24 +0000 Andy Wingo ext/raw1394/gstdv1394src.c: Make interruptible, so it won't block forever in a read(). Original commit message from CVS: 2005-10-07 Andy Wingo * ext/raw1394/gstdv1394src.c: Make interruptible, so it won't block forever in a read(). 2005-10-07 13:17:53 +0000 Andy Wingo ext/raw1394/gstdv1394src.c: Clean up for style before doing some hacking. The only change should be that the state ch... Original commit message from CVS: 2005-10-07 Andy Wingo * ext/raw1394/gstdv1394src.c: Clean up for style before doing some hacking. The only change should be that the state change stuff was put into basesrc's start() and stop() routines, which coalesces some steps. 2005-10-07 11:30:41 +0000 Tim-Philipp Müller configure.ac: Add check for mmap Original commit message from CVS: * configure.ac: Add check for mmap * gst/debug/Makefile.am: Only compile efence plugin on systems that have mmap. 2005-10-05 16:36:57 +0000 Christian Schaller * gst-plugins-good.spec.in: add latest files Original commit message from CVS: add latest files 2005-10-05 11:38:29 +0000 Tim-Philipp Müller gst/debug/: Port progressreport, navseek, navigationtest, testsink and breakmydata. Original commit message from CVS: * gst/debug/Makefile.am: * gst/debug/breakmydata.c: * gst/debug/gstdebug.c: * gst/debug/gstnavigationtest.c: * gst/debug/gstnavseek.c: * gst/debug/gstnavseek.h: * gst/debug/progressreport.c: * gst/debug/testplugin.c: Port progressreport, navseek, navigationtest, testsink and breakmydata. 2005-10-05 11:15:23 +0000 Edward Hervey ext/dv/gstdvdemux.c: Fixes for better conversion Original commit message from CVS: * ext/dv/gstdvdemux.c: (gst_dvdemux_src_convert), (gst_dvdemux_src_query): Fixes for better conversion 2005-10-04 17:58:40 +0000 Michael Smith gst/autodetect/: Set state of elements to NULL before removing from bins. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset), (gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset), (gst_auto_video_sink_find_best), (gst_auto_video_sink_detect): Set state of elements to NULL before removing from bins. Set state of test element to NULL if we failed to move it to READY 2005-10-04 17:44:43 +0000 Edward Hervey ext/dv/: Added DEFAULT <==> BYTES, TIME conversions on srcpad, Original commit message from CVS: * ext/dv/Makefile.am: * ext/dv/gstdvdemux.c: (gst_dvdemux_src_query), (gst_dvdemux_src_conver): Added DEFAULT <==> BYTES, TIME conversions on srcpad, Corrected the query function for position so it doesn't forget what format was asked, and calls the conversion functions on the correct pad. 2005-10-03 17:59:18 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: back to head Original commit message from CVS: back to head === release 0.9.3 === 2005-10-03 17:48:57 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * README: * configure.ac: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: release time Original commit message from CVS: release time 2005-10-02 23:08:35 +0000 Andy Wingo ext/flac/gstflacdec.c (gst_flacdec_write): Deal with pad_alloc error returns. Original commit message from CVS: 2005-10-03 Andy Wingo * ext/flac/gstflacdec.c (gst_flacdec_write): Deal with pad_alloc error returns. 2005-10-02 15:33:14 +0000 Andy Wingo configure.ac (GST_PLUGIN_LDFLAGS): Change to be like -base. Original commit message from CVS: 2005-10-02 Andy Wingo * configure.ac (GST_PLUGIN_LDFLAGS): Change to be like -base. * ext/flac/gstflacenc.c: Ported to 0.9. * ext/flac/gstflacdec.c (gst_flacdec_loop): Handle errors better. * ext/flac/Makefile.am: Add the GST_PLUGINS_BASE cflags and libs, and link to gsttagedit. Enable flacenc. * ext/flac/gstflacdec.c: Re-enable tag reading. 2005-09-30 16:36:49 +0000 Wim Taymans gst/rtp/: Various class and caps fixes from Andre Magalhaes (andrunko) Original commit message from CVS: * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps): * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtph263penc.c: * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init), (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property): * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer): Various class and caps fixes from Andre Magalhaes (andrunko) 2005-09-29 13:08:41 +0000 Wim Taymans gst/level/level-example.c: Update for new bus API. Original commit message from CVS: * gst/level/level-example.c: (main): Update for new bus API. 2005-09-28 13:38:02 +0000 Wim Taymans gst/qtdemux/qtdemux.c: No need to take stream lock here. Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header): No need to take stream lock here. 2005-09-28 09:45:00 +0000 Tim-Philipp Müller configure.ac: Fix unexpanded autoconf macro GST_DOC, which has been renamed to GST_DOCBOOK_CHECK (see common/m4/gst-d... Original commit message from CVS: * configure.ac: Fix unexpanded autoconf macro GST_DOC, which has been renamed to GST_DOCBOOK_CHECK (see common/m4/gst-doc.m4) (#316202). 2005-09-27 15:12:45 +0000 Tim-Philipp Müller sys/oss/gstosssink.c: Fix playback of mono streams (bytes_per_sample should be set from the sample width and the numb... Original commit message from CVS: * sys/oss/gstosssink.c: (gst_oss_sink_prepare): Fix playback of mono streams (bytes_per_sample should be set from the sample width and the number of channels negotiated, and not just be set to 4) (#317338) 2005-09-26 14:59:10 +0000 Christian Schaller * gst-plugins-good.spec.in: add auparse to plugins list Original commit message from CVS: add auparse to plugins list 2005-09-26 14:42:09 +0000 Wim Taymans gst/rtp/gstrtpmpaenc.c: Set buffer duration correctly. Original commit message from CVS: * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush), (gst_rtpmpaenc_handle_buffer): Set buffer duration correctly. 2005-09-26 13:06:27 +0000 Tim-Philipp Müller gst/avi/gstavidemux.c: Don't crash when encountering a stream with an unknown fourcc or codec id. Instead, create a p... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_base_init), (gst_avi_demux_class_init), (gst_avi_demux_parse_stream), (gst_avi_demux_change_state): Don't crash when encountering a stream with an unknown fourcc or codec id. Instead, create a pad of type video/x-avi-unknown or audio/x-avi-unknown, which as a side-effect also results in less confusing error messages in players ('no decoder' vs. 'no streams'); minor fixes to state change function and class_init function. 2005-09-24 13:34:46 +0000 Thomas Vander Stichele * check/Makefile.am: * tests/check/Makefile.am: set up plugin paths properly Original commit message from CVS: set up plugin paths properly 2005-09-24 13:10:52 +0000 Wim Taymans gst/autodetect/: These are sinks. Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init): These are sinks. 2005-09-24 12:10:02 +0000 Thomas Vander Stichele check/elements/level.c: fix test for new GstClockTime use Original commit message from CVS: * check/elements/level.c: (GST_START_TEST): fix test for new GstClockTime use * gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps), (gst_level_transform_ip): * gst/level/gstlevel.h: fix up the decay peak, ensuring the decay peak is never lower than the peak for that interval 2005-09-23 18:23:04 +0000 Thomas Vander Stichele * ChangeLog: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-rtp.xml: * gst/level/gstlevel.c: updating docs Original commit message from CVS: updating docs 2005-09-23 18:15:51 +0000 Thomas Vander Stichele * ChangeLog: * Makefile.am: * check/elements/level.c: * common: * gst/level/Makefile.am: * gst/level/gstlevel.c: * gst/level/gstlevel.h: * gst/level/level-example.c: * tests/check/elements/level.c: convert to using GstClockTime for all time values, finally. Original commit message from CVS: convert to using GstClockTime for all time values, finally. 2005-09-23 15:01:00 +0000 Thomas Vander Stichele * gst/goom/Makefile.am: fix build of goom Original commit message from CVS: fix build of goom 2005-09-23 14:20:01 +0000 Thomas Vander Stichele * common: * gst/level/gstlevel.c: we handle more than two channels Original commit message from CVS: we handle more than two channels 2005-09-23 04:23:00 +0000 Thomas Vander Stichele * ChangeLog: * configure.ac: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/esd/Makefile.am: * ext/flac/Makefile.am: * ext/gconf/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/jpeg/Makefile.am: * ext/ladspa/Makefile.am: * ext/libcaca/Makefile.am: * ext/libmng/Makefile.am: * ext/libpng/Makefile.am: * ext/mikmod/Makefile.am: * ext/pango/Makefile.am: * ext/raw1394/Makefile.am: * ext/shout2/Makefile.am: * ext/speex/Makefile.am: * gst/alpha/Makefile.am: * gst/auparse/Makefile.am: * gst/auparse/gstauparse.c: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debug/Makefile.am: * gst/effectv/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/law/Makefile.am: * gst/matroska/Makefile.am: * gst/median/Makefile.am: * gst/monoscope/Makefile.am: * gst/multipart/Makefile.am: * gst/oldcore/Makefile.am: * gst/rtp/Makefile.am: * gst/rtsp/Makefile.am: * gst/smoothwave/Makefile.am: * gst/smpte/Makefile.am: * gst/videobox/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * sys/oss/Makefile.am: * sys/osxaudio/Makefile.am: fix build and use of GST_LIBS Original commit message from CVS: fix build and use of GST_LIBS 2005-09-22 22:38:48 +0000 Edgard Lima * ChangeLog: * PORTED_09: * configure.ac: * gst/auparse/gstauparse.c: * gst/auparse/gstauparse.h: Auparse ported to 0.9. Tested with filesrc ! auparse ! osssink and alsasink Original commit message from CVS: Auparse ported to 0.9. Tested with filesrc ! auparse ! osssink and alsasink 2005-09-22 14:13:36 +0000 Wim Taymans gst/rtp/: Use is_filled to both check MTU and max-ptime of base class. Original commit message from CVS: * gst/rtp/TODO: * gst/rtp/gstrtpdec.c: (gst_rtpdec_getcaps): * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init), (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property): * gst/rtp/gstrtpmp4venc.h: * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer): * gst/rtp/gstrtpmpaenc.h: Use is_filled to both check MTU and max-ptime of base class. 2005-09-22 11:28:23 +0000 Wim Taymans gst/rtp/gstrtpmp4venc.c: Don't fragment packets with multiple frames. Original commit message from CVS: * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init), (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property): Don't fragment packets with multiple frames. 2005-09-22 10:39:11 +0000 Wim Taymans gst/rtp/: Remove g_print. Original commit message from CVS: * gst/rtp/TODO: * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps): * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init), (gst_rtpmp4venc_init), (gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property), (gst_rtpmp4venc_get_property): * gst/rtp/gstrtpmp4venc.h: Remove g_print. Update TODO Make payload encoder a bit smarter and more correct with timestamps. Added option in payloader to include config string in-band. 2005-09-21 19:41:45 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: Strip spaces for key/value pairs. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send): Strip spaces for key/value pairs. 2005-09-21 17:53:26 +0000 Wim Taymans gst/rtsp/gstrtspsrc.c: More SDP parsing and caps setting. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send), (gst_rtspsrc_change_state): More SDP parsing and caps setting. Do NO_PREROLL differently. add pads only after negotiated. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_getcaps): Implement the getcaps function. 2005-09-21 17:50:29 +0000 Wim Taymans gst/rtp/gstrtpamrdec.c: Handle multiple AMr packets per payload. Handle CRC and parse ILL/ILP. Original commit message from CVS: * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain): Handle multiple AMr packets per payload. Handle CRC and parse ILL/ILP. * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps): Make caps params strings for easy SDP mapping. * gst/rtp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps): Handle capsnego better. * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps): * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_new_caps): Generate and parse config string in the caps. 2005-09-21 12:19:24 +0000 Wim Taymans gst/rtp/README: Update README Original commit message from CVS: * gst/rtp/README: Update README * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps): Make extra params as strings. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state), (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send): Make state change return NO_PREROLL as this is a live source. * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property): Don't unref old caps when NULL. 2005-09-20 17:35:11 +0000 Wim Taymans gst/rtsp/: Add URI handler. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type), (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_uri_get_type), (gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri), (gst_rtspsrc_uri_handler_init): * gst/rtsp/sdpmessage.c: (sdp_media_get_format): * gst/rtsp/sdpmessage.h: Add URI handler. Parse SDP and create caps. 2005-09-20 17:19:43 +0000 Christian Schaller * gst-plugins-good.spec.in: more spec file fixoring Original commit message from CVS: more spec file fixoring 2005-09-20 17:04:33 +0000 Christian Schaller * gst-plugins-good.spec.in: * gst-plugins.spec.in: fix spec files Original commit message from CVS: fix spec files 2005-09-20 10:51:51 +0000 Thomas Vander Stichele * gst/rtp/README: * gst/rtp/gstrtpamrdec.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrenc.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pdec.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263penc.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpmp4vdec.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4venc.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadec.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpaenc.c: * gst/rtp/gstrtpmpapay.c: don't use underscores Original commit message from CVS: don't use underscores 2005-09-20 07:30:31 +0000 Stefan Kost gst/alpha/gstalpha.c: fix element description Original commit message from CVS: * gst/alpha/gstalpha.c: fix element description 2005-09-19 17:57:06 +0000 Thomas Vander Stichele * docs/plugins/gst-plugins-good-plugins.prerequisites: prereqs as well Original commit message from CVS: prereqs as well 2005-09-19 17:53:42 +0000 Thomas Vander Stichele * docs/plugins/.gitignore: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.signals: commit result of scanobj step Original commit message from CVS: commit result of scanobj step 2005-09-19 17:03:55 +0000 Wim Taymans gst/rtp/gstrtph263pdec.c: Don't check payload for now. Original commit message from CVS: * gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_chain): Don't check payload for now. 2005-09-19 16:43:56 +0000 Thomas Vander Stichele * Makefile.am: add check-valgrind target Original commit message from CVS: add check-valgrind target 2005-09-19 16:26:30 +0000 Wim Taymans gst/wavparse/gstwavparse.*: Fix wavparse some more. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_parse_file_header), (gst_wavparse_stream_init), (gst_wavparse_handle_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data), (gst_wavparse_loop), (gst_wavparse_pad_convert), (gst_wavparse_pad_query), (gst_wavparse_srcpad_event), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: Fix wavparse some more. 2005-09-19 11:48:13 +0000 Wim Taymans check/elements/level.c: Fix for bus API change. Original commit message from CVS: * check/elements/level.c: (GST_START_TEST): Fix for bus API change. 2005-09-19 11:38:10 +0000 Wim Taymans gst/level/level-example.c: Fix for new bus API. Original commit message from CVS: * gst/level/level-example.c: (main): Fix for new bus API. * gst/udp/gstudpsrc.c: (gst_udpsrc_set_property): Set caps on pads. 2005-09-19 11:07:40 +0000 Wim Taymans ext/lame/gstlame.c: Set caps on outgoing buffers. Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_chain): Set caps on outgoing buffers. 2005-09-19 11:06:05 +0000 Thomas Vander Stichele * gst/debug/Makefile.am: disable flags for unbuilt plugins Original commit message from CVS: disable flags for unbuilt plugins 2005-09-19 08:21:29 +0000 Thomas Vander Stichele * common: * docs/plugins/scanobj-build.stamp: normal builds shouldn't scan gobjects Original commit message from CVS: normal builds shouldn't scan gobjects 2005-09-16 16:04:28 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: * ext/lame/gstlame.h: clean up further so we don't try to set up five times for a simple pipeline Original commit message from CVS: clean up further so we don't try to set up five times for a simple pipeline 2005-09-16 00:38:50 +0000 Thomas Vander Stichele * check/Makefile.am: * common: * tests/check/Makefile.am: remove gst-register Original commit message from CVS: remove gst-register 2005-09-15 13:57:56 +0000 Wim Taymans * ChangeLog: * common: * gst/rtp/Makefile.am: * gst/rtp/README: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpamrdec.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrenc.c: * gst/rtp/gstrtpamrenc.h: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmenc.h: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmparse.h: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pdec.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263penc.c: * gst/rtp/gstrtph263penc.h: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4vdec.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4venc.c: * gst/rtp/gstrtpmp4venc.h: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadec.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpaenc.c: * gst/rtp/gstrtpmpaenc.h: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpapay.h: Updates to payloader/depayloaders, make payloaders use the base classes. Original commit message from CVS: Updates to payloader/depayloaders, make payloaders use the base classes. Updated README with suggested RTP caps and how to convert to/from SDP. Added config descriptor in mp4v payloader. 2005-09-15 10:47:58 +0000 Andy Wingo gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best): gst/autodetect/gstautovideosink.c Original commit message from CVS: 2005-09-15 Andy Wingo * gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best): * gst/autodetect/gstautovideosink.c (gst_auto_video_sink_find_best): Update for new registry API. 2005-09-14 20:51:47 +0000 Thomas Vander Stichele common/: a simple py script to generate valid xml from a C example probably also need to strip an MIT license when we... Original commit message from CVS: * common/c-to-xml.py: * common/gtk-doc-plugins.mak: a simple py script to generate valid xml from a C example probably also need to strip an MIT license when we decide * docs/plugins/Makefile.am: * gst/level/Makefile.am: * gst/level/gstlevel.c: (gst_level_init): * gst/level/level-example.c: (message_handler), (main): add an example to level that will show up in the docs * gst/rtp/TODO: add a note for the future 2005-09-14 11:44:11 +0000 Michael Smith gst/wavenc/gstwavenc.c: Actually define the debug object being used in wavenc. Fixes #316205 Original commit message from CVS: * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init): Actually define the debug object being used in wavenc. Fixes #316205 2005-09-14 11:23:44 +0000 Michael Smith * ChangeLog: * gst/smpte/Makefile.am: Link smpte plugin against GST_BASE_LIBS, to get libgstbase; needed to build on win32 as this plugin uses collectpads ... Original commit message from CVS: Link smpte plugin against GST_BASE_LIBS, to get libgstbase; needed to build on win32 as this plugin uses collectpads (bug 316204) 2005-09-12 16:37:05 +0000 Jan Schmidt * ChangeLog: Fix up bogus ChangeLog entry Original commit message from CVS: Fix up bogus ChangeLog entry 2005-09-12 16:14:48 +0000 Andy Wingo autogen.sh (package): Now type 'make' to build gst-plugins-good. Original commit message from CVS: 2005-09-12 Andy Wingo * autogen.sh (package): Now type 'make' to build gst-plugins-good. 2005-09-11 17:52:09 +0000 Thomas Vander Stichele * common: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-fdsrc.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-wavparse.xml: add source module to docs; reinspect Original commit message from CVS: add source module to docs; reinspect 2005-09-09 17:56:43 +0000 Jan Schmidt Move fdsrc back into gstreamer core elements. Original commit message from CVS: * configure.ac: * gst/fdsrc/Makefile.am: * gst/fdsrc/gstfdsrc.c: * gst/fdsrc/gstfdsrc.h: Move fdsrc back into gstreamer core elements. * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_transform_ip): * gst/videobox/gstvideobox.c: (gst_video_box_set_property): Basetransform changes. 2005-09-09 16:11:48 +0000 Thomas Vander Stichele * ChangeLog: * ext/jpeg/gstsmokeenc.c: * ext/jpeg/smokecodec.c: fix compiler warnings Original commit message from CVS: fix compiler warnings 2005-09-09 11:09:49 +0000 Thomas Vander Stichele gst-plugins-good.spec.in: spec file fixes Original commit message from CVS: * gst-plugins-good.spec.in: spec file fixes * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init), (gst_multiudpsink_render), (gst_multiudpsink_add), (gst_multiudpsink_clear): it actually helps to actually stream if we hook up the add signal to an actual implementation * gst/udp/gstudpsrc.c: (gst_udpsrc_start): some debugging 2005-09-08 16:58:40 +0000 Flavio Oliveira * ext/jpeg/Makefile.am: * ext/jpeg/gstjpeg.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokeenc.c: jpgenc ported to GSTreamer 0.9 Original commit message from CVS: jpgenc ported to GSTreamer 0.9 2005-09-08 16:26:17 +0000 Flavio Oliveira * ChangeLog: jpegenc ported to GStreamer 0.9 Original commit message from CVS: jpegenc ported to GStreamer 0.9 2005-09-07 13:49:37 +0000 Stefan Kost ext/: gsttaginterface.h -> gsttagsetter.h Original commit message from CVS: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/speex/gstspeexenc.c: gsttaginterface.h -> gsttagsetter.h 2005-09-06 23:30:03 +0000 Jan Schmidt Port to 0.9 and re-enable efence plugin. Original commit message from CVS: * configure.ac: * gst/debug/Makefile.am: * gst/debug/efence.c: (gst_efence_class_init), (gst_efence_init), (gst_efence_chain), (gst_efence_buffer_alloc), (plugin_init), (gst_fenced_buffer_finalize), (gst_fenced_buffer_copy), (gst_fenced_buffer_alloc), (gst_fenced_buffer_class_init), (gst_fenced_buffer_init), (gst_fenced_buffer_get_type): Port to 0.9 and re-enable efence plugin. 2005-09-06 21:31:25 +0000 Tim-Philipp Müller ext/flac/gstflacdec.*: Add support for flac files with 24/32 bits per sample; and misc. minor clean-ups. Seeking is s... Original commit message from CVS: * ext/flac/gstflacdec.c: (flac_caps_factory), (raw_caps_factory), (gst_flacdec_write), (gst_flacdec_convert_src): * ext/flac/gstflacdec.h: Add support for flac files with 24/32 bits per sample; and misc. minor clean-ups. Seeking is still partly broken (for me at least). 2005-09-06 15:50:58 +0000 Wim Taymans gst/rtp/: Added mpeg4 video payload encoder/decoder. Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_get_type), (gst_rtpmp4vdec_base_init), (gst_rtpmp4vdec_class_init), (gst_rtpmp4vdec_init), (gst_rtpmp4vdec_setcaps), (gst_rtpmp4vdec_chain), (gst_rtpmp4vdec_set_property), (gst_rtpmp4vdec_get_property), (gst_rtpmp4vdec_change_state), (gst_rtpmp4vdec_plugin_init): * gst/rtp/gstrtpmp4vdec.h: * gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_get_type), (gst_rtpmp4venc_base_init), (gst_rtpmp4venc_class_init), (gst_rtpmp4venc_init), (gst_rtpmp4venc_setcaps), (gst_rtpmp4venc_flush), (gst_rtpmp4venc_chain), (gst_rtpmp4venc_set_property), (gst_rtpmp4venc_get_property), (gst_rtpmp4venc_change_state), (gst_rtpmp4venc_plugin_init): * gst/rtp/gstrtpmp4venc.h: * gst/rtp/gstrtpmpadec.c: (gst_rtpmpadec_chain): * gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush): Added mpeg4 video payload encoder/decoder. Added some docs in mpa payloader. 2005-09-06 14:06:47 +0000 Thomas Vander Stichele * configure.ac: back to HEAD Original commit message from CVS: back to HEAD === release 0.9.1 === 2005-09-06 14:05:33 +0000 Thomas Vander Stichele * ChangeLog: * NEWS: * README: * RELEASE: * autogen.sh: * common: * configure.ac: releasing 0.9.2 Original commit message from CVS: releasing 0.9.2 2005-09-05 17:20:28 +0000 Jan Schmidt * gst/videocrop/gstvideocrop.c: * sys/v4l2/gstv4l2element.c: * sys/v4l2/gstv4l2src.c: Fix up all the state change functions. Original commit message from CVS: Fix up all the state change functions. 2005-09-05 16:28:16 +0000 Andy Wingo ext/dv/gstdvdemux.c (gst_dvdemux_chain): Move the pad adding here from the state change handler, so we fire signals w... Original commit message from CVS: 2005-09-05 Andy Wingo * ext/dv/gstdvdemux.c (gst_dvdemux_chain): Move the pad adding here from the state change handler, so we fire signals without holding the state lock. 2005-09-05 15:10:18 +0000 Thomas Vander Stichele * gst/qtdemux/qtdemux.c: cleaning up bad Original commit message from CVS: cleaning up bad 2005-09-05 13:18:42 +0000 Thomas Vander Stichele * docs/.gitignore: * docs/plugins/.gitignore: maintenance commits Original commit message from CVS: maintenance commits 2005-09-04 15:09:33 +0000 Thomas Vander Stichele * configure.ac: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/inspect-build.stamp: * docs/plugins/inspect.stamp: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-fdsrc.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videoflip.xml: * docs/plugins/inspect/plugin-wavparse.xml: distcheck fixes Original commit message from CVS: distcheck fixes 2005-09-04 11:50:47 +0000 Thomas Vander Stichele * Makefile.am: * autogen.sh: * common: * docs/plugins/Makefile.am: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: fix distcheck Original commit message from CVS: fix distcheck 2005-09-02 15:56:52 +0000 Thomas Vander Stichele * gst-plugins-good.spec.in: various spec fixes Original commit message from CVS: various spec fixes 2005-09-02 15:44:50 +0000 Andy Wingo * check/elements/level.c: * examples/gstplay/player.c: * examples/stats/mp2ogg.c: * ext/aalib/gstaasink.c: * ext/cairo/gsttextoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: * ext/gconf/gstgconfvideosink.c: * ext/gdk_pixbuf/gstgdkanimation.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/ladspa/gstsignalprocessor.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/mikmod/gstmikmod.c: * ext/pango/gsttextoverlay.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautovideosink.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/debug/breakmydata.c: * gst/debug/gstnavigationtest.c: * gst/effectv/gstquark.c: * gst/fdsrc/gstfdsrc.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/matroska/ebml-read.c: * gst/matroska/ebml-write.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16enc.c: * gst/rtp/gstrtpL16parse.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdec.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrenc.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdec.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmenc.c: * gst/rtp/gstrtpgsmparse.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pdec.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263penc.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpmpadec.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpaenc.c: * gst/rtp/gstrtpmpapay.c: * gst/rtsp/gstrtspsrc.c: * gst/smoothwave/gstsmoothwave.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/videomixer/videomixer.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * po/af.po: * po/az.po: * po/cs.po: * po/en_GB.po: * po/hu.po: * po/it.po: * po/nb.po: * po/nl.po: * po/or.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * sys/oss/gstossmixerelement.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * tests/check/elements/level.c: All plugins updated for element state changes. Original commit message from CVS: 2005-09-02 Andy Wingo * All plugins updated for element state changes. 2005-09-02 15:43:54 +0000 Andy Wingo * ext/lame/gstlame.c: All plugins updated for element state changes. Original commit message from CVS: 2005-09-02 Andy Wingo * All plugins updated for element state changes. 2005-09-01 21:24:57 +0000 Thomas Vander Stichele * ext/aalib/Makefile.am: fix build after cleaning up my vomit Original commit message from CVS: fix build after cleaning up my vomit 2005-09-01 21:23:09 +0000 Thomas Vander Stichele * ext/aalib/Makefile.am: fix build after cleaning up my vomit Original commit message from CVS: fix build after cleaning up my vomit 2005-09-01 21:20:45 +0000 Thomas Vander Stichele * gst/smpte/Makefile.am: fix build after cleaning up my vomit Original commit message from CVS: fix build after cleaning up my vomit 2005-09-01 21:15:30 +0000 Thomas Vander Stichele * gst/smpte/Makefile.am: fix build after cleaning up my vomit Original commit message from CVS: fix build after cleaning up my vomit 2005-09-01 20:23:22 +0000 Thomas Vander Stichele * ChangeLog: * Makefile.am: * check/.gitignore: * check/Makefile.am: * check/elements/.gitignore: * check/elements/level.c: * common: * configure.ac: * gst/level/gstlevel.c: * gst/level/gstlevel.h: * tests/check/.gitignore: * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/level.c: Andrewio Patrickoforus Wingonymus - 5 additional tests for your sins Original commit message from CVS: Andrewio Patrickoforus Wingonymus - 5 additional tests for your sins Add a regression test for level and fix a casting bug that made the additional channels turn out wrong 2005-09-01 17:55:14 +0000 Thomas Vander Stichele add docs to build Original commit message from CVS: * Makefile.am: * configure.ac: add docs to build * common/plugins.xsl: wrap Description into a refsect2 * docs/Makefile.am: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/goom/Makefile.am: * gst/goom/gstgoom.c: (gst_goom_get_type), (gst_goom_base_init), (gst_goom_class_init), (gst_goom_init), (gst_goom_dispose), (gst_goom_sink_setcaps), (gst_goom_src_setcaps), (gst_goom_src_negotiate), (gst_goom_event), (gst_goom_chain), (gst_goom_change_state): * gst/goom/gstgoom.h: GstGOOM -> GstGoom add an example launch line * gst/level/gstlevel.h: * gst/monoscope/gstmonoscope.c: cleanups 2005-08-31 16:28:05 +0000 Thomas Vander Stichele * gst/dvdlpcmdec/.gitignore: * gst/dvdlpcmdec/Makefile.am: * gst/dvdlpcmdec/gstdvdlpcmdec.c: * gst/dvdlpcmdec/gstdvdlpcmdec.h: remove dvdlpcmdec, it's dvd stuff Original commit message from CVS: remove dvdlpcmdec, it's dvd stuff 2005-08-30 19:41:12 +0000 Thomas Vander Stichele * Makefile.am: * gst-libs/gst/gettext.h: * gst-libs/gst/gst-i18n-plugin.h: add some i18n headers Original commit message from CVS: add some i18n headers 2005-08-30 19:24:37 +0000 Thomas Vander Stichele * docs/plugins/.gitignore: ignore more Original commit message from CVS: ignore more 2005-08-30 19:24:03 +0000 Thomas Vander Stichele * docs/Makefile.am: Makefile.am Original commit message from CVS: Makefile.am 2005-08-30 19:20:02 +0000 Thomas Vander Stichele * docs/upload.mak: * docs/version.entities.in: commit new stuff Original commit message from CVS: commit new stuff 2005-08-30 19:01:18 +0000 Thomas Vander Stichele * ChangeLog: * common: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.types: document elements and plugins. Shazam ! Original commit message from CVS: document elements and plugins. Shazam ! 2005-08-30 17:37:00 +0000 Thomas Vander Stichele * .gitignore: * COPYING: * RELEASE: * gst-plugins-good.spec.in: add some files Original commit message from CVS: add some files 2005-08-17 19:05:51 +0000 Wim Taymans configure.ac: Added mpegaudioparse Original commit message from CVS: * configure.ac: Added mpegaudioparse * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_src_setcaps), (gst_lame_sink_setcaps), (gst_lame_sink_event), (gst_lame_chain): Some cleanups. Fix memleak. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init), (gst_mp3parse_init), (gst_mp3parse_chain), (gst_mp3parse_change_state): * gst/mpegaudioparse/gstmpegaudioparse.h: Ported mpegaudioparse 2005-08-16 16:12:15 +0000 Wim Taymans Fix compile warning. Original commit message from CVS: * configure.ac: * ext/amrnb/amrnbparse.c: (gst_amrnbparse_read_header): Fix compile warning. * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_src_getcaps), (gst_lame_src_setcaps), (gst_lame_sink_setcaps), (gst_lame_init), (gst_lame_sink_event), (gst_lame_chain), (gst_lame_change_state): * ext/lame/gstlame.h: Port lame plugin 2005-07-05 10:51:49 +0000 Andy Wingo Way, way, way too many files: Remove crack comment from the 2000 era. Original commit message from CVS: 2005-07-05 Andy Wingo * Way, way, way too many files: Remove crack comment from the 2000 era. 2005-07-05 10:51:41 +0000 Andy Wingo Way, way, way too many files: Remove crack comment from the 2000 era. Original commit message from CVS: 2005-07-05 Andy Wingo * Way, way, way too many files: Remove crack comment from the 2000 era. 2004-10-26 11:36:52 +0000 Iain Holmes * ext/lame/gstlame.c: Memory leak fixes Original commit message from CVS: Memory leak fixes Allow level to take mono or stereo audio 2004-08-26 00:32:00 +0000 Zaheer Abbas Merali ext/lame/gstlame.*: Added new media support to lame Original commit message from CVS: 2004-08-26 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_init), (gst_lame_chain): * ext/lame/gstlame.h: Added new media support to lame 2004-08-19 22:44:50 +0000 Zaheer Abbas Merali Only enable lame presets if version of lame has presets in API Original commit message from CVS: 2004-08-19 Zaheer Abbas Merali * configure.ac: * ext/lame/Makefile.am: * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup): Only enable lame presets if version of lame has presets in API 2004-08-15 13:47:00 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: describe the enum values for vbr mode and presets more verbosely Original commit message from CVS: 2004-08-15 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type), (gst_lame_preset_get_type), (gst_lame_class_init): describe the enum values for vbr mode and presets more verbosely 2004-08-13 15:22:49 +0000 Zaheer Abbas Merali ext/lame/gstlame.*: add preset property to lame so it can use lame presets Original commit message from CVS: 2004-08-13 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_mode_get_type), (gst_lame_quality_get_type), (gst_lame_padding_get_type), (gst_lame_preset_get_type), (gst_lame_class_init), (gst_lame_init), (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup): * ext/lame/gstlame.h: add preset property to lame so it can use lame presets 2004-08-13 14:55:27 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: whoops forgot break, thanks teuf Original commit message from CVS: 2004-08-13 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_get_property): whoops forgot break, thanks teuf 2004-08-13 14:41:02 +0000 Zaheer Abbas Merali ext/lame/gstlame.*: fix lame's broken vbr stuff, allow it to resample if need be, and also make xing header optional Original commit message from CVS: 2004-08-13 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_vbrmode_get_type), (gst_lame_class_init), (gst_lame_src_getcaps), (gst_lame_sink_link), (gst_lame_init), (gst_lame_set_property), (gst_lame_get_property), (gst_lame_setup): * ext/lame/gstlame.h: fix lame's broken vbr stuff, allow it to resample if need be, and also make xing header optional 2004-08-12 17:22:30 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: added getcaps function so samplerate doesntget fixated to silly values Original commit message from CVS: 2004-08-12 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_src_getcaps), (gst_lame_init): added getcaps function so samplerate doesntget fixated to silly values 2004-08-12 16:44:14 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: revert previous fix Original commit message from CVS: 2004-08-12 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_src_link): revert previous fix 2004-08-12 16:12:00 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: made source pad link function check if sinkpad is ok..fixes the problem where core fixates the ou... Original commit message from CVS: 2004-08-12 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_src_link): made source pad link function check if sinkpad is ok..fixes the problem where core fixates the output rate of lame stupidly 2004-08-12 15:48:50 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: set default compression ratio paramter to 0.0 so bitrate parameter works :) Original commit message from CVS: 2004-08-12 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_class_init), (gst_lame_init): set default compression ratio paramter to 0.0 so bitrate parameter works :) 2004-08-09 09:22:12 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: fix add debugging Original commit message from CVS: fix add debugging 2004-08-02 11:39:17 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: gearing up for release Original commit message from CVS: gearing up for release 2004-08-02 09:16:14 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: add link function. fixes @148986 Original commit message from CVS: add link function. fixes @148986 2004-07-28 20:26:31 +0000 Zaheer Abbas Merali ext/lame/gstlame.c: send tag events downstream Original commit message from CVS: 2004-07-28 Zaheer Abbas Merali * ext/lame/gstlame.c: (gst_lame_chain): send tag events downstream * ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type), (gst_shout2send_get_type), (gst_shout2send_set_clock), (gst_shout2send_class_init), (gst_shout2send_init), (set_shout_metadata), (gst_shout2send_set_metadata), (gst_shout2send_chain), (gst_shout2send_set_property), (gst_shout2send_get_property), (gst_shout2send_connect), (gst_shout2send_change_state): * ext/shout2/gstshout2.h: - fix for sending mp3 audio to icecast2 server, if pad link function not called before PAUSED state - added option to use GStreamer clock sync (as opposed to libshout's own sync) - added tagging support for mp3 audio broadcasted * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init): debug info 2004-07-27 21:51:30 +0000 Steve Lhomme * gst/audiofx/gststereo.c: fix local includes and 64 bits constants Original commit message from CVS: fix local includes and 64 bits constants 2004-07-26 15:42:18 +0000 Benjamin Otte ext/lame/gstlame.c: add debugging category, add error checks like checking return values of setup calls, make sure it... Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_sink_link), (gst_lame_init), (gst_lame_chain), (gst_lame_setup), (gst_lame_change_state), (plugin_init): add debugging category, add error checks like checking return values of setup calls, make sure it still works after PLAYING=>NULL=>PLAYING, fix encoding of mono streams 2004-06-14 10:58:27 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: sync mp3 caps Original commit message from CVS: sync mp3 caps 2004-06-14 10:52:35 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: add comment Original commit message from CVS: add comment 2004-05-21 23:28:57 +0000 Stéphane Loeuillet * ext/lame/gstlame.c: second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ... Original commit message from CVS: second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in gst-plugins/ext/ this time) 2004-05-09 14:37:15 +0000 Benjamin Otte ext/: \1/Codec, (fixes #142193) Original commit message from CVS: reviewed by Benjamin Otte * ext/a52dec/gsta52dec.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: (gst_faad_base_init): * ext/ivorbis/vorbisfile.c: * ext/lame/gstlame.c: * ext/libfame/gstlibfame.c: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/sidplay/gstsiddec.cc: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: correct klasses. Mostly s,Codec/(Audio|Video),\1/Codec, (fixes #142193) 2004-05-07 00:43:50 +0000 Benjamin Otte ext/lame/gstlame.c: simplify Original commit message from CVS: * ext/lame/gstlame.c: (gst_lame_chain): simplify * ext/mad/gstmad.c: (gst_mad_handle_event): fix event leak * gst/typefind/gsttypefindfunctions.c: (mp3_type_find): be able to detect mp3 files < 4096 bytes 2004-05-03 16:46:10 +0000 Stéphane Loeuillet * ext/lame/gstlame.c: don't trust lame_init to set good values as defaults Original commit message from CVS: don't trust lame_init to set good values as defaults 2004-03-15 19:32:27 +0000 Thomas Vander Stichele * gst/audiofx/gststereo.c: don't mix tabs and spaces Original commit message from CVS: don't mix tabs and spaces 2004-03-15 19:32:25 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: don't mix tabs and spaces Original commit message from CVS: don't mix tabs and spaces 2004-03-15 16:32:54 +0000 Johan Dahlin *.h: Revert indenting Original commit message from CVS: * *.h: Revert indenting 2004-03-15 16:32:53 +0000 Johan Dahlin *.h: Revert indenting Original commit message from CVS: * *.h: Revert indenting 2004-03-14 22:34:33 +0000 Thomas Vander Stichele * gst/audiofx/gststereo.c: * gst/audiofx/gststereo.h: gst-indent Original commit message from CVS: gst-indent 2004-03-14 22:34:30 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: * ext/lame/gstlame.h: * ext/lame/test-lame.c: gst-indent Original commit message from CVS: gst-indent 2004-02-22 15:14:24 +0000 Benjamin Otte configure.ac: export [_]*{gst,Gst,GST}.* symbols from plugins Original commit message from CVS: 2004-02-22 Benjamin Otte * configure.ac: export [_]*{gst,Gst,GST}.* symbols from plugins 2004-02-22 Christophe Fergeau reviewed by: Benjamin Otte * ext/lame/gstlame.c: (add_one_tag): * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value), (gst_vorbisenc_metadata_set1): * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: (gst_vorbis_tag_add): apply fixes from bugs #135042 (lame can't write tags) and #133817 (add GST_ALBUM_VOLUME_{COUNT,NUMBER} tags) 2004-02-19 22:19:55 +0000 Benjamin Otte ext/: use gst_tag_list_insert when you want to insert tags Original commit message from CVS: 2004-02-19 Benjamin Otte * ext/lame/gstlame.c: (gst_lame_chain): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_chain): use gst_tag_list_insert when you want to insert tags 2004-02-02 17:23:32 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: change NULL to (NULL) for GST_ELEMENT_ERROR Original commit message from CVS: change NULL to (NULL) for GST_ELEMENT_ERROR Make sure errors end with "." 2004-01-29 23:20:44 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: GST_ELEMENT_ERROR Original commit message from CVS: GST_ELEMENT_ERROR 2004-01-18 21:46:58 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: use new error signal and classification Original commit message from CVS: use new error signal and classification 2003-12-22 01:47:08 +0000 David Schleef * ext/lame/gstlame.c: Merge CAPS branch Original commit message from CVS: Merge CAPS branch 2003-12-07 14:47:09 +0000 Christophe Fergeau * ext/lame/gstlame.c: * ext/lame/gstlame.h: Uses new tagging framework Original commit message from CVS: Uses new tagging framework 2003-12-04 10:37:38 +0000 Andy Wingo * gst/audiofx/gststereo.c: remove copyright field from plugins Original commit message from CVS: remove copyright field from plugins 2003-12-04 10:37:35 +0000 Andy Wingo * ext/lame/gstlame.c: remove copyright field from plugins Original commit message from CVS: remove copyright field from plugins 2003-12-02 02:28:12 +0000 David Schleef * ext/lame/test-lame.c: change _connect to _link Original commit message from CVS: change _connect to _link 2003-11-16 22:02:23 +0000 Leif Johnson * gst/audiofx/gststereo.c: + checking in plugin category changes Original commit message from CVS: + checking in plugin category changes 2003-11-07 12:47:02 +0000 Ronald S. Bultje * gst/audiofx/gststereo.h: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... Original commit message from CVS: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files 2003-11-07 12:46:51 +0000 Ronald S. Bultje * ext/lame/gstlame.h: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... Original commit message from CVS: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files 2003-11-02 22:34:11 +0000 Benjamin Otte * gst/audiofx/gststereo.c: fix for new plugin system Original commit message from CVS: fix for new plugin system 2003-11-02 00:13:26 +0000 Iain Holmes * ext/lame/gstlame.c: Fixed lame too Original commit message from CVS: Fixed lame too 2003-10-09 09:04:23 +0000 Ronald S. Bultje * ext/lame/gstlame.c: Fix typo in Andy's commit Original commit message from CVS: Fix typo in Andy's commit 2003-10-08 16:08:19 +0000 Andy Wingo * gst/audiofx/gststereo.c: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. Original commit message from CVS: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. 2003-10-08 16:08:10 +0000 Andy Wingo * ext/lame/gstlame.c: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. Original commit message from CVS: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. 2003-09-30 19:48:39 +0000 Ronald S. Bultje * ext/lame/gstlame.c: Input and output samplerate are *not* necessarily the same in lame. This fixes the output caps Original commit message from CVS: Input and output samplerate are *not* necessarily the same in lame. This fixes the output caps 2003-09-16 10:00:00 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: reverting error patch before making a branch. Original commit message from CVS: reverting error patch before making a branch. 2003-09-15 01:08:38 +0000 Benjamin Otte * ext/lame/gstlame.c: converted gst_element_error to new format in ext/ - gettext pending Original commit message from CVS: converted gst_element_error to new format in ext/ - gettext pending 2003-09-12 11:35:23 +0000 Ronald S. Bultje * ext/lame/gstlame.c: Fix tiny caps error in lame caps - mpegversion(1) was missing Original commit message from CVS: Fix tiny caps error in lame caps - mpegversion(1) was missing 2003-08-10 00:01:58 +0000 David Schleef * ext/lame/Makefile.am: Remove redundant plugindir definition Original commit message from CVS: Remove redundant plugindir definition 2003-07-10 15:39:11 +0000 Christian Schaller * ext/lame/README: * ext/lame/gstlame.c: fix license field of lame plugin to say LGPL, lame is LGPL. Add Readme with info Original commit message from CVS: fix license field of lame plugin to say LGPL, lame is LGPL. Add Readme with info 2003-07-06 20:49:50 +0000 Ronald S. Bultje * ext/lame/gstlame.c: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri... Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs 2003-07-05 22:48:58 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: * ext/lame/gstlame.h: patch from hadess, modified Original commit message from CVS: patch from hadess, modified 2003-06-29 19:46:13 +0000 Benjamin Otte * gst/audiofx/gststereo.c: compatibility fix for new GST_DEBUG stuff. Original commit message from CVS: compatibility fix for new GST_DEBUG stuff. Includes fixes for missing includes for config.h and unistd.h I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately. 2003-06-29 19:46:09 +0000 Benjamin Otte * ext/lame/gstlame.c: compatibility fix for new GST_DEBUG stuff. Original commit message from CVS: compatibility fix for new GST_DEBUG stuff. Includes fixes for missing includes for config.h and unistd.h I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately. 2003-06-07 00:34:51 +0000 Ronald S. Bultje * ext/lame/gstlame.c: * ext/lame/gstlame.h: Another duration patch from Joshua (slightly modified by me) Original commit message from CVS: Another duration patch from Joshua (slightly modified by me) 2003-05-29 19:32:39 +0000 Ronald S. Bultje * ext/lame/gstlame.h: Fix build prob Original commit message from CVS: Fix build prob 2003-05-29 12:41:42 +0000 Wim Taymans * ext/lame/gstlame.c: - copy offset from input buffer Original commit message from CVS: - copy offset from input buffer 2003-05-13 12:28:16 +0000 Ronald S. Bultje * ext/lame/gstlame.c: * ext/lame/gstlame.h: Get timestamping somewhat better Original commit message from CVS: Get timestamping somewhat better 2003-05-12 20:08:17 +0000 Zeeshan Ali * ext/lame/gstlame.c: Hacked lame to make it copy the timestamp on the source buffer to the sink buffer Original commit message from CVS: Hacked lame to make it copy the timestamp on the source buffer to the sink buffer 2003-01-10 13:38:27 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: PadConnect -> PadLink Original commit message from CVS: PadConnect -> PadLink 2003-01-10 10:22:24 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t... Original commit message from CVS: another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so that I don't spend double the time resolving conflicts 2002-12-08 17:20:44 +0000 Iain Holmes * ext/lame/gstlame.c: Replace audio/mp3 with audio/x-mp3 and audio/x-flac with application/x-flac Original commit message from CVS: Replace audio/mp3 with audio/x-mp3 and audio/x-flac with application/x-flac 2002-12-08 14:50:04 +0000 Thomas Vander Stichele * ext/lame/Makefile.am: parallel install fixes Original commit message from CVS: parallel install fixes 2002-12-08 02:44:00 +0000 Wim Taymans * ext/lame/gstlame.c: cleanups Original commit message from CVS: cleanups 2002-11-20 21:02:40 +0000 Wim Taymans * ext/lame/gstlame.c: Remove redundant properties. Original commit message from CVS: Remove redundant properties. 2002-11-02 05:39:21 +0000 David I. Lehn * ext/lame/Makefile.am: use AM_CFLAGS instead of CFLAGS Original commit message from CVS: use AM_CFLAGS instead of CFLAGS 2002-10-02 08:04:00 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: api change Original commit message from CVS: api change 2002-09-18 19:02:52 +0000 Christian Schaller * gst/audiofx/gststereo.c: plugins part of license field patch Original commit message from CVS: plugins part of license field patch 2002-09-18 19:02:46 +0000 Christian Schaller * ext/lame/gstlame.c: plugins part of license field patch Original commit message from CVS: plugins part of license field patch 2002-09-10 09:31:38 +0000 Ronald S. Bultje * ext/lame/test-lame.c: This updates all plugins to the new API for gst_pad_try_set_caps Original commit message from CVS: This updates all plugins to the new API for gst_pad_try_set_caps 2002-09-01 15:40:39 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: small updates Original commit message from CVS: small updates 2002-07-08 19:32:49 +0000 Wim Taymans * ext/lame/gstlame.c: unref event Original commit message from CVS: unref event 2002-07-07 14:17:00 +0000 Wim Taymans * ext/lame/gstlame.c: Don't free uninitialized pointers Original commit message from CVS: Don't free uninitialized pointers 2002-07-07 14:06:38 +0000 Wim Taymans * ext/lame/gstlame.c: Lame should accept events even when not negotiated yet. Original commit message from CVS: Lame should accept events even when not negotiated yet. 2002-06-08 09:26:09 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: reorder Original commit message from CVS: reorder 2002-04-20 21:42:51 +0000 Andy Wingo * gst/audiofx/gststereo.c: a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br... Original commit message from CVS: * a hack to work around intltool's brokenness * a current check for mpeg2dec * details->klass reorganizations * an element browser that uses details->klass * separated cdxa parse out from the avi directory 2002-04-11 20:42:26 +0000 Andy Wingo * gst/audiofx/gststereo.c: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. Original commit message from CVS: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. also, some -Werror fixes. 2002-04-11 20:42:25 +0000 Andy Wingo * ext/lame/gstlame.c: * ext/lame/test-lame.c: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. Original commit message from CVS: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. also, some -Werror fixes. 2002-03-30 17:06:26 +0000 Wim Taymans * ext/lame/gstlame.c: * ext/lame/test-lame.c: Changed to the new props API Original commit message from CVS: Changed to the new props API Other small tuff. 2002-03-27 04:02:38 +0000 Andy Wingo * ext/lame/gstlame.c: update g_value stuff to match property types Original commit message from CVS: update g_value stuff to match property types 2002-03-24 22:07:03 +0000 Andy Wingo * ext/lame/gstlame.c: filter newlines out of GST_DEBUG statements to reflect new core behavior fixes to adder's caps, again Original commit message from CVS: * filter newlines out of GST_DEBUG statements to reflect new core behavior * fixes to adder's caps, again 2002-03-20 21:45:04 +0000 Andy Wingo * gst/audiofx/gststereo.c: * gst/audiofx/gststereo.h: s/Gnome-Streamer/GStreamer/ Original commit message from CVS: s/Gnome-Streamer/GStreamer/ 2002-03-20 21:45:03 +0000 Andy Wingo * ext/lame/gstlame.c: * ext/lame/gstlame.h: s/Gnome-Streamer/GStreamer/ Original commit message from CVS: s/Gnome-Streamer/GStreamer/ 2002-03-19 17:14:57 +0000 Andy Wingo * ext/lame/gstlame.c: fix compile error (untested) Original commit message from CVS: fix compile error (untested) 2002-03-19 04:10:06 +0000 Andy Wingo * gst/audiofx/gststereo.c: removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with... Original commit message from CVS: * removal of //-style comments * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct, and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory. 2002-03-19 04:10:05 +0000 Andy Wingo * ext/lame/Makefile.am: * ext/lame/gstlame.c: removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with... Original commit message from CVS: * removal of //-style comments * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct, and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory. 2002-03-19 01:39:42 +0000 Andy Wingo * ext/lame/Makefile.am: s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi... Original commit message from CVS: s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagically, and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag 2002-03-03 00:53:24 +0000 Andy Wingo * ext/lame/gstlame.c: get up-to-date with the gst_caps_debug api improved capsnego in mad improved capsnego in adder improved capsnego in i... Original commit message from CVS: * get up-to-date with the gst_caps_debug api * improved capsnego in mad * improved capsnego in adder * improved capsnego in intfloat plugins * unbroke capsnego in stereomono plugins * fix cothread stack allocation within the main thread in new cothreads 2002-02-21 17:33:59 +0000 Thomas Vander Stichele * ext/lame/Makefile.am: uncomment lame test until we can get the register to work Original commit message from CVS: uncomment lame test until we can get the register to work 2002-02-21 17:20:35 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: use gst-debuginfo.m4 macro so plugins are actually compiled with debug info some more debug output for lame Original commit message from CVS: * use gst-debuginfo.m4 macro so plugins are actually compiled with debug info * some more debug output for lame 2002-02-21 14:04:02 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: on sink connect, check if the current pad is compatible with the given caps cleaned up debug output change pad templa... Original commit message from CVS: * on sink connect, check if the current pad is compatible with the given caps * cleaned up debug output * change pad template to only accept allowed sample rates if these changes are considered ok by others then the same should be applied to other encoding plugins (notably the compatibility check) 2002-02-19 20:49:52 +0000 Thomas Vander Stichele * ext/lame/test-lame.c: ok, this works Original commit message from CVS: ok, this works 2002-02-19 20:35:42 +0000 Wim Taymans * ext/lame/test-lame.c: Always bring the elements to READY before trying to do capsnego. fix the caps as lame doesn't accept law==1 Original commit message from CVS: Always bring the elements to READY before trying to do capsnego. fix the caps as lame doesn't accept law==1 2002-02-19 20:19:36 +0000 Thomas Vander Stichele * ext/lame/test-lame.c: still does not work ;( Original commit message from CVS: still does not work ;( 2002-02-19 18:28:05 +0000 Thomas Vander Stichele * ext/lame/Makefile.am: * ext/lame/test-lame.c: adding a test for lame stuff Original commit message from CVS: adding a test for lame stuff 2002-02-19 17:29:55 +0000 Wim Taymans * ext/lame/gstlame.c: Added event handling. Original commit message from CVS: Added event handling. Fix flush Fix state change. Convert to gobject deep_notify 2002-02-19 12:55:16 +0000 Thomas Vander Stichele * ext/lame/gstlame.c: somebody help me fix lame ;) Original commit message from CVS: somebody help me fix lame ;) I commented out the state change function because it is called before lame has the right caps. Is the state change function still necessary ? in any case, at least now lame actually listens to osssrc re: rate and channels 2002-01-31 17:08:46 +0000 David I. Lehn * ext/lame/gstlame.h: Revert lame include dir change. Upstream uses $prefix/include/lame/lame.h. Original commit message from CVS: Revert lame include dir change. Upstream uses $prefix/include/lame/lame.h. 2002-01-30 11:25:58 +0000 Thomas Vander Stichele * ext/lame/gstlame.h: I checked lame packages and source code and they seem to want lame.h in prefix/include/lame.h so I fixed stuff accord... Original commit message from CVS: I checked lame packages and source code and they seem to want lame.h in prefix/include/lame.h so I fixed stuff accordingly. Do any systems have lame in include/lame/lame.h ? If so, mail me and we'll work it out. 2002-01-18 02:05:25 +0000 Wrobell * ext/lame/Makefile.am: - plugins are built without versioning info Original commit message from CVS: - plugins are built without versioning info 2002-01-13 22:27:24 +0000 Wim Taymans * ext/lame/gstlame.c: Bring the plugins in sync with the new core capsnego system. Original commit message from CVS: Bring the plugins in sync with the new core capsnego system. Added some features, enhancements... 2002-01-12 03:34:26 +0000 David I. Lehn * ext/lame/Makefile.am: s/filter/plugin/ link plugins to GST_LIBS rearrange rules to a common format Original commit message from CVS: * s/filter/plugin/ * link plugins to GST_LIBS * rearrange rules to a common format 2001-12-22 23:26:33 +0000 Andy Wingo * gst/audiofx/gststereo.c: * gst/audiofx/gststereo.h: Initial revision Original commit message from CVS: Initial revision 2001-12-21 12:47:09 +0000 Wim Taymans * ext/lame/gstlame.c: * ext/lame/gstlame.h: Lame cleanup Original commit message from CVS: Lame cleanup Added EOS, flush, error reporting etc. 2001-12-20 23:48:55 +0000 Thomas Vander Stichele * ext/lame/Makefile.am: * ext/lame/gstlame.c: * ext/lame/gstlame.h: adding lame Original commit message from CVS: adding lame 2001-12-17 18:37:01 +0000 Thomas Vander Stichele building up speed Original commit message from CVS: building up speed