diff --git a/ChangeLog b/ChangeLog index 99a3669c1b..5a1d9b75ec 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,376 @@ -=== release 1.7.2 === +=== release 1.7.90 === -2016-02-19 Sebastian Dröge +2016-03-01 Sebastian Dröge * configure.ac: - releasing 1.7.2 + releasing 1.7.90 + +2016-03-01 16:53:05 +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/hr.po: + * po/hu.po: + * po/id.po: + * po/it.po: + * po/ja.po: + * po/lt.po: + * po/lv.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: Update translations + +2016-01-28 16:26:47 +0100 Tom Deseyn + + * gst/tcp/gstmultisocketsink.c: + multisocketsink: handle client close correctly and EWOULDBLOCK + Fixes 100% cpu usage when client disconnects. Commit 6db2ee56 + would just make multisocketsink ignore reads of 0 bytes without + removing the client, so we'd get woken up over and over again + for the client. + Fix the original issue differently by handling the non-fatal error code. + https://bugzilla.gnome.org/show_bug.cgi?id=761257 + https://bugzilla.gnome.org/show_bug.cgi?id=743834 + +2016-02-27 00:11:02 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/video-orc-dist.c: + * gst-libs/gst/video/video-orc-dist.h: + video: update disted orc backup file + https://bugzilla.gnome.org/show_bug.cgi?id=761851 + +2016-02-11 11:27:57 +0100 Göran Jönsson + + * gst-libs/gst/video/video-converter.c: + * gst-libs/gst/video/video-orc.orc: + video-converter: add direct UYVY to GRAY8 conversion function + https://bugzilla.gnome.org/show_bug.cgi?id=761851 + +2016-02-04 16:01:00 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opus: fix mono<->stereo up/down-mixing + https://bugzilla.gnome.org/show_bug.cgi?id=761588 + +2016-02-26 17:09:06 +0800 Lim Siew Hoon + + * gst-libs/gst/pbutils/encoding-profile.c: + pbutils: docs: Remove the empty lines in between and + They are converted into by gtk-doc... + https://bugzilla.gnome.org/show_bug.cgi?id=762674 + +2016-02-26 12:41:01 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From b64f03f to 6f2d209 + +2016-02-26 00:53:05 +0000 Tim-Philipp Müller + + * ext/opus/gstopusenc.c: + opusenc: remove deprecated "cbr", "audio", and "constrained-vbr" properties + They have been replaced by "audio-type" and "bitrate-type". + https://bugzilla.gnome.org/show_bug.cgi?id=756282 + +2016-02-26 00:37:57 +0000 Tim-Philipp Müller + + * docs/plugins/gst-plugins-base-plugins-docs.sgml: + * docs/plugins/gst-plugins-base-plugins-sections.txt: + * docs/plugins/gst-plugins-base-plugins.args: + * docs/plugins/gst-plugins-base-plugins.hierarchy: + * docs/plugins/gst-plugins-base-plugins.interfaces: + * docs/plugins/inspect/plugin-opus.xml: + docs: add Opus to docs + +2016-02-26 00:20:10 +0000 Tim-Philipp Müller + + * configure.ac: + * ext/Makefile.am: + * ext/opus/Makefile.am: + * ext/opus/gstopus.c: + * tests/check/Makefile.am: + * tests/check/elements/.gitignore: + opus: move Opus audio decoder and encoder from -bad to -base + Hook into build system after moving history. + https://bugzilla.gnome.org/show_bug.cgi?id=756282 + +2016-02-25 23:51:42 +0000 Tim-Philipp Müller + + Merge branch 'plugin-move-opus' + Move Opus decoder and encoder from -bad to -base. + https://bugzilla.gnome.org/show_bug.cgi?id=756282 + +2016-02-25 23:13:39 +0000 Tim-Philipp Müller + + * tools/gst-play-1.0.1: + * tools/gst-play.c: + tools: gst-play: add 'n' and 'b' as additional shortcuts for next/previous item + < and > are composed with shift + something else on many keyboards + layouts, so don't work well when injecting them via windowing systems + which will send them as shift key press and separate other key, and + we the don't combine that to < or > properly. n/b are easier. + +2016-02-26 00:02:49 +0200 Sebastian Dröge + + * tests/check/Makefile.am: + * tests/check/libs/baseaudiovisualizer.c: + audiovisualizer: Use the library instead of including the source file + Fixes build now that the shader enum GType has moved to a different file. + +2016-02-25 20:39:04 +0200 Sebastian Dröge + + * gst-libs/gst/pbutils/gstaudiovisualizer.c: + audiovisualizer: Let GstAudioVisualizerShader enum GType be autogenerated by glib-mkenums + That happens automatically already anyway. + +2016-02-25 17:46:31 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/video-frame.c: + video: flesh out docs for gst_video_frame_map() + +2016-02-25 10:47:17 +0000 Luis de Bethencourt + + * gst-libs/gst/pbutils/gstaudiovisualizer.c: + visual: correct type name + Base class type name should not reference libvisual since not all child + elements use this. This was an oversight when merging audiovisualizers into + a common base class. + +2016-02-24 14:05:03 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-quantize.c: + audio-quantize: fix feedback dither + Make sure we allocated enough extra space in the error buffer to + store the feedback error. + +2016-02-24 12:54:39 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-converter.c: + audio-converter: perform dithering on the current format + Use the current (intermediate) format to decide how to set up dithering + instead of the input format. + +2016-02-23 18:23:45 +0200 Sebastian Dröge + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Handle gst_pad_get_current_caps() returning NULL gracefully + +2016-02-23 09:35:14 +0100 Edward Hervey + + * gst/playback/gstplaysink.c: + Revert "playsink: Properly mark pending blocked pads" + This reverts commit 62053852de01fb324a915b27c00f5b8dc0f66fb3. + The issue that the patch fixes is only noticeable when using decodebin3, + which isn't yet in master. + +2015-12-10 15:32:06 +0100 Adam Miartus + + * gst-libs/gst/tag/gstid3tag.c: + tag: id3v2: read conductor tag + ID3v2 features the TPE3 info frame, which contains information + about the conductor. + https://bugzilla.gnome.org/show_bug.cgi?id=762451 + +2016-02-20 11:31:43 +0000 Tim-Philipp Müller + + * ext/theora/gsttheoradec.c: + * gst-libs/gst/video/video-frame.c: + * gst/videoconvert/gstvideoconvert.c: + * gst/videoscale/gstvideoscale.c: + * sys/ximage/ximage.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvcontext.c: + * sys/xvimage/xvimage.c: + * sys/xvimage/xvimagesink.c: + Fix use of undeclared core debug category symbols + libgstreamer currently exports some debug category + symbols GST_CAT_*, but those are not declared in any + public headers. + Some plugins and libgstvideo just use GST_DEBUG_CATEGORY_EXTERN() + to declare and use those, but that's just not right at + all, and it won't work on Windows with MSVC. Instead look + up the categories via the API. + +2016-02-20 10:05:17 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/audio.def: + * gst-libs/gst/audio/audio.vcproj: + * gst-libs/gst/audio/audiofilter.vcproj: + * gst-libs/gst/riff/riff.def: + * gst-libs/gst/riff/riff.vcproj: + * gst-libs/gst/video/video.vcproj: + * gst/adder/adder.vcproj: + * gst/audioconvert/audioconvert.vcproj: + * gst/audiorate/audiorate.vcproj: + * gst/tcp/tcp.vcproj: + * gst/typefind/typefindfunctions.vcproj: + * gst/videoconvert/videoconvert.vcproj: + * gst/videorate/videorate.vcproj: + * gst/videoscale/videoscale.vcproj: + * gst/videotestsrc/videotestsrc.vcproj: + * gst/volume/volume.vcproj: + * win32/MANIFEST: + * win32/vs6/grammar.dsp: + * win32/vs6/gst_plugins_base.dsw: + * win32/vs6/libgstadder.dsp: + * win32/vs6/libgstaudio.dsp: + * win32/vs6/libgstaudioconvert.dsp: + * win32/vs6/libgstaudiorate.dsp: + * win32/vs6/libgstaudioresample.dsp: + * win32/vs6/libgstaudioscale.dsp: + * win32/vs6/libgstaudiotestsrc.dsp: + * win32/vs6/libgstdecodebin.dsp: + * win32/vs6/libgstdecodebin2.dsp: + * win32/vs6/libgstdirectsound.dsp: + * win32/vs6/libgstfft.dsp: + * win32/vs6/libgstgdp.dsp: + * win32/vs6/libgstinterfaces.dsp: + * win32/vs6/libgstogg.dsp: + * win32/vs6/libgstpbutils.dsp: + * win32/vs6/libgstplaybin.dsp: + * win32/vs6/libgstriff.dsp: + * win32/vs6/libgstrtp.dsp: + * win32/vs6/libgstrtsp.dsp: + * win32/vs6/libgstsdp.dsp: + * win32/vs6/libgstsinesrc.dsp: + * win32/vs6/libgstsubparse.dsp: + * win32/vs6/libgsttag.dsp: + * win32/vs6/libgsttheora.dsp: + * win32/vs6/libgsttypefindfunctions.dsp: + * win32/vs6/libgstvideo.dsp: + * win32/vs6/libgstvideorate.dsp: + * win32/vs6/libgstvideoscale.dsp: + * win32/vs6/libgstvideotestsrc.dsp: + * win32/vs6/libgstvolume.dsp: + * win32/vs6/libgstvorbis.dsp: + * win32/vs7/gst-plugins-base.sln: + * win32/vs7/libgstadder.vcproj: + * win32/vs7/libgstaudio.vcproj: + * win32/vs7/libgstaudioconvert.vcproj: + * win32/vs7/libgstaudiorate.vcproj: + * win32/vs7/libgstaudioresample.vcproj: + * win32/vs7/libgstaudiotestsrc.vcproj: + * win32/vs7/libgstdecodebin.vcproj: + * win32/vs7/libgstinterfaces.vcproj: + * win32/vs7/libgstogg.vcproj: + * win32/vs7/libgstplaybin.vcproj: + * win32/vs7/libgstriff.vcproj: + * win32/vs7/libgstsubparse.vcproj: + * win32/vs7/libgsttag.vcproj: + * win32/vs7/libgsttcp.vcproj: + * win32/vs7/libgsttheora.vcproj: + * win32/vs7/libgsttypefind.vcproj: + * win32/vs7/libgstvideo.vcproj: + * win32/vs7/libgstvideorate.vcproj: + * win32/vs7/libgstvideoscale.vcproj: + * win32/vs7/libgstvideotestsrc.vcproj: + * win32/vs7/libgstvolume.vcproj: + * win32/vs7/libgstvorbis.vcproj: + * win32/vs8/gst-plugins-base.sln: + * win32/vs8/libgstadder.vcproj: + * win32/vs8/libgstaudio.vcproj: + * win32/vs8/libgstaudioconvert.vcproj: + * win32/vs8/libgstaudiorate.vcproj: + * win32/vs8/libgstaudioresample.vcproj: + * win32/vs8/libgstaudiotestsrc.vcproj: + * win32/vs8/libgstdecodebin.vcproj: + * win32/vs8/libgstinterfaces.vcproj: + * win32/vs8/libgstogg.vcproj: + * win32/vs8/libgstplaybin.vcproj: + * win32/vs8/libgstriff.vcproj: + * win32/vs8/libgstsubparse.vcproj: + * win32/vs8/libgsttag.vcproj: + * win32/vs8/libgsttcp.vcproj: + * win32/vs8/libgsttheora.vcproj: + * win32/vs8/libgsttypefind.vcproj: + * win32/vs8/libgstvideo.vcproj: + * win32/vs8/libgstvideorate.vcproj: + * win32/vs8/libgstvideoscale.vcproj: + * win32/vs8/libgstvideotestsrc.vcproj: + * win32/vs8/libgstvolume.vcproj: + * win32/vs8/libgstvorbis.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. + +2016-02-19 12:38:24 +0200 Sebastian Dröge + + * configure.ac: + Back to development + +=== release 1.7.2 === + +2016-02-19 11:48:30 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * docs/plugins/gst-plugins-base-plugins.args: + * docs/plugins/gst-plugins-base-plugins.hierarchy: + * docs/plugins/gst-plugins-base-plugins.interfaces: + * docs/plugins/gst-plugins-base-plugins.prerequisites: + * docs/plugins/inspect/plugin-adder.xml: + * docs/plugins/inspect/plugin-alsa.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-audioconvert.xml: + * docs/plugins/inspect/plugin-audiorate.xml: + * docs/plugins/inspect/plugin-audioresample.xml: + * docs/plugins/inspect/plugin-audiotestsrc.xml: + * docs/plugins/inspect/plugin-cdparanoia.xml: + * docs/plugins/inspect/plugin-encoding.xml: + * docs/plugins/inspect/plugin-gio.xml: + * docs/plugins/inspect/plugin-libvisual.xml: + * docs/plugins/inspect/plugin-ogg.xml: + * docs/plugins/inspect/plugin-pango.xml: + * docs/plugins/inspect/plugin-playback.xml: + * docs/plugins/inspect/plugin-subparse.xml: + * docs/plugins/inspect/plugin-tcp.xml: + * docs/plugins/inspect/plugin-theora.xml: + * docs/plugins/inspect/plugin-typefindfunctions.xml: + * docs/plugins/inspect/plugin-videoconvert.xml: + * docs/plugins/inspect/plugin-videorate.xml: + * docs/plugins/inspect/plugin-videoscale.xml: + * docs/plugins/inspect/plugin-videotestsrc.xml: + * docs/plugins/inspect/plugin-volume.xml: + * docs/plugins/inspect/plugin-vorbis.xml: + * docs/plugins/inspect/plugin-ximagesink.xml: + * docs/plugins/inspect/plugin-xvimagesink.xml: + * gst-plugins-base.doap: + * win32/common/_stdint.h: + * win32/common/audio-enumtypes.c: + * win32/common/audio-enumtypes.h: + * win32/common/config.h: + * win32/common/video-enumtypes.c: + Release 1.7.2 2016-02-19 10:31:05 +0200 Sebastian Dröge @@ -181,6 +548,35 @@ of the video area. https://bugzilla.gnome.org/show_bug.cgi?id=761251 +2016-02-03 16:28:42 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opus: fix FEC + FEC may only be used when PLC is enabled on the audio decoder, + as it relies on empty buffers to generate audio from the next + buffer. Hooking to the gap events doesn't work as the audio + decoder does not like more buffers output than it sends. + The length of data to generate using FEC from the next packet + is determined by rounding the gap duration to nearest. This + ensures that duration imprecision does not cause quantization + to 2.5 milliseconds less than available. Doing so causes the + Opus API to fail decoding. Such duration imprecision is common + in live cases. + The buffer to consider when determining the length of audio + to be decoded is the previous buffer when using FEC, and the + new buffer otherwise. In the FEC case, this means we determine + the amount of audio from the previous buffer, whether it was + missing or not (and get the data either from this buffer, or + the current one if the previous one was missing). + +2016-02-02 15:20:48 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: fix wrong buffer being checked for missing data + This caused a decoding error if the resulting (wrong) buffer size + was passed to the Opus decoding API. + https://bugzilla.gnome.org/show_bug.cgi?id=758158 + 2016-01-28 13:29:39 +0100 Sebastian Dröge * gst/audiorate/gstaudiorate.c: @@ -1022,6 +1418,16 @@ * docs/plugins/inspect/plugin-xvimagesink.xml: docs: update to git +2015-12-14 11:09:46 +0900 Vineeth TM + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.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 + 2015-12-14 13:59:02 -0500 Nicolas Dufresne * ext/alsa/gstalsasrc.c: @@ -1426,6 +1832,12 @@ * gst-libs/gst/tag/id3v2.c: tags: id3: make sure to register private-id3v2-frame tag before using it +2015-11-17 15:23:17 -0800 Reynaldo H. Verdejo Pinochet + + * ext/opus/gstopusenc.c: + Remove unnecessary NULL checks before g_free() + g_free() is NULL-safe + 2015-11-17 17:07:37 +0100 Ognyan Tonchev * gst-libs/gst/rtsp/gstrtspconnection.c: @@ -1468,6 +1880,14 @@ in_samples is >= 0 is never going to be false. Removing it. CID 1338689 +2015-11-12 12:21:54 +0000 Luis de Bethencourt + + * ext/opus/gstopusenc.c: + opusenc: avoid potential overflow expression + The result of the two expressions will be promoted to guint64 anyway, + perform all the arithmetic in 64 bits to avoid potential overflows. + CID 1338690, CID 1338691 + 2015-11-11 14:44:55 +0900 Vineeth TM * tests/check/libs/video.c: @@ -1701,6 +2121,14 @@ done. Also clarify this in the documentation. API: gst_audio_channel_get_fallback_mask() +2015-11-05 12:11:19 +0100 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Update sink pad templates + We always require the channel-mapping-field. If it's 0 we require nothing + else, otherwise we need channels, stream-count and coupled count to be + available. + 2015-11-05 11:34:07 +0100 Thibault Saunier * gst/volume/gstvolume.c: @@ -1738,6 +2166,86 @@ configurations https://bugzilla.gnome.org/show_bug.cgi?id=681447 +2015-11-04 00:12:52 +0200 Sebastian Dröge + + * tests/check/elements/opus.c: + opus: Remove invalid unit test + Opus headers should never be in-band, so don't test for correct + handling of that. + +2015-11-04 00:12:22 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Create an empty taglist if there is none + There always have to be 2 buffers in the streamheaders, even if + the comment buffer is basically empty. + +2015-11-03 14:50:53 +0200 Sebastian Dröge + + * ext/opus/Makefile.am: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + opus: Add proper support for multichannel audio + https://bugzilla.gnome.org/show_bug.cgi?id=757152 + +2015-11-02 17:33:53 +0200 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Handle GstAudioClippingMeta instead of the pre-skip field in the OpusHead + oggdemux is outputting the meta now, and only outputs if it should really + apply to the current buffer. Previously we would skip N samples also if we + started the decoder in the middle of the stream. + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-11-02 16:52:28 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Add GstAudioClippingMeta to buffers that need to be clipped + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-11-02 10:30:52 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Disable granule position calculations by the base class + It is doing the wrong thing because of the Opus pre-skip: while the timestamps + are shifted by the pre-skip, the granule positions are not shifted. + oggmux is doing the right thing here already. + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-10-31 15:02:50 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Add some FIXME comments about calculating padding with LPC + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-10-30 20:57:37 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: Encode exactly the amount of samples we got as input and put correct timestamps on it + The first frame has lookahead less samples, the last frame might have some + padding or we might have to encode another frame of silence to get all our + input into the encoded data. + This is because of a) the lookahead at the beginning of the encoding, which + shifts all data by that amount of samples and b) the padding needed to fill + the very last frame completely. + Ideally we would use LPC to calculate something better than silence for the + padding to make the encoding as smooth as possible. + With this we get exactly the same amount of samples again in an + opusenc ! opusdec pipeline. + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-10-30 20:47:20 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + opusenc: Put lookahead/pre-skip into the OpusHead header + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + 2015-11-03 16:51:47 +0200 Sebastian Dröge * ext/ogg/gstoggstream.c: @@ -1942,6 +2450,17 @@ audiofilter: Clip input buffers to the segment before handling them https://bugzilla.gnome.org/show_bug.cgi?id=757068 +2015-11-01 23:34:32 +0200 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Assume 48kHz if no sample rate is given in the header + +2015-10-30 20:59:41 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Place 48kHz first in the caps + For all the other sample rates the encoder will have to resample internally. + 2015-11-01 23:05:10 +0000 Tim-Philipp Müller * gst/audioconvert/gstaudioconvertorc-dist.c: @@ -2740,6 +3259,15 @@ Thanks to John Chang for reporting. https://bugzilla.gnome.org/show_bug.cgi?id=755098 +2015-09-15 15:39:11 -0300 Thiago Santos + + * ext/opus/gstopusdec.c: + opusdec: remove check for number of channels + opus decoder can convert from different number of channels, no + need to check, just let it negotiate and create a new decoder if + needed. + https://bugzilla.gnome.org/show_bug.cgi?id=755059 + 2015-09-15 15:26:44 +0100 Tim-Philipp Müller * gst-libs/gst/app/gstappsink.c: @@ -2760,6 +3288,18 @@ When context creation fails, error is getting leaked. https://bugzilla.gnome.org/show_bug.cgi?id=754973 +2015-09-11 11:22:35 +0200 Miguel París Díaz + + * ext/opus/gstopusenc.c: + opusenc: improve deprecated properties docs + https://bugzilla.gnome.org/show_bug.cgi?id=754819 + +2015-09-11 11:11:09 +0200 Miguel París Díaz + + * ext/opus/gstopusenc.c: + opusenc: do not throw g_warning when getting deprecated properties + https://bugzilla.gnome.org/show_bug.cgi?id=754819 + 2015-09-11 23:28:37 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: @@ -3119,6 +3659,14 @@ set the GError, so the error can be printed and notified. https://bugzilla.gnome.org/show_bug.cgi?id=753701 +2015-08-16 07:18:34 -0300 Thiago Santos + + * ext/opus/gstopusenc.c: + audioencoders: use template subset check for accept-caps + It is faster than doing a query that propagates downstream and + should be enough + Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc + 2015-08-17 11:18:25 +0900 Vineeth TM * tools/gst-discoverer.c: @@ -3219,6 +3767,15 @@ We were using the wrong variable ... CID #1316477 +2015-08-15 12:58:40 -0300 Thiago Santos + + * ext/opus/gstopusdec.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 + 2015-05-04 11:19:28 +0200 Edward Hervey * gst/playback/gstdecodebin2.c: @@ -3544,6 +4101,13 @@ the format ourselves and thus would have to drop the overlays. Otherwise we should prefer what downstream wants here. +2015-07-27 18:39:13 +0530 Nirbheek Chauhan + + * ext/opus/gstopuscommon.c: + opuscommon: Use GString instead of snprintf for concating + Safer, easier to understand, and more portable. Also, skip + all this if the log level is too low. + 2015-07-23 15:28:42 -0400 Nicolas Dufresne * ext/pango/gstbasetextoverlay.c: @@ -3686,6 +4250,19 @@ merged into a new GstVideoOverlayComposition and passed down downstream. https://bugzilla.gnome.org/show_bug.cgi?id=751157 +2015-04-20 15:04:56 +0200 Carlos Rafael Giani + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: Fix PLC frame size calculations + Previously, PLC frames always had a length of 120ms, which caused audio + quality degradation and synchronization errors. Fix this by calculating an + appropriate length for the PLC frame. + The length must be a multiple of 2.5ms. Calculate a multiple of 2.5ms that + is nearest to the current PLC length. Any leftover PLC length that didn't + make it into this frame is accumulated for the next PLC frame. + https://bugzilla.gnome.org/show_bug.cgi?id=725167 + 2015-07-10 12:49:01 -0400 Nicolas Dufresne * gst-libs/gst/rtp/gstrtpbasedepayload.c: @@ -4392,6 +4969,12 @@ Prevent a double free crash when the demuxer is being finalized. https://bugzilla.gnome.org/show_bug.cgi?id=751000 +2015-06-15 13:43:53 +0200 Mersad Jelacic + + * ext/opus/gstopusenc.c: + opusenc: Add bitrate to the tags + https://bugzilla.gnome.org/show_bug.cgi?id=750992 + 2015-06-19 19:51:25 +0900 Vineeth T M * tools/gst-play.c: @@ -5052,6 +5635,17 @@ * gst-libs/gst/pbutils/codec-utils.c: codec-utils: Add AAC channel configurations 11, 12 and 14 and levels 6 and 7 +2015-06-04 11:54:24 +0200 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: If channel/rate negotiation fails, fall back to stereo and 48kHz + +2015-06-04 11:45:05 +0200 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: gst_structure_fixate_field_nearest_int() only works if the structure has this field + Just set the rate/channels directly if the caps don't have this field. + 2015-06-02 16:14:39 +0200 Edward Hervey * tests/check/generic/clock-selection.c: @@ -5075,6 +5669,13 @@ 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 16:02:37 +0200 Edward Hervey + + * tests/check/elements/opus.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 12:47:50 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: @@ -5718,6 +6319,14 @@ * docs/libs/gst-plugins-base-libs-sections.txt: docs: add new video API to docs +2015-05-04 10:35:55 +0200 Jose Antonio Santos Cadenas + + * ext/opus/gstopusheader.c: + opusheader: Do not include rate in caps if it is 0 + As expressed in gst_opus_header_create_caps, value 0 means unset. + Setting rate value to 0 make negotiation with decoder fail. + https://bugzilla.gnome.org/show_bug.cgi?id=748875 + 2015-05-04 02:18:22 +1000 Jan Schmidt * gst-libs/gst/video/video-info.c: @@ -5795,6 +6404,22 @@ it https://bugzilla.gnome.org/show_bug.cgi?id=747245 +2015-04-28 17:24:04 +0100 Tim-Philipp Müller + + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opus: fix includes and compilation against opus in non-standard prefix + https://bugzilla.gnome.org/show_bug.cgi?id=748594 + +2015-04-28 16:58:21 +0200 Mersad Jelacic + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opus: don't use deprecated gst_buffer_new_and_alloc + Use the helper function available in the base class instead. + https://bugzilla.gnome.org/show_bug.cgi?id=748585 + 2015-04-28 12:01:02 +0200 Wim Taymans * gst-libs/gst/video/video-info.c: @@ -6444,6 +7069,11 @@ older Gtk+ version is present on the system. https://bugzilla.gnome.org/show_bug.cgi?id=747283 +2015-04-03 11:46:12 +0530 Arun Raghavan + + * ext/opus/gstopusenc.c: + opus: Fix incorrect fall-through condition in property getter + 2014-12-09 13:18:42 +0100 Thibault Saunier * gst/videorate/gstvideorate.c: @@ -6685,6 +7315,13 @@ { bias - dither, bias + dither - 1 } https://bugzilla.gnome.org/show_bug.cgi?id=746661 +2015-03-24 15:13:52 +0000 Luis de Bethencourt + + * ext/opus/gstopusenc.c: + opusenc: fall through switch statement + Adding a comment makes coverity happy and quells the issue. + CID 1291629 + 2015-02-16 09:25:03 +1000 Duncan Palmer * gst/playback/gstdecodebin2.c: @@ -6707,6 +7344,52 @@ * gst-libs/gst/allocators/gstfdmemory.c: fdmemory: freed pointer will always be 0 +2015-03-23 13:15:30 +0100 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Set output format immediately after creating the encoder instance + We know the caps by then, there's no need to wait until we actually receive + the first buffer. + +2015-03-23 13:13:35 +0100 Sebastian Dröge + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: Remove another unused variable + +2015-03-23 13:11:42 +0100 Sebastian Dröge + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + * ext/opus/gstopusheader.c: + opusenc: Remove useless headers and header_sent variables from the instance struct + They are only used inside a single function. + +2015-03-23 12:09:25 +0100 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Take channels and sample rate from the caps if we have no stream header + +2015-03-23 12:07:52 +0100 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Reset the decoder if the caps change + +2015-03-23 11:57:09 +0100 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Take output sample rate from the stream headers too + This way we let opusdec do the resampling if needed and don't carry + around buffers with a too high sample rate if not required. + While Opus always uses 48kHz internally, this information from the + header specifies which frequencies are safe to drop. + +2015-03-23 11:56:09 +0100 Sebastian Dröge + + * ext/opus/gstopusheader.c: + opusheader: Put number of channels and sample rate into the caps + https://bugzilla.gnome.org/show_bug.cgi?id=746617 + 2015-03-20 17:45:03 +0900 Wonchul Lee * ext/ogg/gstoggdemux.c: @@ -6977,6 +7660,14 @@ Add GstVideoChroma, GstVideoDither, GstVideoScaler and friends to the docs. Remove and clean up a few obsolete/deleted refs and typos +2015-03-12 12:49:40 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: replace cbr and constrained-vbr properties with an enum + It was deemed confusing before. + https://bugzilla.gnome.org/show_bug.cgi?id=744909 + 2015-03-12 12:17:11 +0000 Sebastian Dröge * gst/playback/gstplaybin2.c: @@ -7248,6 +7939,15 @@ don't reuse the source memory directly. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745207 +2015-03-04 09:24:27 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: fix latency query in FEC case + The max latency parameter is "the maximum time an element + synchronizing to the clock is allowed to wait for receiving all + data for the current running time" (docs/design/part-latency.txt). + https://bugzilla.gnome.org/show_bug.cgi?id=744338 + 2015-03-03 16:36:20 -0500 Nicolas Dufresne * ext/pango/gstbasetextoverlay.c: @@ -7652,6 +8352,13 @@ * gst/playback/gsturidecodebin.c: uridecodebin: Let the latency query fail if one of the source queries fails +2015-02-18 17:41:25 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + opusenc: Remove g_warnings() for the deprecated audio property + Otherwise there are g_warnings() already when just using gst-inspect or + dumping a pipeline graph. + 2015-02-18 11:34:15 +0000 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: @@ -7835,6 +8542,14 @@ handling. https://bugzilla.gnome.org/show_bug.cgi?id=744106 +2015-02-11 14:16:21 +0100 Sebastian Dröge + + * ext/opus/gstopusdec.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 13:32:25 +0100 Wim Taymans * gst-libs/gst/video/video-converter.c: @@ -8265,6 +8980,14 @@ flushing to the caller, rather than emit a flow error. https://bugzilla.gnome.org/show_bug.cgi?id=722442 +2015-01-28 16:43:59 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: change audio property to audio-type + This is now an enum with values generic (default) and voice. + https://bugzilla.gnome.org/show_bug.cgi?id=740891 + 2015-01-28 17:44:57 +0100 Sebastian Dröge * gst-libs/gst/video/video-converter.c: @@ -8831,6 +9554,12 @@ Automatic update of common submodule From ef1ffdc to f2c6b95 +2014-12-17 21:52:13 -0300 Thiago Santos + + * ext/opus/gstopusenc.c: + opusenc: plug ref leak of template caps + the pad template caps is already a new ref. No need to copy. + 2014-12-17 19:14:38 -0300 Thiago Santos * gst-libs/gst/audio/gstaudioencoder.c: @@ -10430,6 +11159,14 @@ into a rectangle in the destination frame. Add an option to add a border and border color. +2014-06-10 09:33:40 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: 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. + 2014-06-05 14:50:15 +0100 Vincent Penquerc'h * ext/vorbis/gstvorbisenc.c: @@ -11399,6 +12136,11 @@ Use the object lock to avoid concurrent processing which leads to small disasters (assertions or crashes) +2014-09-10 17:24:39 +0100 Tim-Philipp Müller + + * ext/opus/gstopusdec.c: + Fix up one-element lists in template caps + 2014-09-09 11:37:26 +0200 Ognyan Tonchev * gst-libs/gst/rtsp/gstrtspconnection.c: @@ -11814,6 +12556,12 @@ are really flowing. Unit test updated accordingly https://bugzilla.gnome.org/show_bug.cgi?id=650652 +2014-08-08 14:08:19 +0200 Sebastian Rasmussen + + * ext/opus/gstopusenc.c: + opusenc: Unref pad template caps after usage + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734517 + 2014-08-11 10:57:43 +0200 Sebastian Dröge * gst/playback/gstdecodebin2.c: @@ -11880,6 +12628,12 @@ check generically whether it's a derived GstMemory type, as boxed types don't allowe derivation. +2014-08-09 14:24:59 +0200 Sebastian Rasmussen + + * ext/opus/gstopusheader.c: + opus: Improve annotation of internal function + https://bugzilla.gnome.org/show_bug.cgi?id=734543 + 2014-08-09 14:14:48 +0200 Sebastian Rasmussen * gst-libs/gst/audio/gstaudioencoder.c: @@ -12415,6 +13169,15 @@ dmabuf: share the mapping with shared copies of the memory https://bugzilla.gnome.org/show_bug.cgi?id=730441 +2014-07-10 15:52:46 +0100 Philip Withnall + + * ext/opus/gstopusheader.c: + opus: Fix a double-unref in the Opus header code + The headers were never getting reffed when being added to the headers + list, which is later unreffed-and-freed by the caller (e.g. + gst_opus_parse_parse_frame()). + https://bugzilla.gnome.org/show_bug.cgi?id=733013 + 2014-07-11 08:51:58 +0200 Sebastian Dröge * po/vi.po: @@ -13051,6 +13814,15 @@ from gstutils that does the correct combination of flow returns. https://bugzilla.gnome.org/show_bug.cgi?id=709224 +2014-05-10 18:32:28 +0200 Miguel París Díaz + + * ext/opus/gstopusenc.c: + opusenc: Use aux vars to minimize critical region + This avoid dead lock between gst_audio_encoder_finish_frame() and + gst_opus_enc_get_property(). + Also, now bytes var is set into protected section. + https://bugzilla.gnome.org/show_bug.cgi?id=729882 + 2014-05-23 19:21:35 +0100 Tim-Philipp Müller * tools/gst-play.c: @@ -14058,6 +14830,12 @@ * gst/playback/gstdecodebin2.c: decodebin: In adaptive streaming mode, only have a fixed buffer limit for the non-buffering multiqueue +2014-04-09 11:02:00 +0100 Vincent Penquerc'h + + * ext/opus/gstopusheader.c: + opus: add missing va_end in variadic function + Coverity 1139944 + 2014-04-08 15:43:50 +0200 Wim Taymans * gst-libs/gst/sdp/gstsdpmessage.c: @@ -14863,6 +15641,11 @@ audiosrc: Fix typo in docs We read *from* the audio device, not to it. +2014-02-08 20:08:29 +0100 Sebastian Dröge + + * tests/check/elements/opus.c: + opus: Remove unused variable from unit test + 2014-02-08 17:11:54 +0100 Sebastian Dröge * tests/check/elements/videoscale.c: @@ -15771,6 +16554,14 @@ makes it easier to use the reserved bits of the structs later. https://bugzilla.gnome.org/show_bug.cgi?id=720810 +2013-12-27 14:29:46 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: increase max payload size to 4000 bytes + 1275 is the maximum size of a frame, but the encoder may return + up to 3 frames, and we need a few extra bytes for TOC, etc. We + use 4000, which is a bit more, and suggested in the libopus docs. + 2013-12-20 19:48:06 -0300 Reynaldo H. Verdejo Pinochet * gst-libs/gst/audio/gstaudiobasesrc.c: @@ -16136,6 +16927,11 @@ clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success. Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684 +2013-12-05 12:04:59 +0100 Sebastian Dröge + + * ext/opus/gstopusdec.c: + opusdec: Require caps to be set before any data processing + 2013-12-05 11:39:07 +0100 Sebastian Dröge * ext/theora/gsttheoradec.c: @@ -19910,6 +20706,11 @@ Automatic update of common submodule From 5edcd85 to 098c0d7 +2013-05-15 10:18:01 +0200 Sebastian Dröge + + * tests/check/elements/opus.c: + opus: Fix event handling in unit test + 2013-05-15 09:26:56 +0200 Sebastian Dröge * gst-libs/gst/audio/audio-info.c: @@ -20521,6 +21322,11 @@ * gst-libs/gst/app/Makefile.am: app: Don't use $(GST_PLUGIN_LIBTOOLFLAGS) for real libraries +2012-10-24 12:16:39 +0200 Sebastian Dröge + + * ext/opus/Makefile.am: + gst: Add better support for static plugins + 2012-10-24 12:10:44 +0200 Sebastian Dröge * configure.ac: @@ -21071,6 +21877,15 @@ uridecodebin: remove commented code This is dead since ~6 years. +2013-03-27 22:24:03 +0000 Tim-Philipp Müller + + Merge SBC decoder and encoder from bluez + https://bugzilla.gnome.org/show_bug.cgi?id=690582 + +2007-08-23 19:12:23 +0000 Marcel Holtmann + + sbc: Add SBC encoder and decoder skeletons for GStreamer + 2013-03-12 08:10:23 +0100 Stefan Sauer * gst/audiotestsrc/gstaudiotestsrc.c: @@ -21819,6 +22634,14 @@ Decoders that get unparsed input are internally leaking nearly every incoming buffer. This checks that case. +2013-02-11 11:06:32 +0100 Wim Taymans + + * ext/opus/gstopusdec.c: + opusdec: clear the state of the decoder + Set the channels and rate back to their default values in _stop because they + are used to renegotiate when needed. + See https://bugzilla.gnome.org/show_bug.cgi?id=692950 + 2013-02-09 16:50:05 +0000 Tim-Philipp Müller * tests/check/elements/streamsynchronizer.c: @@ -21980,6 +22803,12 @@ Automatic update of common submodule From a942293 to 2de221c +2013-01-28 14:12:56 +0000 Tim-Philipp Müller + + * ext/opus/gstopusenc.c: + opusenc: fix crash when setting "cbr" property when encoder is not running yet + https://bugzilla.gnome.org/show_bug.cgi?id=692698 + 2013-01-27 09:45:59 +0530 B.Prathibha * tests/check/pipelines/basetime.c: @@ -22255,6 +23084,12 @@ We need to initialize this variable because we can't be sure that the subclass will set it. +2012-12-18 16:56:28 +0100 Thijs Vermeir + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opus: use appropriate printf format for gsize + 2012-12-18 15:34:42 +0100 Thijs Vermeir * ext/vorbis/gstvorbisdec.c: @@ -23062,6 +23897,21 @@ rtsp: fix GstRTSPMessage g-i annotations for out parameters https://bugzilla.gnome.org/show_bug.cgi?id=687620 +2012-11-03 20:38:00 +0000 Tim-Philipp Müller + + * ext/opus/gstopus.c: + * ext/opus/gstopuscommon.c: + * ext/opus/gstopuscommon.h: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + * tests/check/elements/opus.c: + Fix FSF address + https://bugzilla.gnome.org/show_bug.cgi?id=687520 + 2012-11-03 23:05:09 +0000 Tim-Philipp Müller * COPYING: @@ -23786,6 +24636,14 @@ * win32/common/config.h: Back to feature development +2012-10-24 23:40:20 +0200 Carlos Rafael Giani + + * ext/opus/gstopusdec.c: + opusdec: fixed buffer unmapping bug + When the decoder received a NULL buffer, it tried to + unmap a not mapped buffer. + https://bugzilla.gnome.org/show_bug.cgi?id=686829 + === release 1.0.2 === 2012-10-25 00:54:24 +0100 Tim-Philipp Müller @@ -24050,6 +24908,14 @@ * gst-libs/gst/audio/gstaudiocdsrc.h: audiocdsrc: mention TOCs in docs +2012-10-17 17:34:26 +0100 Tim-Philipp Müller + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.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 16:54:14 +0100 Tim-Philipp Müller * ext/theora/gsttheoradec.c: @@ -24598,6 +25464,11 @@ * ext/ogg/gstoggmux.c: oggmux: send stream-start event +2012-09-20 18:42:50 -0400 Olivier Crête + + * ext/opus/gstopus.c: + opusenc: Rank as Primary + 2012-09-22 16:07:35 +0100 Tim-Philipp Müller * common: @@ -24802,6 +25673,12 @@ * tests/check/libs/xmpwriter.c: replace gst_tag_list_free with gst_tag_list_unref +2012-09-14 17:08:49 +0200 Mark Nauwelaerts + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + replace gst_element_class_set_details_simple with gst_element_class_set_metadata + 2012-09-14 17:02:53 +0200 Mark Nauwelaerts * ext/theora/gsttheoradec.c: @@ -25027,6 +25904,17 @@ * tests/check/elements/videotestsrc.c: tests: port to the new GLib thread API +2012-09-12 09:10:35 +0200 Peter Korsgaard + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opus + jpegformat: unbreak non-debug build + opus + jpegformat plugin builds fail when gstreamer is configured with + --disable-gst-debug as they are checking the GST_DISABLE_DEBUG symbol + instead of GST_DISABLE_GST_DEBUG. + Signed-off-by: Peter Korsgaard + https://bugzilla.gnome.org/show_bug.cgi?id=683850 + 2012-09-12 10:12:25 +0200 Wim Taymans * tests/check/elements/videoscale.c: @@ -25042,6 +25930,16 @@ video: Add support for 4:2:2 10 bit video. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683838 +2012-09-11 18:02:28 -0400 Olivier Crête + + * tests/check/elements/opus.c: + test: Flush opus encoder between tests + +2012-09-11 18:01:58 -0400 Olivier Crête + + * tests/check/elements/opus.c: + test: Flush opus encoder between tests + 2012-09-11 20:53:16 +0100 Tim-Philipp Müller * gst-libs/gst/tag/gsttagdemux.c: @@ -25070,6 +25968,12 @@ ... and therefore will never unblock the other streams. Fixes blocking issue when using playbin suburi feature +2012-09-11 14:31:49 +0200 Mark Nauwelaerts + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: port to the new GLib thread API + 2012-09-11 12:53:01 +0200 Wim Taymans * gst-libs/gst/video/video-info.c: @@ -26470,6 +27374,12 @@ * gst-libs/gst/riff/riff-read.c: riff: fix build on big endian systems +2012-08-04 16:31:30 +0100 Tim-Philipp Müller + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + gst_tag_list_free -> gst_tag_list_unref + 2012-07-29 00:49:31 -0300 Thiago Santos * gst-libs/gst/app/gstappsrc.c: @@ -28186,6 +29096,21 @@ * gst-libs/gst/video/gstvideodecoder.h: videodecoder: Add GstVideoDecoder::propose_allocation() vfunc +2012-06-15 10:32:39 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: add missing mutex unlock on error path + +2012-06-15 10:24:24 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + * ext/opus/gstopusheader.h: + opus: set author to myself, and update copyright notices + because as slomo noted, in fact pretty much all the code in there is mine. + 2012-06-14 23:08:54 +0100 Tim-Philipp Müller * tests/examples/playback/playback-test.c: @@ -28907,6 +29832,12 @@ * gst-libs/gst/video/video.h: video: add support for premultiplied alpha +2012-05-29 17:24:02 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: read gain from the right place in the header + It's at byte offset 16, not 14. + 2012-05-29 17:48:45 +0200 Wim Taymans * gst/videotestsrc/gstvideotestsrc.c: @@ -29061,6 +29992,11 @@ When we need to add borders, take the pixel stride into account to move to the right horizintal offset. +2012-05-27 23:41:24 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: do not assert on bad header, error out instead + 2012-05-26 19:56:48 +0100 Tim-Philipp Müller * tests/check/libs/tag.c: @@ -29125,6 +30061,24 @@ do not currently support) needs it to be specified in bytes. Thanks to Julien Moutte for pointing this out. +2012-05-24 22:12:56 +0100 Vincent Penquerc'h + + * ext/opus/gstopusheader.c: + opus: reject major version number above what we grok + +2012-05-24 21:58:44 +0100 Vincent Penquerc'h + + * ext/opus/gstopusheader.c: + opus: bump written version from 0 to 0x01 + as per the spec update at https://wiki.xiph.org/OggOpus#ID_Header + +2012-04-30 14:40:02 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: fix lost packet handling for FEC/PLC + The base audio decoder sends zero size packets, not NULL buffers, + to signal dropped packets. + 2012-05-24 13:43:06 +0200 Wim Taymans * gst/playback/gstplaybin2.c: @@ -29815,6 +30769,11 @@ * gst-libs/gst/video/gstvideodecoder.h: videodecoder: Change configure_buffer_pool() vfunc to decide_allocation() with same semantics as in basetransform +2012-04-04 11:51:28 +0200 Edward Hervey + + * ext/opus/gstopusheader.c: + opus: Handle GstByteWriter return values + 2012-04-19 14:41:40 +0200 Stefan Sauer * tests/check/pipelines/streamheader.c: @@ -30459,6 +31418,17 @@ pbutils: Link against internal gst video Link pbutils and encoding tests against internal version of libgstvideo. +2012-04-10 17:24:05 +0200 Mark Nauwelaerts + + * tests/check/elements/opus.c: + tests: port some more to 1.0 + +2012-04-10 17:22:44 +0200 Mark Nauwelaerts + + * ext/opus/gstopusdec.c: + opusdec: tweak caps negotiation + ... so as to avoid leaking caps or manipulating NULL caps. + 2012-04-10 00:45:16 +0100 Tim-Philipp Müller * ext/alsa/gstalsamixerelement.c: @@ -30566,17 +31536,33 @@ * po/zh_CN.po: po: update for new translatable strings +2012-04-06 14:52:12 +0200 Mark Nauwelaerts + + Merge remote-tracking branch 'origin/0.10' + Conflicts: + gst/h264parse/gsth264parse.c + gst/videoparsers/gsth264parse.c + 2012-04-06 10:54:04 +0200 Sebastian Dröge * gst/playback/gstdecodebin.c: playback: Remove gstdecodebin.c, which is nowaday unused anyway +2012-04-05 17:15:11 -0400 Thibault Saunier + + Merge remote-tracking branch 'origin/0.10' + 2012-04-05 18:42:42 +0200 Sebastian Dröge * common: Automatic update of common submodule From 7fda524 to 464fe15 +2012-04-05 18:02:56 +0200 Sebastian Dröge + + * ext/opus/gstopus.c: + gst: Update for GST_PLUGIN_DEFINE() API changes + 2012-04-05 15:11:05 +0200 Sebastian Dröge * ext/alsa/gstalsaplugin.c: @@ -30780,6 +31766,15 @@ * win32/common/config.h: gst: Update versioning +2012-04-04 14:41:22 +0200 Sebastian Dröge + + * ext/opus/Makefile.am: + gst: Update versioning + +2012-04-04 12:06:08 +0200 Sebastian Dröge + + Merge remote-tracking branch 'origin/0.10' + 2012-04-04 09:33:30 +0200 Wim Taymans * gst-libs/gst/rtp/gstrtpbuffer.c: @@ -30822,6 +31817,12 @@ * gst/videoconvert/gstvideoconvert.c: videoconvert: plug caps leak +2012-04-02 15:31:38 +0200 Sebastian Dröge + + Merge remote-tracking branch 'origin/0.10' + Conflicts: + gst/mpegtsdemux/tsdemux.c + 2012-04-02 14:23:16 +0200 Mark Nauwelaerts * gst-libs/gst/audio/gstaudiodecoder.h: @@ -30924,6 +31925,11 @@ * tests/examples/app/appsrc-stream2.c: update for buffer api change +2012-03-30 17:09:34 +0200 Mark Nauwelaerts + + * ext/opus/gstopusenc.c: + opusenc: fixup merge + 2012-03-30 16:56:45 +0200 Mark Nauwelaerts * tests/check/elements/appsrc.c: @@ -30993,6 +31999,11 @@ * gst-libs/gst/audio/gstaudiodecoder.h: audiodecoder: Rename ::event() to ::sink_event() and add ::src_event() +2012-03-30 12:22:48 +0200 Sebastian Dröge + + * ext/opus/gstopusenc.c: + ext: Update for GstAudioEncoder API changes + 2012-03-30 12:13:40 +0200 Edward Hervey * gst-libs/gst/tag/gstexiftag.c: @@ -31054,6 +32065,10 @@ Which is telling more about what this actually does and is more consistent with the video base classes. +2012-03-29 18:04:36 +0200 Sebastian Dröge + + Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-bad + 2012-03-29 17:41:55 +0200 Edward Hervey * tests/check/libs/libsabi.c: @@ -31070,6 +32085,89 @@ * docs/design/draft-hw-acceleration.txt: design: First go at hardware-acceleration design doc +2012-03-29 17:41:53 +0200 Sebastian Dröge + + Merge remote-tracking branch 'origin/0.10' + Conflicts: + NEWS + RELEASE + common + configure.ac + docs/libs/gst-plugins-bad-libs-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-adpcmdec.xml + docs/plugins/inspect/plugin-adpcmenc.xml + docs/plugins/inspect/plugin-assrender.xml + docs/plugins/inspect/plugin-audiovisualizers.xml + docs/plugins/inspect/plugin-autoconvert.xml + docs/plugins/inspect/plugin-bayer.xml + docs/plugins/inspect/plugin-bz2.xml + docs/plugins/inspect/plugin-camerabin2.xml + docs/plugins/inspect/plugin-celt.xml + docs/plugins/inspect/plugin-dataurisrc.xml + docs/plugins/inspect/plugin-debugutilsbad.xml + docs/plugins/inspect/plugin-dtmf.xml + docs/plugins/inspect/plugin-dtsdec.xml + docs/plugins/inspect/plugin-dvbsuboverlay.xml + docs/plugins/inspect/plugin-dvdspu.xml + docs/plugins/inspect/plugin-faac.xml + docs/plugins/inspect/plugin-faad.xml + docs/plugins/inspect/plugin-gsm.xml + docs/plugins/inspect/plugin-h264parse.xml + docs/plugins/inspect/plugin-mms.xml + docs/plugins/inspect/plugin-modplug.xml + docs/plugins/inspect/plugin-mpeg2enc.xml + docs/plugins/inspect/plugin-mpegdemux2.xml + docs/plugins/inspect/plugin-mpegtsdemux.xml + docs/plugins/inspect/plugin-mpegvideoparse.xml + docs/plugins/inspect/plugin-mplex.xml + docs/plugins/inspect/plugin-pcapparse.xml + docs/plugins/inspect/plugin-rawparse.xml + docs/plugins/inspect/plugin-rtpmux.xml + docs/plugins/inspect/plugin-rtpvp8.xml + docs/plugins/inspect/plugin-scaletempo.xml + docs/plugins/inspect/plugin-schro.xml + docs/plugins/inspect/plugin-sdp.xml + docs/plugins/inspect/plugin-segmentclip.xml + docs/plugins/inspect/plugin-shm.xml + docs/plugins/inspect/plugin-videomaxrate.xml + docs/plugins/inspect/plugin-videoparsersbad.xml + docs/plugins/inspect/plugin-vp8.xml + docs/plugins/inspect/plugin-y4mdec.xml + ext/celt/gstceltdec.c + ext/dts/gstdtsdec.c + ext/modplug/gstmodplug.cc + ext/opus/gstopusenc.c + gst-libs/gst/video/gstbasevideocodec.c + gst-libs/gst/video/gstbasevideocodec.h + gst-libs/gst/video/gstbasevideodecoder.c + gst-libs/gst/video/gstbasevideodecoder.h + gst-libs/gst/video/gstbasevideoencoder.c + gst-libs/gst/video/gstbasevideoencoder.h + gst/adpcmdec/Makefile.am + gst/audiovisualizers/gstbaseaudiovisualizer.c + gst/h264parse/gsth264parse.c + gst/mpegdemux/mpegtsparse.c + gst/mpegtsdemux/mpegtsbase.c + gst/mpegtsdemux/mpegtspacketizer.c + gst/mpegtsdemux/mpegtsparse.c + gst/mpegtsdemux/tsdemux.c + gst/mpegtsdemux/tsdemux.h + gst/mxf/mxfdemux.c + gst/rawparse/gstaudioparse.c + gst/videoparsers/gsth263parse.c + gst/videoparsers/gsth264parse.c + sys/d3dvideosink/d3dvideosink.c + sys/decklink/gstdecklinksink.cpp + sys/dvb/gstdvbsrc.c + sys/shm/gstshmsrc.c + sys/vdpau/h264/gstvdph264dec.c + sys/vdpau/mpeg/gstvdpmpegdec.c + tests/examples/opencv/gst_element_print_properties.c + win32/common/config.h + 2012-03-29 17:14:48 +0200 Mark Nauwelaerts * gst-libs/gst/rtp/gstrtpbasepayload.c: @@ -31100,6 +32198,12 @@ * gst/gdp/gstgdppay.c: update for buffer changes +2012-03-27 15:13:24 -0400 Olivier Crête + + * ext/opus/gstopus.c: + opus: Rank rtp pay/depay + This way they can be auto-plugged. + 2012-03-27 18:16:53 +0200 Mark Nauwelaerts * gst-libs/gst/tag/gsttagmux.c: @@ -31624,6 +32728,11 @@ * tests/check/elements/decodebin2.c: tests: update for caps api changes +2012-03-12 17:06:11 +0100 Wim Taymans + + * ext/opus/gstopusdec.c: + opusdec: fix for caps api change + 2012-03-12 16:39:14 +0200 Sreerenj Balachandran * configure.ac: @@ -31773,6 +32882,11 @@ buffers. Users of the bufferpool should do this manually based on the results of the allocation query. +2012-03-08 11:32:27 +0100 Wim Taymans + + * tests/check/elements/opus.c: + tests: fix more caps + 2012-03-08 10:59:48 +0100 Wim Taymans * tests/check/elements/videoscale.c: @@ -31785,6 +32899,16 @@ Simply intersect the format with the supported formats to make the code deal with lists of formats. +2012-03-07 17:14:29 +0100 Mark Nauwelaerts + + * ext/opus/gstopuscommon.c: + * ext/opus/gstopuscommon.h: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + opus: port to updated 0.11 + 2012-03-07 12:45:46 +0000 Tim-Philipp Müller * ext/ogg/gstoggdemux.c: @@ -31801,6 +32925,22 @@ common: update common module For new check-norepeat target. +2012-03-07 12:59:28 +0100 Mark Nauwelaerts + + * ext/opus/gstopusenc.c: + opusenc: only request and process 1 frame at a time + ... since it is specified in _finish_frame that input buffer may be invalidated + after calling it, and is as such not reliably available for further encoding. + Also, requesting or allowing several frames is only useful if subclass intends + to process these "in 1 run" (as in, 1 output buffer), not for having another + (inner) loop in subclass where the baseclass one will do just fine. + +2012-03-07 12:55:43 +0100 Mark Nauwelaerts + + * ext/opus/gstopusenc.c: + opusenc: configure baseclass requested samples really in samples + ... as opposed to bytes. + 2012-03-07 09:04:18 +0100 Edward Hervey * win32/common/libgstaudio.def: @@ -32281,6 +33421,11 @@ * tests/icles/test-colorkey.c: Suppress deprecation warnings in selected files, for g_value_array_* mostly +2012-02-27 13:13:14 +0100 Wim Taymans + + * ext/opus/gstopusenc.c: + audioencoders: chain up to parent event handler + 2012-02-27 13:08:36 +0100 Wim Taymans * gst-libs/gst/audio/gstaudioencoder.c: @@ -32714,6 +33859,12 @@ happy. https://bugzilla.gnome.org/show_bug.cgi?id=670548 +2012-02-21 10:06:16 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + gst/colorspace/colorspace.c + 2012-02-21 10:05:20 +0100 David Schleef * gst/videoconvert/videoconvert.c: @@ -32747,6 +33898,13 @@ videoconvert: clamp intermediates when dithering Port from the colorspace plugin in -bad. +2012-02-20 16:07:50 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/opus/gstopusparse.c + gst/colorspace/colorspace.c + 2012-02-20 15:29:49 +0100 Sebastian Dröge * tests/examples/seek/seek.c: @@ -32940,6 +34098,19 @@ * win32/common/libgstaudio.def: defs: update +2012-02-17 09:01:56 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2012-02-16 14:33:20 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + gst/mpegtsdemux/mpegtsbase.c + gst/mpegtsdemux/mpegtspacketizer.c + gst/mpegtsdemux/tsdemux.c + gst/mve/gstmvedemux.c + 2012-02-16 14:23:28 +0100 Wim Taymans Merge branch 'master' into 0.11 @@ -32969,6 +34140,10 @@ * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: assert some more that subclass parsed frame has proper len +2012-02-15 17:14:34 +0100 Mark Nauwelaerts + + Merge branch 'master' into 0.11 + 2012-02-15 13:42:19 +0100 Wim Taymans * gst-libs/gst/audio/gstaudiodecoder.c: @@ -33026,6 +34201,10 @@ tagdemux: fix src query handler We don't want to blindly forward all queries. +2012-02-14 11:19:04 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2012-02-14 10:50:45 +0100 Wim Taymans * tests/check/elements/decodebin.c: @@ -33243,6 +34422,24 @@ method to get to the padtemplates. Fixes 'GstTagDemux subclass GstTagDemux did not set up a {sink,src} pad template' warnings. +2012-02-10 16:46:50 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/chromaprint/gstchromaprint.c + ext/mpeg2enc/Makefile.am + ext/voaacenc/gstvoaacenc.c + gst/dvbsuboverlay/gstdvbsuboverlay.c + gst/mpegtsdemux/mpegtsbase.c + gst/sdp/gstsdpdemux.c + gst/videoparsers/gsth264parse.c + sys/d3dvideosink/d3dvideosink.c + tests/examples/camerabin/gst-camera-perf.c + tests/examples/camerabin/gst-camerabin-test.c + tests/examples/camerabin2/gst-camerabin2-test.c + tests/examples/mxf/mxfdemux-structure.c + tests/examples/scaletempo/demo-main.c + 2012-02-10 15:41:36 +0100 Wim Taymans * tests/check/elements/videoscale.c: @@ -33495,6 +34692,14 @@ audioencoder: don't unref caps parameter Fix refcounting on incomming caps to make sure we don't unref it too much. +2012-02-03 00:50:33 +0000 Tim-Philipp Müller + + * ext/opus/Makefile.am: + build: fix CFLAGS order and LIBS order + _BAD_CFLAGS should always come first, then GST_PLUGINS_BASE_CFLAGS, + then GST_BASE_CFLAGS then GST_CFLAGS. Same for libs: first plugins + base libs, then GST_BASE_LIB then GST_LIBS. + 2012-01-07 23:09:23 -0500 Ryan Lortie * autogen.sh: @@ -33743,6 +34948,12 @@ * sys/v4l/v4lsrc_calls.c: v4l: include the glib compatiblity header for the deprecated mutex API +2012-01-27 14:49:58 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopusenc.c: + plenty: fixup glib deprecations + 2012-01-27 15:12:25 +0100 Sebastian Dröge Merge branch 'master' into 0.11 @@ -33864,6 +35075,15 @@ Automatic update of common submodule From c463bc0 to 7fda524 +2012-01-25 13:22:43 +0100 Sebastian Dröge + + Merge branch 'master' into 0.11 + Conflicts: + configure.ac + ext/kate/gstkateenc.c + gst/colorspace/colorspace.c + gst/mpegvideoparse/mpegvideoparse.c + 2012-01-25 12:50:44 +0100 Edward Hervey * gst/adder/gstadder.c: @@ -34497,6 +35717,10 @@ it seems pretty certain it's the right thing to do, but I'll put this caveat here in case someone checks in the future. +2012-01-13 00:11:54 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + 2012-01-12 23:35:44 +0000 Tim-Philipp Müller * gst-libs/gst/tag/gstvorbistag.c: @@ -34563,6 +35787,11 @@ freed data with chained and normal files, both with gst-launch playbin2 and Totem. +2012-01-11 13:32:36 +0000 Vincent Penquerc'h + + * tests/check/elements/opus.c: + tests: fix buffer leaks in opus tests + 2012-01-11 12:52:17 +0000 Vincent Penquerc'h * gst-libs/gst/pbutils/gstdiscoverer-types.c: @@ -34657,6 +35886,24 @@ * gst/playback/gststreamsynchronizer.c: streamsynchronizer: Don't unref the parent in the event function +2012-01-10 15:50:37 +0100 Sebastian Dröge + + Merge branch 'master' into 0.11 + Conflicts: + gst/mpegtsdemux/tsdemux.c + gst/videoparsers/gsth264parse.c + tests/check/elements/camerabin2.c + +2012-01-10 13:38:50 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix slist leak + +2012-01-10 13:38:42 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix caps leak + 2012-01-10 13:15:12 +0100 Sebastian Dröge Merge branch 'master' into 0.11 @@ -35279,6 +36526,39 @@ gst/playback/gstsubtitleoverlay.c tests/check/libs/tag.c +2011-12-30 11:49:27 +0100 Edward Hervey + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + tests/examples/camerabin2/Makefile.am + +2011-12-30 11:41:17 +0100 Edward Hervey + + 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 + 2011-12-28 16:25:37 +0100 Edward Hervey * tests/check/libs/video.c: @@ -36424,6 +37704,11 @@ A more robust way would be to find a good place to reinject the headers when a seek fails, but I can't seem to get this to work. +2011-12-15 16:42:20 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opus: fix bad merge (stray unmap, undeclared var) + 2011-12-15 11:01:01 -0300 Thiago Santos * gst-libs/gst/tag/gstexiftag.c: @@ -36537,6 +37822,18 @@ * po/sr.po: po: update translations +2011-12-09 17:25:41 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.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 + 2011-12-09 19:21:09 +0100 Wim Taymans * gst-libs/gst/rtsp/gstrtsptransport.c: @@ -36557,6 +37854,40 @@ * tests/check/libs/video.c: tests: disable composition tests in video unit test for now +2011-12-07 15:13:11 -0200 Danilo Cesar Lemes de Paula + + * ext/opus/Makefile.am: + * ext/opus/gstopus.c: + 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 + +2011-12-08 19:47:55 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + opus: include streams count in caps + https://bugzilla.gnome.org/show_bug.cgi?id=665078 + +2011-12-08 18:45:27 +0000 Vincent Penquerc'h + + * ext/opus/gstopuscommon.c: + * ext/opus/gstopuscommon.h: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + opus: properly create channel mapping tables + There are two of them, unintuitively enough; the one passed + to the encoder should not be the one that gets written to the + file. The former maps the input to an ordering which puts + paired channels first, while the latter moves the channels + to Vorbis order. So add code to calculate both, and we now + have properly paired channels where appropriate. + https://bugzilla.gnome.org/show_bug.cgi?id=665078 + 2011-12-09 15:03:41 +0000 Tim-Philipp Müller * gst-libs/gst/rtp/gstrtpbuffer.h: @@ -36643,6 +37974,24 @@ caps and not random caps, and it's hard to imagine a situation where someone would want to rely on the previous behaviour. +2011-12-07 00:06:11 -0500 Olivier Crête + + * ext/opus/gstopusdec.c: + opusdec: header cleanup + https://bugzilla.gnome.org/show_bug.cgi?id=665078 + +2011-12-07 00:06:11 -0500 Olivier Crête + + * ext/opus/gstopusdec.c: + opusdec: Truncate caps first + https://bugzilla.gnome.org/show_bug.cgi?id=665078 + +2011-11-28 19:47:34 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: default to stereo 48000 Hz if possible when no headers seen + https://bugzilla.gnome.org/show_bug.cgi?id=665078 + 2011-12-06 21:57:32 +0000 Tim-Philipp Müller * gst/videorate/gstvideorate.c: @@ -36967,6 +38316,12 @@ * sys/xvimage/xvimagesink.c: update for basesink event handler changes +2011-11-28 19:38:34 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: guard against decoding 0 samples + https://bugzilla.gnome.org/show_bug.cgi?id=665078 + 2011-12-02 11:10:17 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 @@ -37400,6 +38755,20 @@ * gst-libs/gst/video/video.h: libgstvideo: Add force key unit events +2011-11-28 23:20:58 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + +2011-11-28 23:20:32 +0000 Tim-Philipp Müller + + Merge commit '7521b597f4dc49d8d168f368f0e7ebaf98a72156' into 0.11 + +2011-11-28 23:20:02 +0000 Tim-Philipp Müller + + Merge commit '26d6add9457f00ce8ec13844368466f0e3816e5d' into 0.11 + Conflicts: + ext/rtmp/gstrtmpsink.c + 2011-11-28 21:25:11 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 @@ -37586,6 +38955,13 @@ various: fix pad template leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 +2011-11-28 13:08:27 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + various: fix pad template ref leaks + https://bugzilla.gnome.org/show_bug.cgi?id=662664 + 2011-09-07 16:04:14 +0100 Vincent Penquerc'h * ext/theora/gsttheoradec.c: @@ -37633,12 +39009,25 @@ If highres-timestamp is 0, try lowres and if that fails fallback to system clock timestamps. +2011-11-27 23:33:45 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + 2011-11-27 20:14:08 +0100 Matej Knopp * gst/playback/gsturidecodebin.c: uridecodebin: fix debug message printf format compiler warning https://bugzilla.gnome.org/show_bug.cgi?id=662607 +2011-11-26 15:37:25 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + ext/opus/gstopusdec.c + ext/opus/gstopusparse.c + gst-libs/gst/video/gstbasevideodecoder.c + gst-libs/gst/video/gstbasevideodecoder.h + 2011-11-26 12:12:59 +0000 Tim-Philipp Müller Merge remote-tracking branch 'origin/master' into 0.11 @@ -37654,6 +39043,41 @@ oggmux: set collectpads2 not to wait on sparse streams https://bugzilla.gnome.org/show_bug.cgi?id=663174 +2011-11-25 11:41:19 -0200 Danilo Cesar Lemes de Paula + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opusenc: Fixing "Unused var" compiling error for opus codec + https://bugzilla.gnome.org/show_bug.cgi?id=664815 + +2011-11-25 14:00:18 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + opusenc: only use mono streams for > 2 channels + I'm getting odd results with packing streams into stereo + streams, and using only mono streams is enough in all cases. + +2011-11-25 12:47:42 +0000 Vincent Penquerc'h + + * ext/opus/gstopuscommon.c: + * ext/opus/gstopuscommon.h: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opus: add some more debug information about channel mapping + +2011-11-25 12:40:31 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: do not cause the decoder to apply the channel mapping again + Since we already reorder channels, we do not want to write that + reordering in the header, or the decoder will do it again. + +2011-11-25 12:39:20 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: fix bogus assertion + 2011-11-25 15:35:39 +0100 Josep Torra * gst/playback/gstplaysinkconvertbin.c: @@ -37674,6 +39098,16 @@ * gst/playback/gstplaybin2.c: docs: mention explicitly that playbin2 signals are emitted from a streaming thread +2011-11-25 12:48:58 +0100 Edward Hervey + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + ext/faac/gstfaac.c + ext/opus/gstopusdec.c + ext/opus/gstopusenc.c + gst/audiovisualizers/gstspacescope.c + gst/colorspace/colorspace.c + 2011-11-25 11:11:12 +0100 Sebastian Dröge * gst/playback/gstdecodebin2.c: @@ -37755,6 +39189,45 @@ This new property will force the output framerate to a specific value and can be changed during playback. +2011-11-24 13:38:59 +0000 Vincent Penquerc'h + + * ext/opus/gstopusheader.c: + opus: pre-skip and output gain are little endian, remove reminder note + +2011-11-24 13:29:56 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopuscommon.c: + * ext/opus/gstopuscommon.h: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + opus: multichannel support + +2011-11-23 17:49:58 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opus: switch to multistream API + It's very similar to the basic API, and is a superset ot it, + which will allow encoding and decoding more than 2 channels. + +2011-11-23 17:32:03 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: shuffle supported sample rates to favor 48000 + +2011-11-23 16:36:54 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: remove useless setup field + 2011-11-24 12:38:54 +0100 Sebastian Dröge * gst/playback/gstplaysinkconvertbin.c: @@ -37834,22 +39307,110 @@ * ext/vorbis/gstvorbisenc.c: vorbisenc: do not accept 256 channels, 255 is the max vorbis supports +2011-11-23 13:22:12 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: implement replay gain + It would ideally be better to leave this to a rgvolume element, + but we don't control the pipeline. So do it by default, and allow + disabling it via a property, so the correct volume should always + be output. + +2011-11-23 11:58:54 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: add in-band FEC support + This allows reconstruction of lost packets if FEC info is included + in the next packet, at the cost of extra latency. Since we do not + know if the stream has FEC (and this can change at runtime), we + always incur the latency, even if we never lose any frame, or see + any FEC information. Off by default. + 2011-11-23 11:10:31 +0100 Wim Taymans * ext/ogg/gstoggstream.c: ogg: fix compilation +2011-11-23 11:08:39 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/opus/gstopusdec.c + ext/opus/gstopusenc.c + ext/opus/gstopusparse.c + gst/audiovisualizers/gstwavescope.c + gst/filter/Makefile.am + gst/filter/gstfilter.c + gst/filter/gstiir.c + gst/playondemand/gstplayondemand.c + 2011-11-23 10:50:53 +0100 Wim Taymans Merge branch 'master' into 0.11 Conflicts: ext/ogg/gstoggmux.c +2011-11-22 20:27:50 +0000 Tim-Philipp Müller + + * ext/opus/gstopusenc.c: + opusenc: mark properties changeable at runtime with GST_PARAM_MUTABLE_PLAYING + +2011-11-22 18:33:17 +0000 Vincent Penquerc'h + + * tests/check/elements/opus.c: + opus: add test + +2011-11-22 17:04:09 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: allow setting most properties at PLAYING time + Opus allows these to be changed during encoding, transparently + to the decoder. + +2011-11-22 16:14:06 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: bound the bitrate to more sensible values + Go from the bounds mentioned in the spec, and allow some more + variation. + In particular, don't allow silly low bitrates, and allow reaching + the maximum useful bitrate. + +2011-11-22 15:33:20 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opusenc: fix crash on pathological parameters + Asking for 1 bit/s would select a 0 byte buffer, leading + to a crash. Buffer size is now controlled by a max-payload-size + property, which can't be less than 2. + 2011-11-22 13:29:10 +0000 Vincent Penquerc'h * ext/ogg/gstoggstream.c: oggstream: extract opus comments if available +2011-11-21 17:48:54 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + opus: move header magic testing to gstopusheader + +2011-11-21 17:01:49 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: skip pre-skip samples + +2011-11-21 12:50:22 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: read pre-skip from first header if available + 2011-11-22 13:15:33 +0000 Vincent Penquerc'h * ext/ogg/gstoggstream.c: @@ -37921,12 +39482,62 @@ xvimagebufferpool: Use the default ::free_buffer() implementation Which does exactly the same thing +2011-11-21 12:02:28 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: reset tagsetter interface on stop + +2011-11-21 11:44:01 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: handle NULL packets (used for PLC) + +2011-11-21 11:28:10 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: light cleanup + +2011-11-20 09:58:06 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: do not push header buffers + Opus headers appear only when muxed in Ogg, so only place them + on the caps, where oggmux will find them, but other elements will + be blithely unaware of them. + +2011-11-20 09:52:46 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusheader.c: + * ext/opus/gstopusheader.h: + opus: make opusparse set headers on caps + Header-on-caps code moved to a new shared location to avoid + duplicating the code. + 2011-11-19 16:06:09 +0000 Vincent Penquerc'h * ext/ogg/gstoggmux.c: * ext/ogg/gstoggstream.c: ogg: add opus support +2011-11-19 15:58:09 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix terminating NUL being written in signature + +2011-11-16 19:40:20 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: make frame-size an enum + It only supports a set number of specific values (including + a non integer one). + +2011-11-16 19:22:44 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: the encoder might not make use of all the bytes + 2011-11-18 17:58:58 +0100 Wim Taymans * ext/gio/gstgiosrc.c: @@ -37949,6 +39560,17 @@ * gst-libs/gst/audio/gstaudiobasesink.c: fix for scheduling mode rename +2011-11-17 17:32:42 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/celt/gstceltdec.c + ext/opus/gstopusdec.c + ext/opus/gstopusdec.h + ext/opus/gstopusenc.c + ext/opus/gstopusenc.h + ext/opus/gstopusparse.c + 2011-11-17 17:07:41 +0100 Wim Taymans Merge branch 'master' into 0.11 @@ -37999,6 +39621,24 @@ * gst/adder/gstadder.c: collectpads: port API changes +2011-11-16 18:49:03 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: do not include variable fields in caps + Those can vary from one packet to the next, so have no reason + to be in the caps. + +2011-11-16 18:43:53 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix constrained-vbr property name typo + +2011-11-16 18:35:29 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: let the base class handle all timing + 2011-11-16 19:00:44 +0100 Mark Nauwelaerts * ext/vorbis/gstvorbisenc.c: @@ -38011,6 +39651,39 @@ ... which ensures nothing subsequently tries to slip past _chain and into a possibly improperly setup subclass. +2011-11-15 19:53:33 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopus.c: + opusparse: add opusparse element + A very simple element that parses Opus streams from the ad hoc + framing used by the Opus test vectors. + +2011-11-16 17:24:20 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: allow negotiation of rate/channels with downstream + Since an opus stream may be decoded to any (sensible) rate, + and either stereo or mono, we try to accomodate downstream. + +2011-11-16 17:05:17 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: rewrite logic + Parameters such as frame size, etc, are variable. Pretty much + everything can change within a stream, so be prepared about it, + and do not cache parameters in the decoder. + +2011-11-16 16:56:43 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opus: port to base audio encoder/decoder + 2011-11-15 13:29:31 +0000 Vincent Penquerc'h * gst-libs/gst/audio/gstaudiodecoder.c: @@ -38048,6 +39721,36 @@ * gst/subparse/gstsubparse.c: add parent to query function +2011-11-16 13:26:35 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: allow negotiation of rate/channels with downstream + Since an opus stream may be decoded to any (sensible) rate, + and either stereo or mono, we try to accomodate downstream. + +2011-11-16 01:14:32 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: rewrite logic + Parameters such as frame size, etc, are variable. Pretty much + everything can change within a stream, so be prepared about it, + and do not cache parameters in the decoder. + +2011-11-15 23:00:32 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + opusdec: remove buffer pool, buffers are not constant size + +2011-11-15 19:53:33 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopus.c: + opusparse: add opusparse element + A very simple element that parses Opus streams from the ad hoc + framing used by the Opus test vectors. + 2011-11-16 12:37:44 +0100 Wim Taymans * ext/libvisual/visual.c: @@ -38055,6 +39758,11 @@ Use the _check_reconfigure method instead of checking flags. Don't need to ref the parent anymore, core does that. +2011-11-15 17:49:48 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix pointer mismatch in memcpy on drain + 2011-11-15 17:58:19 +0100 Wim Taymans * gst-libs/gst/audio/gstaudiodecoder.c: @@ -38128,6 +39836,15 @@ This allows flacdec to not emit audio for headers, while allowing the base audio decoder to keep its timestamps in sync. +2011-11-14 13:41:58 +0000 Vincent Penquerc'h + + * ext/opus/Makefile.am: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opus: port to encoder/decoder base classes + 2011-11-14 12:45:31 +0100 Robert Swain * gst-libs/gst/audio/gstaudiodecoder.c: @@ -38474,6 +40191,13 @@ Indent Add padding +2011-11-11 17:46:41 +0000 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + opus: port to 0.11 + 2011-11-11 18:23:22 +0100 Wim Taymans * gst-libs/gst/fft/gstfftf32.c: @@ -38667,12 +40391,20 @@ remove bogus files They got somehow commited in 7012e88090e69339c60a4eb9449f7a7e39ca6aa3 +2011-11-11 10:39:17 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-11-10 23:02:35 +0200 Stefan Sauer * gst/volume/gstvolume.c: * tests/icles/audio-trickplay.c: controller: port controller api changes +2011-11-10 18:34:48 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-11-10 18:32:39 +0100 Wim Taymans * ext/libvisual/visual.c: @@ -38687,6 +40419,11 @@ * tests/check/libs/gstlibscpp.cc: tests: fix build after removal of base64 lib +2011-11-10 17:13:40 +0000 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix bandwidth property type mismatch + 2011-11-10 17:52:36 +0100 Wim Taymans * gst-libs/gst/video/gstvideosink.h: @@ -38788,6 +40525,10 @@ pbutils: Fix introspection annotations Fixes #663689 +2011-11-10 12:14:19 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-11-10 11:42:10 +0100 Edward Hervey * tests/check/libs/struct_arm.h: @@ -38805,6 +40546,16 @@ * gst/playback/gstsubtitleoverlay.c: upates for new ACCEPT_CAPS query +2011-11-09 12:24:37 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-09 12:19:04 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + gst/colorspace/colorspace.c + 2011-11-09 12:11:59 +0100 Wim Taymans Merge branch 'master' into 0.11 @@ -39003,6 +40754,26 @@ gst/playback/gstplaysinkvideoconvert.c gst/playback/gstplaysinkvideoconvert.h +2011-10-05 18:25:58 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix latency query + This makes live 'audiosrc ! opusenc ! opusdec ! audiosink' pipelines + actually work without all audio being dumped. + https://bugzilla.gnome.org/show_bug.cgi?id=660999 + +2011-10-05 15:47:06 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: use debug level for debug info, not error + https://bugzilla.gnome.org/show_bug.cgi?id=660999 + +2011-09-29 14:22:33 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: fix calculation of filler data size + https://bugzilla.gnome.org/show_bug.cgi?id=660469 + 2011-05-02 13:05:28 +0300 Felipe Contreras * gst-libs/gst/audio/gstbaseaudiosink.c: @@ -39081,6 +40852,10 @@ Some found by Havard Graff. Signed-off-by: Felipe Contreras +2011-11-07 10:02:00 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-11-04 22:00:43 +0100 Stefan Sauer * gst/adder/gstadder.c: @@ -39137,6 +40912,10 @@ * ext/ogg/gstoggdemux.c: oggdemux: fix somtimes pad +2011-11-04 11:01:42 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-11-04 10:48:50 +0100 Wim Taymans * ext/ogg/gstoggmux.c: @@ -39431,6 +41210,10 @@ * gst-libs/gst/video/video.h: video: Add convenience macros for accessing GstVideoInfo flags +2011-11-02 10:31:24 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-31 02:39:48 +0100 Wim Taymans * gst-libs/gst/netbuffer/gstnetbuffer.c: @@ -39505,6 +41288,10 @@ Update for pad API changes GstProbeType, GstProbeReturn and GstActivateMode -> GstPad* +2011-10-31 14:51:32 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + 2011-10-31 14:26:09 +0000 Tim-Philipp Müller * gst/playback/gstsubtitleoverlay.c: @@ -39659,6 +41446,10 @@ * gst/typefind/gsttypefindfunctions.c: fix compilation +2011-10-27 16:13:56 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-27 15:44:58 +0200 Wim Taymans Merge branch 'master' into 0.11 @@ -39829,6 +41620,10 @@ baseaudiosink: fix unused variable compiler warning if debugging in core is disabled https://bugzilla.gnome.org/show_bug.cgi?id=660150 +2011-10-18 14:32:05 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-18 13:00:29 +0200 René Stadler * gst/playback/gstsubtitleoverlay.c: @@ -39875,6 +41670,10 @@ * gst-libs/gst/audio/audio.c: audio: Indent and doc fixes +2011-10-16 15:28:31 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-13 08:53:34 +0200 Wim Taymans Merge branch 'master' into 0.11 @@ -39994,6 +41793,10 @@ * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: handle empty input by discarding +2011-10-08 11:17:11 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-08 11:05:29 +0200 Wim Taymans * ext/vorbis/gstvorbisdec.c: @@ -40073,6 +41876,10 @@ and falling back to a prefix check if nothing was found. https://bugzilla.gnome.org/show_bug.cgi?id=657261 +2011-10-06 14:05:42 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-04 21:17:37 -0300 Thiago Santos * gst/encoding/gstencodebin.c: @@ -40127,6 +41934,10 @@ The video-sink property allows manual specification via g_object_set () of the video sink element to be used. +2011-10-04 13:29:21 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-10-03 15:20:06 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: @@ -40159,6 +41970,49 @@ a similar way to add other streams (eg, subtitles). https://bugzilla.gnome.org/show_bug.cgi?id=642878 +2011-10-03 11:24:04 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-09-28 14:57:02 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: fix decoding + A simple ... opusenc ! opusdec ... pipeline now works. + https://bugzilla.gnome.org/show_bug.cgi?id=660364 + +2011-09-28 14:56:18 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: moan if we get an unexpected amount of data + https://bugzilla.gnome.org/show_bug.cgi?id=660364 + +2011-09-28 14:22:02 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opus: properly setup caps and init state from caps + https://bugzilla.gnome.org/show_bug.cgi?id=660364 + +2011-09-28 13:25:21 +0100 Vincent Penquerc'h + + * ext/opus/gstopusenc.c: + opusenc: use the same frame size setup as the opus test code + https://bugzilla.gnome.org/show_bug.cgi?id=660364 + +2011-09-28 13:24:52 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + opusdec: opus supports a select set of sampling rates + https://bugzilla.gnome.org/show_bug.cgi?id=660364 + +2011-09-28 13:24:21 +0100 Vincent Penquerc'h + + * ext/opus/gstopusdec.c: + * ext/opus/gstopusenc.c: + opus: make it build against current, and remove cruft + https://bugzilla.gnome.org/show_bug.cgi?id=660364 + 2011-09-27 00:26:29 +0100 Vincent Penquerc'h * ext/alsa/gstalsasrc.c: @@ -40426,6 +42280,10 @@ * docs/libs/gst-plugins-base-libs-sections.txt: docs: minor docs fix +2011-09-26 22:31:17 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-09-26 21:11:14 +0200 Wim Taymans * gst-libs/gst/audio/gstaudioencoder.c: @@ -41056,6 +42914,10 @@ audio: update audio format enums to match changes in 0.11 And add new audio format info stuff to docs. +2011-09-06 16:13:28 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-09-06 15:40:02 +0200 Stefan Sauer * common: @@ -41904,6 +43766,12 @@ * gst/audiotestsrc/gstaudiotestsrc.c: audiotestsrc: use base class fill method +2011-08-25 12:49:26 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/resindvd/rsnwrappedbuffer.c + 2011-08-24 17:39:11 +0100 Vincent Penquerc'h * ext/ogg/gstoggmux.c: @@ -42534,6 +44402,10 @@ audioresample: fix build without orc https://bugzilla.gnome.org/show_bug.cgi?id=656781 +2011-08-17 19:01:39 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + 2011-08-17 17:24:35 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosrc.c: @@ -43657,6 +45529,20 @@ * gst-libs/gst/video/video.c: video: improve debug +2011-08-04 09:40:46 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-08-04 09:36:07 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + common + configure.ac + gst/colorspace/colorspace.c + gst/colorspace/colorspace.h + gst/colorspace/gstcolorspace.c + 2011-08-03 14:14:55 -0300 Thiago Santos * gst/encoding/gstencodebin.c: @@ -44543,6 +46429,17 @@ tests: remove tests from ancient times They're just noise. +2011-06-05 00:54:19 -0700 David Schleef + + * ext/opus/Makefile.am: + * ext/opus/gstopus.c: + * ext/opus/gstopusdec.c: + * ext/opus/gstopusdec.h: + * ext/opus/gstopusenc.c: + * ext/opus/gstopusenc.h: + opus: duplicate from CELT + Copy the celt plugin and convert it to Opus. Mostly works. + 2011-07-07 11:10:39 +0200 Wim Taymans Merge branch 'master' into 0.11 @@ -98959,3 +100856,9 @@ Original commit message from CVS: releasing 0.10.0 +2001-12-17 18:37:01 +0000 Thomas Vander Stichele + + building up speed + Original commit message from CVS: + building up speed + diff --git a/NEWS b/NEWS index a4a5e76ce2..8f11fc4c19 100644 --- a/NEWS +++ b/NEWS @@ -1,2 +1,2 @@ -This is GStreamer 1.7.2 +This is GStreamer 1.7.90 diff --git a/RELEASE b/RELEASE index fd03ccd0da..f1a9ba7d8e 100644 --- a/RELEASE +++ b/RELEASE @@ -1,16 +1,14 @@ -Release notes for GStreamer Base Plugins 1.7.2 +Release notes for GStreamer Base Plugins 1.7.90 -The GStreamer team is pleased to announce the second release of the unstable -1.7 release series. The 1.7 release series is adding new features on top of +The GStreamer team is pleased to announce the first release candidate of the stable +1.8 release series. The 1.8 release series is adding new features on top of the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release -series of the GStreamer multimedia framework. The unstable 1.7 release series -will lead to the stable 1.8 release series in the next weeks. Any newly added -API can still change until that point. +series of the GStreamer multimedia framework. Binaries for Android, iOS, Mac OS X and Windows will be provided separately -during the unstable 1.7 release series. +during the stable 1.8 release series. This module contains a set of reference plugins, base classes for other @@ -59,30 +57,11 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg) Bugs fixed in this release - * 745880 : sdp: SDP < - > GstCaps helper functions - * 751901 : gst-play: verbose & playbin flags options support - * 755918 : decodebin: Refactor code to remove assertion errors - * 756187 : appsink: Always blocks on EOS until buffers are consumed since 1.6, should be configurable - * 758212 : playbin adds the template caps on autoplug-query - * 759729 : audiofxbad: Name collision with new GstAudioChannelMix API from libgstaudio - * 759855 : build: pbutils needs to link to libgstbase for bytewriter and adapter - * 759890 : audioconvert: creates choppy audio - * 760134 : audioconvert test: doesn't build with clang - * 760204 : videotestsrc: add missing break in set_property switch case - * 760234 : playbin: Assumes recursive accept-caps query, breaks totem - * 760408 : #750013 (streamsynchronizer patches) broke some use cases in GES - * 760477 : playbin: caps intersection autoplugs too early and stream stops - * 760769 : tests:audioconvert: Build error when running make check - * 760938 : audioconvert: crash when executing orc unpack function - * 760949 : decodebin: Correctly expose pads from elements that have directly exposable pads - * 761132 : video-format: fix GstVideoFormatInfo documentation warnings - * 761218 : audio/videodecoder: Use gst_pad_peer_query_caps() instead of using gst_pad_get_allowed_caps() to make negotiated output caps before forwarding GAP event - * 761251 : textoverlay: Expose text rendering dimensions to applications and remove absolute positioning limit - * 761949 : gst-libs/gst/Makefile.am: build audio before rtp - * 761951 : videoencoder: Fix leak when pre_push does not return OK - * 762085 : gst-base 1.7 update created background buzzing noise with audioconvert - * 762239 : matroskademux: Assertions about unmappable memory when demuxing wavpack streams - * 693263 : typefinding: MPEG-2 video ES detected as H.263 + * 761257 : multisocketsink: doesn't handle client close and EWOULDBLOCK + * 761588 : opusdec: no mono/stereo channel conversion + * 761851 : video: add orc UYVY422 to GRAY8 conversion function + * 762451 : id3v2frames: read conductor tag + * 762674 : Compilation error building html in gst-plugins-base-1.7.2 ==== Download ==== @@ -119,27 +98,14 @@ subscribe to the gstreamer-devel list. Contributors to this release - * Arun Raghavan - * Aurélien Zanelli + * Adam Miartus * Edward Hervey - * Evan Callaway - * Havard Graff - * HoonHee Lee - * Hugues Fruchet - * Hyunjun Ko - * Julien Isorce - * Koop Mast - * Lubosz Sarnecki - * Mathieu Duponchelle - * Nirbheek Chauhan - * Reynaldo H. Verdejo Pinochet + * Göran Jönsson + * Lim Siew Hoon + * Luis de Bethencourt * Sebastian Dröge - * Stefan Sauer - * Stian Selnes - * Thiago Santos - * Thibault Saunier * Tim-Philipp Müller - * Vineeth T M - * Vineeth TM + * Tom Deseyn + * Vincent Penquerc'h * Wim Taymans   \ No newline at end of file diff --git a/configure.ac b/configure.ac index 11c46b5c75..058894a8b6 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/prerelease -AC_INIT([GStreamer Base Plug-ins],[1.7.2.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base]) +AC_INIT([GStreamer Base Plug-ins],[1.7.90],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base]) AG_GST_INIT @@ -56,10 +56,10 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 702, 0, 702) +AS_LIBTOOL(GST, 790, 0, 790) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.7.2.1 +GST_REQ=1.7.90 dnl *** autotools stuff **** diff --git a/docs/plugins/gst-plugins-base-plugins.hierarchy b/docs/plugins/gst-plugins-base-plugins.hierarchy index 7b1e99df73..c2d607b48e 100644 --- a/docs/plugins/gst-plugins-base-plugins.hierarchy +++ b/docs/plugins/gst-plugins-base-plugins.hierarchy @@ -128,6 +128,7 @@ GObject PangoFcFontMap PangoCairoFcFontMap GInterface + GDatagramBased GFile GInitable GTypePlugin diff --git a/docs/plugins/inspect/plugin-adder.xml b/docs/plugins/inspect/plugin-adder.xml index 3c1670ff21..f3b334fd5b 100644 --- a/docs/plugins/inspect/plugin-adder.xml +++ b/docs/plugins/inspect/plugin-adder.xml @@ -3,7 +3,7 @@ Adds multiple streams ../../gst/adder/.libs/libgstadder.so libgstadder.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-alsa.xml b/docs/plugins/inspect/plugin-alsa.xml index 90601aeea6..f7bcf6dcfd 100644 --- a/docs/plugins/inspect/plugin-alsa.xml +++ b/docs/plugins/inspect/plugin-alsa.xml @@ -3,7 +3,7 @@ ALSA plugin library ../../ext/alsa/.libs/libgstalsa.so libgstalsa.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-app.xml b/docs/plugins/inspect/plugin-app.xml index 200121ce47..603c6cc9f0 100644 --- a/docs/plugins/inspect/plugin-app.xml +++ b/docs/plugins/inspect/plugin-app.xml @@ -3,7 +3,7 @@ Elements used to communicate with applications ../../gst/app/.libs/libgstapp.so libgstapp.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-audioconvert.xml b/docs/plugins/inspect/plugin-audioconvert.xml index 067f66bbac..77a3e148c6 100644 --- a/docs/plugins/inspect/plugin-audioconvert.xml +++ b/docs/plugins/inspect/plugin-audioconvert.xml @@ -3,7 +3,7 @@ Convert audio to different formats ../../gst/audioconvert/.libs/libgstaudioconvert.so libgstaudioconvert.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-audiorate.xml b/docs/plugins/inspect/plugin-audiorate.xml index 6026a97ed0..847652fc53 100644 --- a/docs/plugins/inspect/plugin-audiorate.xml +++ b/docs/plugins/inspect/plugin-audiorate.xml @@ -3,7 +3,7 @@ Adjusts audio frames ../../gst/audiorate/.libs/libgstaudiorate.so libgstaudiorate.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-audioresample.xml b/docs/plugins/inspect/plugin-audioresample.xml index 0f8e6078c5..38fc04c9a6 100644 --- a/docs/plugins/inspect/plugin-audioresample.xml +++ b/docs/plugins/inspect/plugin-audioresample.xml @@ -3,7 +3,7 @@ Resamples audio ../../gst/audioresample/.libs/libgstaudioresample.so libgstaudioresample.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-audiotestsrc.xml b/docs/plugins/inspect/plugin-audiotestsrc.xml index 74226e19df..46c4bc9220 100644 --- a/docs/plugins/inspect/plugin-audiotestsrc.xml +++ b/docs/plugins/inspect/plugin-audiotestsrc.xml @@ -3,7 +3,7 @@ Creates audio test signals of given frequency and volume ../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so libgstaudiotestsrc.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-cdparanoia.xml b/docs/plugins/inspect/plugin-cdparanoia.xml index c9c51a82ea..f67717ba3b 100644 --- a/docs/plugins/inspect/plugin-cdparanoia.xml +++ b/docs/plugins/inspect/plugin-cdparanoia.xml @@ -3,7 +3,7 @@ Read audio from CD in paranoid mode ../../ext/cdparanoia/.libs/libgstcdparanoia.so libgstcdparanoia.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-encoding.xml b/docs/plugins/inspect/plugin-encoding.xml index 8635c093f1..b51587eda9 100644 --- a/docs/plugins/inspect/plugin-encoding.xml +++ b/docs/plugins/inspect/plugin-encoding.xml @@ -3,7 +3,7 @@ various encoding-related elements ../../gst/encoding/.libs/libgstencodebin.so libgstencodebin.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-gio.xml b/docs/plugins/inspect/plugin-gio.xml index 653e66e7e0..64e8ffc51c 100644 --- a/docs/plugins/inspect/plugin-gio.xml +++ b/docs/plugins/inspect/plugin-gio.xml @@ -3,7 +3,7 @@ GIO elements ../../gst/gio/.libs/libgstgio.so libgstgio.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-libvisual.xml b/docs/plugins/inspect/plugin-libvisual.xml index c0903dca2e..dc4cac6d0f 100644 --- a/docs/plugins/inspect/plugin-libvisual.xml +++ b/docs/plugins/inspect/plugin-libvisual.xml @@ -3,7 +3,7 @@ libvisual visualization plugins ../../ext/libvisual/.libs/libgstlibvisual.so libgstlibvisual.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-ogg.xml b/docs/plugins/inspect/plugin-ogg.xml index 2bbd8067a9..b30d240624 100644 --- a/docs/plugins/inspect/plugin-ogg.xml +++ b/docs/plugins/inspect/plugin-ogg.xml @@ -3,7 +3,7 @@ ogg stream manipulation (info about ogg: http://xiph.org) ../../ext/ogg/.libs/libgstogg.so libgstogg.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-opus.xml b/docs/plugins/inspect/plugin-opus.xml index d4e69dd19f..f41815ee49 100644 --- a/docs/plugins/inspect/plugin-opus.xml +++ b/docs/plugins/inspect/plugin-opus.xml @@ -3,10 +3,10 @@ OPUS plugin library ../../ext/opus/.libs/libgstopus.so libgstopus.so - 1.7.2.1 + 1.7.90 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-pango.xml b/docs/plugins/inspect/plugin-pango.xml index a82e997279..4ebbc9e6ad 100644 --- a/docs/plugins/inspect/plugin-pango.xml +++ b/docs/plugins/inspect/plugin-pango.xml @@ -3,7 +3,7 @@ Pango-based text rendering and overlay ../../ext/pango/.libs/libgstpango.so libgstpango.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-playback.xml b/docs/plugins/inspect/plugin-playback.xml index 2c45241fb0..59d0e6f913 100644 --- a/docs/plugins/inspect/plugin-playback.xml +++ b/docs/plugins/inspect/plugin-playback.xml @@ -3,7 +3,7 @@ various playback elements ../../gst/playback/.libs/libgstplayback.so libgstplayback.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-subparse.xml b/docs/plugins/inspect/plugin-subparse.xml index e140eccdac..a6a1ec8bd8 100644 --- a/docs/plugins/inspect/plugin-subparse.xml +++ b/docs/plugins/inspect/plugin-subparse.xml @@ -3,7 +3,7 @@ Subtitle parsing ../../gst/subparse/.libs/libgstsubparse.so libgstsubparse.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-tcp.xml b/docs/plugins/inspect/plugin-tcp.xml index cbbad4132a..edb8f539af 100644 --- a/docs/plugins/inspect/plugin-tcp.xml +++ b/docs/plugins/inspect/plugin-tcp.xml @@ -3,7 +3,7 @@ transfer data over the network via TCP ../../gst/tcp/.libs/libgsttcp.so libgsttcp.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-theora.xml b/docs/plugins/inspect/plugin-theora.xml index 421d6abbb3..3736f4e232 100644 --- a/docs/plugins/inspect/plugin-theora.xml +++ b/docs/plugins/inspect/plugin-theora.xml @@ -3,7 +3,7 @@ Theora plugin library ../../ext/theora/.libs/libgsttheora.so libgsttheora.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-typefindfunctions.xml b/docs/plugins/inspect/plugin-typefindfunctions.xml index 2e6d33063f..ec03dce5d2 100644 --- a/docs/plugins/inspect/plugin-typefindfunctions.xml +++ b/docs/plugins/inspect/plugin-typefindfunctions.xml @@ -3,7 +3,7 @@ default typefind functions ../../gst/typefind/.libs/libgsttypefindfunctions.so libgsttypefindfunctions.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-videoconvert.xml b/docs/plugins/inspect/plugin-videoconvert.xml index 023356d5cb..eb31462a7a 100644 --- a/docs/plugins/inspect/plugin-videoconvert.xml +++ b/docs/plugins/inspect/plugin-videoconvert.xml @@ -3,7 +3,7 @@ Colorspace conversion ../../gst/videoconvert/.libs/libgstvideoconvert.so libgstvideoconvert.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-videorate.xml b/docs/plugins/inspect/plugin-videorate.xml index 6f7550e5f4..c8340002e5 100644 --- a/docs/plugins/inspect/plugin-videorate.xml +++ b/docs/plugins/inspect/plugin-videorate.xml @@ -3,7 +3,7 @@ Adjusts video frames ../../gst/videorate/.libs/libgstvideorate.so libgstvideorate.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-videoscale.xml b/docs/plugins/inspect/plugin-videoscale.xml index 0b481ef701..aa34ff41bd 100644 --- a/docs/plugins/inspect/plugin-videoscale.xml +++ b/docs/plugins/inspect/plugin-videoscale.xml @@ -3,7 +3,7 @@ Resizes video ../../gst/videoscale/.libs/libgstvideoscale.so libgstvideoscale.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-videotestsrc.xml b/docs/plugins/inspect/plugin-videotestsrc.xml index c93ebbb1e0..5013bb96ff 100644 --- a/docs/plugins/inspect/plugin-videotestsrc.xml +++ b/docs/plugins/inspect/plugin-videotestsrc.xml @@ -3,7 +3,7 @@ Creates a test video stream ../../gst/videotestsrc/.libs/libgstvideotestsrc.so libgstvideotestsrc.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-volume.xml b/docs/plugins/inspect/plugin-volume.xml index 1a72c0d1da..23a0313655 100644 --- a/docs/plugins/inspect/plugin-volume.xml +++ b/docs/plugins/inspect/plugin-volume.xml @@ -3,7 +3,7 @@ plugin for controlling audio volume ../../gst/volume/.libs/libgstvolume.so libgstvolume.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-vorbis.xml b/docs/plugins/inspect/plugin-vorbis.xml index 723bf9655d..e08368e87e 100644 --- a/docs/plugins/inspect/plugin-vorbis.xml +++ b/docs/plugins/inspect/plugin-vorbis.xml @@ -3,7 +3,7 @@ Vorbis plugin library ../../ext/vorbis/.libs/libgstvorbis.so libgstvorbis.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-ximagesink.xml b/docs/plugins/inspect/plugin-ximagesink.xml index b0c72b5b8b..dd479bdf31 100644 --- a/docs/plugins/inspect/plugin-ximagesink.xml +++ b/docs/plugins/inspect/plugin-ximagesink.xml @@ -3,7 +3,7 @@ X11 video output element based on standard Xlib calls ../../sys/ximage/.libs/libgstximagesink.so libgstximagesink.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/docs/plugins/inspect/plugin-xvimagesink.xml b/docs/plugins/inspect/plugin-xvimagesink.xml index 1aca16594d..6d850fce0d 100644 --- a/docs/plugins/inspect/plugin-xvimagesink.xml +++ b/docs/plugins/inspect/plugin-xvimagesink.xml @@ -3,7 +3,7 @@ XFree86 video output plugin using Xv extension ../../sys/xvimage/.libs/libgstxvimagesink.so libgstxvimagesink.so - 1.7.2 + 1.7.90 LGPL gst-plugins-base GStreamer Base Plug-ins source release diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap index 243da2e81d..e6209c10f4 100644 --- a/gst-plugins-base.doap +++ b/gst-plugins-base.doap @@ -34,6 +34,16 @@ A wide range of video and audio decoders, encoders, and filters are included. + + + 1.7.90 + master + + 2016-03-01 + + + + 1.7.2 diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h index 850c1bdbe6..8e4de2c0aa 100644 --- a/win32/common/_stdint.h +++ b/win32/common/_stdint.h @@ -1,8 +1,8 @@ #ifndef _GST_PLUGINS_BASE__STDINT_H #define _GST_PLUGINS_BASE__STDINT_H 1 #ifndef _GENERATED_STDINT_H -#define _GENERATED_STDINT_H "gst-plugins-base 1.7.2" -/* generated using gnu compiler gcc-6 (Debian 6-20160205-1) 6.0.0 20160205 (experimental) [trunk revision 233183] */ +#define _GENERATED_STDINT_H "gst-plugins-base 1.7.90" +/* generated using gnu compiler gcc-6 (Debian 6-20160225-1) 6.0.0 20160225 (experimental) [trunk revision 233690] */ #define _STDINT_HAVE_STDINT_H 1 #include #endif diff --git a/win32/common/config.h b/win32/common/config.h index 93565714d8..df051aadd0 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -90,7 +90,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2016-02-19" +#define GST_PACKAGE_RELEASE_DATETIME "2016-03-01" /* Define if static plugins should be built */ #undef GST_PLUGIN_BUILD_STATIC @@ -230,6 +230,9 @@ /* Define to enable Xiph Ogg library (used by ogg). */ #undef HAVE_OGG +/* Define to enable opus (used by opus). */ +#undef HAVE_OPUS + /* Use Orc */ #undef HAVE_ORC @@ -336,7 +339,7 @@ #define PACKAGE_NAME "GStreamer Base Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Base Plug-ins 1.7.2" +#define PACKAGE_STRING "GStreamer Base Plug-ins 1.7.90" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-base" @@ -345,7 +348,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "1.7.2" +#define PACKAGE_VERSION "1.7.90" /* directory where plugins are located */ #ifdef _DEBUG @@ -379,7 +382,7 @@ #undef USE_TREMOLO /* Version number of package */ -#define VERSION "1.7.2" +#define VERSION "1.7.90" /* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most significant byte first (like Motorola and SPARC, unlike Intel). */