mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Release 1.3.3
This commit is contained in:
Sebastian Dröge
parent
cc429be8eb
commit
988f53ed18
34 changed files with 487 additions and 72 deletions
404
ChangeLog
404
ChangeLog
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@ -1,9 +1,407 @@
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=== release 1.3.2 ===
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=== release 1.3.3 ===
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2014-05-21 Sebastian Dröge <slomo@coaxion.net>
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2014-06-22 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.3.2
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releasing 1.3.3
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2014-06-22 14:23:32 +0200 Sebastian Dröge <sebastian@centricular.com>
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* po/da.po:
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* po/de.po:
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* po/hu.po:
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* po/id.po:
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* po/nl.po:
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* po/pl.po:
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* po/ru.po:
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* po/sr.po:
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* po/uk.po:
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po: Update translations
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2014-06-20 11:00:14 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst-libs/gst/audio/gstaudiodecoder.c:
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* tests/check/libs/audiodecoder.c:
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audiodecoder: Don't be too picky about the output frame counter
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With most decoder libraries, and especially when accessing codecs via
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OpenMAX or similar APIs, we don't have the ability to properly related
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the output buffers to a number of input samples. And could e.g. get
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a fractional number of input buffers decoded at a time.
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Previously this would in the end lead to an error message and stopped
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playback. Change it to a warning message instead and try to handle it
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gracefully. In theory the subclass can now get timestamp tracking
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wrong if it completely misuses the API, but if on average it behaves
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correct (and gst-omx and others do) it will continue to work properly.
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Also add a test for the new behaviour.
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We don't change it in the encoder yet as that requires more internal logic
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changes AFAIU and I'm not aware of a case where this was a problem so far.
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2014-06-12 12:36:26 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
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* gst/tcp/gsttcpserversrc.c:
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tcpserversrc: close the server socket after accepting a connection
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g_socket_accept() is only called once for a server socket. So
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keeping the socket open ist just confusing possible clients.
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https://bugzilla.gnome.org/show_bug.cgi?id=731566
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2014-06-13 10:04:47 +0100 Tim-Philipp Müller <tim@centricular.com>
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* gst/tcp/gsttcpclientsrc.c:
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tcpclientsrc: return FLUSHING when select() is canceled
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https://bugzilla.gnome.org/show_bug.cgi?id=731567
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2014-06-12 13:23:29 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
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* gst/tcp/gsttcpserversrc.c:
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tcpserversrc: return FLOW_FLUSHING instead of an error when accept/select is canceled
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Canceling the accept/select happens when the source is shut down. This is
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not an error and the GST_FLOW_ERROR causes problems when only part of the
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pipeline is shut down.
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https://bugzilla.gnome.org/show_bug.cgi?id=731567
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2014-06-12 11:55:59 +0200 Edward Hervey <bilboed@bilboed.com>
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* gst-libs/gst/sdp/gstmikey.c:
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mikey: Fix Wall to NTP conversion
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We are scaling from a unit in microseconds to a unit in ((1 << 32) per seconds).
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We therefore scale the microseconds values by:
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value of a second in the target unit (1 << 32)
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--------------------------------------------------------------
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value of a second in the origin format (1 000 000 microsecond)
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2014-06-06 12:18:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* ext/ogg/gstoggdemux.c:
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oggdemux: allow unset seek stop time in push mode
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2014-06-11 12:50:23 +0100 Tim-Philipp Müller <tim@centricular.com>
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
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docs: add streamsynchronizer to documentation
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2014-06-11 12:43:35 +0100 Tim-Philipp Müller <tim@centricular.com>
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
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docs: add playsink element to documentation
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2014-06-11 10:53:50 +0100 Tim-Philipp Müller <tim@centricular.com>
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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docs: add navigation interface to docs
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2014-06-10 12:59:53 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst-libs/gst/app/gstappsrc.c:
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appsrc: add send_event handler for flushing
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Adds a send_event handling for allowing appsrc to flush its internal
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data, allowing users to flush the pipeline without setting it to null.
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https://bugzilla.gnome.org/show_bug.cgi?id=724231
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2014-06-09 21:05:00 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst/videoscale/vs_fill_borders.c:
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* gst/videoscale/vs_image.h:
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videoscale: vs_image: strides are a gsize
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The strides that are set from the GstVideoInfo structs are
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a gsize. Using an int can cause overflows when dealing with large
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enough images
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https://bugzilla.gnome.org/show_bug.cgi?id=731195
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2014-06-09 19:44:56 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst-libs/gst/video/video-info.c:
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* tests/check/libs/video.c:
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video: avoid overflows when doing int operations for size
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size is a gsize, so cast the operands to it to avoid overflows
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and setting wrong value to the video size.
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Includes tests.
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https://bugzilla.gnome.org/show_bug.cgi?id=731195
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2014-06-09 10:53:03 +0200 Edward Hervey <bilboed@bilboed.com>
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* ext/theora/gsttheoraenc.c:
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theoraenc: Remove unneeded check
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running timestamps are guaranteed to be positive and valid since the
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GstVideoEncoder base class will clip incoming buffers
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CID #1139797
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2014-06-09 10:38:53 +0200 Edward Hervey <bilboed@bilboed.com>
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* ext/vorbis/gstvorbisenc.c:
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vorbisenc: add missing va_end in variadic function
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Coverity 1139944
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2014-06-06 10:35:31 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* tests/check/libs/videodecoder.c:
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tests: fix uninitialized variable use in video decoder test
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2014-06-05 15:35:31 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/playback/gsturidecodebin.c:
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uridecodebin: Also catch CODEC_NOT_FOUND errors and delay them until all decodebins are done
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2014-06-04 17:00:34 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/playback/gsturidecodebin.c:
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uridecodebin: Ignore missing-plugin messages unless all decodebins post one
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When playing RTSP streams there will be one decodebin per stream. If some of
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them fail because of a missing plugin we should not fail completely but play
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the supported streams at least.
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https://bugzilla.gnome.org/show_bug.cgi?id=730868
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2014-06-04 14:14:14 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/playback/gstdecodebin2.c:
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decodebin: Do async-done on expose errors too
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2014-05-20 12:28:15 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
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* gst-libs/gst/allocators/gstdmabuf.c:
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dmabuf: fix checking mmap flags
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A simple '&' is not sufficiant. With mmapping_flags == PROT_READ and
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prot == PROT_READ|PROT_WRITE the check produces the wrong result.
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Change the check to make sure that prot is a subset of mmapping_flags.
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https://bugzilla.gnome.org/show_bug.cgi?id=730559
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2014-06-03 15:16:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* ext/alsa/gstalsasink.c:
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alsasink: make gst-ident happy
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2014-06-03 15:10:33 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* ext/alsa/gstalsasink.c:
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alsasink: fix occasional crash intersecting invalid values
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When a pipeline using alsasink and push mode upstream fails
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to preroll, the following state will be the case:
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- A loop upstream will be PAUSED, pushing a first buffer
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- alsasink will be READY, pending PAUSED, because async
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On error, the pipeline will switch to NULL. alsasink is in
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READY, so goes to NULL immediately. It zeroes its cached
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caps. Meanwhile, the upstream loop can cause a caps query,
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conccurent with the state change. This will use those cached
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caps. If the zeroing happens between the NULL test and the
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dereferencing, GStreamer will critical down in the GstValue
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code.
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Since it appears that such a gap between states (PAUSED
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and pushing upstream, and NULL downstream) is expected, we
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need to protect the read/write access to the cached caps.
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This fixes the critical.
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See https://bugzilla.gnome.org/show_bug.cgi?id=731121
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2013-10-14 18:56:55 -0300 Thibault Saunier <thibault.saunier@collabora.com>
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* gst-libs/gst/video/gstvideodecoder.c:
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* tests/check/libs/videodecoder.c:
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videodecoder: Keep still meaningfull pending events on FLUSH_STOP
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Only EOS and segment should be deleted in that case.
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+ Add a testcase
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https://bugzilla.gnome.org/show_bug.cgi?id=709868
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2013-10-14 18:48:08 -0300 Thibault Saunier <thibault.saunier@collabora.com>
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* gst-libs/gst/audio/gstaudiodecoder.c:
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* tests/check/libs/audiodecoder.c:
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audiodecoder: Keep still meaningfull pending events on FLUSH_STOP
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Only EOS and segment should be deleted in that case.
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https://bugzilla.gnome.org/show_bug.cgi?id=709868
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2013-10-14 18:45:10 -0300 Thibault Saunier <thibault.saunier@collabora.com>
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* gst-libs/gst/video/gstvideoencoder.c:
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* tests/check/libs/videoencoder.c:
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videoencoder: Keep still meaningfull pending events on FLUSH_STOP
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Only EOS and segment should be deleted in that case.
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https://bugzilla.gnome.org/show_bug.cgi?id=709868
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2013-10-10 18:50:17 -0300 Thibault Saunier <thibault.saunier@collabora.com>
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* gst/encoding/gststreamsplitter.c:
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streamsplitter: Keep still meaningfull pending events on FLUSH_STOP
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Only EOS and segment should be deleted in that case.
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https://bugzilla.gnome.org/show_bug.cgi?id=709868
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2013-10-10 18:48:47 -0300 Thibault Saunier <thibault.saunier@collabora.com>
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* gst-libs/gst/audio/gstaudioencoder.c:
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* tests/check/libs/audioencoder.c:
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audioencoder: Keep still meaningfull pending events on FLUSH_STOP
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Only EOS and segment should be deleted in that case.
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https://bugzilla.gnome.org/show_bug.cgi?id=709868
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2014-06-02 12:40:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* ext/ogg/gstoggstream.c:
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oggstream: consider all opus packets as "keyframes"
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This lets oggdemux determine they are not delta units, and removes
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spurious per packet warnings about being unable to determine the
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packet's keyframeness.
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2014-05-12 17:13:50 +0200 Edward Hervey <bilboed@bilboed.com>
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* gst-libs/gst/sdp/gstmikey.c:
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mikey: Free MikeyPayload in error cases
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CID #1212136
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2014-03-16 14:27:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst/playback/gstdecodebin2.c:
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* tests/check/elements/decodebin.c:
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decodebin: aggregate buffering messages
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Aggregate buffering messages to only post the lower value
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to avoid setting pipeline to playing while any multiqueue
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is still buffering.
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There are 3 scenarios where the entries should be removed from
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the list:
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1) When decodebin is set to READY
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2) When an element posts a 100% buffering (already implemented)
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3) When a multiqueue is removed from decodebin.
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For item 3 we don't need to handle it because this should only
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happen when either 1 is hapenning or when it is playing a
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chained file, for which number 2 should have happened for the
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previous stream to finish
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https://bugzilla.gnome.org/show_bug.cgi?id=726423
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2014-05-28 10:23:24 +0100 Philip Withnall <philip.withnall@collabora.co.uk>
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* gst-libs/gst/audio/audio-format.c:
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audio: Add a missing precondition to gst_audio_format_from_string()
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https://bugzilla.gnome.org/show_bug.cgi?id=730874
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2014-05-26 20:57:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* tests/check/libs/audiodecoder.c:
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* tests/check/libs/videodecoder.c:
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tests: videodecoder: audiodecoder: add tests for eos after segment
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Tests that pushing a buffer after the segment returns EOS
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2014-05-26 21:24:07 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst-libs/gst/video/gstvideodecoder.c:
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videodecoder: actually return the push result in backwards playback
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It was always returning _OK regardless of what downstream returned
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2014-05-26 12:44:48 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst-libs/gst/video/gstvideodecoder.c:
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videodecoder: return EOS when segment is over
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if a buffer is clipped by being completely out of segment, check if this
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buffer is after the end of the segment and return EOS upstream
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https://bugzilla.gnome.org/show_bug.cgi?id=709224
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2014-05-26 12:44:38 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* gst-libs/gst/audio/gstaudiodecoder.c:
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audiodecoder: return EOS when segment is over
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if a buffer is clipped by being completely out of segment, check if this
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buffer is after the end of the segment and return EOS upstream
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https://bugzilla.gnome.org/show_bug.cgi?id=709224
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2014-05-26 11:45:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com>
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* ext/ogg/gstoggdemux.c:
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* ext/ogg/gstoggdemux.h:
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oggdemux: use new gstutils helper GstFlowCombiner
|
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Fixes the handling of GST_FLOW_EOS by using the helper object
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from gstutils that does the correct combination of flow returns.
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https://bugzilla.gnome.org/show_bug.cgi?id=709224
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2014-05-23 19:21:35 +0100 Tim-Philipp Müller <tim@centricular.com>
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* tools/gst-play.c:
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tools: play: use cubic volume factor when adjusting volume
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This is more natural and better-suited for a playback application.
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2014-05-21 13:23:24 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
|
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Back to development
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|
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=== release 1.3.2 ===
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2014-05-21 13:06:34 +0200 Sebastian Dröge <sebastian@centricular.com>
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* ChangeLog:
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* NEWS:
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* RELEASE:
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* common:
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* configure.ac:
|
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* docs/plugins/inspect/plugin-adder.xml:
|
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* docs/plugins/inspect/plugin-alsa.xml:
|
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* docs/plugins/inspect/plugin-app.xml:
|
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* docs/plugins/inspect/plugin-audioconvert.xml:
|
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* docs/plugins/inspect/plugin-audiorate.xml:
|
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* docs/plugins/inspect/plugin-audioresample.xml:
|
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
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* docs/plugins/inspect/plugin-cdparanoia.xml:
|
||||
* docs/plugins/inspect/plugin-encoding.xml:
|
||||
* docs/plugins/inspect/plugin-gio.xml:
|
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* docs/plugins/inspect/plugin-ivorbisdec.xml:
|
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* 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:
|
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* 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:
|
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* gst-plugins-base.doap:
|
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* win32/common/_stdint.h:
|
||||
* win32/common/config.h:
|
||||
Release 1.3.2
|
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|
||||
2014-05-21 12:01:15 +0200 Sebastian Dröge <sebastian@centricular.com>
|
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|
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* po/af.po:
|
||||
* po/az.po:
|
||||
* po/bg.po:
|
||||
* po/ca.po:
|
||||
* po/cs.po:
|
||||
* po/da.po:
|
||||
* po/de.po:
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* po/el.po:
|
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* po/en_GB.po:
|
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* po/eo.po:
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* po/es.po:
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* po/eu.po:
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||||
* po/fi.po:
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* po/fr.po:
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* po/gl.po:
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* po/hr.po:
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* po/hu.po:
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* po/id.po:
|
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* po/it.po:
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||||
* po/ja.po:
|
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* po/lt.po:
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||||
* po/lv.po:
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* po/nb.po:
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||||
* po/nl.po:
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* 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:
|
||||
Update .po files
|
||||
|
||||
2014-05-21 10:50:56 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
|
|
|
|||
26
NEWS
26
NEWS
|
|
@ -1,7 +1,8 @@
|
|||
This is GStreamer Base Plugins 1.3.2
|
||||
This is GStreamer Base Plugins 1.3.3
|
||||
|
||||
Changes since 1.2:
|
||||
|
||||
|
||||
New API:
|
||||
• GstMessageType has GST_MESSAGE_EXTENDED added. All types before
|
||||
that can be used together as a flags type as before, but from
|
||||
|
|
@ -30,6 +31,10 @@ New API:
|
|||
caps.
|
||||
• GstCollectPads has support for flushing and a default handler for
|
||||
SEEK events now.
|
||||
• New GstFlowAggregator helper object that simplifies handling of
|
||||
flow returns in elements with multiple source pads. Additionally
|
||||
GstPad now always stores the last flow return and provides an
|
||||
API to retrieve it.
|
||||
• GstSegment has new API to offset the running time by a specific
|
||||
value and this is used in GstPad to allow positive and negative
|
||||
offsets in gst_pad_set_offset() in all situations.
|
||||
|
|
@ -43,6 +48,7 @@ New API:
|
|||
• Support for tiled, raw video formats has been added.
|
||||
• GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
|
||||
events and merge custom tags into them consistently.
|
||||
• GstBufferPool has support for flushing now.
|
||||
• playbin/playsink has support for application provided audio and video
|
||||
filters.
|
||||
• GstDiscoverer has new and simplified API to get details about missing
|
||||
|
|
@ -54,6 +60,10 @@ New API:
|
|||
DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
|
||||
Wayland and EGL platforms.
|
||||
This replaces eglglessink and also is supposed to replace osxvideosink.
|
||||
• New GstAggregator base class in gst-plugins-bad. This is supposed to
|
||||
replace GstCollectPads in the future and fix long-known shortcomings
|
||||
in its API. Together with the base class some elements are provided
|
||||
already, like a videomixer (compositor).
|
||||
|
||||
|
||||
Major changes:
|
||||
|
|
@ -97,7 +107,8 @@ Major changes:
|
|||
∘ dvbsrc supports more delivery mechanisms and other features
|
||||
now, including DVB S2 and T2 support.
|
||||
∘ The MPEGTS library has support for many more descriptors.
|
||||
∘ Major improvements to tsdemux, especially time related.
|
||||
∘ Major improvements to tsdemux and tsparse, especially time and
|
||||
seeking related.
|
||||
∘ souphttpsrc now has support for keep-alive connections,
|
||||
compression, configurable number of retries and configuration
|
||||
for SSL certificate validation.
|
||||
|
|
@ -110,9 +121,16 @@ Major changes:
|
|||
finish.
|
||||
∘ videoflip can automatically flip based on the orientation tag.
|
||||
∘ openjpeg supports the OpenJPEG2 API.
|
||||
∘ waylandsink was refactored and should be more useful now. It also
|
||||
includes a small library which most likely is going to be removed
|
||||
in the future and will result in extensions to the GstVideoOverlay
|
||||
interface.
|
||||
∘ gst-rtsp-server supports SRTP and MIKEY now.
|
||||
∘ gst-libav encoders are now negotiating any profile/level settings
|
||||
with downstream via caps.
|
||||
∘ Lots of fixes for coverity warnings all over the place.
|
||||
∘ 400+ fixed bug reports, and many other bug fixes and other
|
||||
∘ Negotiation related performance improvements.
|
||||
∘ 500+ fixed bug reports, and many other bug fixes and other
|
||||
improvements everywhere that had no bug report.
|
||||
|
||||
Things to look out for:
|
||||
|
|
@ -120,3 +138,5 @@ Things to look out for:
|
|||
element.
|
||||
• The mfcdec element was removed and replaced by v4l2videodec.
|
||||
• osxvideosink is only available in OS X 10.6 or newer.
|
||||
• The GstDeviceMonitor API will likely change slightly before the
|
||||
1.4.0 release.
|
||||
|
|
|
|||
47
RELEASE
47
RELEASE
|
|
@ -1,8 +1,8 @@
|
|||
|
||||
Release notes for GStreamer Base Plugins 1.3.2
|
||||
Release notes for GStreamer Base Plugins 1.3.3
|
||||
|
||||
|
||||
The GStreamer team is pleased to announce the second release of the unstable
|
||||
The GStreamer team is pleased to announce the third release of the unstable
|
||||
1.3 release series. The 1.3 release series is adding new features on top of
|
||||
the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
|
||||
series of the GStreamer multimedia framework. The unstable 1.3 release series
|
||||
|
|
@ -10,23 +10,15 @@ will lead to the stable 1.4 release series in the next weeks, and newly added
|
|||
API can still change until that point.
|
||||
|
||||
|
||||
This is hopefully the last 1.3 development release and will be followed by
|
||||
the first 1.4.0 release candidate (1.3.90) in 1-2 weeks. Which then hopefully
|
||||
is followed by 1.4.0 soonish in early July.
|
||||
|
||||
|
||||
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
|
||||
during the unstable 1.3 release series.
|
||||
|
||||
|
||||
|
||||
The versioning scheme that is used in general is that 1.x.y is API and
|
||||
ABI backwards compatible with previous 1.x.y releases. If x is an even
|
||||
number it is a stable release series and all releases in this series
|
||||
will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
|
||||
x is odd it is a development release series that will lead to the next
|
||||
stable release series 1.x+1 and contains new features and bigger
|
||||
changes. During the development release series, new API can still
|
||||
change.
|
||||
|
||||
|
||||
|
||||
This module contains a set of reference plugins, base classes for other
|
||||
plugins, and helper libraries. It also includes essential elements such
|
||||
as audio and video format converters, and higher-level components like playbin,
|
||||
|
|
@ -73,15 +65,15 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
|
|||
|
||||
Bugs fixed in this release
|
||||
|
||||
* 720596 : discoverer: Rework the API to make " install missing plugin " feature cleaner
|
||||
* 729514 : rtsp: fails to build on Windows, undefined refs to getsockname and setsockopt
|
||||
* 729515 : W32: playback-test fails to build due to warnings
|
||||
* 729617 : playback-test: crash when setting buffer-size property on playbin
|
||||
* 729632 : rtspconnection: crashing sometimes when addinging a read source
|
||||
* 730010 : gst-play: audio_sink and video_sink strings are not freed
|
||||
* 730368 : Add a read source on write socket when tunnel lost.
|
||||
* 730441 : dmabuf: shared the mapping with shared copies of the memory
|
||||
* 729513 : W32: -base erroneously detects X11 headers from tcl/tk
|
||||
* 709868 : Keep still meaningfull pending events on FLUSH_STOP
|
||||
* 724231 : appsrc: handle flushing from send_event
|
||||
* 730559 : dmabuf: fix checking mmap flags
|
||||
* 730749 : Failed to determine keyframeness of audio/x-opus packet
|
||||
* 730868 : uridecodebin: Does not handle RTSP streams where one of the payload formats is not supported properly
|
||||
* 730874 : audio: Add a missing precondition to gst_audio_format_from_string()
|
||||
* 731121 : alsasink: Race condition causes alsasink to use invalid caps when a pipeline fails to start
|
||||
* 731566 : tcpserversrc: close the server socket after accepting a connection
|
||||
* 731567 : tcpserversrc: return GST_FLOW_FLUSHING instead of GST_FLOW_ERROR when accept is canceled
|
||||
|
||||
==== Download ====
|
||||
|
||||
|
|
@ -118,17 +110,12 @@ subscribe to the gstreamer-devel list.
|
|||
|
||||
Contributors to this release
|
||||
|
||||
* Anuj Jaiswal
|
||||
* Edward Hervey
|
||||
* Göran Jönsson
|
||||
* Luis de Bethencourt
|
||||
* Michael Olbrich
|
||||
* Nicolas Dufresne
|
||||
* Ravi Kiran K N
|
||||
* Philip Withnall
|
||||
* Sebastian Dröge
|
||||
* Thiago Santos
|
||||
* Thibault Saunier
|
||||
* Tim-Philipp Müller
|
||||
* Vincent Penquerc'h
|
||||
* Wim Taymans
|
||||
* Руслан Ижбулатов
|
||||
|
||||
|
|
@ -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],[1.3.2.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
|
||||
AC_INIT([GStreamer Base Plug-ins],[1.3.3],[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, 302, 0, 302)
|
||||
AS_LIBTOOL(GST, 303, 0, 303)
|
||||
|
||||
dnl *** required versions of GStreamer stuff ***
|
||||
GST_REQ=1.3.2.1
|
||||
GST_REQ=1.3.3
|
||||
|
||||
dnl *** autotools stuff ****
|
||||
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adds multiple streams</description>
|
||||
<filename>../../gst/adder/.libs/libgstadder.so</filename>
|
||||
<basename>libgstadder.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>ALSA plugin library</description>
|
||||
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
|
||||
<basename>libgstalsa.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Elements used to communicate with applications</description>
|
||||
<filename>../../gst/app/.libs/libgstapp.so</filename>
|
||||
<basename>libgstapp.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Convert audio to different formats</description>
|
||||
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
|
||||
<basename>libgstaudioconvert.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adjusts audio frames</description>
|
||||
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
|
||||
<basename>libgstaudiorate.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Resamples audio</description>
|
||||
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
|
||||
<basename>libgstaudioresample.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Creates audio test signals of given frequency and volume</description>
|
||||
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
|
||||
<basename>libgstaudiotestsrc.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Read audio from CD in paranoid mode</description>
|
||||
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
|
||||
<basename>libgstcdparanoia.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>various encoding-related elements</description>
|
||||
<filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
|
||||
<basename>libgstencodebin.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>GIO elements</description>
|
||||
<filename>../../gst/gio/.libs/libgstgio.so</filename>
|
||||
<basename>libgstgio.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Vorbis Tremor decoder</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstivorbisdec.so</filename>
|
||||
<basename>libgstivorbisdec.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>libvisual visualization plugins</description>
|
||||
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
|
||||
<basename>libgstlibvisual.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
|
||||
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
|
||||
<basename>libgstogg.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Pango-based text rendering and overlay</description>
|
||||
<filename>../../ext/pango/.libs/libgstpango.so</filename>
|
||||
<basename>libgstpango.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>various playback elements</description>
|
||||
<filename>../../gst/playback/.libs/libgstplayback.so</filename>
|
||||
<basename>libgstplayback.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Subtitle parsing</description>
|
||||
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
|
||||
<basename>libgstsubparse.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>transfer data over the network via TCP</description>
|
||||
<filename>../../gst/tcp/.libs/libgsttcp.so</filename>
|
||||
<basename>libgsttcp.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Theora plugin library</description>
|
||||
<filename>../../ext/theora/.libs/libgsttheora.so</filename>
|
||||
<basename>libgsttheora.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>default typefind functions</description>
|
||||
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
|
||||
<basename>libgsttypefindfunctions.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Colorspace conversion</description>
|
||||
<filename>../../gst/videoconvert/.libs/libgstvideoconvert.so</filename>
|
||||
<basename>libgstvideoconvert.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adjusts video frames</description>
|
||||
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
|
||||
<basename>libgstvideorate.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Resizes video</description>
|
||||
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
|
||||
<basename>libgstvideoscale.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Creates a test video stream</description>
|
||||
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
|
||||
<basename>libgstvideotestsrc.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>plugin for controlling audio volume</description>
|
||||
<filename>../../gst/volume/.libs/libgstvolume.so</filename>
|
||||
<basename>libgstvolume.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Vorbis plugin library</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
|
||||
<basename>libgstvorbis.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>X11 video output element based on standard Xlib calls</description>
|
||||
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
|
||||
<basename>libgstximagesink.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>XFree86 video output plugin using Xv extension</description>
|
||||
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
|
||||
<basename>libgstxvimagesink.so</basename>
|
||||
<version>1.3.2</version>
|
||||
<version>1.3.3</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -32,7 +32,17 @@ A wide range of video and audio decoders, encoders, and filters are included.
|
|||
<location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-plugins-base"/>
|
||||
<browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-plugins-base"/>
|
||||
</GitRepository>
|
||||
</repository>
|
||||
</repository>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
<revision>1.3.3</revision>
|
||||
<branch>1.3</branch>
|
||||
<name></name>
|
||||
<created>2014-06-22</created>
|
||||
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.3.3.tar.xz" />
|
||||
</Version>
|
||||
</release>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
#ifndef _GST_PLUGINS_BASE__STDINT_H
|
||||
#define _GST_PLUGINS_BASE__STDINT_H 1
|
||||
#ifndef _GENERATED_STDINT_H
|
||||
#define _GENERATED_STDINT_H "gst-plugins-base 1.3.2"
|
||||
#define _GENERATED_STDINT_H "gst-plugins-base 1.3.3"
|
||||
/* generated using gnu compiler Debian clang version 3.5.0-2 (trunk) (based on LLVM 3.5.0) */
|
||||
#define _STDINT_HAVE_STDINT_H 1
|
||||
#include <stdint.h>
|
||||
|
|
|
|||
|
|
@ -84,7 +84,7 @@
|
|||
#define GST_PACKAGE_ORIGIN "Unknown package origin"
|
||||
|
||||
/* GStreamer package release date/time for plugins as YYYY-MM-DD */
|
||||
#define GST_PACKAGE_RELEASE_DATETIME "2014-05-21"
|
||||
#define GST_PACKAGE_RELEASE_DATETIME "2014-06-22"
|
||||
|
||||
/* Define if static plugins should be built */
|
||||
#undef GST_PLUGIN_BUILD_STATIC
|
||||
|
|
@ -325,7 +325,7 @@
|
|||
#define PACKAGE_NAME "GStreamer Base Plug-ins"
|
||||
|
||||
/* Define to the full name and version of this package. */
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 1.3.2"
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 1.3.3"
|
||||
|
||||
/* Define to the one symbol short name of this package. */
|
||||
#define PACKAGE_TARNAME "gst-plugins-base"
|
||||
|
|
@ -334,7 +334,7 @@
|
|||
#undef PACKAGE_URL
|
||||
|
||||
/* Define to the version of this package. */
|
||||
#define PACKAGE_VERSION "1.3.2"
|
||||
#define PACKAGE_VERSION "1.3.3"
|
||||
|
||||
/* directory where plugins are located */
|
||||
#ifdef _DEBUG
|
||||
|
|
@ -368,7 +368,7 @@
|
|||
#undef USE_TREMOLO
|
||||
|
||||
/* Version number of package */
|
||||
#define VERSION "1.3.2"
|
||||
#define VERSION "1.3.3"
|
||||
|
||||
/* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most
|
||||
significant byte first (like Motorola and SPARC, unlike Intel). */
|
||||
|
|
|
|||
Loading…
Reference in a new issue