+=== release 1.11.90 ===
+
+2017-04-07 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.11.90
+
+2017-04-07 15:06:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ po: Update translations
+
+2017-04-06 12:01:00 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: streamline and improve AudioSpecificConfig parsing
+ AudioSpecifigConfig is used in a variety of AAC streams but was
+ being parsed differently. Instead, make everyone use the same parsing.
+ * Remove unused 'bits' field (it was always set to 0 if present)
+ * Add proper GAConfig parsing (to know the number of samples per frame
+ if present).
+ Fixes wrong rate/channels configuration in streams coming from qtdemux
+ https://bugzilla.gnome.org/show_bug.cgi?id=780966
+
+2017-04-05 09:46:31 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Fix 32bit only printf format
+ The previous patch was using %llu for 64bits printf, which is 32bit
+ specific. We also trace the latency in time human readable form now.
+
+2016-03-16 16:22:48 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: set streamparm for outputs that support it
+ Without a specified framerate from the sink, the decoder frame interval
+ should be set using the framerate of the encoded video stream.
+ Therefore, the v4l2object should be able to change the framerate on the
+ output if the V4L2 device accepts it.
+ This is also necessary for mem2mem encoders so that their bitrate
+ calculation code may work correctly and they may report the correct
+ frame duration on the capture queue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=779466
+
+2016-03-16 16:24:55 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: only set latency if the frame duration is valid
+ If the duration of the v4l2object is GST_CLOCK_TIME_NONE, because the
+ sink did not specify a framerate in the caps and the driver accepts the
+ framerate, the decoder element uses GST_CLOCK_TIME_NONE to calculate and
+ set the element latency.
+ While this is a bug of the capture driver, the decoder element should
+ not use the invalid duration to calculate a latency, but print a warning
+ instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=779466
+
+2016-11-23 12:17:55 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Block in preroll_wait on unlock
+ The correct behaviour of anything stuck in the ->render() function
+ between ->unlock() and ->unlock_stop() is to call
+ gst_base_sink_wait_preroll() and only return an error if this returns an
+ error, otherwise, it must continue where it left off!
+ https://bugzilla.gnome.org/show_bug.cgi?id=774945
+
+2017-04-05 15:55:20 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: Add warnings for unsupported frame formats
+ At least output an element warning on the bus when we
+ encounter a frame format GStreamer doesn't currently support.
+
+2017-04-04 17:55:13 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Handle Parametric Stereo with HE-AAC(v2)
+ According to ISO/IEC:14496-2:2009 , in the case of HE-AACv2 (audioObjecType
+ 29) parametric stereo is used (a single mono track is used and then
+ transformations are applied to it to provide a stereo output).
+ We therefore report two channels in the case where there is one reported
+ in the audioChannelConfiguration.
+ Fixes the various issues where a demuxer would report two channels, but
+ then the parser would say there's only one channel, and then the decoder
+ would output two channels.
+
+2017-04-04 15:22:25 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Simplify buffer refcounting in add_buffer() and remove unneeded NULL checks
+
+2017-04-04 15:08:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Select the best pad based on the cached last_buf if any
+ last_buf is the one we're going to write next, not buf. As such we
+ should check timestamps against that one if there is one to select the
+ earliest pad.
+ Also remember the currently selected pad in the very beginning when
+ storing the first last_buf.
+ This both solves some edge cases where not the correct next pad was
+ selected corresponding to the target interleave.
+
+2017-04-04 15:07:40 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Error out immediately if a timecode is to be written but downstream return not-OK
+
+2017-04-03 11:34:49 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Update variables before early exit
+ This is an update of d78d5896272d78df41e696fac929e7dfb3bb3dfa
+ We still exit as early as possible in case of non-ok/non-unlinked combined
+ flow, but we first make sure that we update the internal position variables.
+ This ensures that if upstreams "ignores" the flow return (and carries on pushing),
+ we don't end up processing data with completely bogus variables/positions.
+
+2017-03-24 00:11:13 +1300 Douglas Bagnall <douglas@halo.gen.nz>
+
+ * gst/interleave/interleave.c:
+ * gst/interleave/interleave.h:
+ interleave: avoid using uninitialised ordering_map
+ If self->channel_positions == NULL (which seems unlikely),
+ self->default_channels_ordering_map will be used unintialised.
+ We avoid that by keeping track of the channel_mask, which is set when
+ the ordering map is initialised.
+ https://bugzilla.gnome.org/show_bug.cgi?id=780331
+
+2017-03-23 23:56:31 +1300 Douglas Bagnall <douglas@halo.gen.nz>
+
+ * gst/interleave/interleave.c:
+ interleave: don't overflow channel map with >64 channels
+ When there are more than 64 channels, we don't want to exceed the
+ bounds of the ordering_map buffer, and in these cases we don't want to
+ rempa at all. Here we avoid doing that.
+ https://bugzilla.gnome.org/show_bug.cgi?id=780331
+
+2017-03-28 14:23:16 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Use get_pkgconfig_variable instead of calling pkg-config ourself
+ It is avalaible in meson 0.36 which is now are requirement
+
+2017-03-28 14:22:41 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * pkgconfig/gstreamer-plugins-good.pc.in:
+ * pkgconfig/meson.build:
+ pkgconfig: Do not ever build an installed .pc file
+
+2017-03-28 11:15:53 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: test: Fix environment object usage
+
+2017-03-28 11:14:47 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ * pkgconfig/gstreamer-plugins-good.pc.in:
+ * pkgconfig/meson.build:
+ pkgconfig: Generate the pkg-config with meson too
+
+2017-03-27 21:52:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: In gap mode, consider the mdat offset when calculating the remaining mdat size
+ The mdat generally does not start at offset 0, we have to include the
+ size of the moof and whatever else was in front of the mdat.
+
+2017-03-27 11:43:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atomsrecovery.c:
+ atomsrecovery: Error out when fseek() fails instead of silently ignoring
+ CID 1403262
+
+2017-03-23 22:13:05 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Also add videometa if there is padding to the right and bottom
+ https://bugzilla.gnome.org/show_bug.cgi?id=780478
+
+2017-03-21 12:54:27 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: fix output segment and buffer DTS to correspond to the flattened PTS
+ https://bugzilla.gnome.org/show_bug.cgi?id=780347
+
+2017-03-23 17:53:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Remove some unused variables
+
+2017-03-23 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Remove a couple of unneeded levels of indentation
+
+2017-03-22 18:18:40 +0000 Enrique Ocaña González <eocanha@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: distinguish TFDT with value 0 from no TFDT at all
+ TFDTs with time 0 are being ignored since commit 1fc3d42f. They're
+ mistaken with the case of not having TFDT, but those two cases
+ must be distinguished in some way.
+ This patch passes an extra boolean flag when the TFDT is present.
+ This is now the condition being evaluated, instead of checking for
+ 0 time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=780410
+
+2017-03-22 19:15:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Reset current chunk after writing out timecode
+ If we have multiple tracks with timecodes, or it's not the first track
+ that has timecodes, or not the first buffer, we already started a chunk
+ for media data. We now need to "close" that chunk because we wrote data
+ for the timecode track and a new chunk has to be started for the
+ original track the next time it has data.
+
+2017-03-22 18:52:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Do timecode handling per track, not per muxer instance
+ There could be multiple video tracks with timecodes.
+
+2017-03-22 00:38:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/matroska/matroska-demux.c:
+ qtdemux: matroskademux: Ignore repeated seek events
+ Similar to what was done in adaptivedemux, ignore seek
+ events we've already handled - such as when they are received
+ on every srcpad of files with lots of streams.
+
+2017-03-21 14:55:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ dashdemux: Update mdatleft from overall mdatsize and offset when observing a gap
+ Otherwise mdatleft will have a value calculated from the initial
+ mdatsize minus the parts of the stream that we saw, which is not
+ including all the parts of the stream that might've been skipped.
+
+2017-03-20 17:03:32 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ docs: update two references to the removed 'mad' plugin
+ https://bugzilla.gnome.org/show_bug.cgi?id=776140
+
+2017-03-20 12:03:29 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ rtprtxqueue: add basic documentation and example pipelines
+ Mostly explaining the difference between rtprtxqueue and rtprtxsend.
+
+2017-03-17 20:58:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/meson.build:
+ v4l2: Fix meson plugin shared object name
+ It didn't match between AutoMake and Meson, and the Meson name
+ didn't math the plugin name (video4linux2).
+
+2017-03-16 18:20:54 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtprtxreceive: fix example pipelines and improve the documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=771383
+
+2017-03-17 14:10:40 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstflacparse.h:
+ flacparse: fix playback if sample number does not start at 0
+ This reverts commit 29b807685d3c962bbe8afe351c5dca97d59eb5e0, while
+ fixing the original breaking tests/check/pipelines/flacdec.
+
+2017-03-17 11:30:04 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstflacparse.h:
+ Revert "flacparse: fix playback if sample number does not start at 0"
+ This breaks gst-validate on the build server (though not locally),
+ and a unit test, and I can't run unit tests right now for some
+ unrelated reason.
+ This reverts commit 0747b56f8e7f4731d67f8d13a4bdc453dde0fdf7.
+
+2017-03-16 17:44:41 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: print the correct variable in debug statement
+ This debug statement is meant to print the time since the last (early)
+ RTCP transmission, not the last regular RTCP transmission (which also
+ happens to be set a few lines above to current_time, so the debug output
+ is just confusing)
+
+2017-03-16 17:42:27 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: convert LOG message to TRACE
+ This is printed too often (for every chained buffer!) and just clutters the logs.
+
+2017-03-16 14:58:45 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: fix warning message
+ https://bugzilla.gnome.org/show_bug.cgi?id=780105
+
+2017-03-16 13:54:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstflacparse.h:
+ flacparse: fix playback if sample number does not start at 0
+ https://bugzilla.gnome.org/show_bug.cgi?id=777738
+
+2017-03-15 18:58:55 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsource: get clock-rate from pt if needed to generate SR
+ https://bugzilla.gnome.org/show_bug.cgi?id=780105
+
+2017-03-16 13:52:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Include GStreamer souphttpsrc version in default User-Agent string
+
+2017-03-16 00:41:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: fix crash with empty sprops-parameters
+ https://bugzilla.gnome.org/show_bug.cgi?id=780040
+
+2017-03-11 21:20:40 -0800 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/isomp4/atomsrecovery.c:
+ * gst/isomp4/atomsrecovery.h:
+ atomsrecovery: also handle extra atoms after 'mdia' in a 'trak'
+ Take into account the atoms at the end of the 'trak' atom when
+ recovering it. So that its size (already computed and added in the trak
+ size) isn't making offsets wrong.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771478
+
+2017-03-11 12:56:33 -0800 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: avoid fallthrough to moovrecovery failure section
+ Return before that to preserve our successfull results, otherwise no
+ moov recovery information would be written
+ https://bugzilla.gnome.org/show_bug.cgi?id=771478
+
+2017-03-11 12:27:28 -0800 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/isomp4/atomsrecovery.c:
+ atomsrecovery: expect more atom types at the headers
+ Skip more atoms at the header until it finds the 'mdat' to continue the
+ moov recovery
+ https://bugzilla.gnome.org/show_bug.cgi?id=771478
+
+2017-03-14 16:42:25 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * Makefile.am:
+ * configure.ac:
+ * tests/examples/Makefile.am:
+ * tests/examples/pulse/.gitignore:
+ * tests/examples/pulse/Makefile.am:
+ * tests/examples/pulse/pulse.c:
+ pulse example: Remove
+ That example only tested the property probe interface, which has been removed.
+ The same kind of thing can now be done with the generic gst-device-monitor tool.
+
+2017-03-14 16:38:02 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: Remove unused macro
+
+2017-03-14 16:35:25 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: Remove unused definitions
+
+2017-03-14 10:10:19 +0100 Emeric Grange <egrange@gopro.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtmux: add CineForm support
+ https://bugzilla.gnome.org/show_bug.cgi?id=780024
+
+2017-03-14 15:09:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Only create new chunks if we have more than a single stream
+ There's no point in creating multiple chunks otherwise, it only wastes
+ some bytes for storing the chunk offsets.
+
+2017-03-14 10:09:46 +0100 Emeric Grange <egrange@gopro.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add S16L support
+ https://bugzilla.gnome.org/show_bug.cgi?id=780022
+
+2017-03-14 15:48:08 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/splitmux.c:
+ splitmux test: Use passed first/last timestamps
+ Don't hard-code the expected timestamp range, use the
+ values the caller is passing in.
+
+2017-03-12 11:42:25 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * Makefile.am:
+ * docs/plugins/inspect/plugin-soup.xml:
+ Add old plugin names to cruft list
+ This will help fixing uninstalled setup. Also fix missing path
+ correction in one of the plugin xml.
+ https://bugzilla.gnome.org/show_bug.cgi?id=779344
+
+2016-12-15 12:38:40 +0100 Michael Dutka <mail@michael-dutka.de>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph265depay.c:
+ rtph264depay, rtph265depay: remove stray g_debug()
+ https://bugzilla.gnome.org/show_bug.cgi?id=779858
+
+2017-03-10 11:24:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: init fourcc
+ Initialize the fourcc to 0 so that we can detect failure later.
+
+2017-03-08 22:50:52 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/level.c:
+ * tests/check/elements/rglimiter.c:
+ tests: Add missing LDADD for libm in tests using math.h
+ Also, remove the math.h include for the one that just prentend to need
+ it.
+
+2017-03-08 22:15:46 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ Fix shout2 plugin doc generation
+ In the previous patch, we also renamed shout2send to shout2, so it does
+ not clash with it's feature. Though we forgot to rename it in the doc
+ reference. This patch also add a cruft detection on the xml that made me
+ miss this error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=779344
+
+2017-03-04 10:52:47 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-shout2.xml:
+ * ext/pulse/Makefile.am:
+ * ext/pulse/meson.build:
+ * ext/shout2/gstshout2.c:
+ * ext/soup/Makefile.am:
+ * ext/soup/meson.build:
+ * sys/oss4/Makefile.am:
+ Fix plugin filenames to match plugin names
+ - libgstpulse.so becomes libgstpulseaudio.so
+ - libgstsouphttpsrc.so becomes libgstsoup.so
+ - libgstoss4audio.so becomes libgstoss4.so
+ https://bugzilla.gnome.org/show_bug.cgi?id=779344
+
+2017-03-08 16:01:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Free EDTS instead of just clearing it and setting it to NULL
+
+2017-03-08 15:27:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix some memory leaks related to timecode tracks
+
+2017-03-04 00:34:44 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/splitmux.c:
+ splitmux: Add unit test for reverse playback
+ Ensure that reverse playback works and generates the range
+ of timestamps (0-3s) we expect, in monotonically descending order.
+
+2017-02-28 11:50:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Fix reverse playback
+ Fix the check for whether the start time of the segment has
+ been reached when playing in reverse. Otherwise, playback
+ stops after reaching the start of any file part, instead of
+ continuing until all parts within the segment have played
+
+2017-02-22 03:01:31 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't lose crypto info on a new moof
+ We parse the next moof in advance of having pushed
+ all samples from the previous one in some cases, and
+ we'll still need the crypto info from the previous
+ fragment so keep around any unused crypto info entries
+ when adding new ones
+
+2017-02-27 13:55:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Update modification times when sending the moov
+ https://bugzilla.gnome.org/show_bug.cgi?id=779422
+
+2017-03-01 16:11:47 -0800 Michael Smith <mlrsmith@gmail.com>
+
+ * gst/audioparsers/gstsbcparse.h:
+ sbcparse: Fix up values for allocation enumeration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=779389
+
+2017-02-28 13:10:50 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtprtxreceive: fix potential leak of old, unassociated, association requests
+ https://bugzilla.gnome.org/show_bug.cgi?id=722560
+
+2017-02-28 15:47:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Don't increment -1 / unset indices
+ CID 1398545
+
+2017-02-28 15:20:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Protect against NULL pointer dereference for streams without caps
+ CID 1363332
+
+2017-02-28 12:57:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Free mac on errors
+ CID 1212149
+
+2017-02-28 12:45:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: Add missing break to for loop
+
+2017-02-28 11:02:54 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/Makefile.am:
+ check: Fix splitmux test CFLAGS
+ Needs to know where the gstapp headers are
+
+2017-02-27 21:02:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix compilation with gcc 7
+ qtdemux.c: In function ‘qtdemux_parse_samples’:
+ qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context]
+ if (stream->samples_per_frame * stream->bytes_per_frame) {
+ ~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
+
+2017-02-27 21:01:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: Fix compilation with gcc 7
+ gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’:
+ gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
+ memset (mp3parse->xing_seek_table_inverse, 0, 256);
+ ^~~~~~
+ gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_handle_first_frame’:
+ gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
+ memset (mp3parse->xing_seek_table_inverse, 0, 256);
+ ^~~~~~
+
+2017-02-27 19:31:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: When getting new headers, replace the old version of them
+ This prevents storing an infinite amount of e.g. comment headers if they
+ come without a new initialization header in front of them. There can
+ only be one header of each type.
+
+2017-02-27 19:25:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtp-payloading.c:
+ rtp-payloading: Add new test for Vorbis renegotiation
+ Check if encoding, payloading, depayloading and decoding works if the
+ stream configuration (and thus the headers) change.
+
+2017-02-27 19:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ vorbispay: Only replace headers when receiving a new config header
+ If we also replace all headers when receiving any possibly following
+ comments header, we would throw away the config header before being able
+ to make use of it.
+
+2017-02-23 12:11:15 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/splitmux.c:
+ tests: splitmux: add unit test for content with sparse streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=761086
+
+2017-02-22 11:23:19 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxpartreader: ignore sparse streams when calculating the end offset of a part
+ A sparse stream's ending timestamp can be considerably smaller
+ than the ending timestamps of the other streams, which can lead
+ to skipping considerable time from the next part.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761086
+
+2017-02-22 11:21:06 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxpartreader: identify sparse streams
+
+2017-02-25 21:47:03 -0300 Edgard Lima <edgard.lima@gmail.com>
+
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726depay.h:
+ * gst/rtp/gstrtpg726pay.c:
+ * gst/rtp/gstrtpg726pay.h:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmadepay.h:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmapay.h:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtppcmudepay.h:
+ * gst/rtp/gstrtppcmupay.c:
+ * gst/rtp/gstrtppcmupay.h:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexdepay.h:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpspeexpay.h:
+ * sys/v4l2/gstv4l2.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2colorbalance.c:
+ * sys/v4l2/gstv4l2colorbalance.h:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ * sys/v4l2/gstv4l2tuner.c:
+ * sys/v4l2/gstv4l2tuner.h:
+ * sys/v4l2/gstv4l2vidorient.c:
+ * sys/v4l2/gstv4l2vidorient.h:
+ * sys/v4l2/v4l2_calls.c:
+ * sys/v4l2/v4l2_calls.h:
+ Update Edgard Lima's email
+ https://bugzilla.gnome.org/show_bug.cgi?id=779230
+
+2017-02-08 13:36:00 +0000 Andrew <nifigase@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ rtpjitterbuffer: Don't always reset PTS to 0 after a gap
+ In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
+ timestamps is more than (3 * jbuf->clock_rate) we call
+ rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
+ the pipeline (playes, mixers) just skip frames/samples until pts becomes
+ equal to pts before gap.
+ In version 1.10.2 and before this checking was bypassed for packets with
+ "estimated dts", and gaps were handled correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778341
+
+2017-02-24 15:59:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * meson.build:
+ meson: Update version
+
+2017-02-24 15:37:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.2 ===
+
+2017-02-24 15:07:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ Release 1.11.2
+
+2017-02-24 12:50:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2017-02-24 12:44:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ po: Update translations
+
+2017-02-10 20:50:17 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Extract redirection uri on libsoup's restarted callback
+ Let libsoup handle redirection automatically.
+ And then, to figure out redirection uri, extract it on "restarted"
+ callback which will be fired before soup_session_send() is returned.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778428
+
+2017-01-02 19:29:04 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Update image size when extrapolating
+ Update the image size according the amount of data we are going to
+ read/write. This workaround bugs in driver where the sizeimage provided
+ by TRY/S_FMT represent the buffer length (maximum size) rather then the expected
+ bytesused (buffer size).
+ https://bugzilla.gnome.org/show_bug.cgi?id=775564
+
+2017-02-17 15:50:32 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: fix typo in _acquire_format() error messages
+ Fixes:
+ https://bugzilla.gnome.org/show_bug.cgi?id=778815
+
+2017-02-07 17:27:56 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/matroskamux.c:
+ * tests/check/elements/qtmux.c:
+ tests: matroskamux, qtmux: don't add codec_data buffers to template caps
+ streamheader and codec_data buffers fields are only meant to be
+ in the negotiated caps, not the template caps.
+ Fixes false-positive leaks of those buffers detected by the leaks
+ tracer, as template caps are static, and we decided to not include
+ code in gstreamer core to handle this unusual case of template caps
+ having buffers in them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768762
+
+2017-02-09 12:46:54 +0000 Jochen Henneberg <jh@henneberg-systemdesign.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: Update and send out headers when new headers are received
+ The payloader needs to reset and update the vorbis config data which is
+ pushed on the network if it receives new headers, or at least, it may
+ have to do so.
+ Without this, the stream configuration could change without the
+ payloader sending the new configuration to the other side.
+
+2017-02-15 14:48:58 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Change files on incompatible caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-15 16:35:01 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Reset ready_for_output on state change
+ https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-15 15:09:06 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Remove unused next_max_out_running_time
+ https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-15 15:07:32 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Remove unused muxed_out_time
+ https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-17 13:07:05 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "qtdemux: Always snap to the start of the keyframe"
+ This reverts commit 107902ec514bd826aa29d2298107e2c091e1c779.
+ This commit intended to ensure that keyframe seeks land at the
+ start timestamp of a keyframe, rather than in the middle of one,
+ but they cause trouble on files with sparse streams, or with
+ JPEG 'cover art' tracks that have only one or a few JPEG samples
+ with very long durations.
+ That's still desirable for doing seamless cutting of videos,
+ but needs a rethink for implementation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778690
+
+2017-02-17 01:22:11 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/audiofx/audioecho.c:
+ * gst/audiofx/audioecho.h:
+ audiofx/echo: added surround-delay and surround-mask
+ Add a new boolean surround-delay property that makes
+ audioecho just apply a delay to certain channels to create
+ a surround effect, rather than an echo on all
+ channels. This is useful when upmixing from stereo - for example.
+ Add a surround-mask property to control which channels
+ are considered surround sound channels when adding a
+ delay with surround-delay = true
+ Original patch from Jochen Henneberg <jh@henneberg-systemdesign.com>
+
+2017-02-15 00:13:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Use IP_MULTICAST_ALL for filtering IPv4 packets if available
+ This goes around the inefficient control message based filtering and
+ does all the filtering kernel-side. Unfortunately this is Linux-only and
+ there is no IPv6 variant of it (yet).
+
+2017-02-14 19:53:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ meson: dist meson build files
+ Ship meson build files in tarballs, so people who use tarballs
+ in their builds can start playing with meson already.
+
+2017-02-10 10:53:05 +0100 Søren Juul <zpon.dk@gmail.com>
+
+ * gst/icydemux/gsticydemux.c:
+ * tests/check/elements/icydemux.c:
+ icydemux: reset tags on empty value
+ Some radio streams uses StreamTitle='' to reset the title after a
+ track stopped playing, e.g. while the host talks between tracks or
+ during news segments.
+ This change forces an empty tag object to be distributed if
+ StreamTitle or StreamUrl is received with empty value, thus allowing
+ downstream elements to get notified about this.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778437
+
+2017-02-13 11:17:25 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Properly notify missing elements
+ If the srtp elements are not present, post a message on the bus
+ informing about the missing plugins.
+
+2017-02-10 10:32:57 -0300 Juan Pablo Ugarte <ugarte@endlessm.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: mark singleton caps as "may be leaked" objects.
+ Set MAY_BE_LEAKED flag on static pads returned by gst_v4l2_object_get_*_caps()
+ functions. Made functions thread safe by using g_once_init[enter|leave]
+ funtions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778453
+
+2017-02-09 14:18:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Remove now unused done label
+
+2017-02-09 12:55:32 +0100 Nick Kallen <nickkallen@me.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: do not cache caps
+ Upstream elements like videoflip can transform caps, such as changing width and height.
+ When an imagefreeze downstream receives an ACCEPT_CAPS query it will NOW return
+ all caps that it can accept.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778389
+
+2017-02-09 11:29:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Add a comment about how atom_trak_set_elst_entry() works
+
+2014-08-22 09:55:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ qtdemux: demote some log messages to TRACE level
+ Don't spam debug log with uninteresting stuff.
+
+2017-02-08 17:24:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Clear edit lists every time we recalculate them
+ We recalculate them, so any old information has to be forgotten.
+ Otherwise we write invalid edit lists when writing headers multiple
+ times.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778330
+
+2017-02-07 13:10:18 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: Allow for buffers before the segment when measuring
+ Used signed calculations when measuring the max_ts of an input
+ fragment, so as to calculate the correct duration and offset
+ when buffers have timestamps preceding their segment
+
+2017-02-02 12:55:25 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsession: relate received FIRs and PLIs to source
+ This is needed in order to:
+ - Avoid ignoring requests for different media sources.
+ - Add SSRC field in the GstForceKeyUnit event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=778013
+
+2017-01-30 20:20:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: sanity check number of segments in edit list
+ Fixes crash with fuzzed file.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777940
+
+2017-01-02 22:16:39 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Skip seeking query if upstream format is time
+ Don't need to querying byte-format seeking for time-format
+ upstream case
+ https://bugzilla.gnome.org/show_bug.cgi?id=776715
+
+2016-12-01 12:47:08 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use upstream's StreamFlags if there are
+ When multiple demuxer's are used, upstream might want to indicate
+ default streams using GST_STREAM_FLAG_{SELECT, UNSELECT}
+ https://bugzilla.gnome.org/show_bug.cgi?id=775440
+
+2017-01-27 16:14:16 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Timecode track fixes for STSD entry
+ The n_frames field (frames per second) should follow the nominal frame
+ rate for drop-frame timecodes.
+ Also, the trak's timescale (and duration, accordingly) should follow the
+ STSD entry's timescale and frame duration (fps_n and fps_d accordingly),
+ not the other way around.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777832
+
+2017-01-19 11:08:11 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: retry request on early termination from the server
+ Fix a regression introduced by commit 183695c61a54f1 (refactor to use
+ Soup's sync API). The code previously attempted to reconnect when the
+ server closed the connection early, for example when the stream was put
+ in pause for some time.
+ Reintroduce this feature by checking if EOS is received before the
+ expected content size is downloaded. In this case, do the request
+ starting at the previous read position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776720
+
+2017-01-10 09:40:56 -0700 Matt Staples <staples255@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: find_stream_by_channel should ignore unconfigured streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=777101
+
+2017-01-25 18:43:00 +0000 Brendan Shanks <brendan.shanks@teradek.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix debug typo and remove misleading warning
+ https://bugzilla.gnome.org/show_bug.cgi?id=777362
+
+2017-01-25 20:56:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/client-PCMA.c:
+ rtp: Remove unused variable in example
+ client-PCMA.c:84:22: warning: unused variable 'isrc' [-Wunused-variable]
+ GObject *session, *isrc, *osrc;
+ ^
+
+2017-01-09 11:32:35 +0530 Rahul Bedarkar <rahul.bedarkar@imgtec.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: check for not NULL before clearing adapter
+ In case wavparse receives a manually injected FLUSH_STOP event
+ while operating in pull mode we get criticals because we'd try
+ to clear a NULL adapter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777123
+
+2017-01-24 19:23:44 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Properly use ':' for defining keywords
+
+2017-01-17 16:41:58 +0100 Jean-Christophe Trotin <jean-christophe.trotin@st.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: reference memory before the buffer is queued
+ In gst_v4l2_allocator_qbuf(), the memory is referenced after the
+ buffer is queued. Once queued (VIDIOC_QBUF), the buffer might be handled
+ by the V4L2 driver (e.g. decoded) and dequeued (gst_v4l2_allocator_dqbuf),
+ through a different thread, before the memory is referenced (gst_memory_ref).
+ In this case, in gst_v4l2_allocator_dqbuf(), the memory is unreferenced
+ (gst_memory_unref) before having been referenced: the memory refcount
+ reaches 0, and the memory is freed.
+ So, to avoid this crossing case, in gst_v4l2_allocator_qbuf(), the
+ memory shall be referenced before the buffer is queued.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777399
+
+2017-01-24 17:59:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Only write 4 byte zero padding to the Video Sample Description in MOV
+ For MP4 this is not defined, and it actually breaks things for MSE in
+ Chrome if we do this. For MOV this is required by some broken software
+ but the official specification says it's optional:
+ https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=777540
+
+2017-01-02 13:42:04 +0100 Santiago Carot-Nemesio <scarot@twilio.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpstats: Keep number of nacks sent/received per source
+ Currently, the nack packets sent or received are kept at session level,
+ which makes it impossible to distinguish how many of these packages were
+ sent/received per ssrc when several sources are in the same session. This
+ patch is aligned with the https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats
+ https://bugzilla.gnome.org/show_bug.cgi?id=776714
+
+2016-12-08 15:59:33 +0100 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Fix handling of config-interval
+ Insert VPS/SPS/PPS before the first NAL unit containing an I-frame in an
+ access unit only. If an access unit consists of several such NAL units
+ (tiles) VPS/SPS/PPS should only be inserted before the first of them so
+ that parameters are only updated between frames.
+ Do not insert VPS/SPS/PPS before P-frames when config-interval is -1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775817
+
+2017-01-19 12:29:44 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: report a useful error message when soup_session_send fails
+ This helps to understand cases where libsoup doesn't set the message
+ status code after running soup_session_send.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-19 11:05:00 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: properly check that seek range was respected
+ This check must be done only when we are sure the request was
+ successfully sent. soup_session_send() might fail without setting the
+ status code. In this case status code is 0 so we would only catch the
+ error after the seek range check. In this case we would report an error
+ saying that the seek range was not respected, instead of reporting the
+ underlying error that triggered the soup_session_send() failure.
+ https://bugzilla.gnome.org/attachment.cgi?bugid=777222
+
+2017-01-09 21:04:51 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
+ gdkpixbufoverlay: add a positioning coefficient pair
+ ... so as to allow one clearly defined (absolute) positioning mode
+ that can cater for a variety of absolute but also relative positioning
+ with respect to edge or center.
+
+2017-01-21 20:48:22 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufoverlay: update composition in _before_transform
+ ... since we need to determine passthrough mode for buffer preparation before
+ calling into _transform_ip.
+
+2017-01-07 20:11:13 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufoverlay: handle setting NULL gdkpixbuf
+ ... which is a clearer way to clear any current overlay, other than
+ fiddling with alpha or positioning properties to make it virtually go away.
+
+2017-01-20 17:16:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Stop reading a ncdt sub-tag if it goes behind the surrounding tag
+ https://bugzilla.gnome.org/show_bug.cgi?id=777532
+
+2017-01-20 07:58:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix various out of bounds reads when parsing ncdt tags
+ https://bugzilla.gnome.org/show_bug.cgi?id=777500
+
+2017-01-19 13:46:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Increment current stts index whenever we finished one stts entry
+ Otherwise we could read more chunks than there are available, doing an
+ out of bounds read and potentially crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777469
+
+2017-01-19 13:25:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "qtdemux: Increment current stts index in all code paths after reading one chunk"
+ This reverts commit 99d5d7570d0b53dad3bc8eb653b1320ee422aace. It broke
+ playback of various valid files.
+
+2017-01-19 07:52:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Increment current stts index in all code paths after reading one chunk
+ Otherwise we could read more chunks than there are available, doing an
+ out of bounds read and potentially crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777469
+
+2017-01-19 08:37:37 +0100 Edward Hervey <edward@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Initialize return variable
+ In the normal use-case we would end up with ret being unitialized
+ causing havoc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-13 12:27:40 +0000 David Warman <dwarman@manglebit.org>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: avoid XMP tag parsing fatal error.
+ qtdemux_handle_xmp_taglist() requires a writable taglist,
+ but qtdemux->tag_list can become non-writable, specifically
+ after sending global tags (qtdemux.c:958), which adds a
+ second reference. Ensure the list is made writable before
+ calling (make_writable will copy the list if necessary).
+ https://bugzilla.gnome.org/show_bug.cgi?id=766177
+
+2016-05-31 13:17:45 -0300 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: rework taglist handling
+ Keep taglist around during element existance to avoid having to
+ create it at different places before usage. Makes code simpler to handle.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766177
+
+2017-01-16 11:58:02 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: make flow return values handling clearer
+ The flow return values was stored in the element before because the
+ result had to be set from callbacks. This is not the case anymore, we
+ can return the flow result directly from functions, making the code
+ easier to understand.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-13 16:40:43 +0100 Arnaud Vrac <avrac@freebox.fr>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: properly track redirections
+ The current code configures libsoup to handle redirections
+ transparently, without informing the caller, thus preventing the element
+ to record the redirect code and location uri.
+ Fix this by always setting the SOUP_MESSAGE_NO_REDIRECT, preventing
+ libsoup from handling the redirection. When we receive a redirection
+ request and libsoup can safely handle it, return a custom error which
+ triggers a retry with the new URI.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-17 10:53:39 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: add 4444 and 4444xq variants to video/x-prores pad template caps
+ They are handled since commit 7b565475bf551c53b8eed46f7086f3b372f1f6c4
+ (qt: Add support for ProRes 4444 XQ).
+ https://bugzilla.gnome.org/show_bug.cgi?id=777377
+
+2017-01-17 10:48:57 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ matroska: Quiet a WARN when parsing push mode
+ This warning was noisy when returning EOS, which is
+ just used to indicate more data is needed from upstream.
+
+2017-01-16 14:50:22 +0100 Georg Lippitsch <glippitsch@toolsonair.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't write Sync Sample Atom for ProRes
+ https://bugzilla.gnome.org/show_bug.cgi?id=777331
+
+2015-01-28 08:58:26 +0100 Enrico Jorns <ejo@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/v4l2_calls.h:
+ v4l2: Remove usage and definition of LOG_CAPS macro
+ Unlike former definitions of LOG_CAPS, the current implementation simply
+ expands to GST_DEBUG_OBJECT. The LOG_CAPS macro is rarely used and most
+ uses duplicate already existing GST_DEBUG_OBJECT lines. Therefore, the
+ caps are often printed twice which unnecessarily clutters the debug log.
+ Replace LOG_CAPS calls with GST_DEBUG_OBJECT, remove LOG_CAPS calls, and
+ delete the definition of LOG_CAPS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776899
+
+2017-01-16 15:40:43 +0100 Jean-Christophe Trotin <jean-christophe.trotin@st.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: remove duplicated line of code
+ https://bugzilla.gnome.org/show_bug.cgi?id=777330
+
+2017-01-16 15:17:15 +0100 Jean-Christophe Trotin <jean-christophe.trotin@st.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: fix memory type in allocator probe
+ The buffer memory type provided to the VIDIOC_CREATE_BUFS ioctl shall
+ be set with the value ("memory") given as input parameter of the
+ gst_v4l2_allocator_probe() function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777327
+
+2017-01-14 15:27:19 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: fix other icon counter check
+ It's never going to be 0 if we first increment and then check.
+
+2017-01-14 15:16:53 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: boldly assume that first 'covr' image is the front cover
+
+2017-01-14 15:09:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: extract cover art images into GST_TAG_IMAGE not PREVIEW_IMAGE
+ These are usually much bigger than icon size and required by
+ iTunes to be certain fairly large sizes. In qtmux it is also
+ the IMAGE tags which we write out as 'covr' atoms.
+
+2017-01-14 15:05:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: also set PICTURE tag width and height if available
+
+2017-01-14 14:58:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: fix encoder init error with some GST_TAG_PREVIEW_IMAGEs
+ The encoder fails to initialise when we try to set GST_TAG_PREVIEW_IMAGEs
+ sent to use by qtdemux from iTunes-generated m4a files. We should
+ not just blindly translate the PREVIEW tag to file icon image types,
+ but check if the specific conditions required are met (i.e. image
+ type 1 must be a 32x32 PNG icon, and what we're getting is 500x500).
+ https://bugzilla.gnome.org/show_bug.cgi?id=776962
+
+2017-01-13 12:39:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: bump version
+
+2016-12-22 17:40:40 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Add option for timecode-based split
+ If this option is given, it will calculate the next split point based on
+ timecode difference.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774209
+
+2017-01-13 00:01:06 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't reset request pad numbering across uses
+ When reset, don't restart request pad numberings, as
+ request pads can survive across state changes. Only
+ restart at 0 if all request pads are handed back first.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777174
+
+2017-01-11 18:52:28 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxqueue.h:
+ rtxqueue: Expose basic statistics as properties.
+ Statistics about the total number of retransmission requests
+ and the actual number of retransmitted packets can be helpful
+ at application-level.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777182
+
+2017-01-12 17:45:35 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: simplify video/x-h264 caps handling
+ 'stream-format' and 'alignment' are defined in pad template caps so
+ there is no need to check them again here. Also remove bitrate parsing from
+ caps as bitrate in caps doesn't make sense but from tags, which is
+ actually the case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777181
+
+2016-12-08 17:02:22 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: add basic HEVC/H.265 muxing support
+ https://bugzilla.gnome.org/show_bug.cgi?id=736752
+
+2017-01-11 18:29:05 +0100 Georg Lippitsch <glippitsch@toolsonair.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Calculate clean aperture size
+ Calculate clean aperture dimensions by first guessing
+ display aspect ratio based on pixel aspect ratio and
+ frame size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777100
+
+2017-01-10 18:19:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtmux: Write tapt atom for MOV files if PAR not 1/1
+ Needed for QuickTime 7 to properly play files.
+ Also write the clap atom for MOV files always, not only when ProRes is
+ used as a video codec. It's mandatory for MOV.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777100
+
+2017-01-12 16:32:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.1 ===
+
+2017-01-12 15:31:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ Release 1.11.1
+
+2017-01-12 14:38:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2017-01-12 14:36:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ * po/hr.po:
+ * po/id.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2017-01-11 17:53:32 -0800 Andre McCurdy <armccurdy@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: free seqh after calling qtdemux_parse_svq3_stsd_data()
+ The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to
+ be freed by the caller after use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777157
+ Signed-off-by: Andre McCurdy <armccurdy@gmail.com>
+
+2017-01-10 16:01:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ isomp4: Don't spam debug log with knonw/padding atoms
+ Only output WARNING messages for atoms we don't know how to handle
+ instead of for padding/known atoms we don't need to do any processing
+ on
+ https://bugzilla.gnome.org/show_bug.cgi?id=777095
+
+2017-01-09 19:05:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ Fix indentation
+
+2017-01-09 19:04:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: fix compiler warning due to c99-ism
+ rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode
+
+2016-11-11 14:31:03 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/autodetect/gstautodetect.c:
+ autodetect: bring the element state down after success
+ Otherwise some messages that are emitted by the element on NULL->READY
+ will not reach the application.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764947
+
+2017-01-08 01:13:32 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Write tfdt atom into fragmented files.
+ The DASH spec requires that tfdt atoms be present, so
+ write one out. ISO/IEC 23009-1:2014 6.3.4.2
+ https://bugzilla.gnome.org/show_bug.cgi?id=708221
+
+2017-01-07 23:55:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't reset output timestamps when no tfdt
+ If a fragmented stream doesn't have a tfdt, don't
+ reset the output timestamps at each fragment boundary
+ by erroneously using the default value of 0. Introduced
+ by commit 69fc48
+ https://bugzilla.gnome.org/show_bug.cgi?id=754230
+
+2016-12-16 16:51:48 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * ext/vpx/meson.build:
+ * gst/equalizer/meson.build:
+ * gst/isomp4/meson.build:
+ * meson.build:
+ meson: Install presets files
+
+2017-01-03 10:12:30 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix some caps leaks
+ https://bugzilla.gnome.org//show_bug.cgi?id=776789
+
+2016-12-22 17:34:08 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Return a bin with a "location" property as a sink
+ Splitmuxsink might be called with a custom bin as a sink. If it has a
+ "location" property, it can be used.
+
+2016-11-18 22:42:18 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmux: Rewrite buffer collection and scheduling
+ Majorly change the way that splitmuxsink collects
+ incoming data and sends it to the output, so that it
+ makes all decisions about when / where to split files
+ on the input side.
+ Use separate queues for each stream, so they can be
+ grown individually and kept as small as possible.
+ This removes raciness I observed where sometimes
+ some data would end up put in a different output file
+ over multiple runs with the same input.
+ Also fixes hangs with input queues getting full
+ and causing muxing to stall out.
+
+2016-11-17 23:40:27 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ * tests/check/elements/splitmux.c:
+ splitmuxsink: Add format-location-full signal
+ Add a new signal for formatting the filename, which receives
+ a GstSample containing the first buffer from the reference
+ stream that will be muxed into that file.
+ Useful for creating filenames that are based on the
+ running time or other attributes of the buffer.
+ To make it work, opening of files and setting filenames is
+ now deferred until there is some data to write to it,
+ which also requires some changes to how async state changes
+ and gap events are handled.
+
+2016-12-31 01:54:01 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Always snap to the start of the keyframe
+ When performing a key-unit seek, always snap to the start ts
+ of the keyframe buffer we landed on so that the keyframe is
+ entirely within the resulting outgoing segment. That seems
+ the most sensible result, since the user requested snapping
+ to the keyframe position.
+
+2016-12-31 01:48:04 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Omit cslg_shift when snapping seeks
+ Segments times and seek requests are stored and handled
+ in raw 'PTS' time, without the cslg_shift - which only applies
+ to outgoing samples. Omit the cslg_shift portion when
+ extracting PTS to compare for internal seek snaps.
+ If the cslg_shift is included, then keyframe+snap-before seeks
+ generate a segment start/stop time that already includes the
+ cslg_shift, and it's then added a 2nd time, causing the
+ first buffer(s) to have timestamps that are out of segment.
+
+2016-12-30 22:31:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Remove bogus check in atom_stsc_add_new_entry()
+ Remove an old check from atom_stsc_add_new_entry() that
+ extends the last entry in the STSC if the samples per chunk
+ matches, as the new interleave merging logic requires that
+ the final entry by updateable. There's already code
+ below which simply merges the final entry into the previous
+ one when needed, so rely on that instead.
+ Fixes asserts like:
+ ERROR:atoms.c:2940:atom_stsc_update_entry: assertion failed:
+ (atom_array_index (&stsc->entries, len - 1).first_chunk == first_chunk)
+
+2016-04-24 21:38:51 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix key_time in gst_qtdemux_adjust_seek()
+ time in segment should be PTS based (not DTS).
+ https://bugzilla.gnome.org/show_bug.cgi?id=765498
+
+2016-12-28 22:49:27 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Pass seek flags when activating.
+ Pass all seek flags when activating a part
+ based on a seek, so that SNAP flags are preserved.
+
+2016-11-26 01:13:19 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmux: Fix a small race in the splitmuxsrc
+ Make sure the state of the parser is set to
+ collecting streams before chaining up to the
+ parent change_state() method, to close a
+ small window that can cause playback to
+ never commence.
+
+2017-01-02 15:06:33 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/amrparse.c:
+ check: Remove dead code
+
+2016-12-31 09:52:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ multifilesink: refactor max_files handling a bit
+ Use GQueue instead of a GSList so we don't have to traverse
+ the whole list to append something every time. And it also
+ keeps track of the number of items in it for us.
+ Add a function to add filenames to the list of old files and
+ use it in more places, so that memory doesn't build up in
+ other modes either if no max_files limit is specified.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766991
+
+2016-05-29 17:21:47 +0100 Ursula Maplehurst <ursula@kangatronix.co.uk>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: don't leak memory when no max-files limit is set
+ Technically we weren't leaking the memory, just storing it internally
+ and never using it until the element is freed. But we'd still use more
+ and more memory over time, so this is not good over longer periods
+ of time. Only keep track of files if there's actually a limit set,
+ so that we will prune the list from time to time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766991
+
+2016-12-29 12:39:20 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: adjust segment stop for KEY_UNIT negative rate seeking
+
+2016-12-29 12:25:35 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: implement pull mode SNAP flag seeking
+
+2016-12-29 11:26:33 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: tweak KEY_UNIT SNAP seek handling
+ Previously, seeking to position y where y is (strictly) within a keyframe
+ would seek to that keyframe both with SNAP_BEFORE and SNAP_AFTER,
+ where the latter is now adjusted to really snap to the next keyframe.
+
+2016-12-28 13:23:11 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: correctly perform pull mode KEY_UNIT seeking
+ Rather amazingly (and equally unnoticed), keyunit seeking resulted in segments
+ where start != time (which is bogus for simple avi timeline). So, properly
+ adjust the segment (start) rather than fiddling with segment time (only).
+
+2016-12-28 13:04:54 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: restore considering of pull mode KEY_UNIT seeking
+ ... by using the original seek event's flags rather than the corresponding
+ segment flags, which do not have such counterpart flags (and
+ do no longer have them covertly sneaking in nowadays).
+
+2015-05-08 12:44:01 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: only drop actual streamheader buffers with xiph codecs
+ With Xiph codecs the stream header buffers are both in the caps and are
+ usually also at the beginning of each input stream, but it's perfectly
+ possible that the input stream does not have the stream header buffers
+ inline in the data. Matroskamux would drop the first N buffers assuming
+ they're stream headers, but this meant it would drop actual payload data
+ when the stream didn't contain the stream headers inline. Fix this by
+ only dropping leading buffers if they're flagged as stream headers. This
+ fixes issues with streams that are being tapped into after streaming
+ has started.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749098
+
+2016-12-21 17:43:58 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * tests/check/elements/matroskamux.c:
+ matroskamux: adjust unit test to modified behaviour
+ Now matroskamux mark all packets of audio-only streams as keyframes so
+ in test_block_group after pushing the test audio data 4 buffers are produced
+ and not more 2. The last buffer is the original data and must match with what
+ pushed. The remaining ones are matroskamux headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=754696
+
+2016-05-30 01:15:31 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: mark all packets of audio-only streams as keyframes
+ This helps with streaming audio-only streams via multifdsink,
+ tcpserversink and such.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754696
+
+2015-03-28 18:15:36 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: add G722 audio support
+ https://bugzilla.gnome.org/show_bug.cgi?id=746574
+
+2016-12-13 11:11:07 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/udp/gstudpsrc.c:
+ updsrc: Add to join multiple multicast interfaces
+ https://bugzilla.gnome.org/show_bug.cgi?id=776030
+
+2015-03-25 13:51:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ rtpklvdepay: add the SPARSE flag to the outgoing stream-start event
+
+2016-12-14 14:37:45 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpmanager: place content before Since-version API marker
+ Avoids confusing the parser
+
+2016-12-14 14:16:53 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: fix 404 in package origin
+
+2016-12-14 21:45:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check if we have enough data available when parsing edit lists
+ Also consume the data entry by entry to get complicated indexing out of
+ the code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776107
+
+2016-12-14 19:15:03 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't check size in a non-list value
+ After commit 1ea9735a I see these error while using the webcam
+ integrated in my laptop:
+ GStreamer-CRITICAL **: gst_value_list_get_size: assertion 'GST_VALUE_HOLDS_LIST (value)' failed
+ The issue is gst_v4l2src_value_simplify() was doing its job of
+ generating a single value, rather than the original list. That why,
+ when getting the list size, a critical warning was raised.
+ This patch takes advantage of the compiler optimizations to verify
+ first if the list was simplified, thus use it directly, otherwise,
+ if it is a list, verify its size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776106
+
+2016-12-14 10:39:12 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests/jitterbuffer: Major refactoring and cleanups
+ * Changed PCMU->TEST for common macros
+ * Changed verify-functions (lost & rtx) into macros.
+ * Remove option to add marker-bit for test-buffers (not used anywhere)
+ * Add new push_test_buffer function that makes sure there are correlation
+ between dts and the time on the clock. (classic test-mistake)
+ * Established a generic starting-point for tests with the
+ construct_deterministic_initial_state function and use it where
+ applicable, which removes lots of "boilerplate" everywhere.
+ * Add basic lost-event test
+ * Remove as much "magic constants" as possible.
+ * Remove 3 tests that no longer are testing anything that others don't,
+ and was completely unmaintainable.
+ * Remove unnecessary use of the testclock
+ * Verify each test is testing what it actually says it does (and modify
+ where it doesn't)
+ In general, make the tests much smaller, better, more maintainable and
+ readable.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774409
+
+2016-12-14 09:54:11 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * gst-plugins-good.spec.in:
+ Remove generated .spec file
+ Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-14 10:15:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check that the XiTh size is big enough
+ https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-09 20:27:53 +0900 Heekyoung Seo <heekyoung.seo@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check node length of video sample description
+ Add check for node length of video sample description and its fields and
+ for the XiTh atom.
+ Also unify the code a bit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-08 18:50:52 +0900 Heekyoung Seo <heekyoung.seo@lge.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Enable xvid/mp2 codec support
+ Add support for xvid video and mp2 audio, add m2v1 fourcc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-13 22:32:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ * tests/check/elements/rtprtx.c:
+ * tests/check/elements/vp9enc.c:
+ gst: Don't declare variables inside the for loop header
+ This is a C99 feature.
+
+2016-12-11 13:27:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: Ensure to reinit buffers whenever they were not allocated yet
+ That is, whenever we go through start/stop we have to ensure that on the
+ next opportunity the buffers are reallocated again. Otherwise the
+ buffers might be NULL because the element was reused with the same
+ configuration as before (i.e. set_caps() wouldn't have reinited the
+ buffers).
+ https://bugzilla.gnome.org/show_bug.cgi?id=775898
+
+2016-12-10 12:52:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/design/Makefile.am:
+ * docs/design/design-rtpauxiliary.txt:
+ * docs/design/design-rtpcollision.txt:
+ * docs/design/design-rtpretransmission.txt:
+ docs: design: remove, moved to gst-docs
+
+2016-12-09 17:17:35 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * meson.build:
+ meson: Support building without Gst debug
+
+2016-12-09 17:55:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ * gst/flx/gstflxdec.h:
+ flxdec: Only send SEGMENT events after CAPS
+ I.e., don't just forward the event but delay it if we don't have caps on
+ the srcpad yet.
+
+2016-12-09 17:49:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: Unref and unmap buffers in all code paths as needed
+ https://bugzilla.gnome.org/show_bug.cgi?id=775888
+
+2016-12-06 17:42:31 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't set empty interlace-mode list
+ If for some reason we fail to probe formats (all try_fmt calls fail, for
+ example), this is not a critical error, but we end up with an empty list
+ of interlace modes. This causes all subsequent negotiation to fail.
+ This patch fixes interlace-mode setting to be skipped if we failed to
+ detect any.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775702
+
+2016-12-07 17:22:22 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/monoscope/gstmonoscope.c:
+ monoscope: Unref allocation query after finished with it
+ https://bugzilla.gnome.org/show_bug.cgi?id=775752
+
+2016-12-06 07:48:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: Allocate 0-initialized memory for the decoded frame
+ Otherwise we might leak arbitrary information from the uninitialized
+ memory if not every pixel is written.
+ https://scarybeastsecurity.blogspot.gr/2016/12/1days-0days-pocs-more-gstreamer-flic.html
+
+2016-12-05 07:57:19 -0700 Matt Staples <staples255@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix session cleanup when handling redirect on PLAY
+ Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
+ removing code from gst_rtspsrc_send that changed the state varable upon
+ encountering a redirect. Better to let the redirect handlers in
+ gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
+ state-dependent cleanup.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775543
+
+2016-09-07 16:10:27 +0300 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: always send teardown request
+ Allow CMD_CLOSE to cancel all commands not only CMD_PAUSE
+ and ignore CMD_WAIT while closing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748360
+
+2016-12-03 08:19:27 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From f980fd9 to 39ac2f5
+
+2016-12-01 17:08:09 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ jitterbuffer: Don't leak duplicate items
+ When providing items with a seqnum, there is a (very small) probability
+ that an element with the same seqnum already exists. Don't forget
+ to free that item if it wasn't inserted.
+ And avoid returning undefined values when dealing with duplicate items
+
+2016-12-01 11:23:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Sanitize unknown codec caps
+ We might have non-printable characters in the unknown fourcc, replace
+ them with '_', in the same way we do it for unknown tags.
+
+2016-12-01 20:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Free vprp chunk also if it existed but we made no use of it
+ https://bugzilla.gnome.org/show_bug.cgi?id=775479
+
+2016-12-01 17:38:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: Fix memory leak when parsing attachments
+ gst_tag_image_data_to_image_sample() does not take ownership of the
+ passed memory, so don't set it to NULL to allow us to free it later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775472
+
+2016-12-01 14:56:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: Unify zlib/bzip2 decompress loops with the ones from qtdemux
+ Especially, simplify the code a bit.
+
+2016-12-01 14:41:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Increase inflate buffer in bigger steps
+ 1024 bytes is quite small, let's do 4096 bytes (or one page).
+ Also remove redundant if, we're always in that case when getting here.
+
+2016-12-01 14:30:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Ensure that size of the pasp atom is as much as we need
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:30:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Free compressed moov node and it's corresponding decompressed data
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:29:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check size of compressed MOOV header against available data
+ And actually read the size of the cmvd atom from the right position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:27:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix zlib inflate loop
+ Handle errors cleanly, deallocate all memory and return the actual size
+ of the inflated data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 13:38:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Make sure we have enough data in the codec_data to be able to parse it
+ Also error out cleanly if mapping the buffer failed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775450
+
+2016-12-01 13:32:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix out of bounds read in tag parsing code
+ We can't simply assume that the length of the tag value as given
+ inside the stream is correct but should also check against the amount of
+ data we have actually available.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775451
+
+2016-12-01 15:06:06 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ rtp: Fix some memory leaks in usage of gst_pad_get_current_caps()
+ https://bugzilla.gnome.org/show_bug.cgi?id=775071
+
+2016-11-30 17:56:02 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Read interlacing information from 'fiel' atom
+ Read interlacing and TFF/BFF information from the 'fiel' atom and pass it
+ into the caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=775414
+
+2016-11-29 13:55:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix compiler warning
+ qtdemux.c: In function ‘qtdemux_parse_trak’:
+ qtdemux.c:10184:38: error: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 9 has type ‘gint {aka const int}’ [-Werror=format=]
+ GST_DEBUG_OBJECT (qtdemux, "Found jpeg: len %u, need %lu", len,
+ ^
+
+2016-11-28 13:45:24 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Change off_t type to gint
+ off_t is a signed integer type provided by sys/types.h on posix systems.
+ Replace with gint for building on non-posix systems (like windows).
+ https://bugzilla.gnome.org/show_bug.cgi?id=775287
+
+2016-11-22 21:00:25 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: add libm to has_function checks
+ The functions from math.h may be implemented in libm.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774876
+
+2016-10-27 23:02:37 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/meson.build:
+ Revert "meson: dv plugin now works on MSVC"
+ This reverts commit 05a89613feff70cff416367f5aa807a1d5c68b63.
+ Let's not put in stuff that needs unreleased Meson. This can go in
+ for the next cycle.
+
+2016-11-28 13:51:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Ensure that tags are valid UTF-8 before adding them to the taglist
+ https://bugzilla.gnome.org/show_bug.cgi?id=775219
+
+2016-11-28 12:22:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: Post an error message on the bus if we got EOS without having added any pads
+
+2016-11-28 12:00:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Handle non-UTF8 headers and error reasons more gracefully
+ Especially don't put them into GstStructures in one way or another, just
+ ignore them or error out cleanly depending on the importance of their
+ content.
+
+2016-11-28 09:30:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvrawpay.c:
+ vrawpay: Error out cleanly if mapping the video frame fails
+ Instead of later dereferencing NULL and crashing.
+
+2016-11-27 11:14:13 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: Update statistics before pushing
+ If an element queries the number of retransmission buffers pushed
+ *while* the push is still taking place (and before the object lock
+ is taken just after) it would end up with the wrong statistic
+ being reported.
+ Increment it just before the push, avoids races when getting statistics
+ https://bugzilla.gnome.org/show_bug.cgi?id=768723
+
+2016-11-26 11:20:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitmodules:
+ common: use https protocol for common submodule
+ https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-07-28 18:51:24 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ gstv4l2bufferpool: lock flush_stop against regular qbuf
+ These can be called from different threads and both manipulate the
+ pool->buffers array. Lock them properly and let flush_stop move the
+ array contents into a temporary array on the stack to avoid having
+ to call release_buffer under the object lock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775015
+
+2016-11-24 14:25:22 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ gstv4l2bufferpool: remove critical error message when process is called on an inactive pool
+ If the pool is inactive, it is guaranteed to also be flushing, so the
+ following check will return GST_FLOW_FLUSHING anyway.
+ This can happen if a v4l2src is blocking on DQBUF in create and is sent
+ an EOS event on another thread. In that case the pool is set to
+ flushing/inactive without locking, the v4l2src is unblocked, and may
+ call pool_process with a valid buffer on the already inactive pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775014
+
+2016-11-24 14:41:52 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: release buffer if create fails
+ gst_base_src_get_range does not expect a buffer to be returned in
+ the error case, so we are leaking a reference here if create fails.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775014
+
+2016-11-23 18:34:04 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Handle create_session() returning NULL in bundle code
+ CID 1394492.
+
+2016-11-22 16:42:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Make sure to only change DTS of writable buffers
+ And trivial cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 16:42:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Error out much earlier if we don't have a valid PTS
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 16:18:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Only use buffer durations if they are actually valid
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 15:59:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Revert commits that set DTS and duration on buffers unconditionally
+ 39f7e52266fde3b3c035e22cbcbb2bb1fa207b17 was setting the buffer duration
+ to 0 if is not valid, under the assumption that this is "the last"
+ buffer and no others are coming next. This is wrong, last_buf is the
+ previous buffer and not the very last one.
+ 4e3c13c87c258c9c95e2217d32ab314d12b5fffc was setting DTS to 0 if there
+ was none. This will set DTS to 0 for all e.g. audio streams, completely
+ messing up calculations if streams don't start at 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 15:58:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Only write "gap" edit list if there is a non-zero gap
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-23 07:09:06 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/flx/flx_color.c:
+ * gst/flx/flx_fmt.h:
+ * gst/flx/gstflxdec.c:
+ * gst/flx/gstflxdec.h:
+ flxdec: rewrite logic based on GstByteReader/Writer
+ Solves overreading/writing the given arrays and will error out if the
+ streams asks to do that.
+ Also does more error checking that the stream is valid and won't
+ overrun any allocated arrays. Also mitigate integer overflow errors
+ calculating allocation sizes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774859
+
+2016-11-23 11:20:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: Don't unref() parent in the chain function
+ We don't own the reference here, it is owned by the caller and given to
+ us for the scope of this function. Leftover mistake from 0.10 porting.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774897
+
+2016-11-22 20:33:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: libvpx's release buffer is sometimes called with fb->priv==NULL
+ Don't assert on this but just ignore these cases.
+
+2016-11-22 20:24:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Fix cluster searching if we search multiple times in one chunk
+ After finding a cluster id in the byte reader, we skip ahead the reader
+ position by one further byte to be able to continue searching from there
+ inside the same chunk if the cluster candidate was a false positive.
+ We have to accomodate for that additional byte when resuming the search,
+ otherwise all following pulls are off-by-one for every resume and we run
+ into an assertion.
+
+2016-11-22 20:01:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-ids.c:
+ matroska: Add size checks to the parsing of FLAC headers
+
+2016-11-22 23:46:00 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: fix some warnings comparing unsigned < 0
+ bf43f44fcfada5ec4a3ce60cb374340486fe9fac was comparing an unsigned
+ expression to be < 0 which was always false.
+ gstflxdec.c: In function ‘flx_decode_brun’:
+ gstflxdec.c:322:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
+ if ((glong) row - count < 0) {
+ ^
+ gstflxdec.c:332:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
+ if ((glong) row - count < 0) {
+ ^
+ https://bugzilla.gnome.org/show_bug.cgi?id=774834
+
+2016-11-21 16:17:31 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Enable up to 16 unpositioned raw audio channels
+ https://bugzilla.gnome.org/show_bug.cgi?id=774789
+
+2016-11-22 19:05:00 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: add some write bounds checking
+ Without checking the bounds of the frame we are writing into, we can
+ write off the end of the destination buffer.
+ https://scarybeastsecurity.blogspot.dk/2016/11/0day-exploit-advancing-exploitation.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=774834
+
+2016-11-21 15:25:23 +0000 David Evans <bbcrddave@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Be sure not to read off end of FLAC dfLa box
+ https://bugzilla.gnome.org/show_bug.cgi?id=773712
+
+2016-11-21 11:48:58 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: add support for skipping invalid data in push mode
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-21 11:48:29 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskaparse: add support for skipping invalid data
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-18 17:00:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Move to new helper function to parse authentication responses
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-20 14:12:16 +0100 christophecvr <stefansat@telenet.be>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix wrong compiler warning with gcc 6.2
+ | ../../../git/gst/isomp4/qtdemux.c: In function 'qtdemux_parse_tree':
+ | ../../../git/gst/isomp4/qtdemux.c:10224:24: error: 'size' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ | offset += size;
+ | ^~
+ | ../../../git/gst/isomp4/qtdemux.c:10197:25: note: 'size' was declared here
+ | guint32 size, tag;
+ | ^~~~
+ https://bugzilla.gnome.org/show_bug.cgi?id=774747
+
+2016-11-20 16:15:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * configure.ac:
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ win32: remove copies of generated headers
+
+2016-11-20 13:14:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavidemux.h:
+ avidemux: Ensure that raw video have properly aligned buffers
+ That is, aligned to to 32 bytes for video. Fixes crashes if the raw
+ buffers are passed to SIMD processing functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774428
+
+2016-11-20 13:08:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Ensure that raw audio and video have properly aligned buffers
+ That is, aligned to the basic type for audio and to 32 bytes for video.
+ Fixes crashes if the raw buffers are passed to SIMD processing functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774428
+
+2016-11-14 14:44:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Always write edit lists for the tracks to give a more accurate duration
+ Always write an edit list for the whole track. In general this is not
+ necessary except for the case of having a gap or DTS adjustment but
+ it allows to give the whole track's duration in the usually more
+ accurate media timescale.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774403
+
+2016-11-18 22:45:45 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove useless return variable
+ qtdemux_expose_streams() returns flow error immediately, if there is an error.
+ So, the variable for the flow return is not needed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774674
+
+2016-11-17 13:59:48 +0000 David Evans <bbcrddave@gmail.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_dump.c:
+ * gst/isomp4/qtdemux_dump.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Add support for FLAC encapsulated in ISOBMFF
+ As defined by
+ https://git.xiph.org/?p=flac.git;a=blob_plain;f=doc/isoflac.txt
+ https://bugzilla.gnome.org/show_bug.cgi?id=773712
+
+2016-11-17 19:59:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Mark pad as needing reconfiguration again if it failed
+ And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:59:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/monoscope/gstmonoscope.c:
+ monoscope: Mark pad as needing reconfiguration again if it failed
+ And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:58:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Mark pad as needing reconfiguration again if reconfiguration failed
+ And consider negotiation failures on flushing pads as FLUSHING, not as
+ NOT_NEGOTIATED.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:56:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdec.c:
+ dvdec: Fix handling of negotiation failures
+ Return NOT_NEGOTIATED if sending the caps event fails, or FLUSHING if
+ the pad was flushing at that point.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 17:16:26 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: add_global_arguments -> add_project_arguments
+ https://bugzilla.gnome.org/show_bug.cgi?id=774656
+
+2016-11-16 10:53:51 +0530 Vinod Kesti <vinodkesti@yahoo.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: pad request fails for flvmux
+ splitmuxsink requests pad from element using pad template like "video_%u", "audio_%u" and "sink_%d". This is true for most of the muxers.
+ But splitmuxsink not able to request pad to flvmux as flvmux has "audio" and "video" as pad templates.
+ fix: splitmuxsink should fallback to "audio" and "video" when template not found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774507
+
+2016-11-17 10:24:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroskaparse: Add remaining relevant parts from a3a55305 to the parser
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-16 22:39:01 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroskaparse: ignore parsing errors at the end of the file
+ This is the same change as a3a55305 for the parser.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-16 08:56:34 +0100 Philippe Normand <philn@igalia.com>
+
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/rtpbundle.c:
+ * tests/check/meson.build:
+ * tests/examples/rtp/.gitignore:
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-rtpbundle.c:
+ * tests/examples/rtp/server-rtpbundle.c:
+ rtpbin: receive bundle support
+ A new signal named on-bundled-ssrc is provided and can be
+ used by the application to redirect a stream to a different
+ GstRtpSession or to keep the RTX stream grouped within the
+ GstRtpSession of the same media type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772740
+
+2016-11-15 16:52:39 +0530 Vinod Kesti <vinodkesti@yahoo.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: assertion while converting ADTS stream to RAW
+ aacparse resizes input buffer while converting ADTS stream to RAW,
+ During buffer resize buffer write permission is not checked.
+ This throws gst_buffer_is_writable assertion and leads to AV sync issue some times.
+ It is corrected by making buffer writeable using gst_buffer_make_writable
+ https://bugzilla.gnome.org/show_bug.cgi?id=774129
+
+2016-11-15 21:17:51 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't modify upstream TIME segment
+ TIME segment implies that stream/running time is being handled by upstream.
+ So, we shouldn't override it without any clue.
+ This patch is for fixing seek in DASH streaming.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774196
+
+2016-11-14 22:33:27 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * config.h.meson:
+ meson: Add define for v4l2-probe config option
+
+2016-11-14 17:37:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Reset caps accumulator to ANY when resyncing the adapter, not EMPTY
+ The accumulator is filled by intersecting with all the pad caps, as such
+ it must be initialized with ANY (like it is before the iteration is
+ started) and not to EMPTY.
+ Fixes the CAPS query always returning EMPTY caps when resyncing happened
+ during the query, e.g. because pads were added/removed.
+
+2016-11-14 12:13:14 +0100 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: remove redundant saddr unref
+ The g_object_unref (saddr) before receiving message seems to be redundant as it
+ is done just before jumping to retry
+ Though not directly related, part of
+ https://bugzilla.gnome.org/show_bug.cgi?id=772841
+
+2016-11-12 23:34:23 +0100 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: receive control messages only in multicast
+ Control messages are used only in multicast mode - to detect if the destination
+ address is not ours and possibly drop the packet. However in non-multicast
+ modes the messages are still allocated and freed even if not used. Therefore
+ request control messages from g_socket_receive_message() only in multicast
+ mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772841
+
+2016-11-11 10:45:01 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/matroska/matroska-mux.c:
+ Use intermediate guint when handling GstVideoMultiviewFlags
+ The underlying integer type of the enum GstVideoMultiviewFlags is
+ implementation defined and may not have the same size as guint.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-11-11 10:44:18 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/multifile/gstsplitfilesrc.c:
+ splitfilesrc: update uri_get_type to match the prototype in GstURIHandlerInterface
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-10-26 22:37:34 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: don't add_global_arguments when being built as a subproject
+ https://bugzilla.gnome.org/show_bug.cgi?id=773568
+
+2016-10-21 15:49:36 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstflacparse.h:
+ flacparse: fix header rewriting being ignored
+ https://bugzilla.gnome.org/show_bug.cgi?id=727802
+
+2016-11-09 06:25:27 +0000 Sean DuBois <sean@siobud.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Add metadatacreator property
+ Allow users to set metadatacreator value in the meta packet
+ https://bugzilla.gnome.org/show_bug.cgi?id=774131
+
+2016-11-01 19:56:36 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Use first buffer TS as mux start time
+ Do not use last buffer TS + buffer duration because buffer duration
+ might be inaccurate, especially for frame rates like 30fps where a
+ rounding error is observed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773785
+
+2016-11-03 15:03:59 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix timer-reuse bug
+ When doing rtx, the jitterbuffer will always add an rtx-timer for the next
+ sequence number.
+ In the case of the packet corresponding to that sequence number arriving,
+ that same timer will be reused, and simply moved on to wait for the
+ following sequence number etc.
+ Once an rtx-timer expires (after all retries), it will be rescheduled as
+ a lost-timer instead for the same sequence number.
+ Now, if this particular sequence-number now arrives (after the timer has
+ become a lost-timer), the reuse mechanism *should* now set a new
+ rtx-timer for the next sequence number, but the bug is that it does
+ not change the timer-type, and hence schedules a lost-timer for that
+ following sequence number, with the result that you will have a very
+ early lost-event for a packet that might still arrive, and you will
+ never be able to send any rtx for this packet.
+ Found by Erlend Graff - erlend@pexip.com
+ https://bugzilla.gnome.org/show_bug.cgi?id=773891
+
+2016-10-09 15:59:05 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix lost-event using dts instead of pts
+ The lost-event was using a different time-domain (dts) than the outgoing
+ buffers (pts). Given certain network-conditions these two would become
+ sufficiently different and the lost-event contained timestamp/duration
+ that was really wrong. As an example GstAudioDecoder could produce
+ a stream that jumps back and forth in time after receiving a lost-event.
+ The previous behavior calculated the pts (based on the rtptime) inside the
+ rtp_jitter_buffer_insert function, but now this functionality has been
+ refactored into a new function rtp_jitter_buffer_calculate_pts that is
+ called much earlier in the _chain function to make pts available to
+ various calculations that wrongly used dts previously
+ (like the lost-event).
+ There are however two calculations where using dts is the right thing to
+ do: calculating the receive-jitter and the rtx-round-trip-time, where the
+ arrival time of the buffer from the network is the right metric
+ (and is what dts in fact is today).
+ The patch also adds two tests regarding B-frames or the
+ “rtptime-going-backwards”-scenario, as there were some concerns that this
+ patch might break this behavior (which the tests shows it does not).
+
+2016-11-03 16:33:53 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix bug in reschedule_timer
+ The new timeout is always going to be (timeout + delay), however, the
+ old behavior compared the current timeout to just (timeout), basically
+ being (delay) off.
+ This would happen if rtx-delay == rtx-retry-timeout, with the result that
+ a second rtx attempt for any buffers would be scheduled immediately instead
+ of after rtx-delay ms.
+ Simply calculate (new_timeout = timeout + delay) and then use that instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773905
+
+2016-11-03 13:27:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/wavparse.c:
+ * tests/files/Makefile.am:
+ * tests/files/audiotestsrc.wav:
+ tests: wavparse: add test for processing an actual .wav file
+ https://bugzilla.gnome.org/show_bug.cgi?id=773861
+
+2016-11-03 12:34:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't set caps to NULL after setting them on the srcpad
+ We would like to check later on EOS if we found a known stream type or
+ not, to possibly post an error message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773861
+
+2016-11-02 14:33:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't deref NULL pads in debug output
+ That tends to crash.
+
+2016-11-02 11:46:07 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ isomp4: Don't use gst_video_colorimetry_to_string_full()
+ The API was reverted. Just use the plain
+ gst_video_colorimetry_to_string() function.
+
+2016-11-02 11:00:13 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix GObject warnings on shutdown.
+ Commit 83e718 added a pad template to splitmux request
+ pads, which means that GstElement now releases the pads on
+ dispose, but after having removed all elements in the bin
+ and unlinked them. Make sure we can handle cleanup in that case
+ without throwing assertions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773784
+
+2016-11-02 02:25:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multifile/gstsplitmuxsrc.h:
+ splitmuxsrc: Store seek seqnum and send it on EOS / segment events.
+ GES relies on the EOS event having the seqnum of the seek that
+ caused it.
+
+2016-11-02 02:25:00 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Forward a not-linked error on the bus
+ Handle not-linked as for other fatal errors and post it
+ onto the bus so the app knows
+
+2016-11-01 21:00:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix compiler warning
+ qtdemux.c: In function ‘qtdemux_parse_tree’:
+ qtdemux.c:10139:16: error: ‘color_table_id’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ if (color_table_id != 0) {
+ ^
+ qtdemux.c:10121:19: note: ‘color_table_id’ was declared here
+ guint16 color_table_id;
+ ^~~~~~~~~~~~~~
+
+2016-10-20 17:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use a default interleave of 250ms for all codecs
+ https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-10-19 14:33:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use a default interleave when ProRes is used
+ The ProRes guidelines suggest an interleave of 0.5s is common, but
+ specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
+ be used per chunk.
+ It might also make sense to use similar numbers in general.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-10-19 14:25:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Allow configuring the interleave size in bytes/time
+ Previously we were switching from one chunk to another on every single
+ buffer. This wastes some space in the headers and, depending on the
+ software, might depend in more reads (e.g. if the software is reading
+ multiple samples in one go if they're in the same chunk).
+ The ProRes guidelines suggest an interleave of 0.5s is common, but
+ specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
+ be used per chunk. This will be handled in a follow-up commit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-09-30 18:22:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Set compressor name, horizontal/vertical resolution and depth for ProRes
+ This is also required by some software to handle ProRes files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 18:05:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux.c:
+ qt: Add support for ProRes 4444 XQ
+ And also 4444 in the muxer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 17:58:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtmux: Write 'clap' atom for ProRes
+ It's required for ProRes to work with other software.
+ It is also in the MP4 standard, but inventing values here seems a bit
+ tricky for the general case and it does not really give any extra
+ information.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 09:55:58 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Read colorimetry information from colr atom if available
+ https://bugzilla.gnome.org/show_bug.cgi?id=772181
+
+2016-09-29 21:56:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Always write colr atom with the colorimetry information
+ https://bugzilla.gnome.org/show_bug.cgi?id=772181
+
+2016-09-29 18:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix writing of the 'fiel' extension atom
+ This was also wrong for JPEG2000. Also write it for all MOV files and
+ JPEG2000, not only for ProRes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-29 17:40:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Write 4 bytes of zeroes at the end of the sample description extensions
+ This is working around some broken software.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-28 20:55:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ atoms: 'pasp' atom is also part of MP4, write it always
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-07-11 19:30:12 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Write additional atoms for prores video
+ These required atoms are: colorimetry, field information, spatial/temporal
+ quality, and vendor.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2014-06-16 17:20:32 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: Don't drop mode b packets with picture start code
+ Some buggy payloaders, e.g. rtph263pay, may use mode B for packets
+ that starts with a picture (or GOB) start code although it's not
+ allowed. Let's be nice and not drop these packets/frames.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773516
+
+2016-06-22 13:59:35 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263ppay: Fix caps leak
+ Fix leaking caps when downstream has not-fixed caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773515
+
+2016-10-26 16:42:19 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Fix indentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=773514
+
+2016-10-18 11:35:58 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Use GST_TRACE_OBJECT for logging bitstream parsing
+ Bump the bitstream parsing to TRACE log level so it doesn't flood the
+ output when trying to read the more useful DEBUG and LOG messages.
+ Also use GST_DEBUG_OBJECT instead of GST_DEBUG in various places
+ https://bugzilla.gnome.org/show_bug.cgi?id=773514
+
+2016-10-18 11:09:10 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Fix leak for B-fragments
+ Altough commits 6a16be7, 64f9d08 and 0c7e3a8 fixed some issues they
+ introduced others. This patch fixes the leak of one macroblock for every
+ B fragment.
+ Macroblock structures must not be freed immediately after finding the
+ boundaries as they are stored and used later. However the inital dummy
+ structure (used for finding the first boundary) must be freed.
+ CID #1212156
+ https://bugzilla.gnome.org/show_bug.cgi?id=773512
+
+2016-10-20 13:14:13 +0200 Alejandro G. Castro <alex@igalia.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpbin: avoid generating errors when rtcp messages are empty and check the queue is not empty
+ Add a check to verify all the output buffers were empty for the
+ session in a timout and log an error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773269
+
+2016-10-26 13:21:29 +0200 Alejandro G. Castro <alex@igalia.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpbin: pipeline gets an EOS when any rtpsources byes
+ Instead of sending EOS when a source byes we have to wait for
+ all the sources to be gone, which means they already sent BYE and
+ were removed from the session. We now handle the EOS in the rtcp
+ loop checking the amount of sources in the session.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773218
+
+2016-10-21 17:31:00 +0000 Matt Staples <staples255@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Also handle redirect on PLAY
+ https://bugzilla.gnome.org/show_bug.cgi?id=772610
+
+2016-08-30 10:24:43 +0200 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: allow missing control attribute in case of a single stream
+ Improve RFC2326 - chapter C.3 compatibility:
+ In case just a single stream is specified in SDP and the control attribute
+ is missing do not drop the stream but rather assume "a=control:*"
+ https://bugzilla.gnome.org/show_bug.cgi?id=770568
+
+2016-10-08 18:11:17 +0200 William Manley <will@williammanley.net>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2: Warn, don't assert if v4l gives us a buffer with a too large size
+ I've seen problems where the `bytesused` field of `v4l2_buffer` would be
+ a silly number causing the later call to:
+ gst_memory_resize (group->mem[i], 0, group->planes[i].bytesused);
+ to result in this error to be printed:
+ (pulsevideo:11): GStreamer-CRITICAL **: gst_memory_resize: assertion 'size + mem->offset + offset <= mem->maxsize' failed
+ besides causing who-knows what other problems.
+ We make the assumption that this buffer has still been dequeued correctly
+ so just clamp to a valid size so downstream elements won't end up in
+ undefined behaviour.
+ The invalid `v4l2_buffer` I saw from my capture device was:
+ buffer = {
+ index = 0,
+ type = 1,
+ bytesused = 534748928, // <- Invalid
+ flags = 8260, // V4L2_BUF_FLAG_TIMESTAMP_MONOTONIC | V4L2_BUF_FLAG_ERROR | V4L2_BUF_FLAG_DONE
+ field = 01330, // <- Invalid
+ timestamp = {
+ tv_sec = 0,
+ tv_usec = 0
+ },
+ timecode = {
+ type = 0,
+ flags = 0,
+ frames = 0 '\000',
+ seconds = 0 '\000',
+ minutes = 0 '\000',
+ hours = 0 '\000',
+ userbits = "\000\000\000"
+ },
+ sequence = 0,
+ memory = 2,
+ m = {
+ offset = 3537219584,
+ userptr = 140706665836544, // Could be nonsense, not sure
+ planes = 0x7ff8d2d5b000,
+ fd = -757747712
+ },
+ length = 2764800,
+ reserved2 = 0,
+ reserved = 0
+ }
+ This is from gdb with my own annotations added.
+ This was with gst-plugins-good 1.8.1, a Magewell XI100DUSB-HDMI video
+ capture device and kernel 3.13 using a dodgy HDMI cable which is great at
+ breaking HDMI capture devices. I'm using io-mode=userptr and have built
+ gst-plugins-good without libv4l.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769765
+
+2016-10-20 20:41:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use a better default value for the movie header timescale
+ Take the maximum video timescale, or if no video track is present the
+ previous value of 1800.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769041
+
+2016-10-20 20:07:19 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Be more clever with the default video track timescale
+ Use the number of milliframes per second for integral and drop-frame
+ framerates, as suggested by the QT file format specification and other
+ places. We already did that for integral framerates before, but not for
+ drop-frame framerates. This now keeps precision better.
+ For all other framerates, check if it's close to a well-known framerate
+ and use that instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769041
+
+2016-10-10 13:00:01 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: extract interlaced information from jpeg video
+ This information is hidden in a small chunk of data.
+ Format found at https://developer.apple.com/standards/qtff-2001.pdf,
+ page 92, "Video Sample Description", under table 3.1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767771
+
+2016-10-26 12:46:28 +0530 Jagadish <jagadishkamathk@gmail.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufoverlay: Fixing x and y offset computation
+ While computing the x and y offsets, it's the video resolution and
+ resized overlay resolution to be used instead of actual overlay image
+ resoltuion. Due to this, the overlay image used to get wrongly overlayed
+ in undesired location
+ https://bugzilla.gnome.org/show_bug.cgi?id=757292
+
+2016-11-01 18:09:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+2016-10-24 16:56:31 +0000 Enrique Ocaña González <eocanha@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use the tfdt decode time on byte streams when it's significantly different than the time in the last sample
+ We consider there's a sifnificant difference when it's larger than on second
+ or than half the duration of the last processed fragment in case the latter is
+ larger.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754230
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.10.0 ===
+
+2016-11-01 17:57:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.10.0
+
+2016-11-01 17:47:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-11-01 17:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ po: Update translations
+
+2016-10-27 12:01:55 +0200 Tobias Schneider <tobias.schneider@voiceinterconnect.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fix extra-controls leak
+ Gst struct v4l2object->extra_controls is created if user sets appropriate
+ option but it is not freed on destruction of v4l2object.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773580
+
+2016-10-31 18:00:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ Revert "souphttpsrc: reduce reading latency by using non-blocking read"
+ This reverts commit 8816764112408766889c8b680a3af51115df4bf5.
+ It causes issues with the timeouts, and causes connections to be closed
+ without actual reason. Needs further investigation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773509
+
+2016-10-31 09:00:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't try to add srcpad if we don't know valid caps yet
+ Otherwise we'll run into an assertion on specially crafted files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773643
+
+2016-10-27 11:23:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ meson: Remove uselessly duplicated dep checks
+ These checks are done inside the meson.build files for each plugin.
+
+2016-10-27 11:22:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/meson.build:
+ meson: dv plugin now works on MSVC
+ Needs a Meson patch to filter out the useless -lpthread
+ https://github.com/mesonbuild/meson/pull/962
+
+2016-10-27 14:03:48 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: allow resolutions above 4096
+ Modify the caps string to allow width and height greater than 4096.
+ There is no need to restrict it since the matroska format allows the
+ width and height values to be up to eight bytes long.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773582
+
+2016-10-23 17:23:10 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Check for G_PLATFORM_WIN32 for presence of ipi_spec_dest
+ G_OS_WIN32 is only set when not building with cygwin, but
+ ipi_spec_dest is missing both with and without cygwin.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-26 08:51:40 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: reset read_position when reading fails
+ souphttpsrc maintains two variables for the position:
+ * 'request_position' is where we want to be
+ * 'read_position' is where we are
+ During Normal operations both are updated in sync when data arrives. A seek
+ changes 'request_position' but not 'read_position'.
+ When the two positions get out of sync, then a new request is send and the
+ 'Range' header is adjusted to the current 'request_position'.
+ Without this patch, if reading fails, then the source is destroyed. This
+ triggers a new request, but the range remains unchanged. As a result, the
+ old range is used and old data will be read.
+ Changing the 'read_position' to -1 makes it explicitly different from
+ 'request_position' and as a result the 'Range' header is updated correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773509
+
+2016-10-25 08:54:34 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: Don't depend on gstreamer-check-1.0 on windows
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-25 15:24:20 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: reset connection info to non-flushing when closing
+ This solves a hanging mainloop in following scenario:
+ * connect to source
+ * network/server drops
+ * pipeline set to NULL (and connection to flushing as part)
+ * pipeline set to PAUSED/PLAYING (connection to non-flushing, but not recorded)
+ * [connecting still not possible]
+ * pipeline set to NULL => mainloop hangs (since no actual flushing is done)
+
+2016-10-26 14:32:48 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Only allow one video request pad
+ The pacing of the overall muxing is controlled
+ by the video GOPs arriving, so we can only handle
+ 1 video stream, and the request pad is named accordingly.
+ Ignore a request for a 2nd video pad if there's already
+ an active one.
+
+2016-10-26 11:59:32 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Take ownership of floating refs
+ sink the floating ref when handed a muxer or sink to use so
+ we clearly take ownership.
+
+2016-10-25 14:51:52 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Set child elements to NULL when removing.
+ Make sure that elements are in the NULL state when removing.
+ Fixes critical warnings when errors occur early on in starting up.
+
+2016-10-25 14:50:53 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Set pad template on request sink pads
+ Ensure that the ghost pad returned as a request pad
+ has the template that was requested
+
+2016-10-25 10:50:47 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * tests/check/meson.build:
+ Revert "meson: move gstreamer-check-1.0 dependency to tests/check"
+ This reverts commit 46632694662b96fddb848a1f2091a215b28a2d35.
+ Does not actually work. See:
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114#c31
+
+2016-06-08 11:24:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Assume PTS is DTS when PTS is missing
+ This fixes issue for encoders that only sets the DTS. We assume that
+ there was no re-ordering when that happens.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762207
+
+2016-10-24 00:34:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ meson: fix build outside of gst-all
+
+2016-10-21 00:42:54 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * sys/directsound/meson.build:
+ meson: directsound: Add ole32 library dependency
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-21 00:42:18 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ * tests/check/meson.build:
+ meson: move gstreamer-check-1.0 dependency to tests/check
+ https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-20 22:08:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/videomixer.c:
+ tests: videomixer: disable racy flush_start_flush_stop test
+ It's been broken for years, and it's unlikely it will ever
+ be fixed for collectpads/videomixer now that there's compositor
+ which works fine. So let's disable it, since all it does
+ is that it creates noise that distracts from other failures.
+ Also see the corresponding adder bug as it failed in the same way:
+ https://bugzilla.gnome.org/show_bug.cgi?id=708891
+
+2016-10-09 16:56:10 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: Fix souphttpsrc tests without CK_FORK=no
+ It seems that the forked processes all attempt to handle the listening
+ socket from the server, and only one has to shutdown the socket to break
+ the server completely.
+ Create a new server inside each test to avoid this.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772656
+
+2016-10-09 15:23:51 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * tests/check/elements/level.c:
+ tests: Fix level test in CK_FORK=no mode
+ The tests accumulate buffers in GstCheck's buffers list, and the list is
+ not (consistently) reset between tests. Do that and remove the now
+ conflicting unrefs for outbuffers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772644
+
+2016-10-07 13:04:27 +0530 Gaurav Gupta <g.gupta@samsung.com>
+
+ * sys/waveform/gstwaveformsink.c:
+ waveformsink: Fix Memory leak using GST_PTR_FORMAT
+ https://bugzilla.gnome.org/show_bug.cgi?id=772497
+
+2016-10-18 12:23:42 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/monoscope/meson.build:
+ meson: Add missing gstaudio dep to monoscope
+ In file included from ../subprojects/gst-plugins-good/gst/monoscope/gstmonoscope.c:42:0:
+ ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
+ #include <gst/audio/audio-enumtypes.h>
+ ^
+ compilation terminated.
+ https://ci.gstreamer.net/job/GStreamer-master-meson/271/console
+
+2016-10-16 02:18:22 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/multifile/meson.build:
+ meson: Add missing pbutils dependency to multifile
+ Found via the Jenkins CI:
+ FAILED: subprojects/gst-plugins-good/gst/multifile/gstmultifile@sha/gstsplitmuxsink.c.o
+ [...]
+ In file included from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.h:24:0,
+ from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.c:59:
+ ../subprojects/gst-plugins-base/gst-libs/gst/pbutils/pbutils.h:30:43: fatal error: gst/pbutils/pbutils-enumtypes.h: No such file or directory
+ #include <gst/pbutils/pbutils-enumtypes.h>
+ ^
+ compilation terminated.
+ https://ci.gstreamer.net/job/GStreamer-master-meson/263/console
+
+2016-10-15 22:11:08 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ meson: Don't set c_std to gnu99
+ Use the default for each compiler on every platform instead. This
+ improves our compatibility with compilers that don't have gnu99 as
+ a c_std.
+
+2016-10-04 18:04:11 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ * tests/check/getpluginsdir:
+ * tests/check/meson.build:
+ meson: Make use of new environment object and set plugin path to builddir
+ Workaround source_root being the root directory of all projects in the subproject
+ case and remove now unneeded getpluginsdir
+ Bump meson requirement to 0.35
+
+2016-10-06 11:15:54 +0530 Gaurav Gupta <g.gupta@samsung.com>
+
+ * tests/examples/rtp/client-rtpaux.c:
+ tests: Fix memory leak in test rtpaux test
+ https://bugzilla.gnome.org/show_bug.cgi?id=772496
+
+2016-10-03 11:27:54 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Forward latency queries to upstream
+ Without this, latency queries to imagefreeze will fail.
+
+2016-09-30 11:35:39 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * hooks/pre-commit.hook:
+ * meson.build:
+ * tests/check/getpluginsdir:
+ meson: Setup pre commit hook and fix getpluginsdir for standalone case
+
+2016-09-29 04:55:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Handle stop point from segment
+ If the seek stop point (or start, during reverse play)
+ was within the segment we just finished, go EOS immediately
+ instead of proceeding through all other parts and sending
+ 0 length seeks to them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-29 03:21:26 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Drop lock shutting down pads
+ Avoid a sporadic deadlock on shutdown by dropping
+ the splitmux lock around pad shutdown
+ https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-29 02:47:36 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: Fix extra unref handling queries
+ https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-29 04:50:25 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Avoid stall when parts get out of sync
+ When one part moves ahead of the others - due to excessive
+ downstream queueing, or really small input files - then
+ we can end up activating parts more than once. That can lead to
+ effects like shutting down pad tasks prematurely.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-30 11:41:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+=== release 1.9.90 ===
+
+2016-09-30 13:02:19 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.9.90
+
+2016-09-30 12:17:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-09-30 11:43:54 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ po: Update translations
+
+2016-09-30 13:22:32 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * tests/check/pipelines/tagschecking.c:
+ tests: Fix tagschecking failure due to missing PTS
+ qtmux now needs the PTS (commit a993883b7), so let's make sure we
+ produce one with our buffers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772228
+
+2016-09-28 23:03:58 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE
+ Just error out if there is no valid PTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772143
+
+2016-09-29 17:37:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Add JPEG2000 ihdr atom to the list of known ones
+ Otherwise qtdemux is always going to complain about it being unknown.
+
+2016-09-29 10:19:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Always write the default frame duration for VP8/9 too
+ The WebM spec allows this now, and it allows us to guess a framerate.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=772141 and
+ also https://bugzilla.gnome.org/show_bug.cgi?id=654379
+
+2016-09-27 15:26:19 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph265depay.c:
+ rtph26[45]depay: Don't handle NALs inside STAP units twice
+ They've already been handled before pushing them into the adapter.
+
+2016-09-27 12:39:12 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/meson.build:
+ meson: tests: fix vp8 availability checks
+ Those variables are not defined if vp8 was not found.
+
+2016-09-27 10:23:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ Revert "multifilesink: streamline the file-switch code a bit"
+ This reverts commit f1ceaab02f3f557e23b77b14771a575788f92bb4.
+ This broke atomic file writes in "buffer" mode. It did make
+ sure that any streamheaders are prepended to each file in
+ buffer mode as well, but that's not really needed in practice,
+ whereas atomic file writes are, so let's restore the status
+ quo ante for now since this was primarily a code cleanup anyway,
+ and if anyone needs to streamheaders in buffer mode too they
+ can make a patch to implement that differently. Re-implementing
+ the atomic writes in the element also seems way too much work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766990
+
+2016-09-27 10:22:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ Revert "multifilesink: close file on write error with next-file mode is set to buffer"
+ This reverts commit 84e441d2685cf223d348a95be0c5ba693bbf6624.
+ This will no longer be needed once we revert f1ceaab02.
+
+2016-09-26 13:22:29 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Add gst-plugins-base plugins directories to be used by tests
+
+2016-09-26 14:30:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/meson.build:
+ * meson.build:
+ * tests/check/getpluginsdir:
+ * tests/check/meson.build:
+ meson: add unit tests
+ Only works properly in an installed setup currently, most
+ likely won't work with a subprojects setup yet.
+
+2016-09-24 09:36:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * po/meson.build:
+ meson: hook up translations
+
+2016-09-08 17:30:41 +0530 Arun Raghavan <arun@arunraghavan.net>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: Don't negotiate to less than two segments
+ GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
+ sure that if our buffer parameters are such that the maxlength is not at
+ least 2x fragsize, we still request the ringbuffer to keep that much
+ space so it continues to work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770446
+
+2016-09-24 23:22:01 +0530 Arun Raghavan <arun@arunraghavan.net>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsbcpay.h:
+ rtpsbcpay: Fix timestamping
+ We were just picking the timestamp of the last buffer pushed into our
+ adapter before we had enough data to push out.
+ This fixes things to figure out how large each frame is and what
+ duration it covers, so we can set both the timestamp and duration
+ correctly.
+ Also adds some DISCONT handling.
+
+2016-07-12 18:14:52 +0200 Georg Lippitsch <glippitsch@toolsonair.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix fourcc for ProRes Proxy
+ This is apco, according to
+ https://wiki.multimedia.cx/index.php?title=Apple_ProRes
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-18 20:55:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/meson.build:
+ meson: fix build with vpx 1.3.x
+ vpx >= 1.4.0 is optional
+
+2016-09-15 18:19:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Use new bin suppressed flags API for managing the element flags
+
+2016-09-15 09:52:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/jack/gstjackaudioclient.c:
+ * gst/rtp/dboolhuff.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/videofilter/gstvideoflip.c:
+ ext, gst: fix indentation
+
+2016-09-15 09:52:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/flvmux.c:
+ * tests/check/elements/rtph263.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ * tests/check/elements/rtpsession.c:
+ * tests/check/elements/rtpvp9.c:
+ tests: fix indentation
+
+2016-08-11 11:04:22 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
+ Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one
+ definitely lost packets is encountered.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769757
+
+2016-08-11 23:07:44 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: improved rtx-rtt averaging
+ The basic idea is this:
+ 1. For *larger* rtx-rtt, weigh a new measurement as before
+ 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
+ 3. For very large measurements, consider them "outliers"
+ and count them a lot less
+ The idea being that reducing the rtx-rtt is much more harmful then
+ increasing it, since we don't want to be underestimating the rtt of the
+ network, and when using this number to estimate the latency you need for
+ you jitterbuffer, you would rather want it to be a bit larger then a bit
+ smaller, potentially losing rtx-packets. The "outlier-detector" is there
+ to prevent a single skewed measurement to affect the outcome too much.
+ On wireless networks, these are surprisingly common.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-05 12:51:59 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
+ Assuming equidistant packet spacing when that's not true leads to more
+ loss than necessary in the case of reordering and jitter. Typically this
+ is true for video where one frame often consists of multiple packets
+ with the same rtp timestamp. In this case it's better to assume that the
+ missing packets have the same timestamp as the last received packet, so
+ that the scheduled lost timer does not time out too early causing the
+ packets to be considered lost even though they may arrive in time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-27 10:39:50 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Don't request rtx if 'now' is past retry period
+ There is no need to schedule another EXPECTED timer if we're already
+ past the retry period. Under normal operation this won't happen, but if
+ there are more timers than the jitterbuffer is able to process in
+ real-time, scheduling more timers will just make the situation worse.
+ Instead, consider this packet as lost and move on. This scenario can
+ occur with high loss rate, low rtt and high configured latency.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-26 18:01:48 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix lost duration when gap after lost timer
+ This patch fixes an issue with the estimated gap duration when there is
+ a gap immediately after a lost timer has been processed. Previously
+ there was a discrepancy beteen the gap in seqnum and gap in dts which
+ would cause wrong calculated duration. The issue would only be seen with
+ retranmission enabled since when it's disabled lost timers are only
+ created when a packet is received and the actual gap length and last dts
+ is known.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-19 01:11:58 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Expose rtx-deadline as a property
+ The default -1 gives the old behavior.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-11 12:02:19 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Improved expected-timer handling when gap > 0
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-11 11:51:50 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Major improvements for RTX stats
+ Stats should also be collected for unsuccessful packets.
+ rtx-rtt is very important for determining the necessary configured
+ latency on the jitterbuffer. It's especially important to be able to
+ increase the latency when retransmitted packets arrive too late and are
+ considered lost. This patch includes these late packets in the
+ calculation of the various rtx stats, making them more correct and
+ useful.
+ Also in the case where the original packet arrives after a NACK is sent,
+ the received RTX packet should update the stats since it provides useful
+ information about RTT.
+ The RTT is only updated if and only if all requested retranmissions are
+ received. That way the RTT is guaranteed to make sense. If not we don't
+ know which request the packet is a response to and the RTT may be bogus.
+ A consequence of this patch is that RTT is not updated for a request
+ when one of the RTX packets for that seqnum is lost, but that since
+ measured RTT will be more accurate.
+ The implementation store the RTX information from the timed out timers
+ and use this when the retransmitted packet arrives. For performance
+ these timers are stored separately from the "normal" timers in order to
+ not impact performance (see attached performance test).
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-11 11:02:44 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Add and expose more stats and increase testing of it
+ Add num-pushed and num-lost.
+ Expose num-late, num-duplicates and avg-jitter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-07 10:20:02 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtxreceive: Set buffer flag for retransmitted packets
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-09 23:47:41 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Option to disable rtx-delay-reorder
+ When disabled we can save some iterations over timers.
+ There is probably an argument for rtx-delay-reorder to exist, but
+ for normal operations, handling jitter (reordering) is something a
+ jitterbuffer should do, and this variable feels like functionality that
+ is not "in-sync" with what the jitterbuffer is trying to achieve.
+ Example: You have 50ms jitter on your network, and are receiving
+ audio packets with 10ms durations. An audio packet should not be
+ considered late until its rtx-timeout has expired (and hence a rtx-event
+ is sent), but with rtx-delay-reorder, events will be sent pretty much
+ all the time due to the jitter on the network.
+ Point being: The jitterbuffer should adapt its size to the measured network
+ jitter, and then rtx-delay-reorder needs to adapt as well, or simply
+ get out of the way and let the other (better) rtx-mechanisms do their job.
+ Also change find_timer to only use seqnum as an argument, since there
+ will only ever be one timer per seqnum at any given time. In the
+ one case where the type matters, the caller simply checks the type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-09-14 09:58:41 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Fix double free from coverity
+ CID #1372887
+
+2016-09-14 09:58:37 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Indent as per gst-indent
+
+2016-09-14 11:30:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Depend on gstreamer 1.9.2.1
+
+2016-09-14 10:17:02 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/autodetect/gstautodetect.c:
+ autodetect: Use gst_bin_set_suppressed_flags() API
+ https://bugzilla.gnome.org/show_bug.cgi?id=771395
+
+2016-09-09 15:36:12 +0200 Thomas Scheuermann <Thomas.Scheuermann@barco.com>
+
+ * ext/jack/gstjackaudioclient.c:
+ jack: Fix pipeline hang when jack changes sample rate or buffer size
+ If jackd changes the buffer size or sample rate, jackaudiosink hangs
+ and can't be stopped. This also happens if jack is configured as slave
+ and a gstreamer pipeline is started on the slave machine while the jack
+ master isn't running yet. If the the jack master is started it changes
+ the buffer size / sample rate and jackaudiosink can't be stopped.
+ This fix calls jack_shutdown_cb when jack_sample_rate_cb or
+ jack_buffer_size_cb is called.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771272
+
+2016-09-12 20:08:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Fix field ordering for reverse playback
+ And actually calculate the field duration instead of a frame duration so
+ that we can properly timestamp output frames in fields=all mode.
+ This is probably still broken for reverse playback in telecine mode.
+
+2016-09-12 09:02:00 +0000 Thomas Klausner <tk@giga.or.at>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix compilation on NetBSD
+ https://bugzilla.gnome.org/show_bug.cgi?id=771278
+
+2016-09-10 20:51:10 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b18d820 to f980fd9
+
+2016-09-09 14:02:25 +0200 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: offset is irrelevant when no crypto info
+ Cause later it will try to use the crypto info array to get an index and
+ attach on of the positions as buffer's crypto info.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770951
+
+2016-09-10 09:53:57 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From f49c55e to b18d820
+
+2016-09-07 15:33:30 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/osxaudio/Makefile.am:
+ osxaudio: Distribute device provider files
+ Those where missing the the dev release tarballs for 1.9.2 which
+ prevented building from tarball on OSX platform
+
+2016-09-06 09:49:39 +0200 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix crash with no cenc aux offset
+ https://bugzilla.gnome.org/show_bug.cgi?id=770951
+
+2016-09-05 09:39:33 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: parse a bit more of the humongous LOAS data
+ https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:39:08 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: make it clear when a potential LOAS frame is not one
+ https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:38:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: add a few comments to anchor parsing to the spec
+ https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:37:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstaacparse.h:
+ aacparse: improve channel/rate handling
+ Keep track of the last parsed channels/rate fields so they can be
+ used even if the element was not yet configured.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:35:53 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix varlength number reading as per spec
+ https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:35:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: strip uneeded static arrays slack
+ https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-07-18 19:18:58 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4adepay.h:
+ rtpmp4adepay: Only declare a stream to be framed once a marker bit has been seen
+ This may cause a few packets to be processed by the parser, but it's
+ better than never pushing out buffers from a slightly broken stream
+ where no marker bits are set.
+
+2016-09-06 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Fix timestamping in reverse playback mode
+ This is only supported right now if after a demuxer that supports reverse
+ playback, e.g. with DV container inside AVI container.
+
+2016-09-05 12:23:54 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: Bump version to 1.9.2
+
+2015-06-26 20:13:17 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/isomp4/GstQTMux.prs:
+ * gst/isomp4/Makefile.am:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Implement the preset interface.
+ + And provide a "youtube" preset, which based on
+ https://support.google.com/youtube/answer/1722171 sets
+ faststart to True.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751559
+
+2016-09-01 12:27:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.2 ===
+
+2016-09-01 12:27:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.9.2
+
+2016-09-01 11:23:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: Update translations
+
+2016-09-01 10:59:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/equalizer/demo.c:
+ * tests/examples/spectrum/demo-audiotest.c:
+ * tests/examples/spectrum/demo-osssrc.c:
+ tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS
+ We use gdk_cairo_create() which is deprecated since 3.22.
+
+2016-08-31 05:50:44 +1000 Jan Schmidt <jan@centricular.com>
+
+ * sys/osxvideo/Makefile.am:
+ * sys/osxvideo/cocoawindow.h:
+ * sys/osxvideo/osxvideosink.h:
+ osxvideo: Remove QuickTime references.
+ QuickTime.h is no longer available on OS X 10.12 (Sierra),
+ and both the header and the framework seem unnecessary
+ for compilation - at least as of 10.11 (El Capitan).
+ https://bugzilla.gnome.org/show_bug.cgi?id=770526
+
+2016-08-19 11:11:03 -0700 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * gst/avi/gstavidemux.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/flv/gstflvdemux.c:
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/wavparse/gstwavparse.c:
+ Use the new API to post flow ERROR messages on the bus
+ https://bugzilla.gnome.org/show_bug.cgi?id=770158
+
+2016-08-26 21:32:07 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/elements/.gitignore:
+ gitignore: ignore qtdemux, rtph261 and rtpvp9 tests
+
+2016-08-26 21:22:16 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/Makefile.am:
+ tests: use GST_NET_LIBS instead of hardcoded -lgstnet
+ Fixes build in OSX when running 'make check' in gst-uninstalled.
+
+2016-08-26 21:14:47 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: remove a wrong 'const' specifier
+ Fixes "error: duplicate 'const' declaration specifier"
+
+2016-08-26 21:11:59 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ build: silence error about pthread for 'make check' in osx
+ Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2016-08-26 20:31:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/Makefile.am:
+ vp9enc: Fix build of unit test by letting it link to libgstvideo
+
+2016-08-26 12:06:35 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems"
+ This broke API, so we need a better solution!
+ This reverts commit c7579d31a6e9d788e94b83258309063d0aae481e.
+
+2016-06-08 15:06:28 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtpvp9.c:
+ rtpvp9depay: Support flexible mode
+
+2016-06-06 17:03:36 +0200 Stian Selnes <stian@pexip.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/vp9enc.c:
+ vp9enc: Fix leak of vpx_image_t
+
+2016-05-06 13:33:22 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263pdepay: Don't try to push empty frame
+ If the result of depayloading is an empty frame, just drop it. This is
+ likely the result of a buggy payloader.
+
+2016-05-06 16:06:53 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ rtpmux: fix PROP_TIMESTAMP_OFFSET range problems
+ It could not set the offset for the full guint32 range.
+
+2016-05-06 09:44:42 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: introduce max-streams property
+ To be able to cap the number of allowed streams for one session.
+ This is useful for preventing DoS attacks, where a sender can change
+ SSRC for every buffer, effectively bringing rtpbin to a halt.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770292
+
+2016-03-31 00:10:49 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: reordered packets are very normal, and should not be a warning
+
+2016-02-05 14:19:25 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: degrade g_warning to GST_ERROR
+ So we don't blow up while investigating
+
+2016-02-04 14:16:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263pdepay: Fix picture header for non-writable payload
+ Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
+ the payload. In this case the payload modifications will not affect the
+ rtp buffer. So instead of modifying the payload buffer directly we
+ should modify the buffer that actually gets pushed on the adapter.
+
+2015-11-19 11:50:47 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph261depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtph261.c:
+ rtph261depay: Fix check of valid payload length
+ Packets with no H.261 payload should be dropped to avoid invalid
+ write/reads.
+
+2015-11-09 10:06:21 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263pay: Fix double free, invalid reads and leak
+
+2014-06-30 15:43:58 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: sanity check RTT before ignoring PLI/FIR
+
+2014-06-30 15:07:45 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: handle sdes messages with non-utf8 more gracefully
+
+2014-06-17 08:52:50 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: change log level on bitstream parsing messages
+
+2016-07-07 11:13:18 +0200 Mikhail Fludkov <misha@pexip.com>
+
+ * tests/check/elements/rtprtx.c:
+ tests/rtprtx: refactor the tests to use gstharness
+ The functionality of all the tests was kept exactly the same. Some tests
+ were renamed:
+ test_push_forward_seq -> test_rtxsend_rtxreceive
+ test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
+ test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
+ test_rtxreceive_data_reconstruction was testing that retransmitted
+ buffer produced by rtxsend was correctly transformed to the original
+ buffer by rtxreceive. Now we are checking for this in all the tests
+ where both rtxsend & rtxreceive are involved. That's why the test was
+ removed.
+
+2016-08-25 15:52:36 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Set RTP marker bit
+ Set the RTP marker bit on the last RTP packet of an H.265 access unit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770394
+
+2016-07-26 19:39:58 +0200 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst/videofilter/gstvideoflip.c:
+ * gst/videofilter/gstvideoflip.h:
+ videoflip: added GstVideoDirection interface
+ It implements now this interface with its video-direction
+ property. Values are changed to GstVideoOrientationMethod but they have
+ the same value than the originals.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768687
+
+2015-11-06 10:39:16 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ gstrtpsession: refactor duplicate code into a function
+ Less code, easier to read, more consistent.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770293
+
+2016-08-23 17:06:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: fix typo in max-misorder-time property name
+
+2016-08-22 00:05:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix printf format compiler warning in debug message
+ On 32-bit x86: gstsplitmuxsink.c:966:31: warning: format ‘%u’ expects
+ argument of type ‘unsigned int’, but argument 9 has type
+ ‘guint64 {aka long long unsigned int}’
+
+2016-08-12 21:12:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * .gitignore:
+ * config.h.meson:
+ * ext/cairo/meson.build:
+ * ext/dv/meson.build:
+ * ext/flac/meson.build:
+ * ext/gdk_pixbuf/meson.build:
+ * ext/jack/meson.build:
+ * ext/jpeg/meson.build:
+ * ext/libpng/meson.build:
+ * ext/meson.build:
+ * ext/pulse/meson.build:
+ * ext/shout2/meson.build:
+ * ext/soup/meson.build:
+ * ext/speex/meson.build:
+ * ext/taglib/meson.build:
+ * ext/vpx/meson.build:
+ * ext/wavpack/meson.build:
+ * gst/alpha/meson.build:
+ * gst/apetag/meson.build:
+ * gst/audiofx/meson.build:
+ * gst/audioparsers/meson.build:
+ * gst/auparse/meson.build:
+ * gst/autodetect/meson.build:
+ * gst/avi/meson.build:
+ * gst/cutter/meson.build:
+ * gst/debugutils/meson.build:
+ * gst/deinterlace/meson.build:
+ * gst/dtmf/meson.build:
+ * gst/effectv/meson.build:
+ * gst/equalizer/meson.build:
+ * gst/flv/meson.build:
+ * gst/flx/meson.build:
+ * gst/goom/meson.build:
+ * gst/goom2k1/meson.build:
+ * gst/icydemux/meson.build:
+ * gst/id3demux/meson.build:
+ * gst/imagefreeze/meson.build:
+ * gst/interleave/meson.build:
+ * gst/isomp4/meson.build:
+ * gst/law/meson.build:
+ * gst/level/meson.build:
+ * gst/matroska/meson.build:
+ * gst/meson.build:
+ * gst/monoscope/meson.build:
+ * gst/multifile/meson.build:
+ * gst/multipart/meson.build:
+ * gst/replaygain/meson.build:
+ * gst/rtp/meson.build:
+ * gst/rtpmanager/meson.build:
+ * gst/rtsp/meson.build:
+ * gst/shapewipe/meson.build:
+ * gst/smpte/meson.build:
+ * gst/spectrum/meson.build:
+ * gst/udp/meson.build:
+ * gst/videobox/meson.build:
+ * gst/videocrop/meson.build:
+ * gst/videofilter/meson.build:
+ * gst/videomixer/meson.build:
+ * gst/wavenc/meson.build:
+ * gst/wavparse/meson.build:
+ * gst/y4m/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ * sys/directsound/meson.build:
+ * sys/meson.build:
+ * sys/v4l2/meson.build:
+ * sys/ximage/meson.build:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ Add support for Meson as alternative/parallel build system
+ https://github.com/mesonbuild/meson
+ With contributions from:
+ Tim-Philipp Müller <tim@centricular.com>
+ Jussi Pakkanen <jpakkane@gmail.com> (original port)
+ Highlights of the features provided are:
+ * Faster builds on Linux (~40-50% faster)
+ * The ability to build with MSVC on Windows
+ * Generate Visual Studio project files
+ * Generate XCode project files
+ * Much faster builds on Windows (on-par with Linux)
+ * Seriously fast configure and building on embedded
+ ... and many more. For more details see:
+ http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
+ http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
+ Building with Meson should work on both Linux and Windows, but may
+ need a few more tweaks on other operating systems.
+
+2016-08-20 16:59:30 +0800 Jie Jiang <jiangjie@nudt.edu.cn>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ Fixed splitmuxsink 32-bit overflow bug
+ Extend the byte tracking counters to 64-bit on
+ all platforms, instead of using gsize, which overflows
+ after 4GB.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770019
+
+2016-08-19 17:18:16 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4: Fix coverity warning
+ If atom_copy_data fails to write anything, return 0
+ CID #1371458
+
+2016-04-09 07:51:03 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ * sys/v4l2/v4l2-utils.c:
+ v4l2: consistently check #ifdef HAVE_GUDEV instead of #if
+ Both work with autotools but they definitely don't mean the same thing, cause
+ problems with other build systems, and are bad form. Existence should always be
+ checked with #ifdef or #if defined.
+
+2016-04-19 10:53:05 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/directsound/gstdirectsoundsink.h:
+ directsound: port away from old DirectX API
+ D3DX has been deprecated for the last 4 years and latest versions of
+ Windows no longer ship headers for it. This is fine as long as you're
+ building with Cerbero's Wine-based DirectX headers, but sucks if you
+ want to build against the actual Windows SDK.
+ We were just using it to get error strings anyway, so just use the
+ generic error string API.
+
+2016-08-18 12:02:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ Revert "flacparse: Add maximum bitrate tag"
+ This reverts commit c703ab69f526092bb26cce41ca691a896c8383d8.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769392
+
+2016-08-18 09:57:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations
+ Need to set max-misorder-time and max-dropout-time to 0 so the
+ jitterbuffer does not base them on packet rate calculations.
+ If it does, out gap is big enough to be considered a new stream and
+ we wait for a few consecutive packets just to be sure
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2016-08-09 12:55:59 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Add option to split at exactly max-size-time
+ Will try to request a keyframe from the encoder to be sent at the target
+ running time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769664
+
+2016-08-09 20:16:16 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Allow time and bytes to reach their respective thresholds
+ https://bugzilla.gnome.org/show_bug.cgi?id=769664
+
+2016-08-17 09:49:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Allow mimetypes with properties as long as they're application/sdp
+ Some servers add properties like charset, e.g.
+ application/sdp; charset=utf8
+ Ideally we should also parse the charset and do conversion of all messages,
+ but that's for a later time.
+
+2016-06-24 16:32:37 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Added support for writing timecode track
+ https://bugzilla.gnome.org/show_bug.cgi?id=767950
+
+2016-08-11 16:32:21 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: Initialize bytes_sent field.
+ This fixes endpoints not receiving any data intermittently.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769773
+
+2016-08-10 11:45:13 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpstats.c:
+ rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2016-08-10 11:26:17 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Don't warn for duplicate packets
+ This is a normal scenario and should not be a warning. This can
+ happen frequently when re-transmits of lost packets are enabled.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762208
+
+2016-08-08 13:49:19 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: Fix typo converting to running time.
+ Use the correct collected timestamp.
+
+2016-08-08 02:53:48 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ Revert "splitmuxsink: Use GstBin async-handling instead of our own."
+ This reverts commit fa008f271a52f82dededc28bd81b020ca7939b47.
+ async-handling in GstBin causes the pipeline to spin at 100%
+ CPU as the top-level pipeline tries to change that state
+ to PLAYING constantly. This is a workaround for a core
+ problem, essentially, but an improvement in this case for now.
+
+2016-08-08 00:56:38 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: Recheck state after unlocking mutex.
+ After dropping the splitmux lock, re-check the state,
+ don't just fall through and sleep unconditionally,
+ as we may have already missed the wakeup.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769514
+
+2016-08-03 03:32:07 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Don't stop and error on EOS flow return
+ Don't immediately halt on EOS flow return from downstream
+ due to out of segment. Let the demuxer handle it and send
+ EOS.
+
+2016-08-04 00:36:28 -0300 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: avoid unref of null buffer
+ The current 'l' pointer will be NULL when the loop
+ is interrupted with a 'break' statement. Need to have
+ it advance to the next list item before interrupting.
+
+2016-08-02 14:01:14 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/wavparse/Makefile.am:
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Add tags for container format and bitrate for uncompressed PCM
+ The PCM bitrate is added to help downstream elements (like uridecodebin)
+ figure out a proper network buffer size
+ https://bugzilla.gnome.org/show_bug.cgi?id=769390
+
+2016-08-01 18:52:26 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Add maximum bitrate tag
+ https://bugzilla.gnome.org/show_bug.cgi?id=769392
+
+2016-07-28 17:58:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: When receiving a DISCONT buffer that does not point to a sample, remember the offset
+ And don't just reset everything. This makes sure that we can continue to
+ handle data in the following scenario:
+ moov: discont
+ moof: discont
+ mdat: continuous
+ Previously this would fail because the offset would be the accumulated offset
+ from moov and moof at the mdat position, while the buffer offset might be
+ something completely different.
+
+2016-07-25 13:34:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpbvpay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtpilbcpay.c:
+ rtp: Filter with the filter caps in the payloader's getcaps
+
+2016-03-03 11:35:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: include http-status-code in error message details
+ https://bugzilla.gnome.org/show_bug.cgi?id=763038
+
+2016-07-25 18:20:03 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix debug statement signedness.
+ The ts variable is a GstClockTime, don't print it
+ as a GstClockTimeDiff.
+
+2016-07-17 22:41:02 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Handle negative running time
+ Use signed clock times for running time everywhere
+ so that we handle negative running times without
+ going haywire, similar to what queue and multiqueue
+ do these days.
+
+2016-07-18 00:12:55 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Drop lock when sending dummy event
+ When pushing the dummy event into the multiqueue,
+ drop the splitmux lock or else we might deadlock.
+
+2016-06-30 01:56:41 +1000 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Intersect with filter caps in getcaps function.
+ Always intersect with the filter caps in the getcaps function
+ to make sure we return a subset of what was requested.
+ Other payloaders also have this problem and need fixing
+ in future commits.
+
+2016-07-12 17:30:56 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/qtdemux.c:
+ tests: qtdemux: fix element and pad leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=768739
+
+2016-07-12 16:45:36 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/audiofirfilter.c:
+ * tests/check/elements/audioiirfilter.c:
+ * tests/check/elements/rtp-payloading.c:
+ * tests/check/elements/videobox.c:
+ * tests/check/pipelines/effectv.c:
+ tests: fix bus leaks
+ gst_bus_add_signal_watch() takes a ref on the bus which should be
+ released using gst_bus_remove_signal_watch().
+ https://bugzilla.gnome.org/show_bug.cgi?id=768739
+
+2016-07-14 03:07:11 +0800 Ting-Wei Lan <lantw@src.gnome.org>
+
+ * configure.ac:
+ configure: Call AG_GST_PKG_CONFIG_PATH to set GST_PKG_CONFIG_PATH
+ GST_PKG_CONFIG_PATH is used in docs/plugins directory, so
+ AG_GST_PKG_CONFIG_PATH must be called to set it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768787
+
+2016-07-12 07:39:58 +0200 Edward Hervey <edward@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Don't drop final bytes of a range request
+ At the end of a range request, we don't want to return GST_FLOW_EOS otherwise
+ the last bytes we just read will be dropped by basesrc.
+ Instead just return GST_FLOW_OK (which was set just before) and let basesrc
+ handle the fact we are at the end of the segment.
+
+2016-07-11 18:30:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2provider: Fix device type detection
+ The type detection would lead to assertion as it would try
+ to create a device without having found any type for it. It
+ also didn't detect MPLANE devices properly.
+
+2016-07-11 18:29:01 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't assert when used by the monitor
+ The monitor sets the object->element object as a GstObject. This
+ works for debug traces, but will assert for ELEMENT_ERROR. This
+ was the only case where that could happen. Add a check for that.
+
+2016-07-11 17:38:00 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Indent very long line
+
+2016-07-12 00:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: At the end of a range request, read another time to finalize the request
+ If we're at the end of a range request, read again to let libsoup
+ finalize the request. This allows to reuse the connection again later,
+ otherwise we would have to cancel the message and close the connection.
+
+2016-07-11 21:13:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f363b32 to f49c55e
+
+2016-07-11 19:57:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Fix keep-alive handling
+ We have to get rid of the message on EOS when the complete stream is read to
+ remember that we successfully finished handling this specific message.
+ Otherwise we will cancel it later and close the connection instead of reusing
+ it at a later time.
+ It might also make sense to reuse connections if a non-200 response is
+ received. As long as there was no connection error, the HTTP connection should
+ be re-usable.
+
+2016-07-11 12:05:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ Also enable V4L2 probe on aarch64 (aka ARM 64bit)
+
+2016-07-11 11:59:19 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/examples/rtp/client-PCMA.c:
+ rtp example: Fix leak
+ Also stop fetching the internal source as this
+ functionality has been broken.
+
+2016-07-08 14:58:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ Enable v4l2 probe on Linux/ARM
+ Most of those have V4L2 drivers these days enabling it make sure that it
+ this code is enabled in major distribution, hence that HW accelerated
+ decoder/encoder can be used on platforms that support it. The probes are
+ slightly increasing the first init of gstreamer library, though the
+ result is cached in the registry for later use.
+
+2016-07-11 09:46:49 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ * tests/check/elements/rtp-payloading.c:
+ rtph265pay: Accept array_completeness=1
+ When parsing NAL unit type in codec_data, check the 6bits of
+ NAL_unit_type only and do not require the array_completeness bit to be
+ 0, since the default and mandatory value of array_completeness is 1 for
+ hvc1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768653
+
+2016-07-10 21:35:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Also copy device_caps in gst_v4l2_dup
+ This fixes regression where M2M error out saying they have no output
+ format (the V4L2 CAPTURE side).
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-10 21:30:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Use correct in6_pktinfo struct instead of in_pktinfo
+ Fixes the build on FreeBSD, which does not have the latter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768623
+
+2016-07-08 17:28:19 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: fix multiplanar capture
+ After switching to using V4L2_CAP_DEVICE_CAPS we lost support for
+ multiplanar device types. After some research, it looks like
+ vcap.capabilities treated the multiplanar flag of output and capture
+ devices equally, but not the new device_caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-08 14:56:30 +0200 Mats Lindestam <matslm@axis.com>
+
+ * gst/multipart/multipartmux.c:
+ * gst/multipart/multipartmux.h:
+ multipartmux: Use PTS and DTS instead of timestamp
+ And pass-through both of them.
+ Based on a patch by Göran Jönsson <goranjn@axis.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=767900
+
+2016-06-30 14:40:40 +0200 Thomas Scheuermann <Thomas.Scheuermann@barco.com>
+
+ * ext/jack/gstjackaudioclient.c:
+ jack: don't wait for callbacks if the jack server shut down
+ Otherwise we'll wait forever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747275
+
+2016-06-23 15:30:19 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Let upstream events go through upstream
+ There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+ Some elements might want to have that information.
+
+2016-06-23 15:22:56 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Let upstream events go through upstream
+ There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+ Some elements might want to have that information.
+
+2016-06-23 15:17:36 +0200 Edward Hervey <edward@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Let upstream events go through upstream
+ There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+ Some elements might want to have that information.
+ Also remove downstream-only CAPS event handling and minimize code
+
+2016-07-07 23:53:54 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2: fix v4l2 probe build error
+ A typo in gst_v4l2_probe_and_register() caused a build error when building
+ with --enable-v4l2-probe. Fixing it.
+ gstv4l2.c: In function 'gst_v4l2_probe_and_register':
+ gstv4l2.c:150:25: error: 'struct v4l2_capability' has no member named 'capabilitites'
+ device_caps = vcap.capabilitites;
+
+2016-07-01 22:53:33 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: use gst_caps_intersect_full in negotiate()
+ Instead of reimplementing the GST_CAPS_INTERSECT_FIRST
+ interection mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-02 01:56:07 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2radio.c:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: use opened device caps instead of physical device ones
+ The same physical device can export multiple devices. In
+ this case, the capabilities field now contains a union of
+ all caps available from all exported V4L2 devices alongside
+ a V4L2_CAP_DEVICE_CAPS flag that should be used to decide
+ what capabilities to consider. In our case, we need the
+ ones from the exported device we are using.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-07 18:24:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Remove suspicious checks for pads being active and linked
+ We should add all pads, no matter if they are linked or active or not at this
+ point. Skipping some that are not will cause different behaviour than with
+ other muxers.
+
+2016-07-07 18:23:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Error out if we start writing data with some pads not having a codec id yet
+ This can only happen if a) upstream somehow gets around the CAPS event failing
+ or b) there never being any CAPS event.
+ The following code assumes that all pads have a codec-id.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768509
+
+2016-07-07 18:14:43 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Consistently use gst_matroska_mux_set_codec_id() for setting the codec id
+
+2016-07-04 09:50:11 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtph265pay/depay: Sync against RFC 7798
+ Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
+ sprop-parameter-sets.
+ rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
+ handles profile-id, tier-flag and level-id in caps query.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-07-06 09:25:00 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flvdemux: Push nominal bitrate tags
+ Add per-stream tag lists, which are used to send nominal
+ bitrate tags. When remuxing FLV => FLV, this now passes
+ through the upstream bitrate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768440
+
+2016-07-06 09:24:49 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flvdemux: Refactor metadata tag handling
+ The FLV header cannot be trusted to indicate video or
+ audio presence, as the comments already mention. Don't
+ delay pushing tags waiting for streams that might never
+ appear.
+ Tags are now pushed immediately after they change:
+ - After parsing an onMetaData script object
+ - After negotiating caps on a pad
+ https://bugzilla.gnome.org/show_bug.cgi?id=768440
+
+2016-07-06 12:44:10 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix AAC codec_data values
+ As seen in the parent switch for object_type_id, the 4 possible values are
+ 0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values.
+ Looks like it was a typo making them decimal instead of hexadecimal.
+ CID 1363328
+
+2016-07-06 13:51:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:06:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.9.1
+
+2016-07-06 11:46:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-07-06 11:22:53 +0300 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix error messages to first convert to doubles before division
+
+2016-07-06 10:18:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/da.po:
+ * po/hr.po:
+ * po/pt_BR.po:
+ * po/sk.po:
+ po: Update translations
+
+2016-07-05 21:11:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
+ There's a small window for a race condition otherwise.
+
+2016-07-04 17:45:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/qtmux.c:
+ qtmux: Use complete AAC caps with codec_data in the tests
+
+2016-07-04 16:58:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Reject raw AAC if no codec_data is found in the caps
+ If necessary, a demuxer will have to invent something here but this is only a
+ problem with non-conformant files anyway.
+
+2016-07-04 16:55:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Invent AAC codec_data if none is present
+ Without, raw AAC can't be handled and we have some information available in
+ the decoder that most likely allows us to decode the stream in one way or
+ another. This is the same code already used by matroskademux for the same
+ reasons, and ffmpeg/vlc play such files just fine too by guesswork.
+
+2016-07-04 14:54:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Reject raw AAC caps without codec_data
+ The resulting file is not going to be playable without guesswork and raw caps
+ should always have codec_data.
+
+2016-05-10 15:48:49 +0200 Edward Hervey <edward@centricular.com>
+
+ qtdemux: Handle upstream GAP in push-mode/time segment
+ This is to handle cases where upstream handles the fragmented streaming in TIME
+ segments and sends us data with gaps within fragments. This would happen when dealing
+ with trick-modes.
+ When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
+ it must obey the following rules:
+ * The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
+ * The buffers containing the first sample after a gap:
+ * MUST start at the beginning of a sample,
+ * MUST have the DISCONT flag set,
+ * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767354
+
+2016-07-01 11:54:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/v4l2-utils.c:
+ v4l2: fix potential double-free of error debug string
+ gst_v4l2_clear_error() doesn't work like g_clear_error(), it
+ doesn't NULLify the pointer, so set freed debug string to NULL
+ so it doesn't get freed again if gst_v4l2_clear_error() is
+ called twice on the error.
+ CID 1362901
+
+2016-07-01 10:05:00 +0000 Brad Lackey <blackey@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't disable UDP protocols on redirecting
+ https://bugzilla.gnome.org/show_bug.cgi?id=768232
+
+2016-07-01 17:28:17 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Push caps only when it was updated
+ Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps
+ event per moof without consideration of duplication.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768268
+
+2016-06-30 15:01:46 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: fix invalid memory access
+ 10 bytes was allocated for stream_format but size of "byte-stream" is
+ more. Use g_strdup() instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-29 23:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: Use a non-timer GstPoll
+ Otherwise set_flushing() will have undefined semantics and nowadays causes a
+ g_critical() to warn about that.
+
+2016-06-19 02:08:25 -0300 Thiago Santos <thiagossantos@gmail.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: dynamically adjust blocksize
+ Update the blocksize depending on how much is obtained from a read
+ of the input stream. This avoids doing too many reads in small chunks
+ when larger amounts of data are available and also prevents using
+ a very large memory area to read a small chunk of data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767833
+
+2016-06-28 16:44:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo
+
+2016-06-28 15:15:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS
+
+2016-06-28 15:08:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Move #includes around to a) work around broken glibc header and b) Windows
+
+2016-06-28 14:25:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix compilation on Windows and *BSD/OSX
+
+2016-06-23 20:21:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Filter out multicast packets that are not for our multicast address
+ https://bugzilla.gnome.org/show_bug.cgi?id=767980
+
+2016-06-28 10:57:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
+ If we consider the RTSP state, what can happen is that it is PLAYING but the
+ element already asynchronously tried to PAUSE and it just did not happen yet.
+ We would then override this setting to PAUSED (while the element actually is
+ in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
+ to produce packets while the sinks are all PAUSED, piling up thousands of
+ packets in the rtpjitterbuffer and other elements and finally failing.
+
+2016-06-27 09:20:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Add comment about H263/MPEG4P2 being non-standard for FLV
+ They are however supported by ffmpeg and apparently used out there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768006
+
+2016-06-24 14:48:53 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Add support for H263 and MPEG4 part2
+ https://bugzilla.gnome.org/show_bug.cgi?id=768006
+
+2016-06-21 17:10:56 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ Update plugins doc
+ This is partly automated using "make update" in docs/plugins, but also
+ required manual merge. Additionally, missing plugins and elements have
+ been added.
+
+2016-06-21 17:51:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/splitmux.c:
+ tests: splitmux: skip tests if theora or ogg plugins are not available
+ https://bugzilla.gnome.org/show_bug.cgi?id=767861
+
+2016-06-21 11:46:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From ac2f647 to f363b32
+
+2016-06-21 07:40:42 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ gstrtpj2kpay: use tile bit and tile number to determine if there are multiple tiles in packet
+ Now we don't have to rely on a special value for the tile number.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767817
+
+2016-06-21 09:34:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: fix compiler warning on OS/X
+ gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535
+ https://bugzilla.gnome.org/show_bug.cgi?id=767817
+
+2016-06-21 09:34:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: update
+
+2016-05-16 17:31:58 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/capssetter.c:
+ * tests/check/elements/icydemux.c:
+ * tests/check/elements/jpegenc.c:
+ * tests/check/elements/level.c:
+ * tests/check/elements/multifile.c:
+ * tests/check/elements/qtmux.c:
+ * tests/check/elements/rtprtx.c:
+ * tests/check/elements/udpsrc.c:
+ fix buffer leaks in tests
+ Need to call gst_check_drop_buffers() to release the buffers exchanged
+ during the test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-17 12:52:43 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/interleave.c:
+ interleave: fix message leaks in test
+ Flush the bus when cleaning up so pending messages are destroyed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-17 12:58:06 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/videomixer.c:
+ videomixer: fix event leaks in test
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-13 15:12:22 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/deinterleave.c:
+ deinterleave: fix leaks
+ - Flush the bus so messages aren't leaked
+ - Fix pad leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-06-17 15:29:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Deprecated sprop-parameter-set property
+ This is supposed to be either in the codec_data (avc stream format) or inside
+ the stream, and we extract it from there. It should not be set from a
+ property as it's stream specific.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767789
+
+2016-06-17 12:16:32 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: make all srtp encoder properties explicit
+ The Session Data Protocol doesn't allow specifying a cipher for the
+ SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
+ "aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
+ an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
+ cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767799
+
+2016-06-17 19:59:13 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/soup/gstsoup.c:
+ soup: work around frequent deadlocks in GLib type initialisation
+ .. by registering the types from the plugin init function. This
+ seems to help, but we'll see if it's enough (might need similar
+ things elsewhere).
+ https://bugzilla.gnome.org/show_bug.cgi?id=693911
+ https://bugzilla.gnome.org/show_bug.cgi?id=674885
+
+2016-06-17 16:08:08 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: The prores variant is stored in the variant field, not format
+ And the caps in the sink pad template already used variant (only).
+
+2016-06-17 13:00:48 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtph265pay: Remove sprop-parameter-sets property
+ There is no valid use case when this property is needed since the values
+ must be in either codec_data or buffer data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-10 16:17:26 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * docs/plugins/scanobj-build.stamp:
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Read NALU type the same way everywhere
+ Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the
+ same way as in other places.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-17 13:58:33 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ rtpjitterbuffer: fix RTPJitterBufferMode documentation
+ Documentation lacks '@' before each enum values and there was an extra
+ line after symbol section which confuses GTK-Doc parser.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767788
+
+2016-05-23 10:18:48 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: take the lock when changing stats
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-06-15 11:19:43 +0200 Jürgen Slowack <jurgen.slowack@barco.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection
+ Fixes sps/pps/vps insertion via the config-interval property.
+ https://bugzilla.gnome.org//show_bug.cgi?id=767680
+
+2016-06-11 12:16:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/pipelines/simple-launch-lines.c:
+ simple-launch-lines: Use correct JPEG2000 caps
+
+2016-06-10 13:43:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix indentation
+
+2016-06-10 13:42:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix date parsing when there are trailing spaces
+ Fixes parsing of "Thu May 11 15:57:46 2006 ".
+ https://bugzilla.gnome.org/show_bug.cgi?id=767496
+
+2016-05-13 15:08:24 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kcommon.h:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ gstrtpj2k: set sampling field required by RFC
+ This field is now required in the sink caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766236
+
+2016-06-09 09:30:48 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Fix unref assertion failure
+ Fix unref assertion failure
+ https://bugzilla.gnome.org/show_bug.cgi?id=767424
+
+2016-05-14 14:46:17 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Work with non-TIME segments
+ With non-time segments, it now assumes that the arrival time of packets
+ is not relevant and that only the RTP timestamp matter and it produces
+ an output segment start at running time 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766438
+
+2016-06-07 20:53:34 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: Wait for segment event before checking it
+ The heuristic to choose between packetise or not was changed to use the
+ segment format. The problem is that this change is reading the segment
+ during the caps event handling. The segment event will only be sent
+ after. That prevented the decoder to go in packetize mode, and avoid
+ useless parsing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2016-06-06 17:00:22 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Wait for segment event before checking it
+ The heuristic to choose between packetise or not was change to use the
+ segment format. The problem is that this change is reading the segment
+ during the caps event handling. The segment event will only be sent
+ after. That prevented the decoder to go in packetize mode, and avoid
+ useless parsing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2016-06-07 16:42:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Keep part of the input buffer
+ Instead of completely getting rid of the input buffer, copy
+ the metadata, the flags and the timestamp into an empty buffer.
+ This way the decoder base class can copy that information again
+ to the output buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758424
+
+2016-06-07 16:41:58 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Coding style fixes
+
+2016-06-07 16:09:23 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Coding style fixes
+
+2016-06-07 16:04:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Add an error return to _try/_set_format
+ This way one can easily ignore errors. Previously, error were always
+ posted ont he bus.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766172
+
+2016-06-07 16:01:55 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/v4l2-utils.c:
+ * sys/v4l2/v4l2-utils.h:
+ v4l2-util: Introduce GstV4l2Error
+ This is to allow returning an error that can easily be sent as
+ message to the application if the element needs it. Using this
+ also allow ignoring errors.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766172
+
+2016-06-07 12:41:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Avoid decide allocation on active pool
+ v4l2src will renegotiate only if the format have changed. As of now,
+ it's not possible to change the allocationw without resetting the
+ camera. To avoid unwanted side effect, simply keep the old allocation
+ if no renegotiation is taking place. This fixes assertion and possible
+ failures in USERPTR or DMABUF import mode (when using downstream pools).
+ https://bugzilla.gnome.org/show_bug.cgi?id=754042
+
+2016-04-28 13:44:49 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Show state name in debugging
+ Makes it easier to trace what's going on
+
+2016-05-10 15:45:42 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove useless variable
+ That variable is only needed for a debug statement, move it there
+
+2016-05-10 15:10:36 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Add/Fix comments on the various structure variables
+ No variables were added/removed. This was just a good excuse to:
+ * Comment what most variables are used for (and when)
+ * Order them in such a way as to show first the common variables used
+ in all cases, followed by those only used in push-mode
+
+2016-05-10 15:07:40 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove unused structure
+ Let's just remove it, been commented for 7+ years :)
+
+2015-09-02 11:48:29 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: use decoder stop command instead of queueing empty buffers
+ Only if the decoder stop command fails, keep queueing empty buffers to
+ signal end of stream as before.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733864
+
+2014-12-12 14:31:36 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: add gst_v4l2_decoder_cmd helper
+ https://bugzilla.gnome.org/show_bug.cgi?id=733864
+
+2016-06-01 20:28:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Forward segments directly if we are operating in PUSH mode on fragmented streams
+ We shouldn't go through segment activation as we will only have a limited
+ understanding of how the whole stream timeline looks like from the moof. We
+ only know about the current fragment, while upstream knows about the whole
+ stream.
+ This fixes seeking in DASH streams, both for seeks after the current moof and
+ for seeks into the current moof. The former would fail because the moof ends
+ and we can't activate any segment, the latter would cause a segment that stops
+ at the moof end, and no further fragments would be played because we end up
+ being EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-06-06 17:54:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: Use looser caps for upstream
+ When we fixate for upstream, try to not introduce new fields when not
+ needed. This was imported from videoconvert element.
+
+2015-01-28 12:07:58 +0100 Enrico Jorns <ejo@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ gstv4l2transform: format fixation for preferring passthrough
+ * If outgoing format is unfixated, try to set it to input format.
+ * Call gst_caps_fixate () at end of fixation routine
+ https://bugzilla.gnome.org/show_bug.cgi?id=766719
+
+2016-05-20 12:49:53 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: allow to change pixel aspect ratio
+ Scalers may change width and height independently,
+ allow to change pixel aspect ratio.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766712
+
+2016-05-20 12:32:25 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix scaling in case of fixed pixel aspect ratio
+ To change pixel aspect ratio from DAR to PAR, the necessary scaling factor
+ is DAR/PAR, not DAR*PAR.
+ For good measure, add debug output similar to the fixed-width and
+ fixed-height cases.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766711
+
+2016-05-13 16:39:25 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fill colorimetry in gst_v4l2_object_acquire_format
+ Instead of relying on the default colorimetry chosen by
+ gst_video_info_set_format(), set info.colorimetry from the
+ values returned by G_FMT. This allows decoders to propagate
+ their input colorimetry downstream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766383
+
+2016-05-18 10:17:12 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: refactor gst_v4l2_object_get_colorspace to take a v4l2_format parameter
+ Move the extraction of colorimetry parameters from struct v4l2_format and the
+ setting of the identity matrix for RGB formats into the function to avoid code
+ duplication.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766383
+
+2016-05-13 14:58:41 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: use visible size, not coded size, for downstream negotiation filter
+ gst_v4l2_probe_caps() returns the coded size, not the visible size. Subtract
+ the known padding from probed caps with the coded size before using them as
+ filter for caps negotiation with downstream elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766382
+
+2016-05-13 14:45:02 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: use G_SELECTION instead of G_CROP in gst_v4l2_object_acquire_format
+ The gst_v4l2_object_acquire_format() function is used by v4l2videodec to obtain
+ the currently set capture format. Since G_FMT returns the coded size, the
+ visible size needs to be obtained from the compose rectangle in order to
+ negotiate it with downstream elements. The G_CROP call hasn't worked on mem2mem
+ capture queues for a long time. Instead use the G_SELECTION call to obtain the
+ compose rectangle and only fall back to G_CROP for ancient kernels.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766381
+
+2016-01-27 09:57:38 +0100 Andreas Naumann <anaumann@ultratronik.de>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Use V4L2_BUF_TYPE_VIDEO_OUTPUT_OVERLAY if driver advertises it.
+ On modern kernels, the G/S_FMT ioctls will always fail using
+ V4L2_BUF_TYPE_VIDEO_OVERLAY with VFL_DIR_TX (e.g. real overlay out drivers)
+ since this is not the intented use (rather rx, according to v4l2 API doc).
+ Probably this is why the Video Output Overlay interface was created, so if
+ the driver advertises it we might as well use.
+ For old kernels (pre 2012) the old way might still work so keeping this for
+ compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761165
+
+2016-06-06 18:52:01 +0100 Kieran Bingham <kieran@bingham.xyz>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Use non-deprecated V4L2 type for RGB15
+ Support for the updated V4L2_PIX_FMT_XRGB555 was added in commit
+ 2538fee2fd8fdb74b05f0a511281bc4707e7cc44 however, when setting the format
+ for use in v4l2 ioctls, the old deprecated format is still used. Convert
+ this to the new accepted format type, as the preferred format.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767300
+
+2016-05-04 14:50:32 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: preserve seek flags
+ Without this some flags get lost in streaming mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767194
+
+2016-06-06 10:47:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/Makefile.am:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ Revert "WIP revert soup"
+ This reverts commit fdac3a7a231f3848665636cf8122f96103b46e3b.
+ Was not supposed to be pushed but a local workaround for
+ https://bugzilla.gnome.org/show_bug.cgi?id=693911#c13
+
+2016-06-03 13:09:35 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: complete warn log with SSRC
+ https://bugzilla.gnome.org/show_bug.cgi?id=767195
+
+2016-05-31 15:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/Makefile.am:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ WIP revert soup
+
+2016-06-03 13:18:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Unref seek event in any case
+ It would be leaked if no seek handler was currently set.
+
+2016-06-03 10:49:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: Properly set event/message sequence numbers based on the previous seek
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-03 10:36:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: Remember if upstream had a time segment and if not properly create time segments
+ Previously the segment.time was wrong, and the position was not updated
+ correctly, resulting in seeks in PUSH mode with upstream providing a BYTES
+ segment to not work at all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-03 09:54:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Implement SEEKING query so we can actually seek if upstream can't seek in TIME
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 14:19:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Recalculate the frame offsets at the beginning of each BYTE segment and whenever upstream gives us a timestamp
+ This fixes seeking in DV streams where upstream operates in PUSH mode with a
+ TIME segment (e.g. avidemux). Without this, we would generate wrong durations
+ and timestamps after a seek.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 13:53:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: Pass-through buffer DISCONT flags
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 16:16:45 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ rtpvp9depay: Don't assert on flexible mode packets
+ Instead just post a warning on the bus for now.
+
+2016-06-02 15:03:17 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ tests: rtpbin: fix caps leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=767156
+
+2016-06-02 15:00:01 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/amrparse.c:
+ tests: amrparse: clean up test
+ - use GST_CHECK_MAIN() to reduce boilerplate
+ - unref the input caps using a teardown function to prevent leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=767156
+
+2016-05-20 15:22:35 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Ensure DISCONT flag is properly propagated
+ The output of deinterlace at startup, or when receiving a new DISCONT
+ buffer, should have the DISCONT flag set on the first buffer.
+
+2016-05-31 21:34:04 +0200 Josep Torra <adn770@gmail.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2src: check for valid size on raw video buffers
+ Discard buffers that doesn't contain enough data when dealing
+ with raw video inputs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767086
+
+2016-05-31 17:10:36 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use the demuxer segment instead of a new one for MSS streams
+ Upstream might have told us something about the to be expected segment, so
+ let's use that information instead of coming up with a [0,-1] segment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 17:04:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Only activate segments and send SEGMENT events if we have streams
+ But in that case also remove the pending newsegment event, otherwise we would
+ later send a possibly outdated event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 16:53:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: In PULL mode, nothing is ever going to send us a SEGMENT event
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 16:38:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't override TIME segments from upstream that we just saw
+ The point of d8fb7a9c96b108814beeaa0e63f818d4648c7fe9 was to not have any
+ spurious segments stored for later if we do BYTES->TIME conversion, but
+ overriding any TIME segments from upstream does not make any sense.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=763165
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2015-07-16 09:48:46 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/multifile/gstmultifilesrc.c:
+ multifilesrc: set position as offset from start-index
+ query position in GST_FORMAT_BUFFER returns
+ offset from start-index rather than index.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752462
+
+2016-05-27 12:49:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/files/Makefile.am:
+ * tests/files/gradient.j2k:
+ tests: add unit test for JPEG-2000 rtp payloader leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=766870
+
+2016-05-25 17:11:13 +0200 Pierre Lamot <pierre.lamot@openwide.fr>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: Fix buffer memory leak
+ Input buffer memory was not unmapped
+ https://bugzilla.gnome.org/show_bug.cgi?id=766870
+
+2016-05-18 12:12:15 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fix caps leak
+ gst_v4l2_object_probe_caps() was taking an extra ref on the returned
+ caps for no reason.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766610
+
+2016-05-22 20:14:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ videocrop mark crop properties as mutable in playing state
+
+2016-05-20 16:47:35 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: fix buffer leak when flushing
+ When early returning in gst_soup_http_src_read_buffer() because the
+ element is FLUSHING, we need to unmap and unref the buffer which was just created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766718
+
+2016-05-20 11:15:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Set seek event seqnum on all SEGMENT events
+ Some were forgotten.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-20 11:12:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavidemux.h:
+ avidemux: Pass through seek event seqnums in all SEGMENT/EOS events and SEGMENT_DONE messages/events
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-20 10:56:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Set seek event seqnum in EOS and SEGMENT_DONE messages/events
+ Also actually store the seqnum in pull mode seeks.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-17 13:40:38 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: fix caps leak
+ The caps returned by gst_pad_get_current_caps() was never unreffed when
+ not early returning.
+ Fix a leak with the elements/deinterlace test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766558
+
+2016-01-25 16:25:51 +0100 Mikhail Fludkov <misha@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: don't act on suspicious BYE RTCP
+ Some endpoints (like Tandberg E20) can send BYE packet containing our
+ internal SSRC. I this case we would detect SSRC collision and get rid
+ of the source at some point. But because we are still sending packets
+ with that SSRC the source will be recreated immediately.
+ This brand new internal source will not have some variables incorrectly
+ set in its state. For example 'seqnum-base` and `clock-rate` values will be
+ -1.
+ The fix is not to act on BYE RTCP if it contains internal or unknown
+ SSRC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762219
+
+2015-11-15 14:54:28 +0100 Mikhail Fludkov <misha@pexip.com>
+
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Add test for locking of the stats signal
+ Keeping the lock while emitting the stats signal introduces potential
+ deadlock in those situations when the signal callback wants the access
+ to rtpsession's properties which also requre the lock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762216
+
+2016-05-19 15:36:57 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: don't hold object lock whilst pushing out headers
+ matroskademux would take the GST_OBJECT_LOCK in
+ - gst_matroska_demux_push_codec_data_all()
+ - gst_matroska_demux_query()
+ Some parse element such as FLAC checks upstream seekability, and
+ there is some use cases that matroska-demux is linked to a parse element
+ (e.g.,FLAC format) without intermediate elements (e.g., queue).
+ In this case, matroska-demux never returns from _push_codec_data_all()
+ because the parser can return only after it receives the response to
+ the upstream query, but that's not going to happen because it's
+ deadlocked.
+ Elements must not hold the object lock whilst pushing out events
+ or data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766645
+
+2016-05-19 12:43:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Set sent_buffers and streamheader_buffers to NULL after freeing
+ Otherwise we might use an already freed list later and crash or worse.
+
+2016-05-18 18:32:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: fix Since version for new "loop" property
+
+2016-05-16 16:18:37 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/rtsp/gstrtpdec.c:
+ rtpdec: fix clock leak
+ gst_system_clock_obtain() returns a new ref.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766521
+
+2016-05-17 05:33:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: add doc blurb with since marker for new "loop" property
+
+2015-11-13 15:52:35 +0100 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: add support for png
+ https://bugzilla.gnome.org/show_bug.cgi?id=758059
+
+2016-05-15 22:07:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: Connect to demux signals before activating
+ Fix a race in splitmuxsrc by properly connecting to the
+ demuxer signals we're interested in *before* setting it running.
+
+2016-05-15 13:31:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ docs: Update for git master
+
+2016-05-15 12:16:23 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4gpay.h:
+ rtpmp4gpay: Don't produce timestamps based on byte count
+ The GST_BUFFER_OFFSET of output buffers returned to GstRtpBasePayload
+ should reflect the number of "samples" in the unit of the RTP clock in this
+ buffer. If this is not true, then it shouldn't be set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761943
+
+2016-05-15 12:24:03 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Fix strcmp usage
+ Just use g_strcmp0 which can handle NULL entries
+
+2016-03-04 10:14:00 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Use audio/x-unaligned-raw instead of audio/x-raw for L16 data
+ Directly setting audio/x-raw caps leads to problems when the delivered
+ data blocks do not align properly at sample boundaries (for example, a
+ data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
+ let a parser be autoplugged.
+ https://bugzilla.gnome.org/show_bug.cgi?id=689460
+
+2016-05-12 11:52:09 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Parsing elst box based on version
+ segment_duration and media_time should be parsed based on version
+ of elst box. Specification defines that an elst box with version 1
+ has uint64 and int64 values for segment_duration and media_time,
+ respectively.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766301
+
+2016-05-14 12:57:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: check if request was cancelled when sending message
+ It might be that the request was aborted by the application and
+ we can return immediatelly
+
+2016-05-14 12:43:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: proxy resolver is on by default
+ Remove from the session creation parameters
+
+2016-05-14 12:15:48 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/Makefile.am:
+ soup: update build to warn about newer deprecated functions
+ We already depend on 2.48
+
+2016-05-14 11:09:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: reduce reading latency by using non-blocking read
+ Non-blocking read will return the amount of data available without
+ blocking to wait for the full requested size.
+ The downside is that now it souphttpsrc needs to have a waiting
+ mechanism in case there is no data available yet to avoid busy
+ looping arond the inputstream.
+
+2016-05-15 12:30:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Take the lock already when reading the other stats, not just for the hash table
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-14 17:04:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ matroska: use math-compat.h for NAN define
+
+2016-05-14 23:39:22 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Use GstBin async-handling instead of our own.
+ Set the async-handling property on GstBin to let it manage
+ async-handling instead of the local handling from the previous
+ commit. Works because of #174a5e in core
+
+2016-05-13 10:17:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: refactor to use Soup's sync API
+ Replace the async API with the sync API to remove all the extra mainloop
+ and context handling. Currently it blocks reading until 'blocksize'
+ bytes are available but that can be improved by using:
+ https://developer.gnome.org/gio/unstable/GPollableInputStream.html#g-pollable-input-stream-read-nonblocking
+ https://bugzilla.gnome.org/show_bug.cgi?id=693911
+
+2016-05-14 04:50:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: replace deprecated API
+ Avoid using soup_server_run_async and old get_port() APIs,
+ replace with me soup_server_listen and get the port through the
+ URIs list returned from the server.
+
+2016-05-14 12:34:10 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Upgrade debug message to error
+ It causes the entire pipeline to fail, it should be easier to find.
+
+2016-05-14 18:32:52 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Hide internal async state changes.
+ When switching fragments, hide the async-start/async-done
+ messages from the parent bin, as otherwise we sometimes (very rarely)
+ hang in PAUSED instead of returning / continuing to PLAYING
+ state.
+
+2016-05-13 21:20:28 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Remove stray carriage-return from debug
+
+2016-05-13 16:43:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ rtp: Ship gstrtpj2kcommon.h file to fix distcheck
+
+2015-04-30 14:43:04 +0200 Jesper Larsen <knorr.jesper@gmail.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Do not write index and header if idx is NULL
+ Fixes criticals with e.g.
+ videotestsrc num-buffers=1 ! identity drop-probability=1.0 ! avimux ! fakesink
+ https://bugzilla.gnome.org/show_bug.cgi?id=748700
+
+2016-05-12 08:43:39 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: manage T tile invalidation bit correctly, update tile id in header correctly.
+ 1. according to RFC, T bit is only set when either the RTP packet only contains the J2K main header, or the packet contains tile parts from multiple tiles. This is now being managed correctly in the code. The second scenario cannot happen with our payloader, since tile headers are always placed in their own RTP packet, and so a packet cannot contain tile parts from multiple tiles.
+ However, I have added code to track if multiple tile parts are included in a single RTP packet, in case in the future we want to put header and data in same packet.
+ 2. Old code would set the tile id to zero for all J2K packets. This is now set correctly to the appropriate tile id.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-12 08:41:51 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: manage fragmented headers correctly
+ J2K main header framentation across multiple RTP packets is now handled correctly
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-11 15:04:26 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kcommon.h:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kdepay.h:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpj2kpay.h:
+ rtpj2k: move common code to shared header, code clean up
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-11 15:01:32 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2k: update documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-12 14:43:43 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/auparse/gstauparse.c:
+ * gst/auparse/gstauparse.h:
+ auparse: Fix sticky event misordering warning
+ Make sure that src pad has caps before sending segment event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766359
+
+2016-05-11 09:28:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Don't notify about stats property changes while taking the session lock
+ The signal handlers might want to actually get the value of the stats
+ property, which would take the session lock again and deadlock.
+ This was introduced by 2e960e70750a0cb7e1117d0c09d08597866a29ee.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-03 13:59:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: improve edts segment handling after seeks in push mode
+ Properly handle edts segments for push-based operation seeking.
+ We only support edts that a single segment that has media at the end,
+ being preceeded by any number of gap segments.
+ This also allows the qt segment rate to be respected after seeks
+ https://bugzilla.gnome.org/show_bug.cgi?id=765669
+
+2016-05-03 10:41:06 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: properly activate segment with rate != 1.0
+ Also use the qt rate to identify the position within a qt segment
+ to properly translate playback time to qt media time
+ https://bugzilla.gnome.org/show_bug.cgi?id=765669
+
+2016-05-03 11:45:01 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix stall when receiving already lost packet
+ When a packet arrives that has already been considered lost as part of a
+ large gap the "lost timer" for this will be cancelled. If the remaining
+ packets of this large gap never arrives, there will be missing entries
+ in the queue and the loop function will keep waiting for these packets
+ to arrive and never push another packet, effectively stalling the
+ pipeline.
+ The proposed fix conciders parts of a large gap definitely lost (since
+ they are calculated from latency) and ignores the late arrivals.
+ In practice the issue is rare since large gaps are scheduled immediately,
+ and for the stall to happen the late arrival needs to be processed
+ before this times out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765933
+
+2016-05-05 14:18:21 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Take session lock when creating stats
+ The access to the session hash table must happen while the session lock is
+ taken, otherwise another thread might modify the hash table while we're
+ creating the stats.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-03 21:17:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: update segment when new duration is found
+ Otherwise the old segment will have a shorter stop time and would
+ cause the stream to end too early.
+
+2016-05-04 11:37:29 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: dismember activate_segment into 2 parts
+ One that updates and push a new segment, the other will move the
+ stream to the new segment starting position
+
+2016-05-04 09:30:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdec.c:
+ * ext/dv/gstdvdemux.c:
+ dv: Use correct pixel-aspect-ratio values
+ The previous ones resulted in odd display aspect ratios and were different
+ from the ones used by e.g. ffmpeg. The new ones now result in display aspect
+ ratios of 4:3 and 16:9.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765946
+
+2015-11-09 17:55:09 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/splitmux.c:
+ tests: add splitmuxsrc test for new "format-location" signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=753625
+
+2015-11-09 17:51:12 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: add a format-location signal that allows bypassing the location property
+ This signal allows a user to directly return a sorted list of
+ files to be joined, so that they don't have to follow the
+ filename pattern that the "location" property expects.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753625
+
+2016-05-04 11:15:20 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix deadlock case when source reaches EOS
+ https://bugzilla.gnome.org/show_bug.cgi?id=765072
+
+2016-05-03 22:59:27 -0700 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: simplify and correct header scanning
+ The wav spec tells that 'fmt' (and 'bext' if present) must come before 'data'.
+ There is no requirement for 'fmt' to be first. We already had a list of chunks
+ to skip, but it is easier to just skip any chunk while seeking for 'fmt'.
+ This fixes reading files generated by ProTools.
+
+2016-04-30 22:15:13 +0900 Hyunjun Ko <zzoon@igalia.com>
+
+ * sys/osxaudio/Makefile.am:
+ * sys/osxaudio/gstosxaudio.c:
+ * sys/osxaudio/gstosxaudiodeviceprovider.c:
+ * sys/osxaudio/gstosxaudiodeviceprovider.h:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ osxaudio: Support audio device provider on osx
+ https://bugzilla.gnome.org/show_bug.cgi?id=753265
+
+2016-05-01 15:09:27 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavimux.c:
+ avimux: set audio header rate according to calculated bps in stop_file
+ ... now that set_fields is no longer called there by
+ e538608b3f90539003de21c1db238f3c9b946e30
+
+2016-04-29 15:04:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Store the segment sequence number in the EOS events and SEGMENT_DONE events/message
+ Also instead of storing it per stream, store it globally in the demuxer. It's
+ the same for each stream anyway.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765806
+
+2016-04-11 10:54:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Always bind to ANY when address is a multicast address and not only on Windows
+ For IPv6 addresses, binding to a multicast group does not work on Linux
+ either. Always bind to ANY and then later join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764679
+
+2016-04-26 17:01:49 +0800 Song Bing <b06498@freescale.com>
+
+ * sys/ximage/ximageutil.c:
+ ximageutil: shouldn't implement transform if don't support it
+ shouldn't implement transform if don't support it. Or gst_buffer_copy_into()
+ will print ERROR log.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765583
+
+2016-04-28 16:24:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Allow MPEG-1 Layer 1 and 2 in addition to 3 in MP4
+ Via the MPEG-4 Part 3 spec we can support the other layers too.
+ Also correct the samples per frame calculation for MP3 if it's MPEG-2 or
+ MPEG-2.5.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765725
+
+2016-04-27 20:46:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Update caps for TCP whenever they change
+ We only changed them for UDP so far, which caused the wrong seqnum-base and
+ other information to be passed to rtpjitterbuffer/etc when seeking. This
+ usually wasn't that much of a problem as the code there is robust enough, but
+ every now and then it causes us to drop up to 32756 packets before we
+ continue doing anything meaningful.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 20:33:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Ensure to not take caps with the wrong pt for getting the clock-rate
+ Especially the caps on the pad might be out of date, and the new caps would be
+ provided for the current pt via the request-pt-map signal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 18:27:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't propagate spurious state change returns from internal elements further
+ We handle them inside rtspsrc and override them in all other cases anyway, so
+ do the same for "internal" state changes like PAUSED->PAUSED and
+ PLAYING->PLAYING.
+ This keeps unexpected NO_PREROLL to confuse state changes in GstBin.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=760532
+ https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 14:09:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Don't override maximum audio chunk size with the scale again just before writing it
+ set_fields() should only be called in the beginning, otherwise we will never
+ remember the maximum audio chunk size and write a wrong block align... which
+ then causes wrong timestamps and other problems.
+
+2016-04-27 13:53:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Actually store the largest audio chunk size for the VBR case of MP2/MP3
+ 3ea338ce271e1f6a96d2ed49d4472b091f6f8b7e changed avimux to do that, but it
+ never actually kept track of the max audio chunk for MP3 and MP2. These are
+ knowing the hdr.scale only after parsing the frames instead of at setcaps
+ time.
+
+2016-04-25 15:03:14 +0200 Mats Lindestam <matslm@axis.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: Allow setting "socket-v6" without setting "socket" too
+ https://bugzilla.gnome.org/show_bug.cgi?id=764897
+
+2016-04-22 15:02:16 +0100 Mario Sanchez Prada <mario@endlessm.com>
+
+ * ext/vpx/gstvpxenc.c:
+ vpxenc: Properly handle frames with too low duration
+ When a frame's duration is too low, calling gst_util_uint64_scale()
+ to scale its value can result into it being truncated to zero, which
+ will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
+ when trying to encode.
+ To prevent this from happening, we simply ignore the duration when
+ encoding if it becomes zero after scaling, logging a warning message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765391
+
+2016-04-22 15:48:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: fix description of linear interlacing method
+
+2016-04-21 14:08:19 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/flv/gstflvmux.c:
+ flv: Handle the case where we do not get any CollectData in handle_buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=765320
+
+2016-04-11 22:41:20 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Do not use unreliable framerate
+ timescale/1 is unreliable value for framerate. Due to downstream
+ element usually use framerate generated by qtdemux, let it be omitted
+ until the framerate can be reliably calculated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764733
+
+2016-04-21 12:53:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ Revert "qtdemux: expose streams with first moof for fragmented format"
+ This reverts commit d8bb6687ea251570c331038279a43d448167d6ad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764733
+
+2016-02-09 17:17:09 +0000 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: support seeking of CENC encrypted streams
+ When playing a stream that has been protected by DASH CENC, playback
+ will fail if a seek is performed. Qtdemux produces the error "stream
+ is protected using cenc, but no cenc protection system information
+ has been found" and playback stops.
+ The problem is that gst_qtdemux_reset() gets called as part of the
+ FLUSH during a seek. This function frees the protection_system_ids
+ array. When gst_qtdemux_configure_protected_caps() is called after the
+ seek has completed, the protection_system_ids array is empty and
+ qtdemux is unable to create the correct output caps for the protected
+ stream.
+ This commit changes it to only free the protection_system_ids on
+ hard resets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761787
+
+2016-04-18 14:33:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: add "retrieve-sender-address" property
+ This allows disabling of sender address retrieval, which might
+ be useful in certain scenarios, like when the socket is connected,
+ or the sender address is not of interest (e.g. when receiving an
+ MPEG-TS stream). Disabling sender address retrieval in those
+ cases can have minor performance advantages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=563323
+
+2015-11-26 13:15:06 +0100 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Change warning handling to break infinite message loop
+ v4l2src can cause an "infinite message loop" when a base control exposed as a
+ property is not provided by the device. In these cases, if in the warning message
+ handling for the bus, the GST_DEBUG_BIN_TO_DOT_FILE* category of functions are used,
+ the src lookup causes a new warning to be posted on the bus, causing a loop.
+ This patch changes the warning for these controls so they are not posted on the bus.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758703
+
+2016-04-15 10:44:02 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ spitmuxsink: Avoid creating small file at EOS
+ When EOS is reached, the current file get closed and the last
+ GOP in the mq was written in a new file.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765072
+
+2016-04-15 19:59:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: S16 uses S32 temporary buffers, float/double their own type
+ Make sure to allocate not only a S16 buffer for S16 but a twice as big one to
+ hold S32.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765116
+
+2016-04-16 02:17:26 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/pulse/pulsesink.c:
+ Revert "pulsesink: uncork if needed upon commit"
+ This reverts commit 0dd46accf6d282ff07065852bd91c85c78af3394.
+ With some audiosinks, starting the ringbuffer on the first commit
+ causes audio glitches at startup by starting to output segments
+ from the ringbuffer before it has been filled / fully prerolled. This
+ doesn't usually happen with pulsesink because we map the pulseaudio
+ ringbuffer directly, but we should keep things consistent with
+ other sinks with regards to startup latency, plus it gives more
+ headway to avoid glitching, should the initial 2nd segment take
+ more than 10ms to generate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=657076
+
+2016-04-15 00:46:56 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: add srtp rollover counters from mikey crypto sessions
+ The server can send multiple crypto sessions, one for each SSRC with its
+ own rollover counter. We parse this information and pass it to the SRTP
+ decoder via the "request-key" signal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730540
+
+2016-04-15 14:35:07 +0000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix debug output when resyncing
+ Don't output the pointer value of the time() function as a timestamp
+ by using the correct variable.
+ Fixes build on Raspberry Pi 3.
+
+2016-04-15 11:36:36 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: If no proxy is set by properties, use the default libsoup proxy resolver
+ That is, use whatever system settings there might exist. This is the same
+ behaviour we use in the HTTP source.
+
+2016-04-14 10:01:28 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to ac2f647
+
+2016-04-13 18:45:07 +0100 Damian Ziobro <damian@xmementoit.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Add max_files_number property
+ https://bugzilla.gnome.org/show_bug.cgi?id=744612
+
+2016-04-13 10:57:03 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: drop reference to videomixer 2
+ Fix a small grammar mistake on "overlayed" while at it.
+
+2016-04-13 09:57:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/ximage/ximageutil.c:
+ ximage: Initialize all fields in the meta explicitly
+ The meta is not allocated with all fields initialized to zeroes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764902
+
+2016-04-12 09:41:00 +0000 Paolo Pettinato <ppettina@cisco.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Forward sticky events on buffer lists too, not only on buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=764933
+
+2016-04-12 15:01:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Drain the field history if the caps are changing
+ Otherwise we will use fields from the old caps with everything set up for the
+ new caps, causing crashes and worse.
+ Also don't do anything if the same caps are set twice.
+
+2016-04-12 15:00:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Instead of confusing crashes later, just error out immediately if mapping a video frame fails
+ This probably still crashes but at least we get some hint about what goes
+ wrong instead of random behaviour later.
+
+2016-04-12 11:38:51 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: check stream is available in PIFF parser
+ qtdemux->streams is an array, it will never evaluate to true when comparing
+ to NULL. Instead we want to check the number of streams to make sure the
+ stream is available.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753614
+ CID 1358389
+
+2016-04-12 11:37:36 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "qtdemux: redundant check in PIFF parser"
+ This reverts commit 41e10524f3babdd92aac8c8c9d5b9cdf184c2d4e.
+
+2016-04-12 11:05:50 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: redundant check in PIFF parser
+ qtdemux->streams is an array of size GST_QTDEMUX_MAX_STREAMS, it will never
+ evaluate to true when comparing to NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753614
+ CID 1358389
+
+2016-04-12 11:56:08 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: avoid leaking GValues
+ unset the GValue if we don't use it any more to avoid leaks.
+
+2016-04-12 10:15:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation
+ The head of the queue is the oldest packet (as in lowest seqnum), the tail is
+ the newest packet. To calculate the fill level, we should calculate tail-head
+ while considering wraparounds. Not the other way around.
+ Other code is already doing this in the correct order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764889
+
+2016-04-11 10:44:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/Makefile.am:
+ rtpmanager: It's GST_LIBS, not GST_LIBS_LIBS
+
+2016-04-11 08:33:17 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix parsing segment duration of empty edit list box
+ For empty edit list, segment-duration in edit list box should not be
+ used for segment event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764870
+
+2016-04-08 13:05:57 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: make timecodescale configurable
+ In some use cases the default timecodescale will produce blocks with the same timestamp
+ https://bugzilla.gnome.org/show_bug.cgi?id=764769
+
+2016-04-07 13:01:52 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jiterbuffer: Move assertion to the right location
+ We shouldn't have "late" lost timers at that point
+
+2016-03-02 14:25:24 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Speed up lost timeout handling
+ When downstream blocks, "lost" timers are created to notify the
+ outgoing thread that packets are lost.
+ The problem is that for high packet-rate streams, we might end up with
+ a big list of lost timeouts (had a use-case with ~1000...).
+ The problem isn't so much the amount of lost timeouts to handle, but
+ rather the way they were handled. All timers would first be iterated,
+ then the one selected would be handled ... to re-iterate the list again.
+ All of this is being done while the jbuf lock is taken, which in some use-cases
+ would return in holding that lock for 10s... blocking any buffers from
+ being accepted in input... which would then arrive late ... which would
+ create plenty of lost timers ... which would cause the same issue.
+ In order to avoid that situation, handle the lost timers immediately when
+ iterating the list of pending timers. This modifies the complexity from
+ a quadratic to a linear complexity.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-03-02 14:23:01 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Don't create lost events if we don't need them
+ When "do-lost" is set to FALSE we don't use/send the lost events.
+ In that case, don't create them to start with :)
+ https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-03-02 13:57:07 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Add tracing of lock usage
+ Helps with debugging lock usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-02-10 19:56:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2: Don't leak v4l2 objects and props on probe errors
+
+2016-04-04 17:42:03 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: add unit test for jpeg depayloader packet loss handling
+ Make sure it always outputs something that looks like a valid
+ JPEG frame, ie. starts with an SOI marker and ends with an EOI
+ marker.
+
+2016-03-15 03:25:26 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtp/gstrtpjpegdepay.c:
+ rtpjpegdepay: Don't send invalid frames downstream after packet loss or a DISCONT
+ After clearing the adapter due to a DISCONT, as might happen when some packet(s)
+ have been lost, the depayloader was pushing data into the adapter (which had no
+ header due to the clear), creating a headerless frame out of it, and sending it
+ downstream. The downstream decoder would then usually ignore it; unless there
+ were lots of DISCONTs from the jitterbuffer in which case the decoder would reach
+ its max_errors limit and throw an element error. Now we just discard that data.
+ It is probaby not worth trying to salvage this data because non-progressive
+ jpeg does not degrade gracefully and makes the video unwatchable even with
+ low packet loss such as 3-5%.
+
+2016-01-05 16:15:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtpjitterbuffer: Add RFC7273 media clock handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=762259
+
+2015-07-10 09:44:15 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: PIFF box detection and parsing support
+ The PIFF data is stored in a custom UUID box which is parsed and the
+ crypto_info of the element is updated accordingly. This allows
+ downstream decryptors to process and decrypt the protected content.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753614
+
+2016-04-01 12:15:05 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbisdepay: remove dead code
+ payload_buffer hasn't been assigned a value before the jumps to
+ switch_failed or packet_short. So the value must be NULL. No need
+ to unmap and unref.
+ CID #1316476
+
+2016-03-31 14:57:20 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: fix leak
+ Free memory of current macroblock once it isn't needed so it isn't leaked
+ by the call of the gst_rtp_h263_pay_B_mbfinder function.
+ if (!(mac = gst_rtp_h263_pay_B_mbfinder (context, gob, mac, mb))) {
+ CID 1212156
+
+2016-03-31 02:15:04 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: Handle a hang draining out at EOS
+ Make sure that all data is drained out when the reference pad
+ goes EOS. Fixes a problem where data that arrives on other
+ pads after the reference pad finishes can stall forever and
+ never pass EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763711
+
+2016-03-18 15:45:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix occasional deadlock when ending file with subtitle
+ Deadlock occurs when splitting files if one stream received no buffer during
+ the first GOP of the next file. That can happen in that scenario for example:
+ 1) The first GOP of video is collected, it has a duration of 10s.
+ max_in_running_time is set to 10s.
+ 2) Other streams catchup and we receive the first subtitle buffer at ts=0 and
+ has a duration of 1min.
+ 3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to
+ 1min. That buffer is blocked in handle_mq_input() because
+ max_in_running_time is still 10s.
+ 4) Since all in_running_time are now > 10s, max_out_running_time is now set to
+ 10s. That first GOP gets recorded into the file. The muxer pop buffers out
+ of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the
+ GstDataQueue is empty.
+ 5) A 2nd GOP of video is collected and has a duration of 10s as well.
+ max_in_running_time is now 20s. Since subtitle's in_running_time is already
+ 1min, that GOP is already complete.
+ 6) But let's say we overran the max file size, we thus set state to
+ SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of
+ previous GOP) arrives in handle_mq_output(), EOS event is sent downstream
+ instead. But since the subtitle queue is empty, that's never going to
+ happen. Pipeline is now deadlocked.
+ To fix this situation we have to:
+ - Send a dummy event through the queue to wakeup output thread.
+ - Update out_running_time to at least max_out_running_time so it sends EOS.
+ - Respect time order, so we set out_running_tim=max_in_running_time because
+ that's bigger than previous buffer and smaller than next.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763711
+
+2015-11-17 18:17:35 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Add new signal 'on-app-rtcp'
+ Similar to the 'on-feedback-rtcp' signal, but emitted for RTCP APP
+ packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762217
+
+2016-03-24 15:57:11 +0900 Minjae Kim <nate.kim@lge.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpmanager: Set to initial value for 'ntpns' in get_current_times()
+ Initialize "ntpns" variable to -1 as the OE compiler for some reason doesn't
+ realize that the variable is set in all code paths.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764119
+
+2016-01-31 11:08:38 +1100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay: Allow different quantization tables for components 2 and 3
+ RFC 2435 mentions in section 4.1 that U/V use table number 1, but this seems
+ just like an example. Some encoders are not following that and there seems to
+ be no reason to reject their streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761345
+
+2016-03-24 19:23:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: Use threads on multi-core systems
+ This is a redo of commit b848c1b6ffd1e508228820a013f94fb445e4777f. The
+ code was lost when the elements where ported to use a baseclass.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764169
+
+2016-02-29 23:40:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * tests/check/elements/splitmux.c:
+ splitmuxsink: only try to create internal sink if it doesn't exist
+ This allows splitmuxsink to be reused after being put to NULL.
+ Test included
+ https://bugzilla.gnome.org/show_bug.cgi?id=762893
+
+2015-10-01 13:41:23 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: probe all colorspace supported by device
+ A device can support more than one colorspace for a given image
+ dimension and pixel format. So we have to probe all the supported
+ colorspace and not only rely on the default one. Otherwise we could end
+ up with negotiation failure if the caps colorimetry field don't match
+ the v4l2 device default one even if the v4l2 could support such
+ colorimetry.
+ This patch enable probing if colorspace for both capture and output
+ device. It really makes sense for output device since the colorspace
+ shall be set by the application and a little less for capture device
+ which, at the moment, shall provide the colorspace; ie: the v4l2
+ specification seems to not take into account the fact that a capture
+ device could do colorspace conversion.
+ As a side effet, probing takes some times and so sligthly delay v4l2
+ initialization. Note that this patch only probe colorspace and not all
+ colorspace, matrix, transfer and range combination to avoid taking too
+ much time, especially with low-speed devices as full probing do 1782
+ ioctl.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755937
+
+2016-03-24 16:21:56 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/flvdemux.c:
+ check: Fix indentation
+
+2016-03-24 16:20:39 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/flvdemux.c:
+ tests: Remove unused variables
+
+2016-03-16 20:26:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Return the current caps on the srcpads on caps queries
+ It's not like we could accept any other caps here. The caps are decided by the
+ upstream caps event.
+ Also keep the filter order intact when filtering the results against the
+ filter caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-24 15:14:23 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix qtdemux memory leak in src_convert function
+ If we don't find the index of the sample correctly in src_convert function,
+ we have to unref about the qtdemux before returning value.
+ So, I have modify it about instead pass qtdemux as a parameter into
+ src_convert function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763973
+
+2016-03-22 13:15:20 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add check condition for fail case in get_duration function
+ Currently, get_duration function always return the TRUE even though
+ it can't be set duration correctly. So, we need to add the else condition
+ about the fail case. Also, we already set the GST_CLOCK_TIME_NONE
+ in this function. So I have modify it which is related code in some
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763968
+
+2016-03-21 10:11:23 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Modify data type of duration in handle_src_query function
+ Data type of duration need to modify from guint64 to GstClockTime
+ for consistency in handle_src_query function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763965
+
+2016-03-18 14:40:58 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Added unit tests for field=auto
+ https://bugzilla.gnome.org/show_bug.cgi?id=763869
+
+2016-03-17 21:21:02 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Added "auto" fields mode
+ The "auto" fields mode will detect the upstream and downstream framerates and
+ will decide to deinterlace all or only top fields.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763869
+
+2016-03-16 20:17:55 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ * tests/check/elements/flvdemux.c:
+ flvdemux: don't emit pad-added until caps are ready
+ In other words, gst_pad_get_current_caps should never return NULL
+ in a pad-added callback from the demuxer.
+ Added tests for the two special cases with AAC and H.264 where this
+ would happen every time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763780
+
+2016-03-04 10:30:12 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/aalib/gstaasink.c:
+ * ext/cairo/gstcairooverlay.c:
+ * ext/dv/gstdvdec.c:
+ * ext/dv/gstdvdemux.c:
+ * ext/flac/gstflacdec.c:
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflactag.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufsink.c:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jpeg/gstjpegdec.c:
+ * ext/jpeg/gstjpegenc.c:
+ * ext/jpeg/gstsmokedec.c:
+ * ext/jpeg/gstsmokeenc.c:
+ * ext/libcaca/gstcacasink.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/libpng/gstpngenc.c:
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesrc.c:
+ * ext/raw1394/gstdv1394src.c:
+ * ext/raw1394/gsthdv1394src.c:
+ * ext/shout2/gstshout2.c:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/speex/gstspeexdec.c:
+ * ext/speex/gstspeexenc.c:
+ * ext/taglib/gstapev2mux.cc:
+ * ext/taglib/gstid3v2mux.cc:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * ext/wavpack/gstwavpackenc.c:
+ * gst/alpha/gstalpha.c:
+ * gst/alpha/gstalphacolor.c:
+ * gst/apetag/gstapedemux.c:
+ * gst/audiofx/audiopanorama.c:
+ * gst/audiofx/gstscaletempo.c:
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ * gst/auparse/gstauparse.c:
+ * gst/autodetect/gstautoaudiosink.c:
+ * gst/autodetect/gstautoaudiosrc.c:
+ * gst/autodetect/gstautovideosink.c:
+ * gst/autodetect/gstautovideosrc.c:
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavimux.c:
+ * gst/avi/gstavisubtitle.c:
+ * gst/cutter/gstcutter.c:
+ * gst/debugutils/breakmydata.c:
+ * gst/debugutils/cpureport.c:
+ * gst/debugutils/gstcapsdebug.c:
+ * gst/debugutils/gstcapssetter.c:
+ * gst/debugutils/gstnavigationtest.c:
+ * gst/debugutils/gstnavseek.c:
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/debugutils/gsttaginject.c:
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/debugutils/testplugin.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/dtmf/gstdtmfsrc.c:
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gstop.c:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstradioac.c:
+ * gst/effectv/gstrev.c:
+ * gst/effectv/gstripple.c:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gststreak.c:
+ * gst/effectv/gstvertigo.c:
+ * gst/effectv/gstwarp.c:
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvmux.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/icydemux/gsticydemux.c:
+ * gst/id3demux/gstid3demux.c:
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/interleave/deinterleave.c:
+ * gst/interleave/interleave.c:
+ * gst/isomp4/gstrtpxqtdepay.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-encode.c:
+ * gst/law/mulaw-decode.c:
+ * gst/law/mulaw-encode.c:
+ * gst/level/gstlevel.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/webm-mux.c:
+ * gst/monoscope/gstmonoscope.c:
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesrc.c:
+ * gst/multifile/gstsplitfilesrc.c:
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multipart/multipartdemux.c:
+ * gst/multipart/multipartmux.c:
+ * gst/replaygain/gstrganalysis.c:
+ * gst/replaygain/gstrglimiter.c:
+ * gst/replaygain/gstrgvolume.c:
+ * gst/rtp/gstasteriskh263.c:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpL24pay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpbvpay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726pay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpilbcpay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtppcmupay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpsirenpay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpstreamdepay.c:
+ * gst/rtp/gstrtpstreampay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9pay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpdtmfmux.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpptdemux.c:
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/rtsp/gstrtpdec.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/shapewipe/gstshapewipe.c:
+ * gst/smpte/gstsmpte.c:
+ * gst/smpte/gstsmptealpha.c:
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ * gst/videobox/gstvideobox.c:
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videofilter/gstgamma.c:
+ * gst/videofilter/gstvideobalance.c:
+ * gst/videofilter/gstvideoflip.c:
+ * gst/videofilter/gstvideomedian.c:
+ * gst/videomixer/videomixer2.c:
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavparse/gstwavparse.c:
+ * gst/y4m/gsty4mencode.c:
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/oss/gstosssink.c:
+ * sys/oss/gstosssrc.c:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxvideo/osxvideosink.m:
+ * sys/sunaudio/gstsunaudiosink.c:
+ * sys/sunaudio/gstsunaudiosrc.c:
+ * sys/waveform/gstwaveformsink.c:
+ * sys/ximage/gstximagesrc.c:
+ * tests/check/elements/autodetect.c:
+ * tests/check/elements/qtmux.c:
+ good: use new gst_element_class_add_static_pad_template()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763076
+
+2016-03-04 09:42:44 +0100 David Buchmann <david.buchmann@gmail.com>
+
+ * tests/check/elements/flvmux.c:
+ flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE
+ https://bugzilla.gnome.org/show_bug.cgi?id=762207
+
+2015-11-04 14:51:19 +0900 Jihae Yi <jihae.yi@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: avoid potentially overflowing expression
+ https://bugzilla.gnome.org/show_bug.cgi?id=757569
+
+2016-03-22 10:43:45 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add the function to get channels and sample rate for AAC
+ Add aac_get_channels and sample_rate function to get these value for
+ AAC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749110
+
+2016-03-24 13:33:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 12:27:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.8.0
+
+2016-03-24 12:02:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-03-16 20:18:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time
+ Doing queries while holding the object lock is a bit dangerous, and in this
+ case causes deadlocks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-17 20:53:27 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Fix typo to not change the input caps but our filtered caps
+ Changing the input caps and not using them anymore afterwards is useless, and
+ it breaks negotiation in pipelines like:
+ gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
+ deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
+ fakesink
+
+=== release 1.7.91 ===
+
+2016-03-15 12:04:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.91
+
+2016-03-15 11:53:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-03-15 11:41:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/hu.po:
+ * po/sr.po:
+ po: Update translations
+
+2016-03-15 03:26:14 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/rtpsource.c:
+ rtpmanager: Some comment and documentation clarifications/fixes
+
+2016-03-13 10:33:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ Revert "flacparse: push tags in pre_push_frame"
+ This reverts commit 4065fcb80a49924b70f0c8fc159dec0ff47943a1.
+ flacparse should not push tags by itself, the base class is going to do that
+ while properly merging in upstream tags. It just didn't because of a bug in
+ the base class, which was hidden by this commit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763553
+
+2016-02-25 05:17:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtp/dboolhuff.c:
+ * gst/rtp/dboolhuff.h:
+ * gst/rtp/gstrtpsbcpay.c:
+ win32: Don't use __attribute__ on MSVC
+ Use MSVC-equivalents for alignment and packing compiler directives when building
+ on MSVC
+
+2016-02-25 05:16:42 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ win32: Don't try to include xmath.h on newer Visual Studio
+
+2016-02-25 05:16:09 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/monoscope/gstmonoscope.c:
+ gst Factor out endian-order RGB formats
+ MSVC seems to ignore preprocessor conditionals inside static pad
+ template macros.
+
+2016-03-08 17:37:17 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ dirctsoundsink: Setting volume should not unmute
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:57:24 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ dirctsoundsink: Fix volume reset on unmute
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:03:55 +0100 Alban Bedel <alban.bedel@avionic-design.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fix capture with bayer formats other than bggr
+ gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8
+ for all bayer formats. This is obviously broken if the device use
+ another ordering. Fix this by properly reading the format parameter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763318
+
+2016-03-07 10:28:06 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: reset pending segment if we are already pushing one
+ When upstream is running in bytes in push-mode, qtdemux will
+ convert seeks from time to bytes and send it upstream. Upstream
+ element will perform a byte seek and send a byte segment to qtdemux
+ that will convert it to time and push it downstream.
+ There is, however, the pending_segment variable that stores a new
+ segment event to be pushed before the next data. When handling seeks
+ as mentioned above this variable was being ignored and, if it contained
+ some segment event, it would override the one resulting from the seek.
+ This would restore a previous segment and would cause the seek segment
+ to be discarded downstream.
+ This patch fixes this issue by unrefing any pending segment as the
+ seek from upstream should contain the latest one that should be
+ used, as requested by the application.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763165
+
+2016-03-07 10:27:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: run gst-indent
+ Otherwise commits will fail with our indent check hook
+
+2016-03-04 15:09:45 +0100 Josep Torra <n770galaxy@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: fix colorimetry for NV12
+ Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior.
+ At #v4l it was described that documentation might be wrong and that
+ we should trust this macro instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762529
+
+2016-03-04 14:07:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix multicast group joining with provided sockets on Windows
+ On Windows the socket will be bound to ANY instead of the multicast group,
+ as binding to a multicast group does not work. Which would mean that we
+ override src->addr to become ANY and won't automatically join a multicast
+ group anymore on Windows.
+ On Linux we would automatically join a multicast group, keep it consistent.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763093
+
+2016-03-02 13:13:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases"
+ This reverts commit a7fb7b53592d87f7983544debb74d364fc3257ad.
+ The mutex is taken by the caller, we should keep it locked when returning so
+ the caller can unlock it again.
+
+2016-03-01 15:01:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: push tags in pre_push_frame
+ Push a tag event before pre-roll if we have tags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762660
+
+=== release 1.7.90 ===
+
+2016-03-01 18:15:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.90
+
+2016-03-01 17:03:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/ca.po:
+ * po/da.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/or.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/tr.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-03-01 16:53:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/bg.po:
+ * po/cs.po:
+ * po/de.po:
+ * po/fr.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2016-03-01 14:14:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases
+
+2016-02-29 10:10:24 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: remove impossible condition
+ It is impossible for a guint to have a negative value, no need to check for
+ this. Introduced in commit 6861d11c49ea0f30d2432cf4ebf6108bc89897f1
+ CID 1354509
+
+2016-02-28 10:12:36 +0100 Petr Viktorin <encukou@gmail.com>
+
+ * gst/alpha/gstalpha.c:
+ alpha: Fix sample pipeline
+ Use the zorder pad property to make sure the semitransparent
+ video is on top of the background.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762809
+
+2016-02-28 13:42:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/replaygain/gstrgvolume.c:
+ * tests/check/elements/rgvolume.c:
+ rgvolume: make tag list writable before modifying it
+ Making the event itself writable is not enough, it won't make
+ the actual taglist in the event writable as well. Instead, just
+ make a copy of the taglist and then create a new tag event from
+ that if required, replacing the old one. Before we would
+ inadvertently modify taglists upstream elements might still
+ be holding on to. Add unit test for this as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762793
+
+2016-02-28 13:01:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Properly error out if binding the UDP sockets fails
+ udpsrc is not returning us a socket in that case.
+
+2016-02-27 20:33:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ goom: Use goom_set_resolution() instead of recreating the goom instance when the resolution changes
+ https://bugzilla.gnome.org/show_bug.cgi?id=762765
+
+2016-02-27 20:32:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ Revert "goom: Initialize the goom struct only once we know width/height and recreate it if those change"
+ This reverts commit cc6e102643c1bae928316dca9f34db028fb9a67e.
+
+2016-02-27 20:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ goom: Initialize the goom struct only once we know width/height and recreate it if those change
+ Fixes crash when the width and/or height is changing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762765
+
+2016-02-26 12:41:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b64f03f to 6f2d209
+
+2016-02-25 22:54:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: add rtpopusdepay and rtpopuspay to documentation
+
+2016-02-17 15:15:11 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ rtp: opus: move Opus RTP payloader/depayloader from -bad to -good
+ https://bugzilla.gnome.org/show_bug.cgi?id=756282
+
+2016-02-17 15:10:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ Merge branch 'plugin-move-rtp-opus'
+ Move Opus RTP depayloader/payloader from -bad to -good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756282
+
+2016-02-25 11:33:13 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: cenc aux info parsing from mdat support in PULL mode
+ This is already supported for PUSH mode but was failing in PULL mode.
+ The aux info is sometimes stored in the mdat before the first sample,
+ so the loop task needs to pull data stored at that location and
+ perform the aux info cenc parsing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761700
+ https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2016-02-24 11:28:09 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: prevent buffer flow if any stream failed to be exposed
+ In some cases the stream configuration can fail, for instance if the
+ stream is protected and no decryptor was found. For those situations
+ the demuxer shouldn't emit any data on the corresponding source pad of
+ the stream and bail out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2016-02-24 09:12:03 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't push encrypted buffer without cenc metadata
+ When the cenc metadata is stored outside of the moof box and the
+ stream is exposed it is possible that the cenc metadata hasn't been
+ processed yet while the first buffer is being pushed. When this
+ happens the buffer can't possibly be decrypted downstream so don't
+ push it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2015-10-21 16:21:45 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: read saio aux_info_type as a FOURCC
+ https://bugzilla.gnome.org/show_bug.cgi?id=756897
+
+2016-02-23 18:27:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/smpte/gstsmpte.c:
+ gst: Handle gst_pad_get_current_caps() returning NULL gracefully
+
+2016-02-23 18:12:54 +0200 Dave Craig <dcraig@brightsign.biz>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
+ Remove calls to gst_pad_has_current_caps() which then go on to call
+ gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
+ use gst_pad_get_current_caps() and check for NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2015-12-16 12:40:39 +0000 Dave Craig <dcraig@brightsign.biz>
+
+ * ext/flac/gstflacenc.c:
+ * gst/flv/gstflvmux.c:
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/shapewipe/gstshapewipe.c:
+ * gst/videocrop/gstaspectratiocrop.c:
+ gst: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
+ Remove calls to gst_pad_has_current_caps() which then go on to call
+ gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
+ use gst_pad_get_current_caps() and check for NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2015-12-16 10:54:17 +0000 Dave Craig <dcraig@brightsign.biz>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Handle gst_pad_get_current_caps() returning NULL gracefully
+ This can happen when the pipeline is currently shutting down.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2016-02-23 15:57:18 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Don't handle seek until ready
+ https://bugzilla.gnome.org/show_bug.cgi?id=762542
+
+2016-02-23 15:55:13 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Unref seek event
+ https://bugzilla.gnome.org/show_bug.cgi?id=762542
+
+2016-02-22 11:01:40 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: close file on write error with next-file mode is set to buffer
+ If we have an error during fwrite call, file stays open and thus next
+ incoming buffer will trigger an assert when trying to opening a new
+ file.
+ This happens if we do not restart element, file is closed at stop, and
+ if application handles the returned GST_FLOW_ERROR to keep bin alive.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762434
+
+2016-02-19 23:44:42 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: don't output empty tags/tag elements
+ Such files will not play on Android, because of bug in libwebm matroska parsing, which is still present in 6.0.1
+ https://bugzilla.gnome.org/show_bug.cgi?id=762349
+
+2016-02-04 15:59:04 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: make up an OpusHead block if possible when missing
+ https://bugzilla.gnome.org/show_bug.cgi?id=761489
+
+2016-02-04 10:43:15 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: make up an OpusHead block if possible when missing
+ This block is needed in the Matroska file, but data coming from
+ RTP may not have one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761489
+
+2016-02-22 13:53:21 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: make stream-id more readable and order-friendly
+ ... as streams are so ordered by id by e.g. decodebin
+ (and as typically already honoured by other demuxers).
+
+2016-02-22 13:25:51 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: remove confusing duplicate track uid field
+
+2016-02-22 14:03:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtpvp9pay.c:
+ rtpvp9pay: add missing break
+ VP9_PAY_PICTURE_ID_7BITS and VP9_PAY_PICTURE_ID_15BITS are mutually
+ exclusive options of the picture-id-mode. We can break after the
+ first case.
+ 1 or 2 bytes need to be added to the header length depending on the
+ PictureID size.
+ https://tools.ietf.org/html/draft-uberti-payload-vp9-00#section-4.2
+ CID 1353479
+
+2016-02-22 09:09:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix buffer memory leak
+ buffer being mapped is not being unmapped in some cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=762420
+
+2015-11-04 10:19:03 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpmanager: Don't warn for duplicate/reordered packets
+ This is a normal scenario and should not be a warning.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762208
+
+2016-02-21 09:47:43 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/alpha/alpha.vcproj:
+ * gst/auparse/auparse.vcproj:
+ * gst/avi/avi.vcproj:
+ * gst/cutter/cutter.vcproj:
+ * gst/debugutils/debug.vcproj:
+ * gst/debugutils/navigationtest.vcproj:
+ * gst/effectv/effectv.vcproj:
+ * gst/flx/flxdec.vcproj:
+ * gst/goom/goom.vcproj:
+ * gst/goom2k1/goom.vcproj:
+ * gst/interleave/interleave.vcproj:
+ * gst/isomp4/qtdemux.vcproj:
+ * gst/law/alaw.vcproj:
+ * gst/law/mulaw.vcproj:
+ * gst/matroska/matroska.vcproj:
+ * gst/multipart/multipart.vcproj:
+ * gst/rtp/rtp.vcproj:
+ * gst/smpte/smpte.vcproj:
+ * gst/spectrum/spectrum.vcproj:
+ * gst/udp/udp.vcproj:
+ * gst/videobox/videobox.vcproj:
+ * gst/videocrop/videocrop.vcproj:
+ * gst/videofilter/gamma.vcproj:
+ * gst/videofilter/videobalance.vcproj:
+ * gst/videofilter/videofilter.vcproj:
+ * gst/videofilter/videoflip.vcproj:
+ * gst/videomixer/videomixer.vcproj:
+ * gst/wavenc/wavenc.vcproj:
+ * gst/wavparse/wavparse.vcproj:
+ * gst/y4m/y4menc.vcproj:
+ * win32/MANIFEST:
+ * win32/vs6/autogen.dsp:
+ * win32/vs6/gst_plugins_good.dsw:
+ * win32/vs6/libgstalaw.dsp:
+ * win32/vs6/libgstalpha.dsp:
+ * win32/vs6/libgstalphacolor.dsp:
+ * win32/vs6/libgstapetag.dsp:
+ * win32/vs6/libgstaudiofx.dsp:
+ * win32/vs6/libgstauparse.dsp:
+ * win32/vs6/libgstautodetect.dsp:
+ * win32/vs6/libgstavi.dsp:
+ * win32/vs6/libgstcutter.dsp:
+ * win32/vs6/libgstdirectsound.dsp:
+ * win32/vs6/libgsteffectv.dsp:
+ * win32/vs6/libgstflx.dsp:
+ * win32/vs6/libgstgoom.dsp:
+ * win32/vs6/libgsticydemux.dsp:
+ * win32/vs6/libgstid3demux.dsp:
+ * win32/vs6/libgstinterleave.dsp:
+ * win32/vs6/libgstjpeg.dsp:
+ * win32/vs6/libgstlevel.dsp:
+ * win32/vs6/libgstmatroska.dsp:
+ * win32/vs6/libgstmedian.dsp:
+ * win32/vs6/libgstmonoscope.dsp:
+ * win32/vs6/libgstmulaw.dsp:
+ * win32/vs6/libgstmultipart.dsp:
+ * win32/vs6/libgstpng.dsp:
+ * win32/vs6/libgstqtdemux.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgstrtsp.dsp:
+ * win32/vs6/libgstsmpte.dsp:
+ * win32/vs6/libgstspeex.dsp:
+ * win32/vs6/libgstudp.dsp:
+ * win32/vs6/libgstvideobalance.dsp:
+ * win32/vs6/libgstvideobox.dsp:
+ * win32/vs6/libgstvideocrop.dsp:
+ * win32/vs6/libgstvideoflip.dsp:
+ * win32/vs6/libgstvideomixer.dsp:
+ * win32/vs6/libgstwaveform.dsp:
+ * win32/vs6/libgstwavenc.dsp:
+ * win32/vs6/libgstwavparse.dsp:
+ * win32/vs7/libgstdirectsound.vcproj:
+ * win32/vs8/gst-plugins-good.sln:
+ * win32/vs8/libgst1394.vcproj:
+ * win32/vs8/libgstaasink.vcproj:
+ * win32/vs8/libgstalaw.vcproj:
+ * win32/vs8/libgstalpha.vcproj:
+ * win32/vs8/libgstalphacolor.vcproj:
+ * win32/vs8/libgstannodex.vcproj:
+ * win32/vs8/libgstapetag.vcproj:
+ * win32/vs8/libgstaudiofx.vcproj:
+ * win32/vs8/libgstauparse.vcproj:
+ * win32/vs8/libgstautodetect.vcproj:
+ * win32/vs8/libgstavi.vcproj:
+ * win32/vs8/libgstcacasink.vcproj:
+ * win32/vs8/libgstcdio.vcproj:
+ * win32/vs8/libgstcutter.vcproj:
+ * win32/vs8/libgstdirectsound.vcproj:
+ * win32/vs8/libgstdv.vcproj:
+ * win32/vs8/libgsteffectv.vcproj:
+ * win32/vs8/libgstflac.vcproj:
+ * win32/vs8/libgstflxdec.vcproj:
+ * win32/vs8/libgstgoom.vcproj:
+ * win32/vs8/libgsticydemux.vcproj:
+ * win32/vs8/libgstid3demux.vcproj:
+ * win32/vs8/libgstjpeg.vcproj:
+ * win32/vs8/libgstladspa.vcproj:
+ * win32/vs8/libgstlevel.vcproj:
+ * win32/vs8/libgstmatroska.vcproj:
+ * win32/vs8/libgstmng.vcproj:
+ * win32/vs8/libgstmonoscope.vcproj:
+ * win32/vs8/libgstmulaw.vcproj:
+ * win32/vs8/libgstmultipart.vcproj:
+ * win32/vs8/libgstpng.vcproj:
+ * win32/vs8/libgstrtp.vcproj:
+ * win32/vs8/libgstrtsp.vcproj:
+ * win32/vs8/libgstshout2.vcproj:
+ * win32/vs8/libgstsmpte.vcproj:
+ * win32/vs8/libgstspeex.vcproj:
+ * win32/vs8/libgsttaglib.vcproj:
+ * win32/vs8/libgstudp.vcproj:
+ * win32/vs8/libgstvideobalance.vcproj:
+ * win32/vs8/libgstvideobox.vcproj:
+ * win32/vs8/libgstvideoflip.vcproj:
+ * win32/vs8/libgstvideomixer.vcproj:
+ * win32/vs8/libgstwavenc.vcproj:
+ * win32/vs8/libgstwavparse.vcproj:
+ win32: remove outdated build cruft
+ This hasn't been touched for generations, doesn't work,
+ and is just causing confusion. We also don't want to
+ maintain these files manually.
+
+2016-02-20 11:51:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: don't use undeclared core debug category symbols
+
+2016-02-06 14:39:05 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: workaround for files with wrong color_table_id value
+ Instead of erroring out, just use the default color table.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761637
+
+2016-02-19 15:02:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/rtp/gstrtpvp9depay.c:
+ flvmux, rtpvp9depay: fix indentation
+
+2016-02-19 15:03:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2src: fix indentation
+
+2015-12-04 00:46:34 +1100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: plug leak(s) in error-scenario
+ https://bugzilla.gnome.org/show_bug.cgi?id=762210
+
+2015-12-04 00:46:12 +1100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix eos event leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=762209
+
+2016-02-19 14:41:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/flvdemux.c:
+ * tests/check/elements/flvmux.c:
+ * tests/check/elements/rtph263.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: fix indentation
+
+2016-02-18 16:09:29 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: port testharness to GstHarness and cleanup/improve
+ Probably found a bug as well, in that there are some timestamps in
+ there that are looking very wrong. (marked with FIXME)
+ https://bugzilla.gnome.org/show_bug.cgi?id=762267
+
+2016-02-18 10:27:19 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: test cleanups/improvements
+ Use fail_unless and friends instead of g_assert
+ Factor seq-num checking out to separate function
+ Check more return-values from push and crank and others
+ https://bugzilla.gnome.org/show_bug.cgi?id=762254
+
+2015-12-03 11:07:05 +0100 Stian Selnes <stian@pexip.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: fix leaks in unit test
+ https://bugzilla.gnome.org/show_bug.cgi?id=762214
+
+2016-02-19 12:38:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.2 ===
+
+2016-02-19 11:49:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.2
+
+2016-02-19 10:31:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: Update translations
+
+2016-02-18 18:33:13 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: plug leaks in cenc aux info parsing
+
+2016-02-18 13:43:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: fix spurious souphttpsrc test timouts
+ Set GSETTINGS_BACKEND=memory, apparently there's something
+ about fork() and the dconf backend (or whatever else that
+ drags in or activates) that messes up locking and causes
+ timeouts due to deadlocks in g_mutex_lock(), since
+ everything works fine with CK_FORK=no as well.
+
+2016-02-18 11:10:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Unmap wavpack header buffer after creating it
+ Otherwise it will be mapped writable all the time and we can't read from it
+ anywhere.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762239
+
+2015-12-08 18:49:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Add test for big seqnum gap handling
+ Make sure that the packets queued when detecting a big gap are pushed
+ after reset (5 consective seqnums) and not dropped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762211
+
+2016-02-17 15:03:13 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtputils.h:
+ rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions
+
+2016-02-09 13:17:00 +0000 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only transform protected caps once
+ Commit 7873bede3134b15e5066e8d14e54d1f5054d2063
+ (https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the
+ behaviour of qtdemux to call gst_qtdemux_configure_stream() for
+ every new moof.
+ When playing a protected stream, gst_qtdemux_configure_stream()
+ calls gst_qtdemux_configure_protected_caps(). The
+ gst_qtdemux_configure_protected_caps() function takes the original
+ media format, puts this in a field called "original-media-type"
+ and then changes the caps to "application/x-cenc".
+ The gst_qtdemux_configure_protected_caps() did not handle the case
+ of being called multiple times, causing it to incorrectly set the
+ caps. The second call was causing the caps to be set to:
+ application/x-cenc, original-media-type"application/x-cenc"
+ This commit makes gst_qtdemux_configure_protected_caps() check that
+ the caps have already been transformed, so that it only gets
+ changed once.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761769
+
+2015-11-03 14:50:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ opus: Add proper support for multichannel audio
+ https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-06-30 13:51:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ opus: Copy metadata in the (de)payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without tags or
+ with only the audio tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-05-04 11:23:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ opusdepay: Set multistream=FALSE on the Opus caps
+ The RTP Opus mapping only allows mono/stereo, and not multistream Opus
+ streams.
+
+2015-03-24 13:57:54 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: Forward stereo preferences from caps upstream
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-24 13:56:21 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: Set the number of channels to 2 as per RFC draft
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-23 12:24:55 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-02-19 14:30:10 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: default encoding name to OPUS
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 14:05:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: make caps writable before truncating them
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-05 10:27:51 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: negotiate the encoding name
+ Chrome uses a different encoding name that gstreamer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2014-11-01 10:10:27 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopus: Use OPUS encoding name
+ Both Firefox and Chrome uses OPUS as the encoding in their SDP.
+ Adding this now defacto standard name remove the need for special
+ case in SDP parsing code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2013-01-31 12:30:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opuspay: fix timestamps
+ Copy timestamps to payloaded buffer.
+ Avoid input buffer memory leak.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929
+
+2012-11-03 20:38:00 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ Fix FSF address
+ https://bugzilla.gnome.org/show_bug.cgi?id=687520
+
+2012-10-22 12:08:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opuspay: remove pointless caps serialization
+ Remove the caps serialization in the rtp caps. the spec nor the receiver
+ does anything with it.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547
+
+2012-10-17 17:34:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ Use gst_element_class_set_static_metadata()
+ where possible. Avoids some string copies. Also re-indent
+ some stuff. Also some indent fixes here and there.
+
+2012-09-20 18:41:24 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: Allocate the rtp buffer correctly
+ Use the right functions to allocate the rtp buffer
+
+2012-09-14 17:08:49 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ replace gst_element_class_set_details_simple with gst_element_class_set_metadata
+
+2012-03-07 17:14:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opus: port to updated 0.11
+
+2011-12-30 11:41:17 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ Merge remote-tracking branch 'origin/master' into 0.11-premerge
+ Conflicts:
+ docs/libs/Makefile.am
+ ext/kate/gstkatetiger.c
+ ext/opus/gstopusdec.c
+ ext/xvid/gstxvidenc.c
+ gst-libs/gst/basecamerabinsrc/Makefile.am
+ gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
+ gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
+ gst-libs/gst/video/gstbasevideocodec.c
+ gst-libs/gst/video/gstbasevideocodec.h
+ gst-libs/gst/video/gstbasevideodecoder.c
+ gst-libs/gst/video/gstbasevideoencoder.c
+ gst/asfmux/gstasfmux.c
+ gst/audiovisualizers/gstwavescope.c
+ gst/camerabin2/gstcamerabin2.c
+ gst/debugutils/gstcompare.c
+ gst/frei0r/gstfrei0rmixer.c
+ gst/mpegpsmux/mpegpsmux.c
+ gst/mpegtsmux/mpegtsmux.c
+ gst/mxf/mxfmux.c
+ gst/videomeasure/gstvideomeasure_ssim.c
+ gst/videoparsers/gsth264parse.c
+ gst/videoparsers/gstmpeg4videoparse.c
+
+2011-12-09 17:25:41 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opusenc: add upstream negotiation for multistream ability
+ This will help elements that cannot deal with multistream,
+ such as the RTP payloader.
+ The caps now do not include a "streams" field anymore, but
+ a "multistream" boolean, since we have no real use for knowing
+ the exact amount of streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
+2011-12-07 15:13:11 -0200 Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ Adding opus RTP payloader/depayloader element
+ Adding OPUS RTP module based on the current draft:
+ http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt
+ https://bugzilla.gnome.org/show_bug.cgi?id=664817
+
+2016-02-17 13:26:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtputils.c:
+ * gst/rtp/gstrtputils.h:
+ rtp: h264/h265: avoid duplication of read_golomb()
+ There is no need to have two identical implementations of the read_golomb
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-17 14:37:44 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Simple implementation of TRICKMODE_KEY_UNITS
+ When the trickmode key-units flag is set on the segment, simply skip
+ any sample on a video stream that isn't a keyframe
+ https://bugzilla.gnome.org/show_bug.cgi?id=762185
+
+2015-08-21 14:15:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: send GAP events for lagging audio and video streams too
+ Send GAP events for non-subtitle streams too if they lag too much
+ behind, but use a higher threshold than for subtitles.
+ This helps with fixing prerolling with a file where one of the
+ audio streams only has data starting from 19s onwards. It's not
+ a complete fix yet, it also requires changes elsewhere, such as
+ in baseparse, to make sure caps are propagated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=614460
+ https://bugzilla.gnome.org/show_bug.cgi?id=753899
+
+2015-12-23 19:54:13 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9depay.h:
+ * gst/rtp/gstrtpvp9pay.c:
+ * gst/rtp/gstrtpvp9pay.h:
+ rtpvp9pay: rtpvp9depay: Initial implementation of draft 01
+ Quick and dirty implementation of an RTP payloader and depayloader
+ for VP9. In particalur it assumes no spatial or temporal layering,
+ non-flexible mode, and some other bits and pieces.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754773
+
+2016-02-16 09:02:30 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix string memory leak
+ codec_name is not being freed in all conditions leading to memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=762117
+
+2015-12-10 12:15:52 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: add "get-session" signal
+ This gets the GstRTPSession element, as compared to the RTPSession object
+ that is returned by get-internal-session.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759293
+
+2016-02-16 00:19:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ rtp: h265: hook up move RTP H.265 payloader/depayloader to build
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-16 00:14:27 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ rtp: h265: use common meta utility functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-05 18:18:31 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.h:
+ * gst/rtp/gstrtph265types.h:
+ rtp: h265: remove codecparser dependency from h265 payloader/depayloader
+ Looks like it just uses the NAL enums and nothing else from
+ the codecparsers, and that's the only reason it had to be
+ moved from -good to -bad when it was originally added. We
+ can probably keep those NAL enums up to date enough, so let's
+ remove the codecparser dependency so it can be moved back into
+ -good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-16 00:24:58 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ Merge branch 'plugin-move-rtp-h265'
+ Move RTP H.265 payloader/depayloader from -bad to -good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-05 15:34:51 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ gstrtph265depay: keep consistency with rtph264depay
+ Use gst_rtp_drop_meta() and the same function prototype for
+ gst_rtp_copy_meta() to keep consistency with the RTP elements in
+ gst-plugins-good
+
+2016-02-05 13:56:34 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: fix termination of access unit
+ Only consider the access unit complete when the next-occurring VCL NAL unit
+ has the first bit after its NAL unit header equal to 1.
+
+2016-01-15 16:10:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: fix unneeded sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then throw it
+ away immediately, just use the original input buffer directly.
+
+2016-01-15 15:56:59 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: add "send VPS/SPS/PPS with every key frame" mode
+ It's not enough to have timeout or event based VPS/SPS/PPS information
+ sent in RTP packets. There are some scenarios when key frames may appear
+ more frequently than once a second, in which case the minimum timeout
+ for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
+ It might also be desirable in general to make sure the VPS/SPS/PPS is
+ available with every keyframe (packet loss aside), so receivers can
+ actually pick up decoding immediately from the first keyframe if
+ VPS/SPS/PPS is not signaled out of band.
+ This commit adds the possibility to send VPS/SPS/PPS with every key frame.
+ This mode can be enabled by setting "config-interval" property to -1. In
+ this case the payloader will add VPS, SPS and PPS before every key (IDR)
+ frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2016-01-15 15:19:41 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtph265pay: change config-interval property type from uint to int
+ This way we can use -1 as special value, which is nicer than MAXUINT.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-08-15 16:22:20 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: make sure we call handle_nal for each NAL
+ Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure
+ we correctly extract the SPS and PPS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730999
+
+2015-08-15 14:45:34 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Copy metadata in the payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-15 11:41:40 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-15 11:30:36 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: fix potential crash when shutting down
+ A race condition in the state change function may cause buffers to be
+ unreffed while they are still used by the streaming thread in
+ gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
+ parent class first in the state change function to make sure streaming
+ has stopped and only then free those buffers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741381
+
+2015-08-14 15:08:08 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: fix buffer leak when using SPS/PPS
+ Fixes a buffer leak that would occur if the pipeline was shutdown while a
+ SPS/PPS header was being created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741271
+
+2015-08-14 11:49:51 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ rtph265depay: copy metadata in the depayloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-12 17:54:52 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: checking if depay has sps/pps nals before insertion
+ Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 17:22:42 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: only update the srcpad caps if something else than the codec_data changed
+ h264parse and gstrtph264depay do the same, let's keep the behaviour
+ consistent. As we now include the codec_data inside the stream, this causes
+ less caps renegotiation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 16:43:48 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: PPS replaces old PPS if it has the same id
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 16:11:00 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: Insert SPS/PPS NALs into the stream
+ rtph264depay does the same and this fixes decoding of some streams with 32
+ SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
+ but the field in the codec_data for the number of SPS or PPS is only 5
+ (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+ This looks like a mistake in the part of the spect about the codec_data.
+
+2015-08-12 15:49:50 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't need to map the
+ input buffer again but can just re-use the mapping the base class has
+ already done.
+ Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 15:14:50 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
+ Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code.
+
+2015-08-12 14:59:53 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: prevent trying to get 0 bytes from adapter
+ This causes an assertion and would lead to getting a NULL instead
+ of a buffer. Without proper checking this would easily lead to a
+ segfault.
+ Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199
+
+2015-07-29 17:29:28 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtp: remove dead assignment
+ Value set to ret will be overwritten at least once at the end of the while
+ loop, removing assignment.
+
+2015-04-24 16:48:23 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ remove unused enum items PROP_LAST
+ This were probably added to the enums due to cargo cult programming and are
+ unused.
+
+2015-03-06 14:54:41 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtp: donl_present variable unused
+ donl_present is not implemented, yet the value is set and checked a few times.
+ Cleaning this.
+ CID #1249687
+
+2015-01-08 15:36:04 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtp: value truncated too short creates dead code
+ type is truncated to 0-31 with "& 0x1f", but right after that it is checks if
+ the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and
+ GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will
+ never be True if the value is maximum 31 after the truncation.
+ The intention of the code was to truncate to 0-63.
+
+2015-01-08 15:27:44 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtp: fix nal unit type check
+ After further investigation the previous commit is wrong. The code intended to
+ check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c
+ does. Type 40 would not be complete.
+
+2015-01-08 13:47:09 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtp: fix dead code and check for impossible values
+ nal_type is the index for a GstH265NalUnitType enum. There are two types of dead
+ code here:
+ First, after checking if nal_type is >= 39 there are two OR conditionals that
+ check if the value is in ranges higher than that number, so if nal_type >= 39
+ falls in the True branch those other conditions aren't checked and if it falls
+ in the False branch and they are checked, they will always also be False. They
+ are redundant.
+ Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41
+ should never be True.
+ Removing this redundant checks.
+ CID 1249684
+
+2014-10-16 10:34:01 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtp: add h265 RTP payloader + depayloader
+
+2016-02-15 11:51:46 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tests/check/elements/rtpmux.c:
+ tests: rtpmux: Fix element memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=762057
+
+2016-02-12 20:57:29 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/monoscope.c:
+ monoscope: rework the scaling code
+ The running average was wrong and the resulting scaling factor was only held in
+ place using the CLAMP. In addtion we are now convering quickly to volume
+ changes.
+ FInally now with this change, we can change the resolution defines and
+ everythign adjusts.
+
+2016-01-28 17:00:55 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/convolve.c:
+ * gst/monoscope/monoscope.c:
+ * gst/monoscope/monoscope.h:
+ monoscope: use constants in the drawing code
+ Make all the drawing ops be based on the constants. This way we can change
+ the fixed size at least at compile time.
+
+2016-01-28 09:51:17 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/gstmonoscope.c:
+ monoscope: replace hardcoded values by constants
+ This at least establishes the relationship.
+
+2016-01-28 09:43:12 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/convolve.c:
+ * gst/monoscope/convolve.h:
+ * gst/monoscope/monoscope.c:
+ * gst/monoscope/monoscope.h:
+ monoscpe: make the convolver use dynamic memory
+ Replace all #defines with members and initialize the convolver with a parameter.
+
+2016-01-28 08:56:44 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/README:
+ monoscope: update README
+ We can already create multiple instances.
+
+2016-01-28 08:53:35 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/convolve.c:
+ * gst/monoscope/monoscope.c:
+ monoscope: code cleanup
+ Use constants more often. Cleanup comments and add more to explain how things
+ work.
+
+2016-02-08 23:41:32 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: remove check for impossible condition
+ Commit bd27a1f30b4458f2edee53c76dd07fb35904b61d added a few error handling
+ memory management checks. These check srccaps to see if it needs to be
+ unreferenced before returning, in the case of invalid_caps this goto jump
+ always happens before srccaps is set, so it will always be NULL in this
+ error label.
+ CID #1352035
+
+2016-02-08 12:48:46 +0100 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES
+ https://bugzilla.gnome.org/show_bug.cgi?id=761705
+
+2016-02-08 15:31:55 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix spelling of reenqueueing
+ To match commit 7d7074cef0272cd5155098bfc2bda6849dd89267. I love the idea
+ of aiming for the maximum number of consecutive vowels.
+
+2016-02-08 10:17:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix spelling of queueing
+ Didn't know which one to choose between queuing and queueing, so I picked
+ the one with the biggest amount of vowels in a row ;-P (both are
+ acceptable apparently)
+
+2016-02-07 15:02:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Don't pass the same data over and over
+ We already pass the entire frame to the decoder. If the decoder ask for
+ more data, don't pass the same data again as this leads to infinit loop.
+ Instead, simply fail the fill function to signal the problem with that
+ frame. It will then be skipped properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761670
+
+2016-02-08 00:10:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/lzo.c:
+ matroska: get rid of _stdint.h include
+
+2016-02-05 20:00:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/Makefile.am:
+ tests: extend the AM_TESTS_ENVIRONMENT from check.mak
+ To get the CK_DEFAULT_TIMEOUT defined for all tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=761472
+
+2016-02-05 18:04:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 86e4663 to b64f03f
+
+2016-01-30 18:43:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers
+ For APP/JPG markers the size is following and we have to skip that. This is
+ not really a problem unless the marker contains e.g. a preview JPEG or
+ something else that we might interprete as another marker.
+
+2016-01-26 22:37:30 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix framerate calculation for fragmented format
+ qtdemux calculates framerate using duration and the number of sample.
+ In case of fragmented mp4 format, however, the number of sample can
+ be figure out after parsing every moof box. Because qtdemux does not
+ parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
+ framerate calculation.
+ This patch will triger gst_qtdemux_configure_stream() for every new moof.
+ Then, framerate will be calculated by using duration and n_samples of the moof.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760774
+
+2016-01-28 22:36:23 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: handling zero segment-duration edit list
+ Based on document ISO_IEC_14496-12, edit list box can have
+ segment duration as zero. It does not imply that media_start equals to
+ media_stop. But, it just indicates a sample which should be presented
+ at the first. This patch derives segment duration using media_time
+ and duration of file. And set derived duration to segment-duration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760781
+
+2016-01-28 21:36:54 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: expose streams with first moof for fragmented format
+ In case of push mode, qtdemux expose streams after got moov box.
+ We can not guarantee that a moov box has sample data such as sample duration
+ and the number of sample in stbl box for fragmented format case.
+ So, if a moov has no sample data, streams will not be exposed until get the first moof.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760779
+
+2016-01-27 18:48:17 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS
+
+2016-01-27 18:44:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps
+ Prevents double-negotiation during startup and in some other cases.
+
+2016-01-27 16:43:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Add negotiation unit tests for all 4 modes
+ These now check the output caps based on the input caps and a following
+ capsfilter and make sure the caps are exactly as expected.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760995
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 17:39:20 +0100 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Do passthrough in auto mode if downstream only supports interlaced
+ If the following conditions are met:
+ 1) upstream and downstream caps are compatible
+ 2) upstream is interlaced
+ 3) downstream doesn't support progressive mode
+ then deinterlace will just do passthrough instead of failing to link.
+ This is done with the following scenario in mind:
+ videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
+ name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
+ queue ! deinterlace name=dein_desktop ! autovideosink
+ In this case, dein_src will do the deinterlacing. However,
+ videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
+ name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
+ queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue !
+ "video/x-raw,interlace-mode=interleaved" ! fakesink
+ In this case, caps auto-negotiation will make dein_file and dein_desktop do
+ the deinterlacing, while dein_src will be passthrough.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760995
+
+2016-01-26 18:05:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Add mode=auto-strict
+ In this mode we will passthrough all progressive caps but interlaced caps must be
+ caps where we actually support deinterlacing.
+ This is the only difference between auto and auto-strict, auto would
+ passthrough all unsupported interlaced caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 17:50:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Implement reconfiguration a bit better
+ And e.g. consider reconfiguration caused by RECONFIGURE events too.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 11:57:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Rewrite caps negotiation
+ Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind
+ of caps were last set, and e.g. if we last had interlaced caps or not. That's
+ just broken.
+ Also previously the handling of non-sysmem caps features was rather random and
+ unusuable.
+ Now the behaviour is the following, depending on the mode property:
+ 1) mode=disabled
+ Completely do passthrough of everything
+ 2) mode=interlaced
+ Only accept formats we can actually deinterlace, and accept interlaced
+ and progressive content and always run the deinterlacer and output
+ progressive content
+ 3) mode=auto (i.e. playbin)
+ Accept all progressive formats as passthrough, accept all formats that we
+ can deinterlace ourselves (which we do then), but also accept everything
+ else for which we then just passthrough. In auto mode, deinterlacing is best
+ effort: If we can, we deinterlace, if we can't we just output interlaced
+ content.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+ https://bugzilla.gnome.org/show_bug.cgi?id=760553
+
+2016-01-26 11:34:40 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Remove unused, obsolete bufferalloc code
+
+2016-01-26 18:50:38 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: use A_AAC instead of A_AAC/MPEGx/y
+ Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete
+ https://bugzilla.gnome.org/show_bug.cgi?id=761144
+
+2016-01-25 17:21:24 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/rtp/gstrtph261pay.c:
+ gst: Fix unintialized variable warnings
+ While cross-compiling with Linaro GCC 5.1-2015.08, it complained
+ about a couple unitialized variables.
+ This patch initializes them to zero.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761094
+
+2016-01-25 15:03:23 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: print potentially negative offset with a sign
+
+2016-01-21 17:41:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Re-add colorimetry field for RGB formats
+ This time, check if it's an RGB format and sets the transformation
+ matrix to identity. The rest of the colorimetry information is
+ meaningfull and shall be kept.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759624
+
+2016-01-22 10:03:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: fix sRGB colorspace definition
+ V4l2 can also use the sRGB colorspace for YUV formats and thus needs a
+ default matrix.
+
+2016-01-21 15:29:46 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/debugutils/gsttaginject.c:
+ taginject: fix sample pipeline in docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=679571
+
+2016-01-21 10:49:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Add adobe colorspace support
+ Use the new primaries and transfer function for Adobe RGB.
+ Explicitly list the colorimetry instead of using the default GStreamer
+ ones. The defaults for BT2020, for example, do not match.
+ Explicitly set the matrix of SRGB to RGB.
+
+2016-01-20 13:41:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Ensure that we always have valid frame user data before using it
+ Otherwise we're going to dereference NULL pointers.
+
+2016-01-20 10:02:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: Unref frame in all code paths of handle_frame()
+ https://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-19 22:49:20 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/vpx/gstvpxenc.c:
+ vpxenc: Unref frame on ERROR
+ All code paths for handle_frame() must somehow take ownership of the frame, be
+ it by actually unreffing, forwarding the frame elsewhere or storing it for
+ later.
+ http://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-20 18:20:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2: Don't free props structure twice.
+ gst_v4l2_device_provider_probe_device() frees the passed props
+ structure, don't free it again in the caller.
+
+2016-01-19 15:15:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Cleanup uneeded return statement
+
+2016-01-19 15:14:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't set colorimetry for non YUV formats
+ Setting colormetry in caps for RGB have no meaning, but worst it
+ confuses the converters downstream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759624
+
+2016-01-19 13:01:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpchannels.c:
+ * gst/rtp/gstrtpchannels.h:
+ rtp: fix compiler warnings with gcc-6
+ In file included from gstrtpL16depay.h:27:0,
+ from gstrtp.c:73:
+ gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
+ static const GstRTPChannelOrder channel_orders[] =
+
+2016-01-19 14:57:03 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't play anything after the end of the data chunk even when seeking
+ Especially in push mode we would completely ignore the size of the data chunk
+ when not stop position is given for the seek. Instead make sure that the end
+ offset is at most the end of the data chunk if known.
+ Without this we would output anything after the data chunk, possibly causing
+ loud noises if the media file is followed by an INFO chunk or an ID3 tag.
+
+2016-01-19 14:55:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
+ We use that to signal "infinity", taking the difference between that and some
+ other value is not going to give us any useful result for the end offsets of
+ segments.
+
+2016-01-18 11:30:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
+ This reverts commit 271501f6576de4d141e7c2f618e28b9e3b1e5b38.
+ It wasn't meant to be pushed yet as the commit message indicates.
+
+2016-01-12 14:01:21 -0800 Aleix Conchillo Flaqué <aconchillo@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
+ We check the stream profile and use the proper RTCP caps:
+ application/x-srtcp if we are using a secure profile and
+ application/x-rtcp otherwise.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760556
+
+2016-01-05 16:15:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ WIP: rtpjitterbuffer: Add RFC7273 media clock handling
+
+2016-01-15 11:36:35 +0000 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/vpx/gstvpxenc.c:
+ vp8enc: Return FLOW_ERROR when an error accures
+ FALSE would mean FLOW_OK
+ https://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-15 03:57:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: break as soon as the device is found
+ No need to loop further if there's no side-effects for it
+
+2016-01-15 03:56:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: Fix error handling when selecting/opening devices
+ Post an element error when the CoreAudio device cannot be selected or opened.
+ Also ensure that we post a GST_ERROR with more detail.
+
+2016-01-13 23:40:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: When flushing on EOS, don't process more data than the "data" size
+ Even if we have more data queued up when flushing than the size of the data
+ chunk, don't process and output it. If the data size is known, this likely
+ contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
+ outputting them as if they were data is going to cause unexpected behaviour
+ and unpleasant audio noises.
+
+2014-08-29 15:40:23 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * tests/check/pipelines/wavenc.c:
+ tests: fix a thinko in the wavenc example
+ The code is supposed to follow somehow what the comment above says, that
+ is to have one channel with a wave of freq 440 and the other channel
+ with a wave of freq 880, but an off by one error results in frequencies
+ of 0 and 440.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735673
+
+2014-08-29 15:07:58 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * gst/interleave/interleave.c:
+ interleave: Fix the example by setting channel-masks in the sink pads
+ The current example does not work, it fails with:
+ ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
+ gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
+ streaming task paused, reason not-negotiated (-4)
+ This is because negotiation with wavenc gets messed up by the missing
+ channel positions configuration.
+ The proper way to define the channel layout when using the interleave
+ element in code would be to set the channel-positions property, but
+ gst-launch-1.0 does not know how to deal with arrays; so the example
+ pipeline works around the issue by setting the channel-masks in the sink
+ pads.
+ Also fix a repetition in the deinterleave example description
+ https://bugzilla.gnome.org/show_bug.cgi?id=735673
+
+2016-01-11 16:29:55 +0000 Tim Sheridan <tim.sheridan@imgtec.com>
+
+ * gst/audioparsers/gstsbcparse.c:
+ sbcparse: Fix frame length calculation
+ SBC frame length calculation wasn't being rounded up to the nearest byte
+ (as specified in the A2DP 1.0 specification, section 12.9). This could
+ cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
+ calculated frame lengths.
+ Incorrect frame length calculation causes frame coalescing to fail, as
+ subsequent frames in the stream aren't found in the expected locations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742446
+
+2016-01-10 22:54:12 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: demote warning on wrong reserved value to fixme
+ We are likely just parsing a backward-compatible stream we
+ don't fully support.
+
+2016-01-08 16:27:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: simplify caps selection
+ The downstream caps query with a filter alraedy gives us the possible
+ intersection so there is no need to check it again with downstream
+ if it is supported. Just try to set it directly.
+
+2016-01-07 20:42:41 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: fix unnecessary sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then
+ throw it away immediately, just use the original input buffer
+ directly.
+
+2016-01-07 20:38:27 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpdvdepay.c:
+ rtpdvdepay: fix unnecessary sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then
+ throw it away immediately, just use the original input buffer
+ directly.
+
+2016-01-07 20:34:05 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpamrdepay.c:
+ rtpamrdepay: fix unnecessary sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then
+ throw it away immediately, just use the original input buffer
+ directly.
+
+2016-01-07 20:27:29 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: fix major memory leak and performance issue
+ We call gst_rtp_buffer_get_payload() which creates a sub-buffer
+ of each input buffer, just to copy over metas, and then leak it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760289
+
+2016-01-08 15:32:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/rganalysis.c:
+ rganalysis: Fix compiler warnings in the unit test
+ elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined
+ [-Werror,-Wshift-negative-value]
+ push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
+ ~~ ^
+ elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined
+ [-Werror,-Wshift-negative-value]
+ push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
+ ~~ ^
+ elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined
+ [-Werror,-Wshift-negative-value]
+ push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
+ ~~ ^
+
+2016-01-05 18:13:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: don't map buffer multiple times when parsing
+
+2016-01-07 18:20:30 +0200 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: Store subtitle stream count in the correct variable
+ And don't override the video stream count instead.
+
+2016-01-05 18:59:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/equalizer/gstiirequalizernbands.c:
+ equalizer: The child-proxy API is GObject based in 1.x
+ Not GstObject anymore.
+
+2015-05-21 17:41:12 +0200 Pablo Anton <pablo.anton@vodalys-labs.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2-*: Configuring output pool correctly for using drivers min_buffer if present.
+ Signed-off-by: Pablo Anton <pablo.anton@vodalys-labs.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=755736
+
+2015-12-31 15:46:31 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: add debug msg on CRC mismatch while validating frame header
+
+2015-12-31 16:00:49 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop unneeded braces at _parse_frame() exit
+ Additionally, drop redundant comment & line break
+
+2015-12-31 15:55:18 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: minor grammar correction
+
+2015-12-31 15:34:57 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: update URLs on pointers to online spec
+
+2015-12-31 14:40:15 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: make buffer DTS setting explicitly unconditional
+ We are setting it to PTS regardless of block_strategy
+
+2015-12-31 14:21:40 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: add actual invalid block type to warning
+ For someone that read the spec is clear the only *invalid*
+ data block type is 127. For the rest, its useful information.
+ Additionally. values 7-126 are currently reserved by the
+ spec so the situation might change in the future.
+
+2015-12-31 14:12:36 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: use shift instead of mask & comp
+ We are only interested on the first bit of the first
+ byte of the metadata block header to figure out whether
+ is marked as the last one. The shift makes it quite
+ clearer.
+
+2015-12-31 12:52:13 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: warn on wishful parsing of weird headers
+ If we get anything from 7 to 126 as type when parsing
+ a metadata block header, we are likely dealing with a
+ FLAC stream version we don't fully understand. Issue
+ a warning if so.
+ Document function assumptions regarding the passed-on
+ type while at this.
+
+2015-12-31 11:33:45 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: show meaningful info on frame CRC check
+ As CRCs are calculated for the comparition already, we
+ might as well (cheaply) inform the user how the numbers
+ differ if a missmatched pair is found.
+ While at it:
+ Rephrase candidate-frame message to make more sense
+
+2015-12-31 02:40:43 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop remaining trailing whitespace
+
+2015-12-31 02:15:06 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop superflous else clauses
+
+2015-12-31 01:09:51 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: factor out buffer time and offset resetting
+ Avoids multiple occurrences of the same resetting pattern
+
+2015-12-31 00:54:48 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: move block handling by type out of _parse_frame()
+
+2015-10-07 18:51:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: replace duplicated codes to call new base sdp apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=745880
+
+2015-12-30 12:16:56 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop redundant return statement on _header_is_valid()
+ Fix the rather vague error message while at it.
+
+2015-12-30 01:56:26 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: rework gst_flac_parse_frame_is_valid()
+ drop unnecessary nesting looking for end of frame
+
+2015-12-30 00:37:04 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: factor out context clearing routine
+
+2015-12-29 18:05:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Guard against no codec data in prores caps creation
+ CID 1346532
+
+2015-12-29 17:58:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: Initialize buffer variable to NULL
+ False positive but trivial to fix and possibly causing compiler warnings at
+ some point in the future too.
+ CID 1346535
+
+2015-07-27 15:53:26 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2deviceprovider: add properties to the device
+ Add properties to the device with exactly the same keys and sematics
+ as what pulseaudio uses as property keys.
+ Also handle the case when a device is probed manually and not through gudev.
+ https://bugzilla.gnome.org//show_bug.cgi?id=759780
+
+2015-12-25 11:41:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: Free the various buffers in GstBaseTransform::stop()
+ Previously we leaked them completely, but as they're specific to the caps
+ freeing them in stop() instead of finalize() makes most sense.
+
+2015-12-24 15:28:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.1 ===
+
+2015-12-24 14:16:21 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.1
+
+2015-12-24 13:19:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-12-24 12:22:32 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/cs.po:
+ * po/de.po:
+ * po/el.po:
+ * po/hu.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-12-21 09:57:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: drop flushes from our own offset seek
+ Prevents downstream from receiving flushes for a seek only in
+ upstream. Those seeks are only to start reading from the right
+ offset when skipping or returning to qt atoms.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758928
+
+2015-11-11 16:53:19 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Always set the channel mask for PCM streams
+ Just use the gst_audio_channel_get_fallback_mask function for now as
+ the specification is too complicated and nobody implements it.
+
+2015-12-21 11:37:26 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Fix sleep for buffer-time lower than 200000
+ https://bugzilla.gnome.org/show_bug.cgi?id=748680
+
+2015-12-21 12:31:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Use -Bsymbolic-functions if available
+ While this is more useful for libraries, some of our plugins with multiple
+ files and some internal API can also benefit from this.
+
+2015-12-18 15:34:52 +0000 William Manley <will@williammanley.net>
+
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/progressreport.h:
+ progressreport: add support for using format=buffers with do-query=false
+ This is useful for investigating and debugging pipelines which are
+ producing buffers at a slower/faster rate than you would expect.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759635
+
+2015-12-18 15:49:43 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Update formats table
+ This change add all the new RGB based format. Those format removes the
+ ambiguity with the ALPHA channel. Some other missing multiplanar format
+ has been added with some additional cleanup.
+
+2015-12-18 05:17:15 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't write invalid edit list start time.
+ Avoid writing a negative number as a large positive
+ integer in an edit list when the first_ts is smaller
+ than the first_dts - which can happen when the first
+ packet received has a PTS but no DTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759615
+
+2015-12-04 23:16:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Only update running time when it increases.
+ Don't increment running time from every buffer. The correct
+ logic to only increment when running time advances is a
+ little further down, so delete this left-over line.
+
+2015-11-18 11:01:20 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Implement prores support
+ https://bugzilla.gnome.org/show_bug.cgi?id=758258
+
+2015-11-18 16:20:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroska-demux: Play ProRes video streams
+ Generate video/x-prores caps for ProRes video streams.
+ Every frame needs an 8 byte header prepended, as described in
+ http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout
+ so do that in a post-processing callback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758258
+
+2015-12-18 10:18:09 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdec.h:
+ dvdec: Remove unused fields
+ Remove unused fields frame_len and space
+ https://bugzilla.gnome.org/show_bug.cgi?id=759614
+
+2015-12-17 16:03:04 +0100 Vincent Dehors <vincent.dehors@openwide.fr>
+
+ * gst/rtp/gstrtpj2kdepay.c:
+ rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=758943
+
+2015-12-16 11:43:58 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ * ext/raw1394/gsthdv1394src.c:
+ dv1394: log error if failed to set socket status flag
+ Log an error message if failed to set write or read socket as
+ non-blocking.
+ CID 1139608
+ CID 1139609
+
+2015-12-15 17:10:00 +0000 Dave Craig <davecraig@unbalancedaudio.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: Check for NULL return value of gst_pad_get_current_caps()
+ https://bugzilla.gnome.org/show_bug.cgi?id=759503
+
+2015-12-16 09:35:53 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ docs: update to git
+
+2015-12-15 14:27:22 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/Makefile.am:
+ vpx: Add missing headers in Makefile.am
+ This fixes distcheck.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-09-24 12:57:00 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * ext/vpx/Makefile.am:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ * ext/vpx/gstvpxenc.c:
+ * ext/vpx/gstvpxenc.h:
+ vpx: created common baseclass GstVPXEnc
+ GstVP8Enc and GstVP9Enc has almost 80% code in common.
+ created common baseclass GstVPXEnc for GstVP8Enc and GstVP9Enc
+ https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-12-15 12:57:53 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxdec.h:
+ vpxdec: Remove unneeded add video_meta
+ This also remove copies for VP8, which was not correctly in place
+ in previous related patch.
+
+2015-12-15 09:49:24 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * ext/vpx/Makefile.am:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8dec.h:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9dec.h:
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxdec.h:
+ vpx: created common base class GstVPXdec for vpx decoders
+ Base class for the vp8dec and vp9dec.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-06-10 09:17:08 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * configure.ac:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: Add GTlsInteraction property
+ https://bugzilla.gnome.org/show_bug.cgi?id=750709
+
+2015-12-14 09:05:06 -0500 Evan Callaway <evan.callaway@ipconfigure.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Retry connection if tunneling needs authentication
+ Leverage response from gst_rtsp_connection_connect_with_response to
+ determine if the connection should be retried using authentication. If
+ so, add the appropriate authentication headers based upon the response
+ and retry the connection.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749596
+
+2015-12-14 14:19:05 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: check port-range format
+ The string could exist but with a wrong format, in that case we still want
+ to reset the values of client_port_range.min and max like we do if there is
+ no string.
+ CID 1139593
+
+2015-12-14 14:55:12 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Check device property and fail if device can't be found
+ Don't use default if a specific device is set but it can't be found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759452
+
+2015-12-14 14:15:00 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Fix handling of the mute property
+ - set mute value at startup
+ - correct set and get mute functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2015-12-11 11:23:13 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Check the return value of GetStatus() too to decide if there was an error
+ If GetStatus() fails, the status itself won't be very meaningful but we also
+ have to look at its return value. This fixes blocking pipelines when removing
+ sound devices or during other errors, where we wouldn't notice the error and
+ then wait forever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734098
+
+2015-12-10 17:41:46 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ isomp4: remove unused parameters in build_*_extension
+ AtomTRAK parameter is not used by build_mov_alac_extension(),
+ build_jp2h_extension(), or build_mov_alac_extension() and can be
+ removed.
+
+2015-12-10 15:11:07 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: replace variable only used once
+ Replace has_shift variable with value since it is only use once.
+
+2015-12-09 12:24:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix packet dropping after a big discont
+ We would queue 5 consective packets before considering a reset and a proper
+ discont here. Instead of expecting the next output packet to have the current
+ seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
+ going to drop all queued up packets.
+
+2015-12-09 11:49:02 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/interleave/interleave.h:
+ interleave: Remove unsed field
+ Remove unused field collect_event in interleave.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759226
+
+2015-12-07 16:33:14 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Stop pushing data as soon as possible in push-mode
+ When working in push-mode, we attempt to push out everything currently
+ buffered in the adapter.
+ This has two pitfalls:
+ * We could stop earlier (the moment we get a non-ok or non-not-linked)
+ * We return the last combined flow return, which might be completely
+ different from the previous combined flow return
+
+2015-12-07 09:08:09 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b319909 to 86e4663
+
+2015-12-07 14:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add a warning if an empty RTCP packet is tried to be sent
+ https://bugzilla.gnome.org/show_bug.cgi?id=759119
+
+2015-11-30 19:20:13 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8dec.h:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9dec.h:
+ vpxdec: Use GstMemory to avoid copies
+ With the VPX decoders it's not simple to use downstream buffer pool,
+ because we don't know the image size and alignment when buffers get
+ allocated. We can though use GstAllocator (for downstream, or the system
+ allocator) to avoid a copy before pushing if downstream supports
+ GstVideoMeta. This would still cause a copy for sink that requires
+ specialized memory and does not have a GstAllocator for that, though
+ it will greatly improve performance for sink like glimagesink and
+ cluttersink. To avoid allocating for every buffer, we also use a
+ internal buffer pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745372
+
+2015-11-30 08:42:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Avoid over-skipping when checking LOAS config
+ There might be multiple LOAS config in a row in a full frame. The first
+ one might be a multi-layer config (which we can't properly parse yet)...
+ but then followed by a valid (single-layer) one.
+ The code was previously skipping whole frames (instead of just the LOAS
+ config we failed to read) resulting in multiple frames (seen up to 6s in
+ some situation) being dropped before finally getting the configuration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758826
+
+2015-11-25 17:08:56 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Properly set SPARSE stream flags for subpicture/subtitle
+ And while we're at it, also detect 'DXSA' as being a variant fourcc
+ of 'DXSB' for XSUB
+
+2015-11-30 21:23:52 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: grammar fix
+
+2015-11-30 21:01:17 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: switch shoutcast stream provider
+ Fixes failing ICY test. Previous provider has
+ streaming disabled outside UK.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758114
+
+2015-11-18 16:10:11 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/avi/gstavimux.c:
+ avimux: don't crash if we never got audio caps before stopping
+ auds.blockalign is set once the first caps arrive. If
+ gst_avi_mux_stop_file() is called before this happens then auds.blockalign
+ is zero and gst_avi_mux_audsink_set_fields() cause a crash:
+ [...]
+ avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign;
+ [...]
+ https://bugzilla.gnome.org/show_bug.cgi?id=758912
+
+2015-12-01 18:20:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: don't block when resurecting a buffer
+ When we are resurecting a buffer, don't block. instead let us copy a
+ buffer.
+
+2015-12-01 00:30:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: remove extra variable to improve readability
+ Makes it easier to see that the event is being replaced/unrefed
+
+2015-12-01 00:22:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: respect seqnum in seek events
+ Propagate the original seek seqnum to events originated from
+ seeking to make sure they have the same value
+
+2015-12-01 00:03:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: flush upstream when seeking in pull mode
+ Makes sure upstream will unblock and return the thread so that
+ seeking can continue
+ https://bugzilla.gnome.org/show_bug.cgi?id=758861
+
+2015-11-27 09:27:29 +0100 Anton Bondarenko <antonbo@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: add "send SPS/PPS with every key frame" mode
+ It's not enough to have timeout or event based SPS/PPS information sent
+ in RTP packets. There are some scenarios when key frames may appear
+ more frequently than once a second, in which case the minimum timeout
+ for "config-interval" of 1 second for sending SPS/PPS is not sufficient.
+ It might also be desirable in general to make sure the SPS/PPS is
+ available with every keyframe (packet loss aside), so receivers can
+ actually pick up decoding immediately from the first keyframe if
+ SPS/PPS is not signaled out of band.
+ This patch adds the possibility to send SPS/PPS with every key frame. This
+ mode can be enabled by setting "config-interval" property to -1. In this
+ case the payloader will add SPS and PPS before every key (IDR) frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-11-27 09:03:51 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtph264pay: change config-interval property type from uint to int
+ This way we can use -1 as special value, which is nicer than MAXUINT.
+ This is backwards compatible even with the GValue API, as shown by
+ a unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-11-26 21:46:11 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for Opus
+ Add support for demuxing Opus encapsulated in MP4 files, based on the
+ following spec: https://www.opus-codec.org/docs/opus_in_isobmff.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=742643
+
+2015-11-25 22:48:32 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: use macro for codec_name
+ Use _codec() macro instead of duplicating code.
+
+2015-03-25 16:32:55 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: videodec: choose format from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=733827
+
+2015-03-27 15:02:33 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: add gst_v4l2_object_probe_caps
+ Add a variant of gst_v4l2_object_get_caps that bypasses the probed_caps cache.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733827
+
+2015-11-19 17:20:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2-probe: Skip devices without supported formats
+
+2015-11-13 12:35:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ * sys/v4l2/gstv4l2.c:
+ v4l2: Track /dev/video* to triggered required probe
+ If something in /dev/video* get added, removed or replaced, we need to
+ probe the devices again in order to ensure the dynamic devices are up to
+ date.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758085
+
+2015-11-25 14:51:40 +1100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: rtpsession: don't send empty RTCP packets
+ generate_rtcp can produce empty packets when reduced size RTCP is turned on.
+ Skip them since it doesn't make sense to push them and they cause errors with
+ elements that expect RTCP packets to contain data (like srtpenc).
+
+2015-11-24 10:57:28 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: restore the segment on case of soft reset
+ When seeking back to restore the mdat position a flush is pushed
+ through and it resets downstream segment information. Make sure
+ that after the flush (that does a soft reset) a segment will
+ be pushed again
+ Fixes regressions spotted at
+ https://ci.gstreamer.net/job/GStreamer-master-validate/2100/
+
+2015-11-20 12:44:22 +0000 Graham Leggett <minfrin@sharp.fm>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: fix spelling of variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=758390
+
+2015-11-20 11:05:51 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: unite duplicate FourCC
+ Unite in fourcc.h the FourCCs that are used twice or more in qtdemux
+
+2015-11-19 15:33:45 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Fix capture/output-io-mode properties
+ There was some miss-match in the implementation. This makes it
+ concistent, though functionally it worked, except the video decoder
+ output-io-mode getter.
+
+2015-11-19 19:48:06 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ atoms: remove unused argument of build_mov_wave_extension()
+ AtomTrak * trak argument of build_move_wave_extension() isn't used.
+ Removing it.
+
+2015-11-19 19:28:20 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: remove duplicate FourCC
+ Use the available FourCCs in fourcc.h instead of duplicating them.
+
+2015-11-19 18:36:39 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ isomp4: centralize all FourCC
+ 10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c
+ already exist in fourcc.h. Don't duplicate these and use them directly.
+ Plus moving 6 to fourcc.h, to centralize them all.
+
+2015-11-19 17:32:12 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/matroska/webm-mux.c:
+ matroska/webmmux: fix outdated example launch lines
+ Update gst-launch-0.10 lines to gst-launch-1.0
+
+2015-11-16 13:26:50 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ isomp4: add support for Opus in mp4mpux
+ Add support for muxing MP4 files containing Opus. Based on the spec
+ detailed here:
+ https://www.opus-codec.org/docs/opus_in_isobmff.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=742643
+
+2015-11-18 19:10:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Replace tabs with spaces
+
+2015-11-18 19:07:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers
+ This fixes seeking if the first entries in the samples table are negative. The
+ binary search would always fail on this as the array would not be sorted if
+ interpreting the negative numbers as huge positive numbers. This caused us to
+ always output buffers from the beginning after a seek instead of close to the
+ seek position.
+ Also add a case to the comparison function for equality.
+
+2015-11-18 16:01:48 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: remove duplicate check
+ We want 1 or 2 streamheaders, the check if (bufarr->len != 1 &&
+ bufarr->len != 2) is enough. Not need to check if bufarr->len is <= 0 or
+ > 255.
+
+2015-11-18 14:48:36 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Fix error leak and handle error
+ g_thread_try_new allows for possiblity of failures. In case it fails,
+ error is not handled and leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758260
+
+2015-11-15 17:16:29 -0800 Josep Torra <n770galaxy@gmail.com>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ rtpgstdepay: Properly handle backward compat for event deserialization
+ Actual code is checking for a NULL terminator and a ';' terminator,
+ for backward compat, in a chained way that cause all events being rejected.
+ The proper condition is to reject the events when terminator isn't
+ in ['\0', ';'] set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758151
+
+2015-11-15 17:11:02 -0800 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: Test for handling of custom events in rtpgst
+ Add a simple test that checks proper serialization/deserialization
+ of custom events with rtpgstpay and rtpgstdepay.
+
+2015-11-16 16:23:43 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vpxdec: Use threads on multi-core systems
+ This adds an automatic mode to the threads property of vpxdec in order to
+ use as many threads as there is CPU on the platform. This brings back
+ GStreamer VPX decoding performance closer to what is achieved by other
+ players, including Chromium.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758195
+
+2015-11-16 10:58:32 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only send initial gaps for non-fragmented streams
+ It would be unusual to have the header segment with an 'edts' atom
+ indicating gaps at the beginning when handling fragmented streams.
+ The header usually doesn't contain any timestamping information, this
+ should come from the playlist/manifest and the segments with media
+ in those scenarios.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758171
+
+2015-11-17 09:41:34 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "Revert "qtdemux: respect qt segments in push-mode for empty starts""
+ This reverts commit d842ff288a9d01214a046becbfd9cbff3a4acea0.
+ This was reverted by accident
+
+2015-11-17 12:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: Add "loop" property for enabling/disabling multicast loopback
+ On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it
+ is a setting for the receiver socket. As such we will need it on udpsrc too to
+ allow filtering out our own multicast packets.
+
+2015-11-16 13:52:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "qtdemux: respect qt segments in push-mode for empty starts"
+ This reverts commit 142d8e2d23e5602e7382977af1043d621625f8c8.
+
+2015-11-16 16:56:04 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix string memory leak
+ The string got using g_strdup_printf will be allocated memory
+ and should be freed after use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758161
+
+2015-11-14 21:51:11 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2/object: remove unnecessary NULL check before g_free()
+
+2015-11-14 21:45:29 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/oss/gstosssrc.c:
+ osssrc: remove unnecessary NULL check before g_free()
+
+2015-11-14 21:43:24 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/sunaudio/gstsunaudiosrc.c:
+ sunaudiosrc: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:36:30 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:31:08 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:26:21 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska/read-common: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:43:10 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4/atoms: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:35:54 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ rtp/theorapay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:33:54 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtp/vorbispay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:31:34 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtp/jpegpay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:27:04 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:22:09 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:14:25 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: remove unnecessary NULL check before g_free()
+
+2015-11-14 20:09:54 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstop.c:
+ effectv/optv: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:05:03 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstshagadelic.c:
+ effectv/shagadelictv: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:01:43 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstripple.c:
+ effectv/ripple: remove unnecessary NULL checks before g_free()
+
+2015-11-14 19:56:57 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstradioac.c:
+ effectv/radioac: remove unnecessary NULL checks before g_free()
+
+2015-11-14 19:52:12 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gststreak.c:
+ effectv/streak: remove unnecessary NULL check before g_free()
+
+2015-11-14 17:04:55 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: remove unnecessary NULL checks before g_free()
+
+2015-11-14 16:57:13 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: remove unnecessary NULL check before g_free()
+
+2015-11-14 16:54:42 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: remove unnecessary NULL check before g_free()
+
+2015-11-14 16:20:33 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: remove unnecessary NULL checks before g_free()
+
+2015-11-13 13:34:02 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: add support of NV16, NV61 and NV24 formats
+ Mapped respectively to V4L2_PIX_FMT_NV16/V4L2_PIX_FMT_NV16M,
+ V4L2_PIX_FMT_NV61,V4L2_PIX_FMT_NV61M and V4L2_PIX_FMT_NV24 v4l2 formats.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758058
+
+2015-11-11 14:10:53 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxpartreader: Fix GCond leak
+ inactive_cond is not being cleared resulting in memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757924
+
+2015-08-06 12:44:20 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: fix output state memory leak
+ When jpeg_finish_decompress is called, output state reference is being created.
+ But if there is any failures in finishing decompress, it jumps to setjmp,
+ and at that point state was not referenced. Resulting in leak of output state.
+ Hence adding another setjmp after output state is referenced.
+ Similarly adding another setjmp to unmap the frame in case error happens before
+ finish_decompress
+ https://bugzilla.gnome.org/show_bug.cgi?id=753087
+
+2015-08-10 11:23:45 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: respect qt segments in push-mode for empty starts
+ In push-mode it is hard to support qt segments overall but it is
+ possible to support when the file isn't heavily edited but just contain
+ a segment to indicate a gap at the beginning. This also allows properly
+ timestamping data that has negative DTS in push-mode.
+ It is relevant to support those for 2 scenarios:
+ 1) fragmented streaming
+ 2) HTTP playback of 'regular' mp4
+ https://bugzilla.gnome.org/show_bug.cgi?id=753484
+
+2015-11-05 18:39:33 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ pulse: Don't leak caps and structures in the device provider
+
+2015-11-04 19:01:20 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: Document properties that are expressed in bits per second
+ This changed in 928cd110bcea5d143cab3ea747991851d52ecbad and
+ 73c0c2920f9aca96982a4de0c20b3417aa148b81 but was not documented.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-11-04 18:51:32 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: Trivial gst-indent fixes
+
+2015-08-12 13:35:40 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: support for cenc auxiliary info parsing outside of moof box
+ When the cenc aux info index is out of moof boundaries, keep track of
+ it and parse the beginning of the mdat box, before the first sample.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755614
+
+2015-11-03 20:33:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Use codecutils helpers for creating Opus caps
+ Also fix up codec data with values from the container.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-11-03 14:51:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: There is no multistream field for Opus anymore
+ https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-11-03 12:42:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/webm-mux.c:
+ matroska/webmmux: Support Opus in webmmux and VP9 in matroskamux
+ https://bugzilla.gnome.org/show_bug.cgi?id=729950
+
+2015-11-03 12:40:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Parse and handle CodecDelay, SeekPreroll and DiscardPadding
+ https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 12:18:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Write CodecDelay, DiscardPadding and SeekPreroll for Opus
+ And also adjust timestamps and durations according to the codec delay, both
+ should include it for whatever reason.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 11:49:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Opus headers are not in-band
+ https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 22:01:07 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2: Set O_CLOEXEC on the device fd
+ This is needed to make sure that child processes don't inherit the video
+ device fd which can cause problems with some drivers.
+
+2015-11-03 14:46:30 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpmanager: switch G_GINT64_FORMAT for GST_STIME_ARGS
+ No need to use G_GINT64_FORMAT for potentially negative values of
+ GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
+ Plus it creates more readable values in the logs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757480
+
+2015-11-03 14:26:29 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff
+ No need to manually handle negative values of diff, GST_STIME_ARGS does
+ exactly this.
+
+2015-11-02 16:53:15 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: use GST_STIME_ARGS for GstClockTimeDiff
+ No need to manually handle negative values of diff, GST_STIME_ARGS does
+ exactly this.
+
+2015-11-02 16:43:46 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: use GST_STIME_ARGS for GstClockTimeDiff
+ No need to manually handle negative values of diff, GST_STIME_ARGS is
+ available for this.
+
+2015-10-30 10:05:37 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audiochebband.c:
+ audiochebband: Fix typo in example pipeline
+ Fix typo in example pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757340
+
+2015-10-28 23:47:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2: fix double-unref in the v4l2 device provider
+
+2015-10-27 10:48:00 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-ids.c:
+ matroskamux: don't drop JPEG frames that only have PTS but no DTS set
+ For the MS/VfW codec ids, we want to write DTS timestamps instead
+ of PTS because that's what everyone else seems to do (and it's also
+ how it is in AVI). So for those input formats we use the buffer DTS
+ instead of the PTS. However, if there's no DTS set but only the PTS
+ then just take the PTS instead of dropping the input buffer. This
+ is useful especially for I-frame only codecs like JPEG and huffyuv,
+ but should also be fine as fallback in general.
+ Fixes regression with input JPEG frames that only have PTS set on them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756967
+
+2015-10-24 23:57:38 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/splitmux.c:
+ tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
+ https://bugzilla.gnome.org/show_bug.cgi?id=753622
+
+2015-10-24 23:57:29 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: do not destroy the multiqueue & muxer when going to NULL
+ Instead, delay it until all request pads have been released. This is
+ because the release_pad() vfunc requires the multiqueue and muxer to
+ be there in order to release their request pads as well. If those
+ elements are destroyed earlier, release_pad() does not work, no
+ pads are released and some resources are leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753622
+
+2015-10-20 15:28:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Read buffer timestamp *after* actually setting it
+ https://bugzilla.gnome.org/show_bug.cgi?id=756809
+
+2015-10-24 17:14:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ * gst/audiofx/gstscaletempo.h:
+ scaletempo: Fix handling of rate < 0
+ We have to reverse all samples in a buffer before processing them to properly
+ have continuous data from one buffer to another. As a result we will have a
+ negative applied rate and a rate of 1.0.
+ Also make sure that input buffers are correctly clipped to the segment,
+ otherwise our calculations are going to go wrong.
+ Also copy over the segment event's sequence number to the output segment while
+ we're at it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757033
+
+2015-10-19 18:04:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: break as soon as non-interlaced if found
+ It looks for a non-interlaced entry on the filter caps, break
+ as soon as one is found to avoid wasting cpu
+
+2015-10-19 17:50:28 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: implement accept-caps
+ Implement accept-caps handler to avoid doing a full caps query
+ downstream to handle it.
+ This commit implements accept-caps as a simplification of the _getcaps
+ function, so it exposes the same limitations that getcaps would.
+ For example, not accepting renegotiation to caps with capsfeatures when
+ it was last configured to a caps that it has to deinterlace.
+
+2015-10-19 17:06:28 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/deinterlace.c:
+ tests: deinterlace: fix small typo in comment
+
+2015-10-26 00:41:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/files/Makefile.am:
+ check: Dist splitvideo0[012].ogg test files.
+
+2015-10-23 20:16:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ * gst/audiofx/gstscaletempo.h:
+ scaletempo: Add support for F64
+
+2015-10-22 17:40:38 -0700 Mischa Spiegelmock <mspiegelmock@gmail.com>
+
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * gst/multipart/multipartdemux.c:
+ * gst/rtp/README:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/udp/gstudpsrc.c:
+ docs: Minor fixes in various places
+ https://bugzilla.gnome.org/show_bug.cgi?id=756996
+
+2015-10-21 17:43:31 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/goom/plugin_info.c:
+ goom: remove compiler trick
+ After commit 2cb6cfed22166b262ae50cb58f3ff11dd8ba91f9 there is no need to
+ trick the compiler anymore about the usage of variable cpuFlavour.
+
+2015-10-21 14:35:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b99800a to b319909
+
+2015-10-21 17:41:38 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audiofxbaseiirfilter.h:
+ audiofx: remove unused variable
+ Remove unsued variable have_coeffs in audiofxbaseiirfilter
+ https://bugzilla.gnome.org/show_bug.cgi?id=756905
+
+2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Use new GST_ENABLE_EXTRA_CHECKS #define
+ https://bugzilla.gnome.org/show_bug.cgi?id=756870
+
+2015-10-21 14:25:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 9aed1d7 to b99800a
+
+2015-10-21 11:53:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: relax creation time parsing
+ Parse wrong timestamps like we used to write as well,
+ e.g. 10:9:42, and the hour might be without a leading
+ zero in any case.
+
+2015-10-21 11:45:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix indentation
+
+2015-10-21 11:44:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: extract both creation date and time
+ Before we only extracted the date part.
+
+2015-10-21 11:16:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: fix writing of creation time
+ Don't write time as e.g. 11:9:42
+
+2015-10-13 12:42:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: update fragment offset
+ It was always being set to 0, making the resulting stream broken
+ for the receiver
+ https://bugzilla.gnome.org/show_bug.cgi?id=756422
+
+2015-10-19 15:36:37 +0300 Ryan Hendrickson <ryan.hendrickson@alum.mit.edu>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't unconditionally use strnlen()
+ It's not available on older OSX and we can as well use memchr() here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756154
+
+2015-10-19 17:38:32 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/auparse/gstauparse.c:
+ auparse: Fix event memory leak
+ Free the event after being handled to prevent memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756799
+
+2015-10-19 09:14:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: unify raw audio caps into a single caps structure
+
+2015-10-14 15:42:50 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for FFV1 coded streams in mov
+ https://bugzilla.gnome.org/show_bug.cgi?id=752495
+
+2015-10-14 15:53:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: EOS immediately if we have an empty seek segment
+ https://bugzilla.gnome.org/show_bug.cgi?id=748316
+
+2015-10-14 10:43:19 +0300 Stavros Vagionitis <stavrosv@digisoft.tv>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Make non-inclusive segment boundaries inclusive
+ The problem is that the filesrc and souphttpsrc are behaving
+ differently regarding the calculation of the segment boundaries. The
+ filesrc is using a non-inclusive boundaries, while the souphttpsrc
+ uses inclusive. Currently the hlsdemux calculates the boundaries as
+ inclusive, so for this reason there is no problem with the souphttpsrc,
+ but there is an issue in the filesrc.
+ The GstSegment is non-inclusive, so the proposed solution is to use
+ non-inclusive boundaries in the hlsdemux in order to be consistent.
+ Make the change in the hlsdemux, will break the souphttpsrc, which
+ will expect inclusive boundaries, but the hlsdemux will offer
+ non-inclusive. This change makes sure that the non-inclusive
+ boundaries are converted to inclusive.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748316
+
+2015-10-11 22:07:54 +0000 Graham Leggett <minfrin@sharp.fm>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpclientsink.h:
+ souphttpclientsink: Add the retry and retry-delay properties
+ These allow a failed request to be retried after the given number of seconds
+ instead of failing the pipeline. Take account of the Retry-After header if
+ present. Add retries parameter that controls the number of times an HTTP
+ request will be retried before failing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756318
+
+2015-10-14 12:03:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix caps leak
+ If the QtDemuxStream are re-used they may already have caps which used
+ to be leaked.
+ Reproduced using the
+ validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
+ scenario.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756561
+
+2015-10-14 09:29:50 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix taglist memory leak
+ Free the stream and its sub items instead of just the stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=756544
+
+2015-10-11 12:06:26 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Allow negotiating to S8 as a raw format but stop making it best choice
+ Negotiation to audio/x-raw,format=S8 was not possible because S8 does
+ not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
+ https://bugzilla.gnome.org/show_bug.cgi?id=756387
+
+2015-10-11 09:18:40 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Add prores support
+ https://bugzilla.gnome.org/show_bug.cgi?id=756388
+
+2015-10-12 18:56:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: add GST_PLUGINS_BASE_LIBS for flvdemux check
+ So it pulls in the right libgsttag-1.0.
+
+2015-10-11 22:27:47 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * gst/goom/Makefile.am:
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom/gstgoom.h:
+ * gst/goom2k1/Makefile.am:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ * gst/goom2k1/gstgoom.h:
+ goom/goom2k1: remove obsolete left over files
+ They now use the new GstAudioVisualizer base class
+ from gst-plugins-base/gst-libs/gst/pbutils
+ Also fixed undefined reference to gst_audio_visualizer_get_type
+ Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-12 10:48:23 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: Fix buffer memory leak during failures
+ mapped buffer is not being unmapped during failures
+ https://bugzilla.gnome.org/show_bug.cgi?id=756231
+
+2015-10-12 11:18:51 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Check if soup message is created
+ If soup message is not created then the same should not be passed
+ on, which is resulting in segfault. Hence throwing a warning message
+ and returning
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:15:15 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Check if location being set is valid
+ Adding a check in set_property to find if the location uri is valid
+ and printing warning if not valid.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:09:30 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Fix memory leaks during failures
+ freeing streamheader_buffers and sent_buffers during failure cases.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:03:17 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Replace redundant free_buffer_list function
+ Removing free_buffer_list and replacing it with already available function
+ g_list_free_full
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-11 16:40:01 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/Makefile.am:
+ check: Don't forget base CFLAGS for flvdemux check
+ elements/flvdemux.c:25:25: fatal error: gst/tag/tag.h: No such file or directory
+
+2015-10-11 11:37:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/ebml-write.c:
+ * gst/matroska/ebml-write.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: Create a TIME segment when creating streamable output
+ Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which
+ does the same for flvmux.
+
+2015-09-23 13:50:52 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/Makefile.am:
+ * gst/flv/gstflvdemux.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/flvdemux.c:
+ flvdemux: output speex vorbiscomment as a GstTagList
+ This is what speexdec expects.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755478
+
+2015-09-22 22:59:16 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ * tests/check/elements/flvmux.c:
+ flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
+ Or else flvdemux don't understand it
+ https://bugzilla.gnome.org/show_bug.cgi?id=754435
+
+2015-09-02 10:44:59 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ * tests/check/elements/flvmux.c:
+ flvmux: use time segment and copy timestamps when streamable
+ Add a basic test using speex data to verify timestamping.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754435
+
+2015-09-23 13:14:03 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: speex is also always 16KHz
+ This is just a cosmetic change for the logs, since the right caps
+ for Speex is being set elsewhere.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755479
+
+2015-07-14 15:19:44 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: Add 'source-stats' to stats and notify
+ Add statitics from each rtp source to the rtp session property.
+ 'source-stats' is a GValueArray where each element is a GstStructure of
+ stats for one rtp source.
+ The availability of new stats is signaled via g_object_notify.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752669
+
+2015-06-05 17:20:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Implement sending of reduced size RTCP packets
+ https://bugzilla.gnome.org/show_bug.cgi?id=750456
+
+2015-10-08 15:01:13 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audiodynamic.h:
+ audiofx: Remove unused variable
+ Remove unused variable 'degree' in audiodynamic
+ https://bugzilla.gnome.org/show_bug.cgi?id=756234
+
+2015-10-08 14:44:07 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix memory leak for corrupted file
+ Free brands before overriding them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756226
+
+2015-10-08 11:44:04 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ gdkpixbufdec: Fix pixbuf_loader leak during failures
+ https://bugzilla.gnome.org/show_bug.cgi?id=756219
+
+2015-10-07 23:23:45 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Add missing break
+
+2015-10-07 13:03:02 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-10-07 13:02:12 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpmanager: add "max-dropout-time" and "max-misorder-time" props
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-10-07 17:14:57 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix date memory leak
+ When getting date from taglist, the memory should be freed after
+ using it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756171
+
+2015-10-05 11:03:38 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix sample memory leak
+ When getting sample from taglist, the memory should be freed after
+ using it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756068
+
+2015-10-05 13:10:56 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/cutter/gstcutter.c:
+ cutter: Fix buffer leak
+ Buffer is added to the internal cache, and pushed only when accumulated
+ buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
+ is not freed. Freeing the cache when the state changes from PAUSED to READY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754212
+
+2015-08-31 21:10:16 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Use default upstream event handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-08-31 21:05:03 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set
+ https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-07-22 09:47:22 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ * tests/check/elements/rtpmux.c:
+ gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
+ By not doing this, the muxer is not effectively a rtpmuxer, rather a
+ funnel, since it should be a single stream that exists the muxer.
+ If not specified, take the first ssrc seen on a sinkpad, allowing upstream
+ to decide ssrc in "passthrough" with only one sinkpad.
+ Also, let downstream ssrc overrule internal configured one
+ We hence has the following order for determining the ssrc used by
+ rtpmux:
+ 0. Suggestion from GstRTPCollision event
+ 1. Downstream caps
+ 2. ssrc-Property
+ 3. (First) upstream caps containing ssrc
+ 4. Randomly generated
+ https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-10-02 22:42:20 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fixup last commit
+
+2015-10-02 22:21:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * gst/udp/gstudpsrc.c:
+ Update GLib dependency to 2.40.0
+
+2015-06-30 16:56:19 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpstats: add utility for calculating RTP packet rate
+
+2015-08-10 18:14:39 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: handle empty segments in seeking adjust
+ If seeking targets an empty segment skip it as there is no media
+ offset to get from it. Instead look for the next one.
+ This doesn't make seeking in push-mode work if you seek to an
+ empty segment but at least won't get you to wrong offsets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753484
+
+2015-04-17 14:25:43 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: post messages when fragments are being opened and closed
+ This can be useful for applications that need to track the created fragments
+ (to log them in a recording database, for example)
+ https://bugzilla.gnome.org/show_bug.cgi?id=750108
+
+2015-04-29 18:23:28 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: allow non-video streams to serve as reference
+ In the absence of a video stream, the first stream will be used as
+ reference.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753617
+
+2015-07-22 17:45:12 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: initialize mux_start_time properly
+ mux_start_time refers to the running_time of the buffer
+ that goes first in the output file. Normally this time is
+ 0, so this variable is initialized to 0 during the state
+ change to PAUSED.
+ However, when dealing with dynamic pipelines and starting
+ a recording while the pipeline has already run for a while,
+ the running_time of the first buffer is > 0 and this causes
+ a problem with detecting the end of the first file(s) when
+ splitting by duration, because the code will later compare
+ the threshold_time with (last buffer running_time - mux_start_time)
+ and will get it wrong until mux_start_time advances enough
+ to make this difference < threshold_time, creating empty files
+ in the meantime.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753624
+
+2015-09-16 16:03:02 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Reverse playback does not consider segment.start
+ During reverse playback, the media should stop playing at segment.start
+ This does not happen, and avidemux continues to process data even when
+ current timestamp is less that segment.start.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755094
+
+2015-09-23 12:39:35 +0900 Manasa Athreya <manasa.athreya@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream
+ If stream has more than one trex box which is not matched to actual
+ track id, it makes qtdemux crashed.
+ Author : Manasa Athreya (manasa.athreya@lge.com)
+ https://bugzilla.gnome.org/show_bug.cgi?id=754864
+
+2015-09-04 14:24:45 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: get size, stride info using VideoInfo
+ Use VideoInfo data to get size stride and
+ offset, instead of hard coded macros.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754558
+
+2015-09-04 14:18:50 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: free mask
+ Free the memory allocated to 'mask' to avoid
+ memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754555
+
+2015-08-20 11:02:58 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ * tests/icles/equalizer-test.c:
+ * tests/icles/gdkpixbufoverlay-test.c:
+ * tests/icles/gdkpixbufsink-test.c:
+ * tests/icles/test-oss4.c:
+ * tests/icles/videocrop-test.c:
+ gstreamer: good: tests: Fix memory leaks when context parse fails.
+ When g_option_context_parse fails, context and error variables are not getting free'd
+ which results in memory leaks. Free'ing the same.
+ And replacing g_error_free with g_clear_error, which checks if the error being passed
+ https://bugzilla.gnome.org/show_bug.cgi?id=753853
+
+2015-10-02 16:18:15 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: doesn't handle probation and rtp gap in case of sender
+ https://bugzilla.gnome.org/show_bug.cgi?id=754548
+
+2015-10-02 16:16:32 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpsession.h:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals
+ Allows for applications to get internal source's RTP statistics.
+ (eg. sender sources for a server/client)
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-10-02 14:17:48 +1000 Jan Schmidt <jan@centricular.com>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Gather and coalesce all damaged areas before retrieving.
+ These days the xserver seems to give us the same damage regions
+ over and over for entire windows, and we retrieve them multiple
+ times, which gives time for more damage to appear. Instead, just
+ quickly gather all damaged areas into a region list and copy
+ out once.
+
+2015-10-01 16:24:32 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/goom2k1/Makefile.am:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: use the new audiovisualizer base class
+ Rebase to have goom using the GstAudioVisualizer base class in
+ gst-plugins-base/gst-libs/gst/pbutils
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-01 16:16:08 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/goom/Makefile.am:
+ * gst/goom/gstgoom.h:
+ goom: use the new audiovisualizer base class
+ Rebase to have goom using the GstAudioVisualizer base class in
+ gst-plugins-base/gst-libs/gst/pbutils
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-09-30 17:35:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ * tests/check/elements/deinterleave.c:
+ deinterleave: implement accept-caps
+ Avoid using default accept-caps handler that will query downstream
+ and is more expensive. Just check if the caps is compatible with
+ the template and check if the channels are the same.
+
+2015-09-30 09:35:39 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/deinterleave.c:
+ tests: deinterleave: also check for caps query results
+
+2015-09-30 12:30:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: use the caps query filter
+ It was being ignored and would lead to wrong results if the
+ element doing the query would rely on the intersection being made.
+
+2015-09-30 10:00:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: implement a caps query handler for the sinkpad
+ It was missing and apparently code relied on having it there
+ for not allowing a change in the number of channels
+
+2015-09-30 09:05:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: fix caps leak
+ Caps from the pad template are being leaked. In any case it is
+ from a static pad template and will 'leak' in the end, just doing
+ the cleanup for the good practice.
+
+2015-09-29 11:15:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/gdkpixbufoverlay.c:
+ tests: gdkpixbufoverlay: add minimal unit test
+ https://bugzilla.gnome.org/show_bug.cgi?id=755773
+
+2015-09-29 11:12:48 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufsink: don't leak old pixel buffer when setting a new overlay
+ https://bugzilla.gnome.org/show_bug.cgi?id=755773
+
+2015-09-28 20:25:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: avoid potential string overflow
+ We don't necessarily have full control over the input tags, so
+ it's possible that the ISRC tag contains a longer string than
+ expected, in which case we'd write over the end of the static-size
+ 13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
+ Make sure to only copy the ISRC if it's not too long, and make
+ sure the buffer we write to is always NUL-terminated by using
+ g_strlcpy().
+ CID 1324931.
+
+2015-09-28 18:03:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Remove leftover assertion from 0.10
+ We now allocate memory via GstAllocator and as such can handle arbitrary
+ alignments, not only <= G_MEM_ALIGN.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755708
+
+2015-09-25 10:01:37 +0200 Guillaume Marquebielle <guillaume.marquebielle@parrot.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix uninitialized variables in LOAS config reading
+ On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
+ "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
+ 'channels' are pointing to uninitialized values can end on setting random values as
+ rate and channels on src caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755611
+
+2015-09-18 00:58:23 +1000 Jan Schmidt <thaytan@noraisin.net>
+
+ * ext/gdk_pixbuf/gstgdkpixbufsink.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ Fix some compiler warnings when building with G_DISABLE_ASSERT
+ Touches rtpmanager and gdkpixbufsink
+
+2015-08-18 14:30:57 +0100 Chris Bass <floobleflam@gmail.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: support timed-text subtitle tracks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752818
+
+2015-09-26 00:12:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ gst: Don't use deprecated gst_segment_to_position()
+
+2015-09-21 13:47:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
+ https://bugzilla.gnome.org/show_bug.cgi?id=755125
+
+2015-09-16 19:28:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpbin/session: Allow RTCP sync to happen based on capture time or send time
+ Send time is the previous behaviour and the default, but there are use cases
+ where you want to synchronize based on the capture time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755125
+
+2015-09-25 23:51:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:15:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.6.0
+
+2015-09-25 22:57:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-09-25 14:08:09 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/smpte/gstsmptealpha.c:
+ smptealpha: Do not set width/height before comparing with old values
+ Otherwise we end up considering the values did not change and we wrongly
+ work with the old video format (which will lead to wrong
+ behaviour/segfaults).
+ https://bugzilla.gnome.org/show_bug.cgi?id=755621
+
+2015-09-23 20:59:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Accumulate segments for edit lists before activating the next segment
+ eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
+ accumulating segments manually when activating a segment. This is only
+ needed when handling edit lists, not when activating a segment because of a
+ seek. Do the accumulation when switching edit list segments instead.
+ This fixes segment seeks again, while keeping edit lists playback working.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755471
+
+2015-09-23 17:43:51 +0530 Vikram Fugro <vikram.fugro@gmail.com>
+
+ * gst/spectrum/gstspectrum.c:
+ spectrum: send phase values in the GstMessage for Phase info
+ https://bugzilla.gnome.org/show_bug.cgi?id=755463
+
+2015-09-22 00:46:01 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Don't output a warning on MONO multiview mode.
+
+2015-09-19 17:02:18 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtptheoradepay.c:
+ rtptheoradepay: Fix memory leaks
+ The same memory leaks were fixed in identical fashion for
+ vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
+
+2015-09-19 17:04:07 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
+ * use g_list_free_full(), don't iterate elements maually when freeing
+ * call gst_rtp_*_pay_clear_packet(), don't duplicate its code
+ * use gst_buffer_unref() to clarify that it is buffers being released,
+ instead of refering directly to gst_mini_object_unref()
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
+
+2015-09-19 18:44:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=755265
+
+=== release 1.5.91 ===
+
+2015-09-18 19:33:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.91
+
+2015-09-18 19:23:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-09-18 11:50:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-09-17 10:50:01 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix taglist leak
+ gst_tag_list_insert() does not take ownership of the inserted taglist.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755138
+
+2015-09-16 07:05:36 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Skip LOAS AAC until a valid config is seen.
+ It's normal when dropping into the middle of a stream to
+ not always have the config available immediately, so skip LOAS
+ until a valid config is seen without either setting invalid
+ caps or erroring out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751386
+
+2015-09-13 15:41:38 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: reset just a bit more upon flush_stop
+
+2015-09-13 15:40:09 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: remove dead struct member
+
+2015-09-11 17:09:28 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: fix GError memory leak when hostname resolution fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=754869
+
+2015-09-10 15:26:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/matroska/ebml-write.c:
+ matroskamux: drop HEADER flag from output buffers
+ Drop HEADER flag from output buffers if they are not indeed
+ headers.
+ Fixes resending of headers in tcp connection handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=754768
+
+2015-09-10 16:00:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/ebml-write.c:
+ matroskamux: fix matroskamux ! matroskademux
+ Don't carry over DISCONT flags from the input buffers to the
+ output buffer, or the demuxer might reset its state when it
+ receives the first data buffer just after parsing the simple
+ block header, and then expect sane data to follow.
+ Fixes matroskamux ! demux erroring out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754768
+ https://bugzilla.gnome.org/show_bug.cgi?id=657805
+
+2015-09-09 12:51:40 -0700 Martin Kelly <martin@surround.io>
+
+ * gst/rtsp/README:
+ rtsp: fix small README typo
+ https://bugzilla.gnome.org/show_bug.cgi?id=754807
+
+2015-09-04 19:45:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstwavpackparse.c:
+ wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
+ https://bugzilla.gnome.org/show_bug.cgi?id=752106
+
+2015-09-04 19:34:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
+ flac contains the sample offset in the frame header, so after a seek
+ without index flacparse will know the exact position we landed on and
+ timestamp buffers accordingly. It only set the pts though, which means
+ the baseparse-set dts which was set to the seek position prevails, and
+ since the seek was based on an estimate, there's likely a discrepancy
+ between where we wanted to land and where we did land, so from here on
+ that dts/pts difference will be maintained, with dts possibly multiple
+ seconds ahead of pts, which is just wrong. The easiest way to fix this
+ is to just set both pts and dts based on the sample offset, but perhaps
+ parsed audio should just not have dts set at all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752106
+
+2015-09-06 16:33:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ docs: remove properties and signals that no longer exist
+ https://bugzilla.gnome.org/show_bug.cgi?id=726443
+
+2013-10-11 15:13:00 +0000 George Chriss <gschriss@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Make the element count in arrays not include end
+ One-line removal of tags_written++
+ This should fix rtmp output to crtmpserver, and hopefully
+ noone is expecting that the element count includes the end
+ element, as different bits of documentation say different
+ things about whether it should or not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=661624
+
+2015-07-30 00:59:15 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Store incoming bitrate tags and send in the metadata
+ Apparently the Microsoft Azure RTMP server requires that the
+ videodatarate and audiodatarate metadata be provided, so
+ set those, even if it's to 0. Use the actual input bitrate
+ tags if available.
+
+2015-09-04 00:06:29 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't parse key data more than needed.
+ When an auxilliary streams are present in the SDP media,
+ there's no need to re-parse the SDP attributes multiple
+ times.
+
+2015-09-03 20:56:55 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
+ In parse_keymgmt(), don't mutate the input string that's been passed
+ as const, especially since we might need the original value again if
+ the same key info applies to multiple streams (RTX, for example).
+ When a resource is 404, and we have auth info - retry with the auth
+ info the same as if we had receive unauthorised, in case the resource
+ isn't even visible until credentials are supplied.
+ Fix a memory leak handling Mikey data.
+ When generating a random keystring, don't overrun the 30 byte
+ buffer by generating 32 bytes into it.
+
+2015-09-04 15:18:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix build with GLib < 2.44
+ G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.
+
+2015-09-04 12:01:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
+ This happens on Windows if we use the same socket for sending packets,
+ and the remote sends ICMP port/host unreachable messages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754534
+
+2015-09-02 21:12:41 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbis/theoradepay: Fix handling of fragmented packets
+ This was broken in b1089fb520 by not considering the full packet length of a
+ fragmented packet but only the length of the first one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754417
+
+2015-09-01 15:39:22 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/dtmf/gstdtmfsrc.c:
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ dtmfsrc: Reply to latency query
+
+2015-08-31 16:42:30 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/elements/rtpsession.c:
+ tests: Fix rtpsession test failure
+ The time of the first RTCP packet is semi-random, so
+ sometimes it was produced before enough packets from
+ the second SSRC were received. First drop queued RTCP
+ packets, then advance the clock enough to ensure
+ that at least one new RTCP packet is produced.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750731
+
+2015-08-31 13:56:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/elements/level.c:
+ level: improve the test for multi-channel mode
+ Change the test to verify the read-index for multiple messages per buffer.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=754144
+
+2015-08-31 12:46:52 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Align raw video frames to 32 bytes
+ Outputting unaligned video frames causes videoscale et al to
+ crash when attempting SIMD-accelerated conversion.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736965
+
+2015-08-26 23:16:46 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/level/gstlevel.c:
+ level: fix level calculations for mutliple channels
+ This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.
+
+2015-08-27 10:28:55 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: Fix memory leak
+ In gst_smpte_collected(), check upfront if input formats are same
+ or not. This avoids allocation of in1 and in2 buffers and
+ subsequent memory leak when input formats do not match.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754153
+
+2015-08-21 11:52:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: don't try to connect to dead radio server
+
+2015-08-21 16:29:16 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Trivial fix to check correct condition
+ When checking for describe method, because of missing parentheses, wrong
+ condition is being checked, which will result in wrong behavior.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753912
+
+2015-08-21 13:19:02 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: read: fix tag list memory leak
+ gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
+ be unref'ed as soon as the tags are merged to the tocentry
+ https://bugzilla.gnome.org/show_bug.cgi?id=753904
+
+2015-08-21 12:20:59 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/wavpack/gstwavpackdec.c:
+ wavpackdec: fix taglist memory leak
+ When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
+ by audiodecoder and the caller should free the taglist being passed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753903
+
+2015-08-20 14:45:33 +0200 Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix pad closing
+ Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=753875
+
+=== release 1.5.90 ===
+
+2015-08-19 13:29:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.90
+
+2015-08-19 12:47:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-08-19 11:29:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-08-13 17:29:58 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesrc.c:
+ multifilesrc: fix regression with starting from index set via index property
+ When we haven't started yet, set the start_index when we set the index property,
+ so that we start at the right index position after the initial seek. The index
+ property was never really meant to be for writing, but it used to work, so let's
+ support it for backwards compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739472
+
+2015-08-18 10:52:11 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix offset calculation when parsing CENC aux info
+ Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
+ DASH common encryption, however commit
+ bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
+ shortly before the CENC commit caused the calculation of the CENC
+ aux info offset to be incorrect.
+ The base_offset was being added if present, but if the base_offset
+ is relative to the start of the moof, the offset was being added twice.
+ The correct approach is to calculate the offset from the start of the
+ moof and use that offset when parsing the CENC aux info.
+
+2015-08-17 14:28:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: actually return true for accept-caps query handling
+
+2015-08-17 14:07:10 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ rtp: copy metadata in the (de)payloaders which is missed before
+ https://bugzilla.gnome.org/show_bug.cgi?id=753706
+
+2015-08-16 15:21:51 -0400 Dustin Spicuzza <dustin@virtualroadside.com>
+
+ * configure.ac:
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/directsound/gstdirectsoundsink.h:
+ directsoundsink: allow specifying audio playback device
+ https://bugzilla.gnome.org/show_bug.cgi?id=753670
+
+2015-08-16 13:51:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: remove single entry if from loop
+ Iterate from the 2nd channel on and create the 1 channel struct
+ outside to make loop structure simpler and only slightly faster.
+
+2015-08-16 13:21:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: implement proper accept-caps
+ Should just compare with what can be immediatelly accepted by
+ the element. flacenc can't renegotiate so if it has a caps already
+ it should only accept if it is that caps otherwise just use the
+ template caps
+
+2015-08-16 13:03:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: improve sink pad template caps
+ Removes the need for custom caps query handling and makes it more
+ correct from the beginning on the template. It is a bit uglier
+ to read because there is 1 entry per channel but makes code easier
+ to maintain.
+
+2015-08-16 12:41:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/y4m/gsty4mencode.c:
+ y4mencode: fix gst-launch version in documentation
+
+2015-08-15 22:32:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/speex/gstspeexenc.c:
+ * ext/wavpack/gstwavpackenc.c:
+ * gst/law/alaw-encode.c:
+ * gst/law/mulaw-encode.c:
+ audioencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: speexenc, wavpackenc, mulawenc, alawenc
+
+2015-08-15 22:29:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/jpeg/gstjpegenc.c:
+ * ext/libpng/gstpngenc.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/y4m/gsty4mencode.c:
+ videoencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
+
+2015-08-16 17:21:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: use new baseparse API to fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-03-17 17:50:37 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: use new base parse API to fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 14:37:53 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: use new baseparse API and fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 13:04:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use signed integer type to be able to check for negative subtraction results
+ CID 1315829
+
+2015-08-16 11:50:34 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbisdepay: remove dead code
+ payload_buffer must be NULL in ignore_reserved. Check will always be false.
+ Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
+ CID #1316476
+
+2015-08-15 22:45:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-encode.c:
+ * gst/law/alaw-encode.h:
+ alawenc: port to AudioEncoder base class
+
+2015-08-15 09:16:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacdec.c:
+ * ext/speex/gstspeexdec.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * gst/law/alaw-decode.c:
+ * gst/law/mulaw-decode.c:
+ audiodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
+
+2015-08-15 08:49:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ videodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: jpegdec, pngdec, vp8dec, vp9dec
+
+2015-08-15 11:31:04 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-decode.c:
+ alawdec: make error handling a bit nicer
+ Print the element along with the debug to make it easier to trace
+ the failures
+
+2015-08-15 11:04:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-decode.h:
+ alawdec: port to audiodecoder base class
+ mulawdec was already ported, alawdec was left behind.
+
+2015-08-15 10:34:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only look for more samples in moofs in pull-mode
+ For playback of some fragmented formats with qtdemux it will
+ try to look for the next moof after finishing one but it is only
+ possible for pull-mode. For playback of streaming fragmented formats
+ such as DASH it should just not try to look for another moof but
+ instead wait for more data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752602
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-08-15 12:58:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstdcaparse.c:
+ dcaparse: Don't look for a second syncword
+ There are streams out there that consistently contain garbage between
+ every frame so we never ever find a second consecutive syncword.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=738237
+
+2015-08-15 11:12:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp8enc, vp9enc: reset multipass file index when stopping encoder
+ Fixes multipass encoding when re-using the same element/pipeline
+ for subsequent encoding runs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-15 11:09:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp9enc: provide support for multiple pass cache files
+ Some files may provide different caps insight of one stream. Since
+ vp9enc support caps reinit, we should support cache reinit too.
+ If more then file cache file will be created, the naming will be:
+ cache cache.1 cache.2 ...
+ Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-14 11:41:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/aacparse.c:
+ tests: aacparse: use caps query instead of accept-caps
+ The accept-caps query just does a shallow check at the current
+ element while at this test we want it to also look at downstream.
+ So use caps query there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 11:40:22 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: enable accept-template flag
+ Do a quick check with the pad template caps as it is enough. Users
+ should have figured the appropriate full caps on a previous caps query
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 15:46:53 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: send the User-Agent header
+ Sometimes it is useful to know this information on the
+ server side. Other popular implementations (vlc, ffmpeg, ...)
+ also send this header on every message.
+ This includes a new "user-agent" property that the user
+ can set to use a custom User-Agent string. The default
+ is "GStreamer/<version>"
+ https://bugzilla.gnome.org/show_bug.cgi?id=750101
+
+2015-08-14 15:42:42 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: wrap gst_rtsp_message_init_request in a local function
+ This will allow adding common request initialization, like the
+ user agent string, in just one place.
+
+2015-08-14 09:36:09 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: make sure buffer gets reallocated if max_delay changes
+ https://bugzilla.gnome.org/show_bug.cgi?id=753490
+
+2015-07-09 09:51:26 +0200 Oleksij Rempel <linux@rempel-privat.de>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ vp8enc: provide support for multiple pass cache files
+ Some files may provide different caps insight of one stream. Since vp8enc
+ support caps reinit, we should support cache reinit too.
+ If more then file cache file will be created, the naming will be:
+ cache
+ cache.1
+ cache.2
+ ...
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-04-15 22:51:51 +0200 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4gdepay.c:
+ rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
+ Use constantDuration to calculate the timestamp of non-first AU in the
+ RTP packet.
+ If constantDuration is not present in the MIME parameters, its value
+ must be calculated based on the timing information from two consecutive
+ RTP packets with AU-Index equal to 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747881
+
+2015-08-14 06:43:13 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: remove unnecessary if, g_free is null safe
+
+2015-08-14 08:33:56 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: add property to set HTTP method
+ To allow souphttpsrc to be use HTTP methods other than GET
+ (e.g. HEAD), add a "method" property that is a string. If this
+ property is not set, GET is used.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752413
+
+2015-08-14 11:13:01 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/generic/states.c:
+ check: Rename states unit test
+ Makes it easier to differentiate from other modules states unit test
+
+2015-08-14 09:21:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ goom: Rename get_type() function of base class to prevent symbol conflicts
+ This is a problem when statically linking.
+
+2015-08-13 16:32:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
+ Otherwise we will just output buffers without timestamps after a reset if no
+ timestamps are provided by upstream, e.g. when using RTSP over TCP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-08-12 17:16:01 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/matroska/matroska-demux.h:
+ * gst/matroska/matroska-parse.h:
+ matroska: Remove unused variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=753556
+
+2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ * gst/rtp/gstrtputils.c:
+ * gst/rtp/gstrtputils.h:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvorbispay.h:
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ rtp: Copy metadata in the (de)payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-10 18:20:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix small typo in comment
+
+2015-08-10 16:19:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/goom2k1/gstgoom.c:
+ goom2k1/doc: Fixup previous commit
+
+2015-08-10 15:55:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1/doc: Use GstGoom2k1 namespace
+ The doc generator isn't happy when we have class name clash. Simply
+ use it's own namespace.
+
+2015-08-10 17:10:42 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: removed unused variable in set_property
+ unused local variable 'delay' is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753450
+
+2015-08-10 12:45:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix suboptimal queue iteration code
+
+2015-08-09 17:25:45 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't use glib 2.44-only API
+
+2015-07-29 14:14:50 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: add support for ISOBMFF Common Encryption
+ This commit adds support for ISOBMFF Common Encryption (cenc), as
+ defined in ISO/IEC 23001-7. It uses a GstProtection event to
+ pass the contents of PSSH boxes to downstream decryptor elements
+ and attached GstProtectionMeta to each sample.
+ https://bugzilla.gnome.org/show_bug.cgi?id=705991
+
+2015-08-10 14:13:50 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: checking if depay has sps/pps nals before insertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753430
+
+2015-08-08 16:44:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: fix outdated comment
+ The default behaviour was changed in the 0.10 -> 1.x
+ transition, but the comment was not updated.
+
+2015-08-08 17:42:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ rtptheorapay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:41:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:34:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
+ We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
+ and 4:4:4 formats.
+
+2015-08-06 17:46:13 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ rtpklv(de)pay: add "RTP" in the klass string
+ GstRTSPMedia uses this classification to detect the real payloader
+ inside a dynpay bin and asserts if it doesn't find it, therefore
+ it is required
+ https://bugzilla.gnome.org/show_bug.cgi?id=753325
+
+2015-08-05 11:13:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: use a dynamic pt in the test
+ 1) Tests that using dynamic PT instead of the default ones work
+ 2) If we ever decide to change the codec here we don't need to
+ worry about change the PT for the default one of the new codec
+ in the test
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-05 10:53:15 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: print valid type where guint32 is expected
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-06 11:33:37 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmupay.c:
+ rtppayload: set standard payload type as default
+ Initialize the PT to the default value of the codec and check if
+ it is still the default before declaring the pt to be dynamic or
+ not when setting the caps.
+ Also use the PT constants from the rtp lib when possible
+ https://bugzilla.gnome.org/show_bug.cgi?id=747965
+
+2015-07-26 12:07:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: store the moof-offset also for push mode
+ It will be used in some cases for getting the correct offsets
+ from trun atoms.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-26 02:09:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.h:
+ qtdemux: handle default-base-is-moof flag
+ Handle the flag from the tfhd that signals the base offset to
+ start from the moof atom
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-29 18:54:35 -0600 Glen Diener <grd@loganmill.net>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskademux: Preserve forward referenced track tags
+ https://bugzilla.gnome.org/show_bug.cgi?id=752850
+
+2015-08-04 18:07:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: fix test failure
+ The RTP PT for alaw is 8.
+ Less than 50 packets are received in the length of this test so it
+ would never drop a buffer or would drop only the last buffer and
+ it would fail sometimes when the received wouldn't receive the
+ retransmission packet in time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpstreamdepay.c:
+ rtpstreamdepay: Only allow activation in push mode
+ We need a proper caps event from upstream with the full RTP caps as we can't
+ create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
+ a filesrc or any other element that supports pull mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753066
+
+2015-08-04 16:28:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ soup: fix typo in translated string
+ https://bugzilla.gnome.org/show_bug.cgi?id=753240
+
+2015-08-04 12:25:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Put the profile and level into the caps
+
+2015-08-04 12:09:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Only update the srcpad caps if something else than the codec_data changed
+ h264parse does the same, let's keep the behaviour consistent. As we now
+ include the codec_data inside the stream too here, this causes less caps
+ renegotiation.
+
+2015-08-04 11:48:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
+ The spec says:
+ When a picture parameter set NAL unit with a particular value of
+ pic_parameter_set_id is received, its content replaces the content of the
+ previous picture parameter set NAL unit, in decoding order, with the same
+ value of pic_parameter_set_id (when a previous picture parameter set NAL unit
+ with the same value of pic_parameter_set_id was present in the bitstream).
+
+2015-08-03 13:45:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: remove extra \n at debug message
+
+2015-08-03 13:42:20 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: prevent deadlock when states change too fast
+ If the GOP is completed, pads have to start gathering for the
+ next one but it is possible that the the state might go to
+ COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
+ thread has a chance to wake up and proceed, leaving it trapped in
+ the check_completed_gop loop and deadlocking the other threads
+ waiting for it to advance.
+ To solve it, this patch also checks that tha input running time
+ hasn't changed to prevent this scenario.
+
+2015-08-03 17:55:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Insert SPS/PPS NALs into the stream
+ h264parse does the same and this fixes decoding of some streams with 32 SPS
+ (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
+ the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
+ As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+ This looks like a mistake in the part of the spec about the codec_data.
+
+2015-07-30 11:29:27 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: handle empty http proxy string
+ 1) If the system http_proxy environment variable is not set
+ or set to an empty string, we must not set proxy to avoid
+ http connection error.
+ 2) In case of proxy property setting, if user want to clear
+ the proxy setting, they should be able to set it to NULL or
+ an empty string again, so this is fixed too.
+ 3) Check if the proxy string was parsed correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752866
+
+2015-07-29 15:46:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: remove unused variable
+ Remove unused variable 'framecount' from dvdemux
+ https://bugzilla.gnome.org/show_bug.cgi?id=753008
+
+2015-07-30 15:32:09 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-30 15:21:20 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtp/gstrtpmp4vdepay.c:
+ rtpmp4vdepay: rtpbuffer is being unref'ed twice
+ process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
+ the refernce should not be removed here
+ https://bugzilla.gnome.org/show_bug.cgi?id=753042
+
+2015-07-29 11:26:46 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 19:28:33 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Support mpegtsmux as a muxer.
+ As a fallback, look for a pad template sink_%d on
+ the muxer when requesting pads, to support mpegtsmux
+ https://bugzilla.gnome.org/show_bug.cgi?id=752999
+
+2015-06-25 01:35:27 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ splitmuxsrc: Use a separate lock to delay typefind.
+ Don't hold the main splitmux part lock over
+ the parent state change function, as it prevents
+ posting error messages that happen. Since the purpose
+ is to prevent typefinding from proceeding, use a
+ separate mutex just for that.
+
+2015-07-29 13:43:50 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: fix memory leak
+ After adding to tag list, key_val is not being free'd
+ resulting in memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=752992
+
+2015-07-27 13:34:14 +0900 Manasa Athreya <manasa.athreya@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
+ 'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
+ be more bits as well, in which case it's just like 'twos'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752613
+
+2015-07-24 15:10:05 +0200 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: Allow framerate to be large then 100pfs
+ This limit was arbitrary. We still fixate near 100pfs for compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752825
+
+2015-07-25 03:25:28 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Stop without posting error on flushing
+ This could just be a normal pipeline shutdown.
+
+2015-07-23 15:00:08 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
+ https://bugzilla.gnome.org/show_bug.cgi?id=752618
+
+2015-07-16 18:09:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/matroskademux.c:
+ tests: add minmal matroskademux test for subtitle output
+ Some of the subtitle chunks will have embedded
+ NUL-terminators (last three), some don't (first three),
+ some will have markup, some won't, some will be valid
+ UTF-8 (all but last), some won't (last stanza).
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-16 18:49:26 +0300 Dimitrios Christidis <dchristidis@mykolab.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix for subtitle buffers with NUL terminators
+ Commit 45892ec8 created a regression where g_utf8_validate() would fail
+ if the subtitle buffer had a NUL terminator as part of the data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-21 13:31:05 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpvp8depay: Check available bytes before copy
+ Need to check that the number of bytes we want to copy from the adapter
+ actually is available and handle the error case gracefully. This error
+ may happen if malformed packets are received and we don't have a
+ complete frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752663
+
+2015-07-16 09:32:36 +0900 Paul Hyunil <paul.hyunil@lge.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Support subtitle when track subtype is fourcc_subt
+ https://bugzilla.gnome.org/show_bug.cgi?id=752655
+
+2015-07-20 16:59:40 +0800 Song Bing <b06498@freescale.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Set timestamp when queue buffer.
+ Should set timestamp when queue buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752618
+
+2015-07-16 15:12:17 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: handle different ssrc's on sinkpads
+ Do this by not putting the ssrc from the src pads in the caps used to
+ probe other sinkpads, and then intersecting with it later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752491
+
+2015-07-16 17:19:03 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/webm-mux.c:
+ Update mailing list address from sourceforge to freedesktop
+
+2015-07-15 13:44:52 +0300 Dimitrios Christidis <dchristidis@mykolab.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix trailing '*' displayed with some text subtitles
+ The subtitle buffer we push out should not include a NUL terminator
+ as part of the data, we just add such a terminator for safety, but
+ it should not be included in the buffer size.
+ A NUL terminator is not valid UTF-8, so checks will fail if it's
+ included in the size, and the NUL will be replaced by the fallback
+ character specified when converting, i.e. '*'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-15 18:23:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ * ext/pulse/pulseutil.c:
+ * ext/pulse/pulseutil.h:
+ pulse: add properties to GstDevice
+ Add the extra properties we get from pulse to the GstDevice we expose
+ with the device monitor
+
+2015-07-15 17:20:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audioinvert.c:
+ * gst/audiofx/audiowsincband.c:
+ audiofx: Fix typo in example pipelines
+ Fix typo in example pipelines of audiowsincband and audioinvert.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752416
+
+2015-04-15 18:27:04 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: add a "format-location" signal that allows better control over filenames
+ In certain applications, splitting into files named after a base
+ location template and an incremental sequence number is not enough.
+ This signal gives more fine-grained control to the application to
+ decide how to name the files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750106
+
+2015-04-15 20:13:27 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudiosrc: no resampling on OS X
+ Unlike Remote IO, AUHAL doesn't have built-in resampling
+ for sources -- confirmed by Core Audio engineer Doug Wyatt:
+ http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-04-15 18:29:14 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudiosrc: avoid get_channel_layout
+ This only produces a warning and serves no purpose.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-04-07 15:40:14 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Avoid making a duplicate structure in caps for mono/stereo case
+ For 1ch or 2ch devices, we just need to set the caps to allow both
+ options since CoreAudio will up/downmix appropriately.
+ Also fixes the condition for the 2ch case to be exact, rather than at
+ least 2 channels since the downmix will not take place in the >stereo
+ case.
+
+2015-04-06 16:22:34 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Don't set the format on an initialized AudioUnit
+ We need to initialize the AudioUnit early to be able to probe the
+ underlying device, but according to the AudioUnitInitialize() and
+ AudioUnitUninitialize() documentation, format changes should be done
+ while the AudioUnit is uninitialized. So we explicitly uninitialize the
+ AudioUnit during a format change and reinitialize it when we're done.
+
+2015-04-06 15:55:59 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Minor spelling fix (unitialize -> uninitialize)
+
+2015-03-21 20:34:25 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Fix lockup in _audio_unit_property_listener
+ _audio_unit_property_listener is called either from a Core Audio thread
+ or as a result of a Core Audio API (e.g. AudioUnitInitialize)
+ from our own thread. In the latter case, osxbuf can be already locked
+ (GStreamer's mutex is not recursive).
+ We introduce the flag cached_caps_valid and use it instead of nullifying
+ cached_caps when we cannot lock on osxbuf.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-12 12:15:12 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Invalidate cached caps on format change
+ Listen for changes in hardware stream format and channel layout, and
+ invalidate cached caps (since they contain the preferred caps).
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-09 23:34:06 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Overhaul of probing caps
+ - Probing caps is unified between source and sink
+ - Hardware stream format is now reported as preferred capabilities
+ (dynamically updated when hardware configuration changes)
+ - Get hardware channel layout from Remote IO just like from HAL
+ - More comprehensive mapping between AudioChannelLabel and
+ GstAudioChannelPosition
+ - Support for unpositioned channel layouts
+ - Announce stereo-mono upmixing/downmixing in caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-09 23:15:56 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: AudioUnitInitialize on open
+ Call AudioUnitInitialize upon open. Otherwise, we cannot get
+ (hardware) stream format nor channel layout from the outer scope.
+
+2015-07-12 14:27:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvp8depay.c:
+ rtp: depayloaders: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't
+ need to map the input buffer again but can just re-use
+ the mapping the base class has already done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-05-27 19:19:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't
+ need to map the input buffer again but can just re-use
+ the map the base class has already done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-07-10 00:13:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix indention
+
+2015-07-09 23:59:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Always estimate DTS from the current clock time
+ Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
+ we would produce wrong DTS. As now the estimated DTS is based on the clock,
+ don't store it in the jitterbuffer items as it would otherwise be used in the
+ skew calculations and would influence the results. We only really need the DTS
+ for timer calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-09 09:26:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/.gitignore:
+ gitignore: ignore rtph263 test
+
+2015-07-08 23:47:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
+ Replace static constants with macros to make gcc happy
+ CC elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
+ elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
+ static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
+ ^
+ elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
+ static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
+ ^
+ elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
+ PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;
+
+2015-07-08 23:40:45 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: run indent and fix some comments
+ Fix indent on this file and break some comment lines into two to make
+ it fit 80 chars per line
+
+2015-07-08 15:02:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: rework segment event handling for adaptive streaming
+ When a new time segment is received upstream is going to restart
+ with a new atom. Make the neededbytes and todrop variables
+ reflect that to avoid waiting too much or dropping the
+ initial bytes that contain the header.
+
+2015-07-08 12:35:55 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: push data from adapter before starting new segment
+ The adapter might have data remaining from the previous segment,
+ push it all before clearing the adapter and starting a new segment.
+ It can accumulate data if it had pushed and got not-linked, returning
+ immediately without processing all the data. Before starting a new
+ segment this data should be handled.
+
+2015-07-08 19:59:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 21:08:36 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix gap-time calculation and remove "late"
+ The amount of time that is completely expired and not worth waiting for,
+ is the duration of the packets in the gap (gap * duration) - the
+ latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
+ that we make a "multi-lost" packet for.
+ The "late" concept made some sense in 0.10 as it reflected that a buffer
+ coming in had not been waited for at all, but had a timestamp that was
+ outside the jitterbuffer to wait for. With the rewrite of the waiting
+ (timeout) mechanism in 1.0, this no longer makes any sense, and the
+ variable no longer reflects anything meaningful (num > 0 is useless,
+ the duration is what matters)
+ Fixed up the tests that had been slightly modified in 1.0 to allow faulty
+ behavior to sneak in, and port some of them to use GstHarness.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738363
+
+2015-06-30 11:21:31 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
+ This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0.
+ The reverted patch is wrong and introduces a regression because there
+ may still be time to receive some of the packets included in the gap
+ if they are reordered.
+
+2015-07-07 23:53:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: flush samples before adding more from moof
+ Avoids accumulating all samples from a fragmented stream that could
+ lead to a 'index-too-big' error once it goes over 50MB of data. It
+ could reach that before 2h of playback so it doesn't take that long.
+ As upstream elements are providing data in time format they should
+ be the ones that have more information about the full media index
+ and should be able to seek if possible.
+
+2015-07-07 23:56:12 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: rename upstream_newsegment to upstream_format_is_time
+ upstream_newsegment isn't really clear on what it means, it is set
+ to TRUE when the upstream element sends a segment in TIME format, so
+ rename it to be more clear about it.
+ It is important to know this because it means that upstream has
+ a notion of time and qtdemux is likely being driven by an upstream
+ element that is reading from a higher level abstraction than a file,
+ such as a DASH, MSS or DLNA element.
+
+2015-07-07 21:31:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix leak by flushing previous sample info from trak
+ In fragmented streaming, multiple moov/moof will be parsed and their
+ previously stored samples array might leak when new values are parsed.
+ The parse_trak and callees won't free the previously stored values
+ before parsing the new ones.
+ In step-by-step, this is what happens:
+ 1) initial moov is parsed, traks as well, streams are created. The
+ trak doesn't contain samples because they are in the moof's trun
+ boxes. n_samples is set to 0 while parsing the trak and the samples
+ array is still NULL.
+ 2) moofs are parsed, and their trun boxes will increase n_samples and
+ create/extend the samples array
+ 3) At some point a new moov might be sent (bitrate switching, for example)
+ and parsing the trak will overwrite n_samples with the values from
+ this trak. If the n_samples is set to 0 qtdemux will assume that
+ the samples array is NULL and will leak it when a new one is
+ created for the subsequent moofs.
+ This patch makes qtdemux properly free previous sample data before
+ creating new ones and adds an assert to catch future occurrences of
+ this issue when the code changes.
+
+2015-07-07 16:46:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix index size check and debug message
+ It is allocating samples_count + n_samples, not only n_samples
+
+2015-07-08 17:02:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Calculate receive time if we don't have any
+ This is required to properly schedule packet loss timers and make
+ sure all our calculations work properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 15:13:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
+ That is, handle DTS==GST_CLOCK_TIME_NONE correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 20:31:42 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix event leak
+ when seek fails in avidemux, event is not being freed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752117
+
+2015-07-08 12:02:22 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtph263.c:
+ rtph263depay: Make sure payload is large enough
+ Plus new unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752112
+
+2015-07-08 08:59:49 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ rtpklvdepay: fix printf format compiler warning
+ v_len is of type guint64, but while print the value(16 + len_size + v_len)
+ G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT
+ https://bugzilla.gnome.org/show_bug.cgi?id=752100
+
+2015-07-07 20:25:47 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: add new RTP elements to docs
+
+2015-07-07 20:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: add basic unit test for KLV payloading
+ Also make it so that the mtu is always set if specified, not
+ only in case of the rather weird bufferlist test code path.
+ This allows us to easily make the payloader fragment a payload
+ across multiple output packets by setting a small MTU on it.
+
+2015-07-07 19:58:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ rtpklvdepay: improve start detection and handle fragmented KLV units
+
+2015-07-05 20:25:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ rtp: add SMPTE 336M KLV metadata depayloader
+ http://tools.ietf.org/html/rfc6597
+
+2014-08-09 10:08:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpklvpay.h:
+ rtp: add SMPTE 336M KLV metadata payloader
+ http://tools.ietf.org/html/rfc6597
+
+2015-07-07 16:59:20 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/atomsrecovery.c:
+ * gst/isomp4/properties.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/rtpmanager/rtpsource.c:
+ docs: fix "Symbol name not found at the start of the comment block"
+ Add symbols or change comment into a regular comment.
+
+2015-07-07 16:58:53 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/audioparsers/gstamrparse.h:
+ docs: remove outdated doc strings
+
+2015-07-03 23:10:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ docs: add missing plugins and ensure master doc is sorted
+
+2015-07-07 15:54:41 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ Revert "imagefreeze: Remove impossible error condition"
+ This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94.
+ pad only handle EOS events but not EOS flow, and will push the buffer again
+ resulting in an assertion error. So we should not handle the buffer
+ and return EOS flow.
+
+2015-07-07 15:50:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpg729depay.c:
+ rtpg729depay: unmap rtp buffer in error path
+
+2015-07-07 15:48:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: fix buffer leak
+ The handle_buffer vfunc takes ownership of the input buffer.
+ Fixes elements/rtp-payloading under valgrind.
+
+2015-07-02 08:52:43 +0200 Tobias Mueller <muelli@cryptobitch.de>
+
+ * gst/goom/goom_core.c:
+ goom: Initialised variables to remove compiler warnings
+ goom_core.c: In function 'goom_update':
+ goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
+ ^
+ goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
+ ^
+ https://bugzilla.gnome.org/show_bug.cgi?id=752053
+
+2015-07-07 09:18:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: fix indentation
+
+2015-07-06 19:11:00 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Fix uninitialized variable compiler error
+ endpos variable does not correctly understand in the
+ 4.6.3 GCC version. So compile error appears when we do
+ compile rtph261pay using jhbuild.
+ This patch is fixed the compile error in 4.6.3 GCC version.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751985
+
+2014-11-12 12:08:58 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Handle seek flags properly
+ Allows for non-keyframe seeks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738570
+
+2015-02-24 10:50:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: avoid looping reading the 'moof' atom forever
+ It gets stuck if it only finds a moof and no mfra/mfro or moov
+ atoms. Skip the moof to continue the parsing to have it either
+ play or error out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745089
+
+2015-06-26 13:24:17 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: improve error handling
+ for files which have corrupted header, libflac is not able to
+ process the metadata properly. We just try to ignore the error
+ and continue with the processing, since metadata parsing is not
+ making much of a difference to libflac
+ https://bugzilla.gnome.org/show_bug.cgi?id=751334
+
+2015-07-06 20:16:38 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: add meta transform function
+ ximage metadata can't be transformed or copied, but provide an empty
+ transformation function instead of NULL to allow unconditional calling
+ of metas' transform functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751778
+
+2014-06-16 16:14:28 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: init debug category
+ https://bugzilla.gnome.org/show_bug.cgi?id=752012
+
+2014-06-20 10:59:14 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpv8depay: ignore reserved bit in payload descriptor
+ Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
+ R: Bit reserved for future use. MUST be set to zero and MUST be
+ ignored by the receiver.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751929
+
+2015-07-04 20:56:42 +0200 Stian Selnes <stian@pexip.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: rtph261depay: Add documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=751982
+
+2015-07-03 21:58:14 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-07-03 14:29:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Fix compiler warning
+ gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
+ gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
+ GObjectClass *gobject_class;
+
+2015-07-03 14:03:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261depay.c:
+ rtph261depay: Let the base class push the buffer so it can deal with the flow return
+
+2015-07-03 14:11:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Remove unused adapter
+
+2015-07-03 13:17:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpspeexpay.c:
+ speexpay: Directly attach payload to the output buffer instead of copying it
+
+2015-07-03 13:07:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ sbcpay: Attach payload directly to the output instead of copying
+
+2014-12-01 14:18:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261depay.h:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph261pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtp: add H.261 RTP payloader and depayloader
+ Implementation according to RFC 4587.
+ Payloader create fragments on MB boundaries in order to match MTU size
+ the best it can. Some decoders/depayloaders in the wild are very strict
+ about receiving a continuous bit-stream (e.g. no no-op bits between
+ frames), so the payloader will shift the compressed bit-stream of a
+ frame to align with the last significant bit of the previous frame.
+ Depayloader does not try to be fancy in case of packet loss. It simply
+ drops all packets for a frame if there is a loss, keeping it simple.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751886
+
+2015-07-03 12:18:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpvdepay.c:
+ rtpmpvdepay: Don't forget to unmap the input buffer
+
+2015-07-03 12:14:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpvpay.c:
+ rtpmpvpay: Create buffer lists instead of pushing each buffer individually
+
+2015-07-03 12:03:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpapay.c:
+ rtpmpapay: Use buffer lists instead of pushing each fragment individually
+
+2015-07-03 10:51:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmp4apay.c:
+ rtpmp4apay: Create buffer lists and don't copy payload memory
+
+2015-06-29 16:14:18 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT
+ When there are a lot of small gaps, we can consider that there is
+ a big gap (too losses) to reset the buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751636
+
+2015-06-29 15:53:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: If possible, always update the current time before looping over all timers
+ If we have a clock, update "now" now with the very latest running time we have.
+ If timers are unscheduled below we otherwise wouldn't update now (it's only updated
+ when timers expire), and also for the very first loop iteration now would otherwise
+ always be 0.
+ Also the time is used for the timeout functions, e.g. to calculate any times
+ for the next timeouts and we would otherwise pass too old times there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751636
+
+2015-07-02 14:34:57 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix memory leak
+ tmp needs to be freed before going out of scope in 'done'.
+ CID #1308954
+
+2015-07-02 12:23:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it
+
+2015-07-02 09:48:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us
+
+2015-07-02 09:17:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pay.h:
+ rtph263pay: Stop using an adapter and directly use the buffer
+ We always pushed one buffer into the adapter, then handled exactly that one
+ buffer and flushed it from the adapter. Now also don't memcpy() the actual
+ payload but just attach the input buffer's data to the output buffer.
+ This code still needs some serious refactoring/rewriting.
+
+2015-07-01 21:57:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgsmpay.c:
+ rtpgsmpay: Remove non-existing includes for now
+ git add -p mistake.
+
+2015-07-01 19:29:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Use the return value of gst_buffer_append()
+
+2015-07-01 19:19:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgsmpay.c:
+ rtpgsmpay: Attach payload to the output buffer instead of copying it
+
+2015-07-01 17:58:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: Attach payload directly to output buffers instead of copying
+
+2015-07-01 17:43:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpg723pay.c:
+ rtpg723pay: Attach payload buffer to the output instead of copying
+
+2015-07-01 17:30:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpdvdepay.c:
+ rtpdvdepay: Map the output buffer once instead of once every 80 bytes
+
+2015-07-01 21:46:46 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix return type of index_entry_offset_search()
+ It's a compare function and may return a negative value,
+ so should for correctness and consistency return a signed
+ integer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751780
+
+2015-07-01 14:12:57 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: refactor handle_next_buffer
+ The goal of this patch is making handle_next_buffer function
+ more readable avoiding unnecesary gotos and adding other
+ cosmetic changes.
+
+2015-07-01 15:40:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpac3pay.c:
+ rtpac3pay: Attach the payload to the output buffer instead of copying it
+ Might also want to produce buffer lists here if needed.
+
+2015-07-01 15:38:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ rtp: Fix indention
+
+2015-07-01 12:37:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-VP8-OPUS.sh:
+ * tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh:
+ rtp: Add examples with VTS/ATS for VP8/OPUS
+ Let's have an example with modern codecs.
+
+2015-06-30 18:11:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
+
+2015-06-30 14:06:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ vp8depay: Don't lock/map every non-keyframe buffer twice
+ Just copy the complete header instead of first looking at the first byte
+ and then at the remaining 10 bytes.
+
+2015-06-29 16:05:44 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: document fallthrough cases
+ Pacify coverity and document fallthrough cases in switch statements.
+ CID #1308948, #1308947, #1308946
+
+2015-06-29 10:36:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout"
+ This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182.
+ If we have multiple immediate timers, we want to first handle the one with the
+ lowest sequence number... which would be broken now.
+ Instead of this we should just use a GSequence for the timers, and have them
+ sorted first by timestamp, and for equal timestamps by sequence number. Then
+ we would always only have to take the very first timer from the list and never
+ have to look at any others.
+
+2015-06-29 10:14:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout
+ If we have lots of such immediate timeouts, we would otherwise have quadratic
+ runtime in the number of timeouts.
+
+2015-06-19 18:01:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: sticky events are sent automatically from the pad
+ No need to send them explicitly from the element
+ https://bugzilla.gnome.org/show_bug.cgi?id=751240
+
+2015-06-19 18:00:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: make sure to push sticky events before adding pad
+ It allows the caps to be set on the pad before being added for
+ dynamic autoplugging to work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751240
+
+2015-06-26 00:05:29 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
+ Enable to use new ntp-time-source property of rtpbin
+ https://bugzilla.gnome.org/show_bug.cgi?id=751496
+
+2015-06-25 23:19:58 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpbin/session: fix description
+ https://bugzilla.gnome.org/show_bug.cgi?id=751496
+
+2015-06-25 10:57:25 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/matroska/matroska-demux.c:
+ * tests/examples/shapewipe/shapewipe-example.c:
+ docs: decodebin2 -> decodebin
+
+2015-06-25 10:47:06 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: update example pipeline
+ Update reference to decodebin2 to decodebin
+
+2015-06-25 10:45:35 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: remove dead assignments
+ Values in fields_required and same_buffer are overwritten before used. Removing
+ assignment
+
+2015-06-25 10:06:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/Makefile.am:
+ * ext/mikmod/Makefile.am:
+ * ext/mikmod/README:
+ * ext/mikmod/drv_gst.c:
+ * ext/mikmod/gstmikmod.c:
+ * ext/mikmod/gstmikmod.h:
+ * ext/mikmod/mikmod_reader.c:
+ * ext/mikmod/mikmod_types.c:
+ * ext/mikmod/mikmod_types.h:
+ * m4/Makefile.am:
+ * m4/libmikmod.m4:
+ * win32/MANIFEST:
+ * win32/vs8/libgstmikmod.vcproj:
+ mikmod: remove ancient unported plugin
+ This hasn't been touched in 11 years, and
+ clearly no one's been missing it.
+
+2015-06-23 20:15:13 +0900 Gilbok Lee <gilbok.lee@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: does not detect orientation
+ Most files don't contain the values for transposing the coordinates
+ back to the positive quadrant so qtdemux was ignoring the rotation
+ tag. To be able to properly handle those files qtdemux will also ignore
+ the transposing values to only detect the rotation using the values
+ abde from the transformation matrix:
+ [a b c]
+ [d e f]
+ [g h i]
+ https://bugzilla.gnome.org/show_bug.cgi?id=738681
+
+2015-06-25 00:04:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.2 ===
+
+2015-06-24 23:30:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.2
+
+2015-06-24 22:56:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-06-24 11:15:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/nl.po:
+ po: Update translations
+
+2015-06-23 18:42:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * tests/check/elements/qtmux.c:
+ qtmux: Correctly test each segments
+ In presence of gaps, qtdemux will emit multiple segments. The
+ second segment start should match the CTTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751361
+
+2015-06-23 17:54:31 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Correctly calculate the elst media start
+ The media start has nothing to do with the shift we have applied
+ but with the value of the first PTS. This is defined as:
+ Dt(0) = 0
+ Ct(0) = Dt(0) + CTTS(0)
+ So the media start is always the first CTTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751361
+
+2015-06-23 11:49:32 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: accumulate previous edts entries into segment.base
+ Allows playing edts editted files with proper synchronization of
+ streams. This patch fixes the regression introduced by
+ bf95f93c0189aa04f18e264b86b6527e431c5d53 that was added to fix
+ segment seeks handling.
+ Having the accumulated_base separated from the main segment.base
+ allows handling both segment seeks and edts editted files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751361
+
+2015-06-23 00:56:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: improve some debug messages
+ Those messages are about the stream, use the pad as the
+ debug object to make it clear from the logs
+ https://bugzilla.gnome.org/show_bug.cgi?id=751361
+
+2015-06-22 22:22:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: store last_dts of the first buffer
+ Buffers need not to start at running-time 0 so the last_dts needs
+ to be the value of the first buffer's dts as it is used to compute
+ the duration of the buffers. If it was left at 0 the first buffer
+ would have a larger duration when it shouldn't
+ https://bugzilla.gnome.org/show_bug.cgi?id=751361
+
+2015-06-23 17:11:57 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: fix possible memory leak
+ when buffer is stored to seektable, and stop gets called due to
+ corrupt flac file, then the seektable is not being released
+ https://bugzilla.gnome.org/show_bug.cgi?id=751364
+
+2015-06-23 16:28:40 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ Revert "splitmuxsink: Mask async-start/done while switching files."
+ This reverts commit d61e5393f110ed482815d77807245d78b52eff46.
+ Causes failures muxing larger GOP sizes for some reason. Reverting
+ while I figure it out
+
+2015-06-18 23:22:06 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Fix startup and shutdown races.
+ Fix 2 startup races when things happen too quickly, and 1
+ at shutdown by holding a ref to the pads in use until the
+ loop functions exit.
+ Handle errors activating file parts and publish them on
+ the bus.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750747
+
+2015-06-18 09:26:13 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Mask async-start/done while switching files.
+ Sometimes, extra async-start/done from the internal sink
+ while the element is still starting up can cause splitmuxsink
+ to stall in PAUSED state when it has been set to PLAYING
+ by the app. Drop the child's async-start/done messages while
+ switching, so they don't cause state changes at the
+ splitmuxsink level.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750747
+
+2015-06-15 16:12:10 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Use gst_video_multiview_guess_half_aspect()
+ Use the gst_video_multiview_guess_half_aspect() utility function
+ to set the half-aspect flag (or not) on stereoscopic frame-packed
+ videos.
+
+2015-06-15 16:10:37 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Move multiview caps calculations, add half-aspect heuristics
+ Move the multiview caps calculations to the configure_stream()
+ function, so the rest of the video info is available, and
+ use the gst_video_multiview_guess_half_aspect() function to
+ determine if the half-aspect flag should be set on frame-packed
+ video.
+
+2015-06-18 16:06:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add cslg support
+ The cslg atom provide information about the DTS shift. This is
+ needed in recent version of ctts atom where the offset can be
+ negative. When cslg is missing, we parse the CTTS table as proposed
+ in the spec to calculate these values.
+ In this implementation, we only need to know the shift. As GStreamer
+ cannot transport negative timestamps, we shift the timestamps forward
+ using that value and adapt the segment to compensate. This patch also
+ removes bogus offset of ctts_soffset, this offset shall be included
+ in the edit list.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751103
+
+2015-06-19 18:37:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * tests/check/elements/qtmux.c:
+ qtmux: Test gaps at start of stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=751242
+
+2015-06-19 18:40:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use PTS to figure-out presence of gaps
+ We need to look at the presentation timestamp in order to conclude if
+ there is a gap at the start of a stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751242
+
+2015-06-19 16:45:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Set edit list to compensate DTS shift
+ We shift DTS forward to avoid negative timestamps which cannot be
+ represented with version 0 of the CTTS table. To stick with that
+ version (backward compatibility), the spec recommend using an
+ edit list entry to move back the presentation time to where it
+ should be.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751242
+
+2015-06-22 14:35:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Insert AVC end of sequence
+ This FLV specific mark is needed to prevent Flow Player (most likely
+ all Flash base player) from going into buffering state when near EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751320
+
+2015-06-22 13:05:29 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ matroska: remove useless check
+ No need to check for context availability while freeing. We are inside
+ inside a code block with a condition that dereferences context.
+ if (context->type == 0 ...
+ https://bugzilla.gnome.org/show_bug.cgi?id=751306
+
+2015-06-22 19:35:57 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/matroska/lzo.c:
+ lzo: fix memory leak
+ the opened file is not being closed during test, which will result
+ in memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751306
+
+2015-06-22 19:30:58 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/mikmod/mikmod_reader.c:
+ mikmod_reader: Possible null pointer dereference:
+ gst_reader variable is being used before actually checking if it
+ allocated properly
+ https://bugzilla.gnome.org/show_bug.cgi?id=751306
+
+2015-06-22 19:45:14 +0900 Sangkyu Park <sk1122.park@samsung.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Minor clean-up
+ 1. Fix the code which is wrong coding style.
+ 2. Fix a typing error of comment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751316
+
+2015-06-22 11:28:13 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Do not try to push NULL buffers
+ If update_receiver_stats() fails, we can't really do anything with this buffer
+ anymore and have to drop it. This happens if there's a big seqnum
+ discontinuity for example.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-06-22 13:10:02 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: trivial cleanup
+ trivial patch to add proper ( while checking for if(G_UNLIKELY())
+ https://bugzilla.gnome.org/show_bug.cgi?id=751306
+
+2015-06-22 13:16:08 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/audioparsers/gstdcaparse.c:
+ dcaparse: initialize size variable
+ size can be used in cleanup without being initialized. Hence
+ setting it to 0 when declaring
+ https://bugzilla.gnome.org/show_bug.cgi?id=751306
+
+2015-06-22 13:13:29 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: initialze bpf variable
+ bpf variable might be used in cleanup without being intialized.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751306
+
+2015-06-19 14:50:59 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ rtprtxqueue: reverse pending list before pushing buffers
+ With this we send the RTX buffers in the same order
+ that they were requested.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751297
+
+2015-06-21 19:22:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Fix DTS validity check
+ This check was up-side-down, causing a bad timestamp at start
+ and then all timestamp being delayed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751298
+
+2015-06-17 15:19:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux_dump.c:
+ * gst/isomp4/qtdemux_dump.h:
+ * gst/isomp4/qtdemux_types.c:
+ cslg: Add Composition Shift Least Greatest Atom
+ This simply add fourcc and dump function for the cslg Atom.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751103
+
+2015-06-17 15:18:38 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ ctts_dump: Fix signess issues
+ It didn't bug, but use correct signess in traces. The number of
+ entries is unsigned while the offset can be signed according to
+ recent spec.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751103
+
+2015-06-16 17:48:08 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6015d26 to f74b2df
+
+2015-06-16 11:43:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: gst_rtp_buffer_ext_timestamp() modifies its first argument, keep a copy around
+
+2015-06-16 10:30:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Compare ext RTP times, not plain RTP time and ext RTP time when calculating elapsed time
+ Otherwise all RTP times after a wraparound would be considered as going
+ backwards, they will always be smaller than the ext RTP time.
+
+2015-06-15 19:25:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: The default rtp-profile should be AVP, not AVPF
+
+2015-06-15 14:32:21 +0900 Sangkyu Park <sk1122.park@samsung.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Minor cleanup
+ 1. Add Null check in 'free_item' function.
+ 2. Fix a typing error of comment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750965
+
+2015-06-12 17:44:51 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ flmux: Make sure best_time is initialized
+
+2015-06-12 23:29:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpsession.h:
+ rtpbin/session: Add new ntp-time-source property and deprecate use-pipeline-clock property
+ The new property allows to select the time source that should be used for the
+ NTP time in RTCP packets. By default it will continue to calculate the NTP
+ timestamp (1900 epoch) based on the realtime clock. Alternatively it can use
+ the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's
+ clock time. The latter is especially useful for synchronizing multiple
+ receivers if all of them share the same clock.
+ If use-pipeline-clock is set to TRUE, it will override the ntp-time-source
+ setting and continue to use the running time plus 70 years. This is only kept
+ for backwards compatibility.
+
+2015-04-07 16:03:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/qtmux.c:
+ tests: qtmux: test for muxing with DTS outside the segment
+ https://bugzilla.gnome.org/show_bug.cgi?id=740575
+
+2015-06-11 17:26:49 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Adjust segment according to ctts offset
+ In presence of a CTTS, the segment start/stop must be offset so
+ the segment start/stop include the PTS. This is needed since the
+ PTS cannot be negative in this format. This fixes issues where the
+ running time of the first buffer isn't at the start.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740575
+
+2015-04-03 20:34:42 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Handle DTS with negative running time
+ As QT works with duration, simply bring back first DTS to 0 and shift
+ forward the PTS of the same amount.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740575
+
+2015-06-10 18:15:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Add negative runtime DTS support
+ This is done by using new feature of the CollectPad clip function
+ which sets the DTS as a gint64 in the collected data. It also simplify
+ the code a bit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740575
+
+2015-06-12 23:06:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Rename some variables and debug output to make more sense
+ Local and remote were mixed up in a few places, and the time we store here is
+ not UNIX time (1970 epoch), but NTP time (1900 epoch) in nanoseconds.
+
+2015-06-12 19:21:10 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: fix latency property query on RemoteIO
+ AudioUnitGetProperty would fail with kParamErr (-50) every time,
+ simply because size wasn't initialized.
+ Now it returns zero latency, but at least it doesn't fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750868
+
+2015-04-07 14:06:16 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * ext/pulse/pulsesrc.c:
+ pulsesrc: Fix mapping of latency parameters to buffer attributes
+
+2015-06-12 01:56:37 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Actually set detected 3D info into output caps.
+ Use the information read from the StereoMode info
+ to configure multiview-mode and multiview-flags in the
+ video caps.
+
+2015-06-11 13:36:54 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Take released-but-not-yet-output bytes into account
+ When deciding whether it's time to switch to a new file, take into
+ account data that's been released for pushing, but hasn't yet
+ been pushed - because downstream is slow or the threads haven't been
+ scheduled.
+ Fixes a race in the unit test and probably in practice - sometimes
+ failing to switch when it should for an extra GOP or two.
+ Also fix a problem in splitmuxsrc where playback sometimes
+ stalls at startup if types are found too quickly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750747
+
+2015-06-11 01:04:51 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ atoms: remove custom gst_buffer_new function in favor of core version
+ Remove a custom specialized version of gst_buffer_new_wrapped by
+ using gst_buffer_new_wrapped_full inside a macro to simplify
+ parameters and give it a more meaningful name.
+ It is only used to create temporary buffers to have its data copied.
+
+2015-06-11 00:14:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ atoms: simplify free form data atoms creation
+ Avoid creating an intermediary buffer or memory area just
+ to copy into an atom's data area.
+
+2015-06-10 22:27:27 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: add AC-3 muxing support
+ Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.
+ One extra feature that was added was the ability to add extension
+ atoms after set_caps as the AC-3 extension atom needs some data
+ that has to be extracted from the stream itself and is not
+ present on caps.
+
+2015-06-10 22:36:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ qtmux: remove unused type MP4S
+
+2015-06-10 22:29:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: remove duplicate attribute value set
+ It is also set a few lines below
+
+2015-06-11 00:22:54 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: Implement basic stereoscopic video support
+ Implement support for the packed video formats WebM
+ uses, not all the values that Matroska might use.
+ In practice, it's really hard to find any samples in the
+ wild of any.
+ Supported in both the muxer and demuxer.
+
+2015-06-10 01:26:15 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_dump.c:
+ * gst/isomp4/qtdemux_dump.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Add basic support for MPEG-A stereoscopic video
+ The MPEG-A format provides an extension to the ISO base media
+ file format to store stereoscopic content encoded with different
+ codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
+ atom declares the presence and storage method for the video.
+ Stereo video information for MPEG-A can also be supplied through
+ the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
+ is not implemented in this patch.
+ Also missing is support for stereo video encoded as separate video tracks
+ for now.
+ Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=611157
+
+2015-06-02 16:15:35 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: Add tls-database property
+ https://bugzilla.gnome.org/show_bug.cgi?id=750298
+
+2015-06-10 14:33:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstasteriskh263.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
+ The mix between all these in the RTP code is confusing, let's try to be
+ consistent.
+
+2015-06-10 14:49:50 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpmanager: clarify negative lost packets in stats
+ Also:
+ - Move notes on units before field documentation.
+ - Unify documentation style.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750653
+
+2015-06-10 06:38:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: fix getter of "ssl-use-system-ca-file"
+ https://bugzilla.gnome.org/show_bug.cgi?id=750298
+
+2015-06-10 09:49:47 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix reverse playback
+ When performing seek, segment->start is being updated with desired_offset,
+ but in case of reverse playback segment->start should be 0 and
+ segment->stop should be updated with desired offset.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750675
+
+2015-01-21 18:09:03 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ gstv4l2bufferpool: handle -EPIPE from DQBUF to signal EOS
+ The V4L2 decoder signals EOS by returning -EPIPE from DQBUF after the
+ last buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743338
+
+2015-06-06 21:09:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add a GTlsInteraction property
+ It can be used for TLS client authentication.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750471
+
+2015-01-09 11:36:11 +0100 Enrico Jorns <ejo@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2: Allow scaling in the v4l2*convert element
+ This is inspired of videoscale and videoconvert elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742917
+
+2015-06-09 19:02:55 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpmanager: document units of stats and arguments
+ Also, minor spelling and style corrections.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750653
+
+2015-06-09 14:42:27 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * Makefile.am:
+ cruft: add the obsolete tmpl dir to cruft-dirs
+
+2015-06-09 11:30:22 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From d9a3353 to 6015d26
+
+2015-06-09 07:04:07 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Fix common version
+ Was accidently downgraded by 87a4884acd8655a6591d735a1d944ecb5ea3de16
+
+2015-06-08 19:11:41 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Also set colorimetry on output devices
+ This completes the code that set the colorimetry on output
+ device.
+
+2015-06-08 19:10:34 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Add missing SMTP240M matrix
+ This is missing in the doc, but was in the header.
+
+2015-06-08 23:00:16 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom/goom_core.c:
+ goom: possible uninitialized variables warning
+ Build fails with the latest snapshot of gcc-4.9 because param1 and param2 might
+ possibly be used uninitialized. They are set depending on the cases of a switch
+ statement and the compiler sees this as not a complete guarantee.
+ Set them to 0 if the switch statement falls down to the default case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750566#c6
+
+2015-06-08 17:24:38 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Fully implement colorimetry support
+ This fixes wrong mapping for sRGB as in GStreamer sRGB correctly
+ apply to RGB formats, while in V4L2 it's an alias for sYCC. Also
+ add support for the new quantization (range), ycbcr_encoding (matrix)
+ and xfer_func (transfer) enumeration.
+
+2015-06-08 17:01:15 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/ext/types-compat.h:
+ * sys/v4l2/ext/v4l2-common.h:
+ * sys/v4l2/ext/v4l2-controls.h:
+ * sys/v4l2/ext/videodev2.h:
+ v4l2: Update kernel headers to latest from media tree
+ This is the latest from media tree. This should enable more development
+ of the v4l2 elements. This includes new flags requires to fix draining
+ path in decoder, colorimetry and much more.
+
+2015-06-08 23:07:55 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From d37af32 to d9a3353
+
+2015-06-08 19:42:30 +0100 Chris Clayton <chris2553@googlemail.com>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8depay: potential access beyond end of array
+ Compiling (with gcc-4.9-20150603) produces an error because of an access beyond
+ the end of an array. This patch fixes the error by initializing the loop
+ control/array index variable (i) to 1 and returning i - 1 when a match is found.
+ Also, because the values stored in the array increase in value as the index
+ increases, the >= test unnecessary, so it is removed.
+
+2015-04-30 02:52:58 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Don't accumulate more than 2 GOPs
+ Don't allow large amounts of data to queue up - we only need
+ the GOP we're writing, and the GOP we're accumulating.
+
+2015-04-16 10:44:49 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: fsync after sending updates in robust mode
+ Use the new GstBuffer SYNC_AFTER flag to trigger an fsync
+ after updating the moov or mdat atom, and after updating the free
+ atom to make it visible.
+
+2015-04-03 00:57:20 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: Only set moov header into streamheader at EOS
+ Only update the moov header into the caps if it's the finalised
+ moov at EOS time. Avoids posting a bogus moov at startup and
+ repeated updates in robust-recording mode
+
+2015-04-03 01:44:15 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/qtmux.c:
+ tests: Update mp4 mux test for mdat placeholder change
+ The mp4 muxer now writes a place-holder mdat as a free
+ atom followed by a 0-byte mdat that covers the rest of the
+ file, making it possible to rewrite it as 64-bit, or leave
+ it as-is if nothing else is written afterward
+
+2015-04-01 11:15:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/atomsrecovery.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ isomp4: Implement robust muxing using ping-pong strategy
+ Implement a robust recording mode, where the output
+ file is always in a playable state, seeking and rewriting
+ the moov header at a configurable interval. Rewriting
+ moov is done using reserved space at the start of
+ the file, and a ping-pong strategy where the moov
+ is replaced atomically so it's never invalid.
+ Track when tags have actually changed, and don't write them into
+ the moov unless they've changed. Clear any existing tags when
+ re-writing them, so we can do progressive moov updating in robust
+ recording mode.
+ Write placeholder mdat as a free atom plus a 32-bit mdat
+ with '0' size, which means "rest of the file" in the spec.
+ Re-write it later to a full 64-bit extended size atom if needed.
+
+2015-04-01 00:58:52 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ isomp4: Update edit list when re-writing moov
+ Correctly update any edit lists each time the moov is recalculated,
+ updating existing table entries if they already exist instead of just
+ adding new ones.
+
+2015-04-08 01:41:18 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: Remove an extra bracket in a comment.
+
+2015-03-19 20:29:44 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Protect total_duration state variable with the object lock.
+ Prevent deadlocks from downstream querying duration from the streaming thread.
+
+2015-06-07 23:06:20 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 21ba2e5 to d37af32
+
+2015-06-07 19:24:20 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom/gstaudiovisualizer.c:
+ goom: clean dereferences of private structure
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-07 19:20:04 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/gstaudiovisualizer.c:
+ goom2k1: clean dereferences of private structure
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-07 17:32:01 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From c408583 to 21ba2e5
+
+2015-06-07 17:01:37 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/Makefile.am:
+ docs: remove variables that we define in the snippet from common
+ This is syncing our Makefile.am with upstream gtkdoc.
+
+2015-06-07 17:16:19 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From d676993 to c408583
+
+2015-06-07 16:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.1 ===
+
+2015-06-07 10:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * gst/deinterlace/tvtime-dist.c:
+ * gst/videomixer/videomixerorc-dist.c:
+ * win32/common/config.h:
+ Release 1.5.1
+
+2015-06-07 10:38:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-06-07 10:32:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property
+
+2015-06-07 09:35:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: Update translations
+
+2015-06-05 15:32:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Don't warn when optional CID are not implement
+ gst_v4l2_get_attributre() shall only be used when the CID is expected
+ to be supported. Otherwise, we get unwanted warning posted to the bus.
+
+2015-06-05 16:43:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc
+ https://bugzilla.gnome.org/show_bug.cgi?id=749581
+
+2015-06-04 14:18:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/interleave/interleave.c:
+ interleave: error when channel-positions-from-input=False
+ self->channels is being incremented only when
+ channel-positions-from-input is set as TRUE. So in case of FALSE
+ self->func is not set and hence creating assertion error.
+ Hence removing the condition to increment self->channels.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744211
+
+2015-06-05 10:33:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Add support for receiving reduced size RTCP
+ It worked before but gave warnings, now we just ignore RTCP
+ packets that don't start with a SR. As all we're interested
+ in here are SRs.
+
+2015-06-03 12:22:42 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ rtpssrcdemux: Add support for reduce size rtcp
+ According to RFC 5506, reduce size packages can be sent, this
+ packages may not be compound, so we need to add support for
+ getting ssrc from other types of packages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750327
+
+2015-06-03 13:14:44 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add support for receiving reduced size rtcp
+ See RFC 5506
+ https://bugzilla.gnome.org/show_bug.cgi?id=750332
+
+2015-06-04 16:09:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Add support for channel configurations 11, 12 and 14 and 7 actually has 8 channels
+ ISO/IEC 14496-3:2009/PDAM 4 added 11, 12 and 14.
+
+2015-06-03 08:57:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtp/gstasteriskh263.c:
+ asteriskh263: Un-rank clashing depayloader
+ This depayloader clash with the standard one for H263p. It produces an
+ H263p stream with a modified header. It uses encoding-name that is the
+ same as H263p (H263-1998) though the resulting ES is not decodable or
+ parsable in GStreamer, making it unsuable in dynamic pipeline. This
+ patch unrank this specialized depayloader since it can only be used in
+ custom pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739935
+
+2015-06-02 18:09:48 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: remove variables not needed anymore
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-02 17:52:46 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/Makefile.am:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: rebase to use the audiovisualizer class
+ Rebase to have goom2k1 using the common GstAudioVisualizer class
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-02 17:29:36 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom/Makefile.am:
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom/gstgoom.c:
+ * gst/goom/gstgoom.h:
+ goom: rebase to use the audiovisualizer class
+
+2015-06-02 16:27:24 +0200 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/aacparse.c:
+ * tests/check/elements/ac3parse.c:
+ * tests/check/elements/apev2mux.c:
+ * tests/check/elements/aspectratiocrop.c:
+ * tests/check/elements/audioamplify.c:
+ * tests/check/elements/audiochebband.c:
+ * tests/check/elements/audiocheblimit.c:
+ * tests/check/elements/audiodynamic.c:
+ * tests/check/elements/audioinvert.c:
+ * tests/check/elements/audiowsincband.c:
+ * tests/check/elements/audiowsinclimit.c:
+ * tests/check/elements/avimux.c:
+ * tests/check/elements/equalizer.c:
+ * tests/check/elements/flacparse.c:
+ * tests/check/elements/id3v2mux.c:
+ * tests/check/elements/jpegdec.c:
+ * tests/check/elements/jpegenc.c:
+ * tests/check/elements/matroskamux.c:
+ * tests/check/elements/mpegaudioparse.c:
+ * tests/check/elements/rganalysis.c:
+ * tests/check/elements/rglimiter.c:
+ * tests/check/elements/rgvolume.c:
+ * tests/check/elements/rtpbin.c:
+ * tests/check/elements/rtpsession.c:
+ * tests/check/elements/spectrum.c:
+ * tests/check/elements/videobox.c:
+ * tests/check/elements/videocrop.c:
+ * tests/check/elements/videofilter.c:
+ * tests/check/elements/wavpackdec.c:
+ * tests/check/elements/wavpackenc.c:
+ * tests/check/elements/wavpackparse.c:
+ * tests/check/elements/y4menc.c:
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/check/pipelines/tagschecking.c:
+ * tests/check/pipelines/wavpack.c:
+ check: Use GST_CHECK_MAIN () macro everywhere
+ Makes source code smaller, and ensures we go through common initialization
+ path (like the one that sets up XML unit test output ...)
+
+2015-05-26 14:47:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Only schedule a timer when we actually have to send RTCP
+ Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
+ RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
+ feedback is actually pending and no regular RTCP has to be sent).
+ This improves CPU usage and battery life quite a lot.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-22 13:44:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Remove useless goto
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-21 12:54:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-H264-rtx.sh:
+ * tests/examples/rtp/client-rtpaux.c:
+ * tests/examples/rtp/server-VTS-H264-rtx.sh:
+ * tests/examples/rtp/server-rtpaux.c:
+ examples: Set RTP profile to AVPF for rtpaux examples
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 16:41:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Set RTP profile on the rtpsession objects
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-21 14:13:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 11:51:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Only put RRs and full SDES into regular RTCP packets
+ If we may suppress the packet due to the rules of RFC4585 (i.e. when
+ below the t-rr-int), we can send a smaller RTCP packet without RRs
+ and full SDES. In theory we could even send a minimal RTCP packet
+ according to RFC5506, but we don't support that yet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 13:51:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Keep track of tp/tn and t_rr_last separately
+ Otherwise we can't properly schedule RTCP in feedback profiles as we need to
+ distinguish the time when we last checked for sending RTCP (tp) but might have
+ suppressed it, and the time when we last actually sent a non-early RTCP
+ packet.
+ This together with the other changes should now properly implement RTCP
+ scheduling according to RFC4585, and especially allow us to send feedback
+ packets a lot if needed but only send regular RTCP packets every once in a
+ while.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 11:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
+ And modify our RTCP scheduling algorithm accordingly. We now can send more
+ RTCP packets if needed for feedback, but will throttle full RTCP packets by
+ rtcp-min-interval (t-rr-int from RFC4585).
+ In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
+ statically set to 1s or 0s by RFC4585. Tmin defines how often we should
+ send RTCP packets at most.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-30 17:41:05 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/law/mulaw-decode.c:
+ mulawdec: Let baseclass estimate bitrate
+ This makes playback directly from a file work with the right caps.
+
+2015-05-27 16:31:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstdynudpsink.h:
+ dynudpsink: keep GCancellable fd around instead of re-creating it constantly
+ And create it only when starting the element.
+
+2015-05-27 15:55:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly
+ Otherwise we constantly create/close event file descriptors,
+ every time we call g_socket_condition_timed_wait() or
+ g_socket_send_message(s)(), i.e. a lot. Which is not
+ particularly good for performance.
+ Can't create GCancellable in ::start() here because it's used
+ in client_new() which may be called via the add-client action
+ signal which may be called before the element is up and running.
+
+2015-05-19 18:13:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: keep GCancellable fd around instead of re-creating it constantly
+ Otherwise we constantly create/close event file descriptors,
+ every single time we call g_socket_condition_timed_wait() or
+ g_socket_receive_message(), i.e. twice per packet received!
+ This was not particularly good for performance.
+ Also only create GCancellable on start-up.
+
+2015-05-26 15:33:37 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: overwritten value assignment
+ curpos is set and immediately after, set again. Remove the redundant
+ assignment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749909
+
+2015-05-23 13:47:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: don't shadow existing outbuf variable
+ And fix unref of the wrong one which will contain NULL
+ in an error code path.
+
+2015-05-23 13:23:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawdepay.h:
+ rtpvrawdepay: map/unmap output frame only once, not for every input packet
+ Map output buffer after creating it and keep it mapped
+ until we're done with it instead of mapping/unmapping
+ it for every single input buffer.
+
+2015-05-25 08:47:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: remove fixme from 2006
+ It has been verified by use over time.
+
+2015-05-23 14:36:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix reverse playback of fragmented media
+ qtdemux creates a samples array and gets the timestamps for buffers by
+ accumulating their durations. When doing reverse playback of fragments,
+ accumulating samples will lead to wrong timestamps as the timestamps
+ should go decreasing from fragment to fragment and the accumulation
+ will produce wrong results.
+ In this case, when receiving a discont for fragmented reverse playback,
+ the previous samples information should be flushed before new data
+ is processed.
+
+2015-05-23 01:03:18 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/multifile/gstsplitfilesrc.c:
+ splitfilesrc: Implement binary search in find_part_for_offset
+ Implement binary search using gst_util_array_binary_search
+ https://bugzilla.gnome.org/show_bug.cgi?id=749690
+
+2015-05-21 13:26:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Don't crash if we receive FIR/PLI from a source we don't know
+
+2015-05-21 09:35:58 +0200 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Fix collection of statistics
+ Stats should be collected on the media rtp source not in the
+ sender one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749669
+
+2015-04-20 10:07:30 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ multifilesink: Add a new max-duration file switching mode
+ This new mode ensures that files will never exceed a certain duration
+ based on incoming buffer PTS (and duration if present)
+ Note:
+ * You need timestamped buffers (duh). If some of the incoming buffers don't
+ have PTS, then it will just accept them in the current file
+
+2015-04-17 16:18:32 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: streamline the file-switch code a bit
+ Use the same functions regardless of the mode we are using
+
+2015-04-02 13:35:18 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ multifilesink: add "aggregate-gops" property to process GOPs as a whole
+ This property can be used in combination with next-file=max-size
+ (and perhaps a future next-file=max-duration) to make sure that
+ each file part starts cleanly with a key frame and the appropriate headers.
+ In order for this property to work correctly, upstream elements should make
+ sure than any headers that need to be written in a standalone file are:
+ 1) in the streamheader caps field
+ 2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER
+ that are just before the keyframe buffer
+ This is useful for MPEG-TS/MPEG-PS file segmenting in
+ combination with mpegtsmux or mpegpsmux.
+ Original patch by: Tim-Philipp Müller <tim@centricular.com>
+
+2015-05-20 16:37:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Use single-include header for the RTSP library
+
+2014-10-24 23:47:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ udp: don't use soon-to-be-deprecated g_cancellable_reset()
+ From the API documentation: "Note that it is generally not
+ a good idea to reuse an existing cancellable for more
+ operations after it has been cancelled once, as this
+ function might tempt you to do. The recommended practice
+ is to drop the reference to a cancellable after cancelling
+ it, and let it die with the outstanding async operations.
+ You should create a fresh cancellable for further async
+ operations."
+ https://bugzilla.gnome.org/show_bug.cgi?id=739132
+
+2015-05-18 20:13:01 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/audiofx/audiochebband.c:
+ * gst/audiofx/audiocheblimit.c:
+ * gst/cutter/gstcutter.c:
+ * gst/equalizer/gstiirequalizernbands.c:
+ * gst/multifile/gstmultifilesink.c:
+ Revert "doc: Workaround gtkdoc issue"
+ This reverts commit 1797c8f8b12d7f4c7a9444c94f34f4d08ec85945.
+ This is fixed by the gtk-doc 1.23 release.
+ <para> cannot contain <refsect2>:
+ http://www.docbook.org/tdg/en/html/para.html
+ http://www.docbook.org/tdg/en/html/refsect2.html
+
+2015-05-18 16:40:21 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtp/gstrtpg726pay.c:
+ rtpg726pay: fix caps leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=749544
+
+2015-05-18 16:34:13 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtp/gstrtpg726depay.c:
+ rtpg726depay: don't leak input buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=749543
+
+2015-05-18 17:38:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Queue bad packets instead of dropping them
+ So we can send them out once we found the next, consecutive sequence number in
+ case one is following.
+
+2015-05-18 17:38:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Use g_queue_foreach() to unref all buffers in queues
+
+2015-05-18 17:19:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Refactor seqnum comparison code a bit
+
+2015-05-18 17:08:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Allow sequence number wraparound during probation
+
+2015-05-18 17:07:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Make sequence number comparison code more readable
+ ... by using gst_rtp_buffer_compare_seqnum() and signed integers
+ instead of implictly using effects of integer over/underflows.
+
+2015-04-22 18:54:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
+ It might just be a late retransmission or spurious packet from elsewhere, but
+ resetting everything would mean that we will cause a noticeable hickup. Let's
+ get some confidence first that the sequence numbers changed for whatever
+ reason.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747922
+
+2015-05-16 23:37:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/audiofx/audiochebband.c:
+ * gst/audiofx/audiocheblimit.c:
+ * gst/cutter/gstcutter.c:
+ * gst/equalizer/gstiirequalizernbands.c:
+ * gst/multifile/gstmultifilesink.c:
+ doc: Workaround gtkdoc issue
+ With gtkdoc 1.22, the XML generator fails when a itemizedlist is
+ followed by a refsect2. Workaround the issue by wrapping the
+ refsect2 into para.
+
+2015-01-23 13:57:40 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: avoid wrong warnings on unknown node types
+ Add 'name' and 'mean' fourccs, as we handle them. Right now each use would
+ trigger a warning.
+
+2015-05-08 19:13:00 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726depay.h:
+ rtpg726depay: add block_align to output caps
+ It is needed to correctly negotiate caps with matroskamux
+ and most other muxers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749129
+
+2015-05-12 13:41:58 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Fix time-domain convolution with >1 channels
+ input_samples is the number of frames, but we used it as the number of
+ samples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747204
+
+2015-05-12 12:13:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp[89]enc: Properly convert between GStreamer and encoder timebase
+ ... by switching numerator and denominator when scaling.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749122
+
+2015-05-11 13:33:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp[89]enc: Don't set timebase from the framerate
+ The framerate very often is just an indication of the ideal framerate, not the
+ actual framerate of the stream. By just using the framerate, we confuse the
+ rate control algorithm algorithm as multiple frames will map to the same PTS
+ or have durations of 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749122
+
+2015-05-10 14:21:04 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * tests/check/elements/wavpackparse.c:
+ tests: wavpackparse: fix unit test
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=738237
+
+2015-05-10 11:05:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/shout2/gstshout2.c:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpL24pay.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/pipelines/wavenc.c:
+ * tests/examples/rtp/client-PCMA.c:
+ * tests/examples/rtp/server-alsasrc-PCMA.c:
+ docs: update example pipelines in element docs
+ Mostly gst-launch -> gst-launch-1.0
+ Use autovideosink/autoaudiosink more often.
+ Sprinkle some converters here and there.
+
+2015-05-09 19:48:55 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES.in
+ https://bugzilla.gnome.org/show_bug.cgi?id=749163
+
+2015-05-10 10:52:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: minor error message clean-up
+ Don't put filename in error message shown to user.
+
+2015-05-07 16:25:36 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: fix buffer leak when stored to seektable
+ Fix a leak with the
+ validate.file.playback.change_state_intensive.samples_multimedia_cx_flac_Yesterday_flac
+ scenario.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749072
+
+2015-05-07 17:10:37 +0900 Paul Hyunil <paul.hyunil@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix example pipeline in docs
+ The gst-launch script for example launch line to test qtdemux is
+ missing a queue before the decodebins, otherwise the gst-launch-1.0
+ command won't work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749054
+
+2015-05-07 14:51:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
+ This reverts commit d22ec496328e6ba8edbf2d071d5608b2af2831e8.
+ Application code might expect that it only gets external sources on those
+ signals, and get confused by this. If anything we would need to add new
+ signals.
+
+2015-03-25 15:27:34 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
+ Without this it seems impossible for an application to easily get notified
+ about the internal ssrcs that are created, e.g. sender sources, and also
+ to know when they are active and produce RTCP packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-05-04 19:26:14 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: fix frame leaks in handle_frame() implementation
+ handle_frame() is supposed to consume @frame, so if we don't call
+ gst_video_decoder_drop_frame() or gst_video_decoder_finish_frame() we have to
+ release it manually.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748909
+
+2015-05-04 16:50:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix up last commit
+
+2015-05-04 16:46:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Only do RTX when using a feedback profile
+
+2015-05-04 13:50:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: The stats min_interval is in seconds, not nanoseconds
+ We have to scale it to compare it against our clock times.
+
+2015-05-04 11:38:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Only return TRUE if early feedback was requested already and it's early enough
+
+2015-04-30 15:42:34 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroska: remove unused property enum items
+
+2015-04-30 12:13:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix buffer leak on eos in push mode
+ Based on patch by Guillaume Desmottes.
+ scenario: validate.http.playback.seek_with_stop.raw_h264_1_mp4
+ https://bugzilla.gnome.org/show_bug.cgi?id=748617
+
+2015-04-29 19:41:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check for sizes of the rdrf (redirect) atom before accessing the data and use g_strndup() instead of g_strdup()
+ Thanks to Ralph Giles for reporting this.
+
+2015-04-29 15:52:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Only enable retransmissions if there is retransmission info in the SDP
+ Otherwise we're going to send early RTCP and NACKs in non-feedback sessions
+ too, which will confuse servers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748627
+
+2015-02-11 18:09:24 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: extract recording time
+ Extracts the recorded time of the dv file from
+ the metadata and puts it into the global tags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743657
+
+2015-04-28 15:59:25 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix seek event leak
+ gst_matroska_demux_handle_seek_event() doesn't consume the
+ event so we have to unref it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748584
+
+2015-04-28 15:42:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Send pending tags when adding a new pad
+ We might've parsed those tags before already and tried to push them to
+ non-existing pads before. Now let's do it for real.
+
+2015-04-23 18:57:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
+ We need to convert the size to bits for our calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-23 18:53:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-14 18:41:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-23 18:49:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-27 16:36:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix RTX unit test
+ The calculations were a bit off everywhere, even before the changes done
+ recently to the delay for RTX of expected future packets. It only worked by
+ accident, but now the calculations are all correct again. Hopefully.
+
+2015-04-27 11:22:11 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/avi/gstavimux.c:
+ * gst/debugutils/breakmydata.c:
+ * gst/debugutils/cpureport.c:
+ * gst/debugutils/gstnavseek.c:
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ * gst/flv/gstindex.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/id3demux/gstid3demux.c:
+ * gst/isomp4/gstrtpxqtdepay.c:
+ * gst/law/mulaw-decode.c:
+ * gst/law/mulaw-encode.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/multifile/gstmultifilesrc.c:
+ * gst/multipart/multipartmux.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videofilter/gstvideotemplate.c:
+ * gst/y4m/gsty4mencode.c:
+ Rename property enums from ARG_ to PROP_
+ Property enum items should be named PROP_ for consistency and readability.
+
+2015-04-25 02:49:58 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix "stats" property docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=748436
+
+2015-04-26 17:54:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Android.mk:
+ * gst/alpha/Makefile.am:
+ * gst/apetag/Makefile.am:
+ * gst/audiofx/Makefile.am:
+ * gst/auparse/Makefile.am:
+ * gst/autodetect/Makefile.am:
+ * gst/avi/Makefile.am:
+ * gst/cutter/Makefile.am:
+ * gst/debugutils/Makefile.am:
+ * gst/deinterlace/Makefile.am:
+ * gst/dtmf/Makefile.am:
+ * gst/effectv/Makefile.am:
+ * gst/equalizer/Makefile.am:
+ * gst/flv/Makefile.am:
+ * gst/flx/Makefile.am:
+ * gst/goom/Makefile.am:
+ * gst/goom2k1/Makefile.am:
+ * gst/icydemux/Makefile.am:
+ * gst/id3demux/Makefile.am:
+ * gst/imagefreeze/Makefile.am:
+ * gst/interleave/Makefile.am:
+ * gst/isomp4/Makefile.am:
+ * gst/law/Makefile.am:
+ * gst/level/Makefile.am:
+ * gst/matroska/Makefile.am:
+ * gst/monoscope/Makefile.am:
+ * gst/multifile/Makefile.am:
+ * gst/multipart/Makefile.am:
+ * gst/replaygain/Makefile.am:
+ * gst/rtp/Makefile.am:
+ * gst/rtpmanager/Makefile.am:
+ * gst/rtsp/Makefile.am:
+ * gst/shapewipe/Makefile.am:
+ * gst/smpte/Makefile.am:
+ * gst/spectrum/Makefile.am:
+ * gst/udp/Makefile.am:
+ * gst/videobox/Makefile.am:
+ * gst/videocrop/Makefile.am:
+ * gst/videofilter/Makefile.am:
+ * gst/videomixer/Makefile.am:
+ * gst/wavenc/Makefile.am:
+ * gst/wavparse/Makefile.am:
+ * gst/y4m/Makefile.am:
+ Remove obsolete Android build cruft
+ This is not needed any longer.
+
+2015-04-24 13:55:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ videocrop: print the property values when set
+ Instead of printing the currently used values. The log is meant
+ to show what the properties changed to, not what is being currently
+ used.
+
+2015-04-24 17:01:10 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/alpha/gstalpha.c:
+ * gst/audiofx/audiokaraoke.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/multifile/gstmultifilesink.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726pay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/smpte/gstsmpte.c:
+ * gst/smpte/gstsmptealpha.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ remove unused enum items PROP_LAST
+ This were probably added to the enums due to cargo cult programming and are
+ unused. Removing them.
+
+2015-04-24 00:30:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/level/gstlevel.c:
+ level: fix infinite loop for very low interval values
+ https://bugzilla.gnome.org/show_bug.cgi?id=745515
+
+2015-04-23 16:08:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 16:08:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 16:06:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Update .gitignore
+
+2015-04-23 09:55:59 +0200 Jesper Larsen <knorr.jesper@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix RTCP caps leak
+ https://bugzilla.gnome.org//show_bug.cgi?id=748353
+
+2015-04-22 20:24:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
+ We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
+ delay. If jitter is very low, this should prevent unnecessary retransmission
+ requests to some degree.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748041
+
+2015-04-22 19:41:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
+ https://bugzilla.gnome.org/show_bug.cgi?id=748041
+
+2015-04-13 11:20:40 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Add "rtx-next-seqnum" property
+ If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
+ future packets based on when they are estimated to arrive.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=748041
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2015-04-22 19:29:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtxreceive: Put debug output for retransmission requests at the right place
+ Before it was only ever printed once for every time a ssrc was associated with
+ a specific stream.
+
+2015-04-22 18:05:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: don't add the same interlace mode twice
+ Some drivers modify the interlace mode to progressive, no matter what
+ input you give them, make sure that we don't add the same interlace mode
+ twice.
+
+2015-04-21 16:34:21 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/equalizer/gstiirequalizer.c:
+ equalizer: fix dynamic changes on bands
+ When we are in passthrough, the transform function doesn't run and if the
+ passthrough check is in this function it will never be deactivated. Fix this by
+ checking directly whenever a gain is changed.
+ Also set the passthrough to TRUE at init because the gains default to 0, so we
+ can passthrough until any gain property is changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748068
+
+2015-04-22 10:30:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * INSTALL:
+ Remove INSTALL file
+ autotools automatically generate this, and when using different versions
+ for autogen.sh there will always be changes to a file tracked by git.
+
+2015-04-22 10:30:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * LICENSE_readme:
+ Remove LICENSE_readme
+ It's completely outdated and just confusing, better if people are
+ forced to look at the actual code in question than trusting this file.
+
+2015-04-21 15:21:33 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: cast unused return to void
+ Quell unchecked return value defect by casting the return value to void and
+ making it explicit it is going to be ignored.
+ CID #206031
+
+2015-04-17 13:08:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vpx/gstvp8dec.c:
+ vp8dec: optimize vpx image to gstbuffer copy when strides match
+ Solving this FIXME. Copy the full plane when strides are the same
+
+2015-04-16 15:11:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: optimize vpx image to gstbuffer copy when strides match
+ Solving this FIXME. Copy the full plane when strides are the same
+
+2015-04-17 13:32:54 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstac3parse.c:
+ ac3parse: fix memory leak
+
+2015-04-17 06:51:46 +0000 Alex O'Konski <alexanderokonski@gmail.com>
+
+ * gst/icydemux/gsticydemux.c:
+ icydemux: Fix segfault if metadata-interval is 0
+ Prevents an extra unref of GstBuffer when passing a non-icy stream through
+ icydemux with metadata-interval set to 0.
+ Reproducible with:
+ gst-launch-1.0 filesrc location=~/testsong.mp3 ! \
+ 'application/x-icy,metadata-interval=(int)0' ! icydemux ! decodebin ! wavenc ! \
+ filesink location=~/testsong.wav
+ https://bugzilla.gnome.org/show_bug.cgi?id=748024
+
+2015-04-17 11:54:23 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audioamplify.c:
+ * gst/audiofx/audiodynamic.c:
+ audiofx: fix typo in example pipelines
+ Fix typo in example pipelines
+ https://bugzilla.gnome.org/show_bug.cgi?id=748022
+
+2015-04-15 18:22:37 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: fix spelling in debug message
+ https://bugzilla.gnome.org//show_bug.cgi?id=747936
+
+2015-04-16 16:33:44 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: selectable amount of bands in equalizer demo
+ Adding an option in the equalizer demo to make the number of bands selectable.
+
+2015-04-16 15:31:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
+ https://bugzilla.gnome.org/show_bug.cgi?id=747394
+
+2015-04-06 12:56:50 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Track RTX ssrc caps
+ This is needed so that we can generate SR for RTX stream correctly (the
+ clock rate is required).
+ https://bugzilla.gnome.org/show_bug.cgi?id=747394
+
+2015-04-14 13:56:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=747394
+
+2015-04-16 16:01:50 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: switch equalizer demo to play from uri
+ Switch the equalizer-nbands demo to use uridecodebin, so users can listen to
+ something more pleasant than white noise. If anybody misses the white noise
+ a uri handler to audiotestsrc can be used.
+
+2015-04-16 11:17:38 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: improve readability of equalizer demo
+ Rename variable name to make it more readable, add comments for the three
+ scales created per block, and set the window title.
+
+2015-04-15 17:32:37 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: add missing license header for equalizer demo
+
+2015-04-16 13:09:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix tag list leaks on error paths
+
+2015-04-16 12:23:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix tag list leak on unknown stream type
+
+2015-04-09 13:19:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/gst-plugins-good.supp:
+ suppressions: ignore an apparent bug in strtod
+ A buffer overread.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747554
+
+2015-04-15 11:07:27 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: do not access property variable without the object lock, use the local stack copy instead
+
+2015-04-14 18:45:44 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: add probe on the multiqueue's sink pad instead of the ghost pad
+ because _release_pad tries to release it from ctx->sinkpad, which is
+ multiqueue's sink pad, and currently fails because the probe is not
+ installed there
+
+2015-04-14 19:08:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtx*: Fix typos
+
+2015-04-14 17:24:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet
+
+2015-04-14 16:27:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Improve debug output a bit if we can't allow early feedback
+
+2015-04-07 18:00:53 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpvp8depay: When dropping intra packet, request keyframe
+ https://bugzilla.gnome.org/show_bug.cgi?id=747208
+
+2015-04-13 20:25:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO
+ This also happens in the very beginning when we receive the first packet, a
+ warning would be very confusing here. In all places where we should warn about
+ this, we would've printed a warning already before.
+
+2015-04-02 13:26:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: minor docs improvement
+
+2014-11-06 12:08:03 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Add "rtx-max-retries" property
+ This property allows to limit the maximum number of retransmission
+ for a specific packet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2014-11-04 15:00:52 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix expected_dts calc in calculate_expected
+ Right above we consider lost_packet packets, each of them having duration,
+ as lost and triggered their timers immediately. Below we use expected_dts
+ to schedule retransmission or schedule lost timers for the packets that
+ come after expected_dts.
+ As we just triggered lost_packets packets as lost, there's no point in
+ scheduling new timers for them and we can just skip over all lost packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2015-03-20 18:21:57 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer
+ Resetting the jitterbuffer drops all packets and other things, and will cause
+ a discontinuity in the packets received by the depayloaders. They should now
+ also flush anything they had pending as the new data will start at a different
+ position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2015-04-10 09:17:26 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Update segment.start after key-unit seek
+ When doing key uint seek, qtdemux calls gst_qtdemux_adjust_seek
+ to get proper offset. And then this offset is set to
+ segment.position and segment.time in gst_qtdemux_perform_seek but
+ segment.start is not updated.
+ After that, application sends segment query,
+ qtdemux sets start and stop to query using gst_segment_to_stream_time. Due
+ to the wrong value in segment.start, the stop position is smaller than
+ it should.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746822
+
+2015-04-07 16:12:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: remove useless variable do_pts
+ We always write the CTTS in qtmux. Ideally we only want to do that
+ for streams that need DTS, it should be present on the track information
+ rather than be decided based on each buffer
+
+2015-04-07 00:53:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: remove subtraction that makes PTS/DTS start from 0
+ As qt uses durations, it doesn't matter, only the difference
+ between consecutive buffers is important. Also, collectpads
+ already replaces PTS/DTS with the running times for them.
+
+2015-04-06 22:36:43 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/qtmux.c:
+ tests: qtmux: add tests to verify it handles non-0 segments
+ Both input streams in this test have a segment.start = 10s, so
+ output should start from 0 anyway.
+ Another test has both starting at non-0 segments, but the running
+ time of both streams should still start from 0
+
+2015-04-06 20:03:19 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/qtmux.c:
+ tests: qtmux: simple muxing test
+ Adds a new simple test that verifies that data is properly muxed
+ and preserved. PTS, DTS, duration and caps are verified.
+
+2015-04-10 10:59:26 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.h:
+ smpte: remove unused fields
+ Remove the fields - format and fps from smpte
+ as they are unused.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747597
+
+2015-04-10 10:29:47 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/alpha.c:
+ tests: add test suite for alpha
+ Added test suite for alpha element with test cases
+ 1. alpha
+ 2. chroma keying
+ https://bugzilla.gnome.org/show_bug.cgi?id=747595
+
+2015-04-09 12:58:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/gst-plugins-good.supp:
+ suppressions: add a well known zlib inflate bug
+
+2015-04-09 12:58:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix mutex leak
+
+2015-04-09 12:58:04 +1000 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/rtprtx.c:
+ tests: Fix rtprtx test by handling buffer lists
+ Commit #1018aa made rtprtxsend handle buffer lists, breaking
+ the test which probes for buffers, but not buffer lists.
+ Use a utility function to run the probe callback on each buffer
+ in the list in turn and remove any buffers that are dropped.
+
+2015-04-01 11:15:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ isomp4: Refactor various state variables into a mux_mode var
+ Instead of checking various state variables around the muxer,
+ track the current muxing mode in a single 'mux_mode' enum.
+ Add some implementation notes about the different mux modes
+
+2015-04-08 16:40:02 +0200 Edward Hervey <edward@centricular.com>
+
+ * common:
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner
+
+2015-04-08 11:17:31 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: Fix framesize parsing
+ The string passed to the parsing function only contains a framesize, and
+ not <pt> + <framesize>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
+
+2015-03-20 12:18:37 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: clip chunk size above the valid maximum (0x7fffffff)
+ https://bugzilla.gnome.org/show_bug.cgi?id=722567
+
+2015-03-20 09:07:35 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: clip chunk length to available data (when known)
+ This prevents silly chunk lengths from possibly overflowing
+ (at least when we know the actual data length).
+ https://bugzilla.gnome.org/show_bug.cgi?id=722567
+
+2015-04-06 20:17:52 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't accumulate segment bases manually
+ gst_segment_do_seek() does that for us already, and doing it twice
+ will break non-flushing seeks in interesting ways. Leftover from 1.0
+ porting.
+ Also copy over segment offset and applied_rate, just in case.
+
+2015-04-06 19:08:10 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/icles/test-segment-seeks.c:
+ icles: Fix waiting for segment-done if it happens too fast
+ Sometimes we can get segment-done before we got async-done. If we waited
+ for async-done only, the segment-done would be dropped and we would wait
+ forever for it a few lines below.
+
+2015-04-06 18:55:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: stbl_index is valid from 0 onwards
+ It indicates the last sample parsed, not the next one to parse.
+ As it starts in -1, any value from 0 onwards means that it has
+ some valid data.
+
+2015-04-05 20:06:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ docs: make GstRTCPSync enum show up in rtpbin docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=747358
+
+2015-04-05 11:45:45 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ docs: add RTPJitterBufferMode enum to rtpbin docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=747358
+
+2015-04-04 11:55:00 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: close files before posting message
+ Makes sure the files were properly flushed and closed before
+ the message reaches the application
+
+2015-03-30 13:54:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/multifile.c:
+ tests: multifile: increment tests to check for multifile messages
+ Also verify that the multifilesink file messages are being correctly
+ posted to the bus
+
+2015-03-30 12:51:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/multifile.c:
+ tests: multifile: handle FIXME for proper checking when test finished
+ Use a GstBus and wait for EOS to finish the tests instead of
+ relying on sleeping
+
+2015-03-30 11:14:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: post file message on EOS
+ When multifilesink is operating in any mode other than one file
+ per buffer, the last file created won't have a file message posted
+ as multifilesink doesn't handle the EOS event.
+ This patch fixes it by using the last position to post a file
+ message when EOS is received. This should ensure at least the
+ time related data and the filename are posted to the application
+ or other elements
+ https://bugzilla.gnome.org/show_bug.cgi?id=747000
+
+2015-04-03 18:57:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-04-03 02:08:50 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Guard against 64-bit overflow
+ For large-file atoms, guard against overflow in the size field,
+ which could make us jump backward in the file and cause
+ infinite loops.
+
+2015-04-01 23:46:13 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ * tests/check/elements/qtmux.c:
+ isomp4: Make non-seekable downstream an error in normal mode
+ When not in fast-start or fragmented mode, we need to be able
+ to rewrite the size of the mdat atom, or else the output just
+ won't be playable - the mdat placeholder with size == 0 will
+ cover the rest of the file, including any moov atom we write out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708808
+
+2014-03-15 15:23:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * tests/check/elements/rtp-payloading.c:
+ rtph263pay/-depay: add framesize SDP attribute
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
+
+2014-03-15 13:33:56 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay/-depay: Remove incorrectly introduced framesize SDP attribute
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726415
+
+2015-03-27 21:09:44 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2src: device sequence/offset correction in case of renegotiation
+ The v4l2 device restarts the sequence counter in case of streamoff/streamon,
+ the GST offset values are supposed to increment strictly monotonic, so
+ adjust the sequence counter/offset values in case of caps
+ renegotiation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745441
+
+2014-11-14 14:18:51 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: add frame loss detection
+ In case of v4l2 driver filled offset/sequence values add frame
+ loss detection (and write a warning message).
+ Move offset meta data setting and frame loss checking after the
+ timestamp adjustment code to get proper timestamps for the
+ warning message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745441
+
+2014-11-14 13:48:51 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: use v4l2 capture device sequence counter
+ Use the v4l2 capture device sequence counter for
+ setting the GstBuffer offset/offset_end values.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745441
+
+2015-03-30 13:12:35 +0200 Tobias Modschiedler <tobias.modschiedler@cetitec.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Ask the driver about its requirements for min_buffers before initiating buffer pool.
+ If propose_allocation() had not been called yet, it was possible that the driver was not asked at all.
+ In buffer pool: Consider minimum number of buffers requested by driver when setting config.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746834
+
+2015-04-01 19:30:27 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8depay.h:
+ rtpvp8depay: Parse width/height/profile from keyframes
+ This makes it possible to mux the result into a container
+ such as matroska.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747208
+
+2015-04-01 19:01:49 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Expose VP8 width/height limitations in the caps template
+ The VP8 format specification (RFC 6386 section 18.1) specifies
+ that the maximum size is 16383x16383.
+
+2015-03-31 00:20:13 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flv: When passing seek event upstream, hold a ref.
+ In case upstream can't handle the seek, make sure we
+ keep a ref on the event to attempt to handle it ourselves.
+
+2015-03-26 13:34:53 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: fix GValue leaks when parsing tags
+ gst_tag_list_add_value() doesn't consume the GValue we pass to it so there is
+ no point copying it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746810
+
+2015-03-23 20:58:25 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-23 20:57:56 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-23 20:56:41 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-27 18:58:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-read-common.c:
+ matroska: store stream tags and push as updated
+ New tags can be found on different parts of the file, so this patch
+ keeps the stream taglists around for the life cycle of the pad
+ and adds those new tags as found. Then a new tag is found, the
+ pad's is marked with a tags changed flag, making the element push
+ a new tag event on the next check. Before this, we were sending
+ only the newly found tags, as the element was losing its taglist
+ when pushing the event.
+
+2015-03-15 14:40:36 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: send global tags incrementally
+ Instead of sending only new tags once they are found, merge the taglist
+ and send them incrementally.
+
+2015-03-14 17:07:05 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskaparse: send global tags
+ Global tags are already being read in matroskaparse, but they are not
+ currently being sent.
+ This patch makes global tags get sent incrementally whenever new ones
+ are found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746242
+
+2015-02-03 10:18:58 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/effectv/gstquark.c:
+ quarktv: fix "planes" property range, a value of 0 is not allowed
+ When planes property is set to 0, the pipeline executes in
+ an infinite loop and never exits. Since planes must never
+ be 0, set the minimum value in the property description
+ to 1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743906
+
+2015-03-26 13:42:02 -0700 David Schleef <ds@schleef.org>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Fix up comments regarding DTS
+
+2015-03-25 15:11:34 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Fix segment in TCP mode
+ It is expected that buffers are time-stamped with running time. Set
+ a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
+ would do. Depayloaders will update the segment to reflect the playback
+ position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-03-26 12:21:25 -0700 David Schleef <ds@schleef.org>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: be more strict about typefinding DTS
+ Code now matches comments.
+
+2015-03-25 15:10:53 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Remove useless function
+ This function didn't do anything special, let's not use a function for
+ that.
+
+2015-03-20 13:03:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitter: Account for rtx_retry in overflow check
+ As rtx_retry is part of the substraction, we need to take it into
+ account, otherwise we may endup with a big value.
+
+2015-03-24 23:15:15 +0000 Julien Isorce <j.isorce@samsung.com>
+
+ * sys/osxvideo/cocoawindow.m:
+ osxvideosink: check for deprecated constants prior to OSX 10.10
+ cocoawindow.m:339:5: error: 'NSOpenGLPFAWindow'
+ is deprecated: first deprecated in OS X 10.9
+ cocoawindow.m:576:7: error: 'NSOpenGLPFAFullScreen'
+ is deprecated: first deprecated in OS X 10.6
+ cocoawindow.m:605:24: error: 'setFullScreen'
+ is deprecated: first deprecated in OS X 10.7
+
+2015-03-24 16:51:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix seeking query
+ The segment start/stop in the query is meant to represent the seekable
+ portion of the stream. It does not match the segment start/stop. Instead
+ export 0 to duration.
+
+2015-03-24 16:18:53 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Only set caps once if they don't change
+ Previously we were setting new caps with the same content for every H264 or
+ AAC codec_data we found in the stream, spamming everything and causing
+ renegotiations.
+
+2015-03-24 12:46:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Don't create AAC/H264 caps without codec_data
+ Instead delay creating the caps until we read the codec_data from the stream,
+ or fail if we get normal data before the codec_data.
+ AAC raw caps and H264 avc caps always need codec_data, setting caps on the pad
+ without them is going to make negotiation fail most of the time. Even if we
+ later set new caps with the codec_data, that's usually going to be too late.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746682
+
+2015-03-24 15:39:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Fix indention
+
+2015-03-22 13:23:44 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Fix string format warning on 32-bit
+ UInt32 (Darwin, not C99's uint32_t) is 'unsigned long' on 32-bit
+ platforms.
+
+2015-03-21 17:50:40 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Fix another instance of sticky event misordering warnings
+ Make sure that the sync_src pad has caps before the segment event.
+ Otherwise we might get a segment event before caps from the receive
+ RTCP pad, and then later when receiving RTCP packets will set caps.
+ This will results in a sticky event misordering warning
+ This fixes warnings in the rtpaux unit test but also in the
+ rtpaux and rtx examples in tests/examples/rtp
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-03-21 17:18:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Also start the RTCP send thread when receiving RTP or RTCP
+ Before we only started it when either:
+ - there is no send RTP stream
+ or
+ - we received an RTP packet for sending
+ This could mean that if the send RTP pads are connected but never receive any
+ RTP data, and the same session is also used for receiving RTP/RTCP, we would
+ never start the RTCP thread and would never send RTCP for the receiving part
+ of the session.
+ This can be reproduced with a pipeline like:
+ gst-launch-1.0 rtpbin name=rtpbin \
+ udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \
+ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
+ rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \
+ rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \
+ fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \
+ rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v
+ Before this change the rtcp_fakesink would never send RTCP for the receiving
+ part of the session (i.e. no receiver reports!), after the change it does.
+ And before and after this change it would send RTCP for the receiving part of
+ the session if the sender part was omitted (the last two lines).
+
+2015-03-19 11:54:12 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: Add support for buffer lists
+
+2015-03-19 11:39:38 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ rtprtxqueue: Implement support for buffer lists
+
+2015-03-18 17:32:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Improve trace readability
+ Change the command number into strings.
+
+2015-01-20 10:18:56 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flvdemux: Don't repeatedly warn after no_more_pads (v2)
+ This can get rather spammy for such a high log level.
+ Only warn once per stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746274
+
+2015-03-16 11:23:52 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Introduce constant for no-more-pads threshold
+ https://bugzilla.gnome.org/show_bug.cgi?id=746274
+
+2015-01-20 10:18:29 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Fix warning to contain 'video'
+ https://bugzilla.gnome.org/show_bug.cgi?id=746274
+
+2015-03-11 21:25:40 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: for dts only stream set pts=dts for intra only formats
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-03-14 16:39:09 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: fix sending of tags
+ * Fix critical when new tags are found after segment event has already
+ been sent.
+ * Send global tags before stream tags.
+ * Split sending of tags out of gst_matroska_demux_send_event() into its
+ own function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745973
+
+2015-03-13 18:26:06 +0000 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: properly escape percent sign in documentation
+
+2015-03-13 18:26:44 +0000 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpdtmfmux.c:
+ rtpdtmfmux: properly escape percent sign in documentation
+
+2015-03-13 18:48:03 +0000 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2src: delay renegotiation until it is likely buffers were reclaimed
+ Allow renegotiation to happen when buffers have returned after an allocation
+ query. As the allocation query is serialized, all buffers from the pool
+ should have returned and we can stop it to create a new one for the
+ new format
+ https://bugzilla.gnome.org/show_bug.cgi?id=682770
+
+2015-03-13 18:47:55 +0000 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2object: add gst_v4l2_object_try_format
+ Similar to set_format but it uses TRY_FMT instead of S_FMT
+ https://bugzilla.gnome.org/show_bug.cgi?id=682770
+
+2015-03-13 18:38:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: fix crash with GST_DEBUG enabled
+ g_inet_socket_address_get_address() does not give
+ us a ref to the address, so don't unref it.
+
+2015-03-12 13:49:56 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/level/gstlevel.c:
+ level: Don't read over the end of the input memory
+ Previously we advanced the in_data pointer by bps for every channel, and then
+ later again for block_size*bps. This caused us to be one sample further than
+ expected if an input buffer covered two analysis frames. And in the end lead
+ to completely bogus values reported by level.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746065
+
+2015-03-12 01:37:08 +1100 Jan Schmidt <jan@centricular.com>
+
+ * sys/oss/gstossdmabuffer.c:
+ Remove a couple of superfluous trailing semi-colons
+
+2015-03-10 09:31:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/alpha/gstalpha.c:
+ * gst/avi/gstavidemux.c:
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/isomp4/gstisoff.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/udp/gstmultiudpsink.c:
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ Fix double semicolons
+
+2015-03-10 15:46:40 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmux: Shut down element before downward state change
+ Make sure the state change won't hang trying to shut down pads
+ by making sure the streaming has stopped before chaining up.
+
+2015-03-09 22:58:05 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: stream format is an SPDIF-only field
+
+2015-03-09 22:53:41 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosrc.h:
+ osxaudio: fix spaces
+
+2015-03-09 22:52:46 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosrc.h:
+ osxaudio: add type check macro
+
+2015-03-09 22:51:51 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: rename gst_core_audio_set_channels_layout()
+ to gst_core_audio_get_channel_layout().
+
+2015-03-09 22:30:28 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ osxaudio: remove unused finalize
+
+2015-03-09 16:25:43 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: remove duplicate declaration of function
+
+2015-03-09 16:22:29 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: remove unused value
+ CID #1226474
+
+2015-03-09 16:14:34 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: fix leak
+ CID 1212156
+
+2015-03-09 15:58:33 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: remove uneeded variable
+ We just need to save the ebit information in case there is an error decoding.
+
+2015-03-09 16:46:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp[89]enc: Reset the encoder when flushing
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 12:51:17 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroska: error mode if can't push buffer
+ If gst_pad_push() fails, inform and return flow error.
+
+2015-03-09 12:13:34 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroska: unused value
+ Value set in ret will be overwritten just before exiting the function.
+ CID #1226469
+
+2015-03-09 11:10:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base
+ These are outside the expected range of sequence numbers and should be
+ clipped, especially for RTSP they might belong to packets from before a seek
+ or a previous stream in general.
+
+2014-02-27 10:52:16 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't include payload type in the caps for framesize
+ When the sdp media attribute framesize are converted to caps
+ the <payload> should not be included.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
+
+2015-03-09 10:05:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams
+
+2015-03-09 11:24:58 +0530 Arun Raghavan <arun@centricular.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Make sure to filter caps in all cases during CAPS query
+ We were skipping the filter step while returning template caps, for
+ example.
+
+2015-03-08 21:15:53 +0000 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Don't update buffer for OUTPUT
+ For output device, we should not update the buffer with flags and
+ timestamp when we dequeue. The information in the v4l2_buffer is not
+ meaningful and it breaks the case where the buffer is rendered at
+ multiple places.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745438
+
+2015-03-08 18:04:34 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Implement cookies property
+
+2015-03-08 18:02:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Implement automatic-redirect property
+
+2015-03-08 17:54:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Implement proxy support
+ The properties were there before, but not used anywhere.
+
+2015-02-21 20:05:24 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: resurrect some flow return handling
+
+2015-03-04 10:27:17 +0100 Nicolas Huet <nicolas.huet@parrot.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix LOAS parsing issue
+ Fix missing index in syncword searching
+ https://bugzilla.gnome.org/show_bug.cgi?id=745585
+
+2015-03-05 17:54:43 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: fix modulo math with ringbuffer parameters
+ To get a multiple of bpf use a subtraction and not an addition
+ https://bugzilla.gnome.org/show_bug.cgi?id=745684
+
+2015-03-07 00:55:47 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Protect property variables with the object lock.
+ Use the object lock instead of the splitmux lock to protect
+ internal property variables, so they're not locked when
+ switching to a new file.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744420
+
+2015-03-06 11:39:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ check: add jitterbuffer unit test
+ See https://bugzilla.gnome.org/show_bug.cgi?id=745539
+
+2015-03-05 09:18:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix handling of interleaved (TCP) streams
+ We need to set up the transport in any case, not just if we have a container
+ stream or a non-interleaved stream. Only if we have an interleaved stream and
+ are retrying, we should not set up the stream again.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745599
+
+2015-03-05 10:00:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vp[89]dec: Drop frames that have no output buffer because of errors
+ finish_frame() assumes that there is an output buffer.
+
+2015-03-05 09:56:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't unref caps we don't own
+
+2015-03-05 09:46:17 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Push RTCP caps on the RTCP pads
+ Otherwise we will get not-negotiated later from rtpbin, and will never be able
+ to send RTCP packets back to the server. Note that error flow returns from the
+ RTCP pads are ignored, that's why it didn't fail more visible before.
+
+2015-03-05 09:35:32 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Make sure to send SEGMENT events on all pads
+
+2015-03-03 16:23:15 +0100 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtp: Add Full Intra Request (FIR) packets to statistics
+ https://bugzilla.gnome.org/show_bug.cgi?id=745587
+
+2015-03-03 16:01:53 +0100 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtp: Add Packet Loss Indication (PLI) to statistics
+ This is helpful to provide statistics in the format defined in
+ http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745587
+
+2015-03-03 19:19:50 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: Remove duration accumulation logic
+ Duration accumulation can cause rounding errors and generate wrong
+ duration with different buffers that share the same timestamp.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-03-03 18:40:16 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: Add an helper method to get buffer timestamps
+ ... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
+ that return PTS or DTS based on stream type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-03-04 11:28:12 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add explanation why we have space for 32 hash tables
+ And also create only one, there's no need yet to create all 32 until
+ we implement RFC2762.
+
+2015-03-04 11:26:57 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ Revert "rtpsession: Do not use an array of maps if they are not being used"
+ This reverts commit 1591adf4cd843d13d8622a30c619425691a84128.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
+ It's the beginning of an implementation of RFC 2762, which is needed for
+ large multicast groups. The implementation is not yet complete but why
+ not leave what is there and implement RFC 2762 instead?
+
+2015-03-04 10:35:12 +0100 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Do not use an array of maps if they are not being used
+ rtpsession declares an array of maps to store srrcs but only the
+ the key 0 is being used. This patch replaces the array of maps
+ for just one map and remove useless parameters in rtpsession
+ https://bugzilla.gnome.org/show_bug.cgi?id=745586
+
+2015-02-27 18:12:09 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: remove not needed code
+ In gst_avi_demux_handle_src_query, there is not needed code.
+ We already check about stream is vbr or not at the upper line.
+ o, we don't need to check this condition becase stream is not
+ vbr 100% in this case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745276
+
+2015-03-03 23:25:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: replace deprecated function
+ Just avoid using the deprecated function entirely,
+ it's easy enough. Defining the macro is not enough.
+
+2015-03-03 19:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: fix compilation against newer gdk-pixbuf
+ gdk_pixbuf_new_from_inline() has been deprecated in favour
+ of GResource.
+
+2015-03-03 18:39:15 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudiosrc: Allow caps renegotiation
+ The ringbuffer does allow renegotiation, so we do not have to report
+ fixed caps once it is acquired (based on a similar patch for the sink
+ side by Ilya Konstantinov <ilya.konstantinov@gmail.com>).
+
+2015-02-21 14:41:08 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Allow renegotiating caps
+ Once osxaudiosink's device is open, it fixates on the initial caps and
+ refuses to accept new caps. This is erroneous since the Audio Unit is
+ can accept a new ASBD, and GstAudioRingBuffer supports reconfiguration
+ as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743925
+
+2015-03-02 12:04:00 +0100 Gwenole Beauchesne <gwenole.beauchesne@intel.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2allocator: fix fd leak in DMABUF import mode.
+ Ensure gst_v4l2_buffer_pool_release_buffer() releases the associated
+ GstV4l2MemoryGroup. In particular, this allows for closing the DMABUF
+ handles prior to instantiating new ones.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745443
+
+2015-03-02 15:06:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Use 0 as duration for the EOS "frame"
+
+2015-03-02 15:02:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp{8,9}enc: Tell the encoder about actual timestamps and durations of frames
+ ... instead of just counting frames. The values are supposed to be in timebase
+ units, not frame units. This fixes various quality problems with VP8/VP9
+ encoding and in general makes the encoder behave better.
+ Thanks to Nirbheek Chauhan for noticing this bug.
+
+2015-03-01 13:56:17 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vpxdec: Fix calculation of width in bytes
+ Right now we only support I420, but vpx seems to support more formats.
+ This will prevent hard to find bug in the future.
+
+2015-03-01 13:52:50 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vpxdec: Don't memcpy in frame map failed
+ This avoid a crash if mapping the frame failed.
+
+2015-03-01 13:48:45 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Add missing break
+ This is cosmetic change.
+
+2015-03-01 13:46:18 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ v4l2: Workaround driver not setting field correctly
+ As it's very common, handle driver not setting field in buffers
+ by using the field value from the format. This workaround a long time
+ bug in UVC driver. For even buggier driver, we simply assume
+ progressive as before. We also only warn once, to avoid spamming.
+
+2015-02-28 18:10:06 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix key unit seek
+ Unlike many other seek flags, the KEY_UNIT seek
+ flag is not copied over into the GstSegment,
+ since it's only relevant for the seek itself,
+ so we need to pass it explicitly to the seek
+ handler here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745339
+
+2015-02-27 09:38:01 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ docs/plugins: Updates
+
+2015-02-26 23:41:47 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ matroskamux/demux: initialize dts_only
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-02-26 23:28:11 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: store DTS for V_MS/VFW/FOURCC streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-02-26 19:48:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ multifile: attempt to fix docs build issue on build bot
+
+2015-02-27 00:41:46 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/interleave/interleave.c:
+ interleave: Drop custom latency query handling
+ This is implemented by the default query handler now.
+
+2015-02-27 00:40:05 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Drop custom latency querying logic
+ This is now implemented in the default latency query handler.
+
+2015-02-26 16:10:41 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: fix payloader description and author e-mail
+ https://bugzilla.gnome.org/show_bug.cgi?id=745226
+
+2014-09-05 16:34:26 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2: query crop configuration after each call of S_CROP
+ S_CROP ioctl is write-only and the device can adjust crop rectangle so
+ we query back the crop configuration after each S_CROP to know what has
+ been done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736133
+
+2015-02-26 02:12:18 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS
+ When such stream is present demuxer should set DTS on buffers instead
+ of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW
+ streams.
+ Sample file
+ https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-02-25 16:45:11 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Check corruption flag on the right buffer
+ We where checking the buffer we are copying to instead of the buffer we
+ are copying from.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740040
+
+2015-01-19 15:29:24 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: set colorspace in caps for capture devices
+ This information is set by the driver for a capture device, and so could
+ be forwarded to pipeline by setting the colorimetry in caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743186
+
+2014-10-06 17:30:06 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2bufferpool: fix import_userptr() in single-planar API when n_planes > 1
+ In the V4L2 single-planar API, when format is semi-planar/planar,
+ drivers expect the planes to be contiguous in memory.
+ So this commit change the way we handle semi-planar/planar format
+ (n_planes > 1) when we use the single-planar API (group->n_mem == 1).
+ To check that planes are contiguous and have expected size, ie: no
+ padding. We test the fact that plane 'i' start address + plane 'i'
+ expected size equals to plane 'i + 1' start address. If not, we return
+ in error.
+ Math are done in bufferpool rather than in allocator because the
+ former is aware of video info.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738013
+
+2015-01-23 10:15:46 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2allocator: let bufferpool calculate image size when importing userptr
+ Offset are relative to the buffer and there is no guarantee substracting
+ them will give us the plane size. So we let bufferpool make the math as
+ it is more aware of video info than allocator and pass a size array to
+ allocator import function.
+ Pointed out by Nicolas Dufresne <nicolas.dufresne@collabora.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=738013
+
+2014-12-11 16:13:15 +0100 Philippe De Muyter <phdm@macqel.be>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: recognize and distinguish all bayer arrangements
+ Up to now, v4l2src recognized only "bggr" amongst the bayer arrangements.
+ Recognize now also the "rggb", "gbrg" and "grbg" arrangements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742363
+
+2015-01-15 16:11:53 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set v4l2_buffer.field when queuing buffer in an output device
+ According to the current specification, application must set this field
+ for an output device.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743013
+
+2015-02-24 05:57:24 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ osxaudiosrc: iOS resampling causes stuttering
+ Fixes stuttering audio when iOS AU is resampling. To make AU resample,
+ one has to request a rate that differs from AVAudioSession's
+ sampleRate. The resampling itself is not the culprit, but rather our
+ API misuse.
+ AudioUnitRender modifies the mDataByteSize members with the
+ actual read bytes count. Therefore, they must be reinitialized
+ before each AudioUnitRender. (The buffers themselves can be
+ preallocated.)
+ The "stutter" was caused by one AudioUnitRender making the buffer
+ too small for other AudioUnitRender invocations, making them fail
+ with -50 (paramErr). By way of luck, when AU didn't resample, all
+ AudioUnitRender invocations read the same number of bytes.
+ (This patch addresses some non-interleaved audio concerns, but
+ at this moment the elements do not support non-interleaved audio
+ and non-interleaved is untested.)
+ https://bugzilla.gnome.org/show_bug.cgi?id=744922
+
+2015-02-22 01:49:52 +0100 Krzysztof Kotlenga <pocek@users.sf.net>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: improve error message when unauthorized
+ Make use of NOT_AUTHORIZED error code instead of falling back to generic
+ READ error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=601733
+
+2015-02-23 20:06:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: remove pointless g_return_val_if_fail()
+ ximage won't ever be NULL here because the dispose
+ function is called via ximage->dispose().
+
+2015-02-23 19:40:25 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: All segment resulting from a seek should have the same seqnum
+ https://bugzilla.gnome.org/show_bug.cgi?id=744983
+
+2015-02-19 23:12:31 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Enable copy when no known allocation params
+ When there is no allocation parameters in the query, enable copy
+ threshold. When this threshold is reached, the buffer pool will start
+ copying when the pool reaches a critical level. If the driver supports
+ CREATE_BUFS, this will be used instead.
+
+2015-02-19 23:08:34 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Update allocator flags
+ When we hit emulated formats, we disable CREATE_BUFS since libv4l2
+ cope very badly with it. Also clear the allocator flags so we will
+ never try to allocate more buffers. This fixes failure when the copy
+ threshold is reached as we where calling CREATE_BUFS, which lead to
+ libv4l2 instability.
+
+2015-02-19 23:07:23 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Use specific debug category
+ The pool has grown enough that it is now handy to seperate v4l2object
+ trace from v4l2bufferpool trace.
+
+2015-02-19 14:29:02 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: default encoding name to VP8
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 14:06:51 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: make caps writable before truncating them
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-05 10:29:26 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: negotiate encoding name
+ Chrome uses a different one than gstreamer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 12:35:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
+ Otherwise we will just send buffers on the pad without any events beforehand
+ and will get g_warnings() about that.
+
+2015-02-19 11:20:51 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/jack/gstjackaudiosrc.c:
+ jack: case missing break statement
+ commit b1098c2ea5eabea7af08ce51d22b867eaed2bbe2 added a new case in
+ gst_jack_audio_src_get_property() but forgot to add the break statement to it.
+
+2015-02-18 19:18:00 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ Revert "v4l2: fraction is reversed"
+ This reverts commit b91fe36644b15ae070d72b9e8a9c7087e82aef12.
+
+2015-02-18 17:49:29 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: fraction is reversed
+ In the fraction 1 / 2. 1 is the numerator and 2 is the denominator.
+ The arguments of fraction gst_value_set_fractions() are value,
+ numerator and denominator.
+ Also, gst_value_set_fraction() fails if denominator is 0 for obvious
+ reasons.
+
+2015-02-17 20:26:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2pool: Deactivate other pool
+ When importing buffers from a downstream pool, we need to deactivate
+ that pool to ensure it will be usable again later. Relying on the
+ refcount to reach zero does not work, since elements like xvimagesink
+ keeps a reference on their proposed pool.
+
+2015-02-18 10:10:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux.c:
+ qtmux: remove not needed condition
+ gst_buffer_replace can handle NULL inputs by itself
+
+2015-02-18 09:40:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate
+ The tfdt should be more accurate as the buffer timestamp is provided
+ by the fragmented format manifest and it might just be an approximation.
+
+2015-02-17 16:57:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
+ We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
+ confuse downstream with buffers that come before such events.
+
+2015-02-17 12:20:57 +0100 hark <hark@puscii.nl>
+
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosink.h:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jack/gstjackaudiosrc.h:
+ jack: Add property port-pattern to specify which JACK ports to connect to
+ https://bugzilla.gnome.org/show_bug.cgi?id=690719
+
+2015-02-17 12:31:06 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/gstisoff.c:
+ * gst/isomp4/gstisoff.h:
+ * gst/isomp4/qtdemux.c:
+ isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_
+ We need different symbol names, because these symbols are also present
+ in the fragmented plugin ... which will cause conflicts when doing
+ static linking
+
+2015-02-16 14:31:05 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/lines.c:
+ goom2k1: use fractional part of float division
+
+2015-02-16 13:59:14 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsin: remove dead code
+ Every instance of goto beach has buf_info equal NULL. Don't check
+ for a condition that never happens.
+ CID #1268399
+
+2015-02-15 21:45:24 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * tests/check/elements/splitmux.c:
+ splitmux-test: Parse error message
+ The test had a function to print the error, but was not parsing it.
+ This was causing warning about dbg_info being used uninitialized. If
+ the test was testing any errors, this would have crashed.
+
+2015-02-15 21:34:28 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/spectrum/gstspectrum.c:
+ spectrum: Fix min and max for bands property
+ The number of FFTs is calculated with the following formula:
+ guint nfft = 2 * bands - 2;
+ nfft is passed to gst_fft_f32_new() as the len argument and is of type
+ unsigned integer. This method required that len is at leas 1, then
+ maximum G_MAXINT, as other values would be negative. If we extrapolate
+ from the formula above it means we need "bands" to be between 2 and
+ ((guint)G_MAXINT + 2) / 2).
+ https://bugzilla.gnome.org/show_bug.cgi?id=744213
+
+2015-02-15 15:51:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix freeing of shared memory
+ When memory (that has been shared using gst_memory_share()) are freed,
+ the memory (or the DMABUF FD) should not bee freed. These memories have
+ a parent. This also removes the extra _v4l2mem_free function and avoid
+ calling close twice on the DMABUF FD.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744573
+
+2015-02-14 11:11:30 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: do not use sparse streams in push-based seeking
+ Using the sparse streams can make the push-based seeking return
+ too far in the stream. It also can lead to issues as the
+ sparse streams will be ignored when restarting playback and,
+ if the sparse stream is the one that has the earliest sample,
+ it will confuse qtdemux's offsets as one stream will have
+ an earlier offset than the demuxer's one which might lead to
+ early EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742661
+
+2015-02-13 19:43:16 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Enhance code readability in pulsesink_query
+ In pulsesink_query function, we use a switch for the query
+ type. In the CAPS case, there is no 'break', instead we
+ return right away. Use a break and return at the end of
+ the function instead for better code readability.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744461
+
+2015-02-13 20:40:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: flag as sink from the start
+
+2015-02-11 15:30:44 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/Makefile.am:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstisoff.c:
+ * gst/isomp4/gstisoff.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Initial 'sidx' atom parsing support
+ Parse the 'sidx' atom and update the total duration according to the
+ parser result. The isoff parser code is imported from
+ gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
+ function was factored out of the gst_isoff_sidx_parser_add_buffer()
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743578
+
+2015-02-11 05:06:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/flv/Makefile.am:
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Use gst_video_guess_framerate()
+ Use gst_video_guess_framerate() from libgstvideo to guess
+ sensible common framerates where possible from the
+ floating point fps in the stream.
+
+2015-02-11 13:53:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/interleave/interleave.c:
+ * gst/rtsp/gstrtpdec.c:
+ * gst/videomixer/videomixer2.c:
+ Improve and fix LATENCY query handling
+ This now follows the design docs everywhere, especially the maximum latency
+ handling.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744106
+
+2015-02-11 10:29:55 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Handle first RTCP packet and early feedback correctly
+ According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
+ an early RTCP packet for the very first one. It must be a regular one.
+ Also make sure to not use last_rtcp_send_time in any calculations until
+ we actually sent an RTCP packet already. In specific this means that we
+ must not use it for forward reconsideration of the current RTCP send time.
+ Instead we don't do any forward reconsideration for the first RTCP packet.
+
+2015-02-10 18:53:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: fix compilation with gcc 5.0
+
+2015-02-10 16:00:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix example pipeline properly
+ x264enc might not have a max-key-int property, but it
+ has a key-int-max property...
+
+2015-02-10 14:57:55 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmux: fix typo
+
+2015-02-10 14:56:23 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: update example pipeline
+ Element x264enc doesn't have a max-key-int property
+
+2015-02-10 13:29:32 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: fix memory leak
+ If execution goes to the beach in line 981, buf_info goes out of scope without
+ the memory being free'd. Handle this case.
+ CID #1268403
+
+2015-02-08 12:03:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix awkward if clause
+
+2015-02-07 01:41:49 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxsink.c:
+ * tests/check/elements/splitmux.c:
+ splitmux: Add unit test for file splitting
+ Add a unit test for file splitting, and fix the leaks in the
+ splitmuxsink it found
+
+2015-02-06 14:43:22 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: fix which stop variable is used in assignment
+ Assignment is done to variable segment.stop when the intention was to assign to
+ local variable stop. Instead of overwriting it, the value is now clamped and
+ segment.stop is set to it soon after.
+ CID #1265773
+
+2015-02-07 00:19:36 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * tests/check/elements/splitmux.c:
+ splitmux: Fix memory leaks until the test valgrinds clean
+
+2015-02-06 06:42:17 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmux: Handle early EOS during part preparation
+ Handle the case where a short file reaches EOS while we're still
+ waiting for no-more-pads, and make sure we continue to the internal
+ READY state for real playback to work properly later.
+
+2015-02-06 05:03:19 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/files/splitvideo00.ogg:
+ * tests/files/splitvideo01.ogg:
+ * tests/files/splitvideo02.ogg:
+ tests: Change splitmux test video files
+ Avoid test failure by changing the stored video resolution
+ from 80x60 to 80x64, which needs bug 741030 to be fixed.
+
+2014-08-01 00:07:53 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * gst/multifile/Makefile.am:
+ * gst/multifile/gstmultifile.c:
+ * gst/multifile/gstsplitfilesrc.c:
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multifile/gstsplitmuxsrc.h:
+ * gst/multifile/gstsplitutils.c:
+ * gst/multifile/gstsplitutils.h:
+ * gst/multifile/test-splitmuxpartreader.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/splitmux.c:
+ * tests/files/splitvideo00.ogg:
+ * tests/files/splitvideo01.ogg:
+ * tests/files/splitvideo02.ogg:
+ splitmux: Implement new elements for splitting files at mux level.
+ Implement 2 new elements - splitmuxsink and splitmuxsrc.
+ splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
+ plus audio/subtitle streams, and starts a new file
+ whenever necessary to avoid overrunning a threshold of either bytes
+ or time. New files are started at a keyframe, and corresponding audio
+ and subtitle streams are split at packet boundaries to match
+ video GOP timestamps.
+ splitmuxsrc is a corresponding source element which handles
+ the splitmux:// URL and plays back all component files,
+ reconstructing the original elementary streams as it goes.
+
+2015-02-04 16:32:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ * tests/files/test-cert.pem:
+ * tests/files/test-key.pem:
+ tests: souphttpsrc: update ssl key/cert pair
+ Our ones were expired. The new ones were copied from libsoup's
+ tests files.
+ Also sets the property to use our own cert to validate the
+ server, otherwise the default system certs would be used
+ and it would fail.
+
+2015-02-04 02:25:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: prevent trying to get 0 bytes from adapter
+ This causes an assertion and would lead to getting a NULL instead
+ of a buffer. Without proper checking this would easily lead to
+ a segfault
+ https://bugzilla.gnome.org/show_bug.cgi?id=737199
+
+2015-02-04 21:50:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Simple implementation of GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS
+ When the trickmode key-units flag is set on the segment, simply skip
+ any sample on a video stream that isn't a keyframe
+
+2015-02-03 17:35:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix container handling
+ We detect a container correctly now so we need to revert the weird
+ check there was before.
+ Use gst_rtspsrc_stream_push_event() to push the caps event on the
+ right pad.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=739391
+
+2015-02-02 19:46:27 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: store and write stream tags
+ Separate global from stream tags storage and write them to the
+ appropriate tags entry in the output
+
+2015-02-02 13:35:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: parse stream tags
+ Keep global and stream tags separately and parse the udta node
+ that can be found under the trak atom. The udta will contain
+ stream specific tags and will be pushed as such
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 14:32:34 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: store stream and container tags separately
+ Tags received via events, when marked as stream tags, will
+ be stored on that stream's trak atom instead of being stored
+ in the main tags atom. This allows the resulting file to have
+ global and stream tags stored.
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 13:14:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: refactor tags functions to accomodata UDTA at trak level
+ Refactor the functions that were bound to the 'moov' atom to
+ directly pass the desired 'udta' that should receive the tags.
+ This allows the tags to be written to 'udta' at the 'moov' or
+ the 'trak' level, creating tags that are for the container or
+ for a stream only.
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 10:47:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: map application name to _swr tag
+ It refers to the application name and version used to create the
+ file
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 02:30:40 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroska: Fix seeking past the end of the file in reverse mode.
+ Snap to the end of the file when seeking past the end in reverse mode,
+ and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling
+ for the stop position by always seeking on a segment in stream time
+
+2015-01-30 18:22:31 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Fix signal name
+ This wasn't meant to be pushed at all yet, but now that it's there
+ already it won't hurt to make it correct at least.
+
+2015-01-30 16:56:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpstats.h:
+ rtpstats: Fix typo in documentation
+
+2015-01-30 16:50:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Add new on-receiving-rtcp signal
+ This will be emitted whenever an RTCP packet is received. Different to
+ on-feedback-rtcp, this signal gets every complete RTCP packet and not
+ just the individual feedback packets.
+
+2015-01-28 14:02:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: simplify segment.base math
+ Remove a fix for heavily edited files added for fixing
+ https://bugzilla.gnome.org/show_bug.cgi?id=345830 to work
+ with seeks and proper gaps playback. The fix was replaced
+ for a more general solution that bases on using previous
+ segment's duration, just like it works for media segments
+ playback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743518
+
+2015-01-27 14:00:35 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/videomixer/videomixerorc-dist.c:
+ videomixer: update orc files
+
+2015-01-26 17:08:12 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix data dropping for fragmented streams
+ For fragmented streams with extra data at the end of the mdat
+ qtdemux was not dropping those bytes and would try to use
+ that extra data as the beginning of a new atom, causing the
+ stream to fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743407
+
+2015-01-25 17:30:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Deprecate rtcp-immediate-feedback-threshold property
+ It had no effect since quite some time and also is not needed in general,
+ especially not to switch between immediate feedback mode and early feedback
+ mode. The latest understanding of the RFC is that from the endpoint point of
+ view, both modes are exactly the same. RTCP is only allowed to use the
+ bandwidth as given by the RFC constraints, as such it is only ever possible
+ to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
+ packets.
+ The difference between immediate feedback mode and early feedback mode is that
+ the former guarantees that an RTCP packet can be sent for every event
+ "immediately", which means that the bandwidth calculations from the RFC have
+ resulted in an RTCP scheduling interval that is small enough. Early feedback
+ mode on the other hand means that we can schedule some packets early to make
+ that happen, but it's not guaranteed at all that it's possible to schedule
+ an RTCP packet per event (i.e. they need to be accumulated or dropped).
+
+2015-01-22 10:29:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Delay the next regular RTCP packet after early RTCP
+ This is required to not exceed the short term average RTCP bitrate when
+ using early feedback as compared to without early feedback.
+
+2015-01-22 10:28:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add new send-rtcp-full signal
+ This indicates with a boolean return value if scheduling a new RTCP packet
+ within the requested delay was possible. Otherwise it behaves exactly like
+ send-rtcp. The only reason for adding a new signal is ABI compatibility.
+
+2015-01-20 00:32:00 +0000 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Free format_info in query_getcaps
+ If we can not create probe stream in query_getcaps function, it will appear
+ memory leakage from format info.
+ The following patch prevent memory leakage in pulsesink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743178
+
+2015-01-23 17:35:51 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: remove unnecessary check
+ No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the
+ flow is OK or not, the check there will be a break from the switch. Removing the
+ check since the outcome is the same.
+ CID #1265762
+
+2015-01-23 15:16:25 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Avoid using freed variable
+ the name variable might have been attributed to pad_name, make sure we
+ free it only *after* pad_name has been used.
+ Coverity CID : 1265774
+
+2015-01-23 15:13:55 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Avoid using freed variable
+ the name variable might have been attributed to pad_name, make sure we
+ free it only *after* pad_name has been used.
+ Coverity CID : 1265775
+
+2014-11-14 12:59:31 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: reuse caps framerate if not overwritten by v4l2 device
+ Enables duration setting in v4l2src.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740403
+
+2015-01-22 10:29:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Fix indention
+
+2015-01-21 17:36:26 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ qtdemux_dump: Bypass even more code if debugging is disabled
+ And avoid using variables that won't exist when debugging is disabled
+
+2015-01-21 15:30:33 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ qtdemux: Only traverse/dump nodes if guaranteed to be used
+ __gst_debug_min is the "global" lowest debug level set. There's no
+ guarantee the qtdemux debug category is actually set at that level.
+
+2014-12-20 17:09:14 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/ebml-read.c:
+ matroska: Avoid debugging below category threshold
+ This part alone was what made the matroska thread take a full core
+ on an android phone ...
+
+2015-01-21 09:55:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstsmptetimecode.c:
+ * ext/mikmod/mikmod_types.c:
+ * gst/audiofx/audiodynamic.c:
+ * gst/audiofx/audiopanorama.c:
+ * gst/effectv/gstradioac.c:
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/videofilter/gstvideotemplate.c:
+ * gst/wavparse/gstwavparse.c:
+ Constify some static arrays everywhere
+
+2015-01-19 17:49:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix deadlock seeking in files without seek entries
+ A mutex unlock was missing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739975
+
+2015-01-19 12:34:25 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/videomixer/blend.c:
+ videomixer: fix illegal memory access in blend function with negative ypos
+ https://bugzilla.gnome.org/show_bug.cgi?id=741115
+
+2015-01-13 16:49:34 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Proxy getcaps
+ Replace the sink_query with new getcaps() virtual and use the proxy
+ helper with the probed caps. This allow upstream element taking decision
+ base on what is supported downstream.
+
+2015-01-13 19:05:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Add support for v210
+
+2015-01-13 18:58:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: v210 is v210, not UYVY and yuv2 is YUY2, not I420
+ Also add a few other raw video formats we support: v308, v216
+ and add comments for a few others we don't support yet.
+ https://developer.apple.com/library/mac/technotes/tn2162/
+
+2015-01-12 15:56:29 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2015-01-10 15:51:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxvideo/cocoawindow.h:
+ * sys/osxvideo/cocoawindow.m:
+ * sys/osxvideo/osxvideosink.h:
+ * sys/osxvideo/osxvideosink.m:
+ osxvideosink: Disable hack for NSApp iteration with a special #define
+ The hack causes deadlocks and other interesting problems and it really
+ can only be fixed properly inside GLib. We will include a patch for
+ GLib in our builds for now that handles this, and hopefully at some
+ point GLib will also merge a proper solution.
+ A proper solution would first require to refactor the polling in
+ GMainContext to only provide a single fd, e.g. via epoll/kqueue
+ or a thread like the one added by our patch. Then this single
+ fd could be retrieved from the GMainContext and directly integrated
+ into a NSRunLoop.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741450
+ https://bugzilla.gnome.org/show_bug.cgi?id=704374
+
+2015-01-08 21:07:05 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: uncork if needed upon commit
+ ... to provide for a running clock.
+
+2015-01-09 16:59:53 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Prevent renegotiation
+ Renegotiation isn't supported, simply prevent it the way we do in
+ v4l2src.
+
+2015-01-06 13:54:25 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Don't unlock the stream lock twice
+
+2015-01-09 11:40:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix stream time conversion
+ Use the right macro to convert to the correct scale or the
+ segment information will be wrong
+ https://bugzilla.gnome.org/show_bug.cgi?id=742572
+
+2015-01-07 18:48:58 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Add protection against driver bug
+ v4l2loopback driver has a this nasty bug that if the queue is larger
+ then 2 buffers, it returns random index on dqbuf. So far we assumed
+ that the index was always right, which would lead to memory being
+ unref twice, and eventually crash.
+
+2015-01-07 17:58:05 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Don't use allocator size to iterate
+ As the buffer array is fixed size and small, it's safer to simply
+ use this static size to cleanup the buffers. This is also more
+ consistent with the rest. The associated method is no longer
+ required and can be dropped.
+
+2015-01-07 17:55:14 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Don't clean buffer array in dispose
+ This should already have been done, plus this code is incorrect
+ and may lead to crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742074
+
+2015-01-07 17:48:31 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Don't ref queued output buffer
+ This partly revert to the old 1.2 behavior. Instead of keeping a
+ reference to the output buffer queued, we simply release them but
+ don't forward it to GstBufferPool. This way, the buffer pool don't
+ need to be flushed to be stopped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742074
+
+2015-01-08 11:37:23 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Never fail on streamoff
+ Failing streamoff prevents allocator from being disposed hence
+ lead to device FD leak. There is no known cases where streamoff
+ may fails for which we'd still be streaming. streamoff is known
+ to fail when a device is being unplugged (in which case errno
+ 19/ENODEV is set).
+ https://bugzilla.gnome.org/show_bug.cgi?id=732734
+
+2015-01-07 21:52:17 -0500 Brad Smith <brad@comstyle.com>
+
+ * configure.ac:
+ v4l2: Add support for detecting the presence of V4L2 support on OpenBSD
+ https://bugzilla.gnome.org/review?bug=742503
+
+2015-01-04 15:57:10 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioparsers/gstac3parse.c:
+ ac3parse: request at least 8 bytes to properly parse header
+ https://bugzilla.gnome.org/show_bug.cgi?id=742325
+
+2015-01-07 16:20:03 -0800 Michael Smith <michael.smith@rdio.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: skip an additional uninteresting chunk type before the fmt chunk.
+
+2015-01-07 18:16:12 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/audiofx/audiodynamic.c:
+ audiodynamic: assert func_index is inside bounds
+ Bringing back the check removed in the previous commit but have that check be a
+ g_assert. Changing the function to static void since return can never be False,
+ because audio format will never be unkown.
+
+2015-01-07 17:31:39 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/audiofx/audiodynamic.c:
+ audiodynamic: remove always-true conditional
+ func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
+ and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
+ The conditional checking if func_index is >= 0 and < 8 will always be true.
+ Removing it.
+ CID 1226442
+
+2015-01-07 18:05:18 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: If we get a gap with a buffer without DTS, error out
+ We (currently?) can't really handle gaps between RTP packets if they're not
+ properly timestamped. The current code would go into calculations with
+ GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
+ better to error out cleanly instead.
+
+2014-11-21 11:39:19 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: set PLAYING state after configuring caps
+ We set to PLAYING after we have configured the caps, otherwise we
+ might end up calling request_key (with SRTP) while caps are still
+ being configured, ending in a crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740505
+
+2014-12-30 18:03:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: remove outdated FIXME
+
+2014-12-30 17:19:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtpcollision.c:
+ tests: rtpcollision: use alawenc/dec in these tests instead of Speex
+ They should always be built, while the speex elements are not.
+ Need to check for a smaller number of buffers then (7->4) because
+ speexenc will add 3 header buffers while alawenc will just output
+ as many buffers as it receives as input.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742098
+
+2014-12-30 16:36:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/simple-launch-lines.c:
+ tests: simple-launch-lines: only run jpeg/png tests if elements are available
+
+2014-12-30 16:26:58 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Don't return a buffer when returning not GST_FLOW_OK
+ basesrc assumes that we don't return a buffer if
+ something else than OK is returned. It will just
+ leak any buffer we might accidentially provide
+ here.
+ This can potentially happen during flushing.
+ Maybe fixes https://bugzilla.gnome.org/show_bug.cgi?id=741993
+
+2014-12-30 14:52:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: use alawenc/dec in these tests instead of Speex
+ They should always be built, while the speex elements are not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742098
+
+2014-12-29 15:35:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Improve detection of being stuck at the same offset
+ Only error out if we read from the same position again and got the
+ same length. Just the same position is not necessarily enough.
+
+2014-12-29 15:00:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Don't get stuck at the same offset when searching for clusters
+ This could happen if there is an invalid cluster with size 0, and in that
+ case just error out instead of looping forever.
+
+2014-12-25 21:32:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: fix ALAC muxing
+ Actually copy the codec data instead of copying nothing
+ and then bombing out because there's no data.
+ Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
+ https://bugzilla.gnome.org/show_bug.cgi?id=741783
+
+2014-12-25 15:48:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpptdemux.c:
+ rtpptdemux: just drop invalid rtp packets instead of erroring out
+ Apparently linphone sends an invalid RTP packet as very
+ first packet. We want to ignore that instead of erroring
+ out (same for any other invalid packets really).
+ https://bugzilla.gnome.org/show_bug.cgi?id=741398
+
+2014-12-25 15:44:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpptdemux.c:
+ rtpptdemux: fix 0.10-ism in docs
+
+2014-12-25 14:58:12 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: use absolute positioning to fix demo
+ https://bugzilla.gnome.org/show_bug.cgi?id=739566
+
+2014-12-25 14:53:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
+ gdkpixbufoverlay: add "positioning-mode" property to allow absolute positions
+ Set positioning-mode=pixels-absolute to allow positioning with
+ absolute coordinates, meaning negative x/y offsets will be
+ interpreted as being to the left/above the video frame instead
+ of being interpreted as relative to the right/bottom edge of
+ the video frame (which is a silly default, but that's how it is).
+ This means we can nicely slide images into and out of the frame,
+ see gdkpixbufoverlay-test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739566
+
+2014-12-22 15:33:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Directly return the ringbuffer's caps if it is acquired
+
+2014-12-22 12:56:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Put all audio formats into the template caps
+ We report the proper caps later from the get_caps() vfunc implementation after
+ probing the selected device.
+
+2014-12-22 12:56:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ osxaudio: Also set the big endian flag for floating point samples
+
+2014-12-22 11:45:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * MAINTAINERS:
+ MAINTAINERS: Update my mail address
+
+2014-12-22 10:23:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Fix deadlock and property change notification in device selection code
+ After creating the ringbuffer we have to set the device on the ringbuffer as
+ it defaults to kAudioDeviceUnknown. At this point it can't have changed to
+ anything else yet and we don't have to notify about changes to the sink/src
+ "device" property. It's also not a good idea because GstAudioBaseSrc has the
+ object lock taken while the ringbuffer is created, which might cause a
+ deadlock if something calls back into the element from "notify::device".
+ Once the base class is done with the NULL_TO_READY state change, it has opened
+ the device via the ringbuffer and this might have chosen a different device.
+ Especially if we initially used kAudioDeviceUnknown. Also notify about this
+ property change as initially intended by this code.
+
+2014-12-19 12:30:03 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2pool: Update configuration size
+ We already update our copy of VideoInfo.size to proper size, now also
+ the configuration so the size matches on release.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741420
+
+2014-12-19 10:57:29 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ matroska-demux: Cache upstream length
+ Instead of constantly querying upstream, just cache the last duration,
+ and in the unlikelyness we might have gone over query again before
+ deciding we are EOS.
+ Cut 15% cpu off matroskademux streaming thread (srsly...)
+
+2014-12-17 17:36:18 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: mux/demux the OpusHead header
+ This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while
+ it is marked as a draft, this part was confirmed to be correct on
+ IRC), and allows one to determine whether a demuxed stream is
+ multistream or not, and thus set the multistream caps field
+ accordingly. In turn, this means downstream does not have to guess.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740744
+
+2014-12-18 11:50:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found
+ CID 1258717
+
+2014-12-18 10:53:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-12 23:06:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ udpsink: allocate scratch space for render functions on the heap
+ and not the stack. Our allocations could get a bit too large
+ to be sure it's not going to cause trouble using the stack.
+
+2014-06-24 01:16:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: re-use send_buffers() code path for render() function
+ It's like rendering a buffer list, just with one buffer.
+ Has the added advantage that if there are multiple clients
+ we can send the buffer to all the clients in one go.
+
+2014-06-24 01:15:25 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: keep client list consistent during removals
+ We unlock and re-lock the client lock while emitting the
+ removed signal, which causes inconsistencies in the client
+ list vs. the client counts. Instead, remove the client from
+ the list already before emitting the signal and put it into
+ a temporary list of clients to be removed. That way things
+ look consistent to the streaming thread, but signal callbacks
+ can still do things like get stats from removed clients.
+
+2014-06-24 00:56:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: fix client count after removal
+
+2014-06-23 18:43:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: keep client list sorted by socket family
+ We make use of in the send_buffers() function if we
+ need to use different sockets to send to IPv4 and
+ IPv6 destinations.
+
+2014-06-20 11:36:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: add sendmmsg-ready render_list function prototype
+ Add prototype for a render_list() function that can use a
+ sendmmsg-style g_socket_send_messages() function once it lands
+ in GLib. We can use this infrastructure to send multiple buffers
+ made up by multiple memories to multiple clients in one go, which
+ drastically reduces the number of syscalls made when sending
+ high-bitrate video streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732152
+
+2014-06-19 19:16:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: make udp client structure refcounted
+ Use the refcount for memory management and keep track
+ of the number of duplicate clients in a separate
+ variable. This will be useful later, and means we
+ don't have to hold the OBJECT_LOCK all the time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732866
+
+2014-06-19 18:31:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: keep count of number of unique and non-unique IPv4 and IPv6 clients
+ This will come in handy later.
+
+2014-12-16 15:00:22 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Disable create_buf with libv4l2
+ Libv4l2 does not work with CREATE_BUFS. Instead of failing on random
+ error caused by libv4l2, disable CREATE_BUFS when an emulated format is
+ detected.
+
+2014-12-09 17:39:12 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Add protection against broken libv4l2
+ It looks like libv4l2 support for CREATE_BUF is incomplete. That
+ combine with existing bugs may lead to crash in GStreamer. These
+ check will make it robust by:
+ - Checking create buf index isn't an already in used index
+ - Checking that the index out of QUERYBUF matches the requested
+ index
+
+2014-12-16 16:37:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Add something to the debug logs if an RTX AUX element can't be added
+ ... because the application already has a signal handler set up here.
+
+2014-11-21 14:13:34 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: add retransmission support according to RFC4588
+ Based on the client-rtpaux example
+
+2014-12-16 13:25:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/osxvideo/osxvideosink.m:
+ osxvideosink: clear rectangle structures before use
+
+2014-12-09 15:09:56 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Always set format
+ Right now we try to be clever by detecting if device format have
+ changed or not, and skip setting format in this case. This is valid
+ behaviour with V4L2, but it's also very error prone. The rational
+ for not setting these all the time is for speed, though I can't
+ measure any noticeable gain on any HW I own. Also, until recently,
+ we where doing get/set on the format for each format we where
+ probing, making it near to impossible that the format would match.
+ This also fixes bug where we where skipping frame-rate setting if
+ format didn't change.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740636
+
+2014-12-15 18:30:01 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ videocrop: Remove todo about caps filter
+ The filter is already interected.
+
+2014-12-15 18:19:05 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocrop.h:
+ videocrop: Make sure new crop is applied
+ Since "basetransform: Fix caps equality check" commit a7f357,
+ set_info() will not be called anymore if crop didn't change
+ the caps. This is fixed by setting "need_update" boolean when
+ cropping properties has been changed, and then applying these
+ if they where not applied before rendering the next frame. This
+ patch also fixed the locking, dropping un-needed custom lock,
+ and no holding needless lock while doing the operation as we
+ already hold the streaming lock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740787
+
+2014-12-12 18:10:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Prefer filter caps order while getting caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-09 13:38:26 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Add some error handling around channel layout parsing
+ For now we just spit a warning and ignore the channel layout if we can't
+ support it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-08 22:38:22 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Take lock around sink/source before accessing the ringbuffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 21:06:27 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudiosrc: Probe channel layout too
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 20:32:04 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Only fix up channels/layout for PCM caps while probing
+ It's unlikely that setting a channel layout will do much for AC3/DTS
+ streams. If we find at some point that it does make sense, we can
+ perform the structure copying unconditionally (i.e., the current code is
+ wrong, since AC3/DTS will get two structures now - one with the channel
+ layout, one without).
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 19:41:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudiosrc: Implement caps probing
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 19:29:57 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: Bind audio device to audio unit early
+ We want to bind the device during open so that subsequent format queries
+ on the audio unit are as specific as possible from that point onwards.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-29 23:16:30 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Fix up caps querying a bit
+ This should make caps queries correct in PAUSED and higher as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 22:32:36 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Move osxaudiosrc-specific code out of the generic path
+ Avoids one layering violation (GstCoreAudio referring to
+ GstOsxAudioSrc).
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 22:23:17 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudioringbuffer.h:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Clean up a GstCoreAudio -> GstOsxAudioSrc/Sink reference
+ Now that device selection has no sink/source-specific bits, we can have
+ generic device selection for this path. We do need to now track state
+ changes so we can look up the final device_id once the device is open,
+ though.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 19:40:52 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Move device caps probing to get_caps()
+ This should be preferred to running the probe at device open time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 18:37:02 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: Make some debug code compile conditionally
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 15:06:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudioringbuffer.h:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Move device selection to ringbuffer->open_device()
+ This is conceptually the right thing to do, and allows us to correctly
+ catch errors in device selection as well, which we could not do while
+ creating the ringbuffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 14:34:34 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Consolidate input and output code paths a bit
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-21 11:54:18 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ Deinterlace: in query_caps return only supported formats if filter is interlaced
+ In some cases the currently set GstVideoInfo is not interlaced, but
+ upstream caps are interlaced and the info is passed in the filter,
+ we should take that info into account and make sure that we do not
+ consider that case as a "pass through" case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741407
+
+2014-12-12 11:06:17 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix debug statement
+ It was using the non-increasing offset variable, which made that statement
+ not so useful :)
+
+2014-12-12 11:03:15 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add macros for the various timescale conversions
+ This helps make the code more readable and avoid future bad usage of
+ scaling function argument order.
+
+2014-12-11 10:16:06 +0100 Patrick Radizi <patrickr@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: fix potential crash when shutting down
+ A race condition in the state change function may cause buffers
+ to be unreffed while they are still used by the streaming thread
+ in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
+ up to the parent class first in the state change function to
+ make sure streaming has stopped and only then free those buffers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741381
+
+2014-12-12 00:42:06 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Copy flags of the overall segment to output segments
+ Preserve the segment flags of the overall demux segment on the output
+ segments for each pad.
+
+2014-12-09 02:43:00 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: use 64bit chunk_offset
+ https://bugzilla.gnome.org/show_bug.cgi?id=741279
+
+2014-12-10 17:39:17 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix rounding errors in duration update
+ Make sure we store updated segment stop/duration with the same
+ granularity as the duration timescale.
+ And add more debug
+
+2014-12-10 16:55:44 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Update duration when we get more information
+ When dealing with fragmented files, we will get more accurate duration
+ information via the mfra and moof atoms.
+ In order for playback to not stop at the initial duration (from the
+ moov atom), we need to check and update the various duration variables
+ when we find more information.
+ Fixes playback of fragmented files in pull mode
+
+2014-12-10 15:08:40 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove variable assignments never read
+ As detected by clang/scan-build
+
+2014-12-10 14:56:06 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Use GstClockTime for nanosecond-based time variables/fields
+ Avoids confusion with timescaled-based variables and bytes (offset)
+ variables.
+ And use GST_CLOCK_TIME_NONE where applicable
+
+2014-12-03 14:47:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/debugutils/gstpushfilesrc.h:
+ pushfilesrc: Add TIME SEGMENT capability
+ Adds a new set of properties to make pushfilesrc output a TIME SEGMENT
+ (instead of the filesrc BYTE SEGMENT).
+ When time-segment is set to True the following will happen:
+ * Seeks are refused (data starts from the beginning of the file)
+ * The BYTE segment will be replaced by a TIME segment with the values
+ specified in the various properties
+ * The first outgoing buffer will have a timestamp set on it (by default
+ it has a value of GST_CLOCK_TIME_NONE)
+
+2014-12-10 11:35:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Also only unref caps if they're not NULL
+
+2014-12-10 11:34:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: gst_pad_get_allowed_caps() will return NULL if there is no peer
+
+2014-12-09 16:38:38 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vpXenc: CLOCK_TIME_NONE is not a valid min_latency value
+ We should just use 0 if we do not have the information
+
+2014-12-03 17:26:56 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Use an empty iterator in iterate_internal_link when no links
+ And not a NULL Iterator, so it is consistent with the way it usually
+ works and avoid user to need a different code paths to handle that.
+
+2014-12-09 14:01:50 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set buffer interlace flags when field is V4L2_FIELD_INTERLACED
+ If v4l2_buffer.field is V4L2_FIELD_INTERLACED, we set corresponding
+ GstVideoBuffer flags depending on the video standard.
+ According to V4L2 specification, M/NTSC transmits the bottom field
+ first, all other standards the top field first.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737603
+
+2014-12-08 21:26:18 +0100 Patrick Radizi <patrickr@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Fixes buffer leak when using SPS/PPS
+ Fixes a buffer leak that would occurr if the pipeline was shutdown
+ while a SPS/PPS header was being created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741271
+
+2014-12-09 04:43:29 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/effectv/gstaging.c:
+ agingtv: fix memcpy when no color aging requested.
+ video_size is the size in pixels, actual size of the memcpy
+ has to be stride * height.
+
+2014-12-07 17:33:51 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Workaround libv4l2 RW emulation bug
+ When libv4l2 emulates RW mode on top of MMAP devices, the queues are
+ only initialized on first read. The problem is that poll() will fail
+ if called before the queues are initialized and streaming. Workaround
+ this by doing a zero size read when pool is started in that IO mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740633
+
+2014-12-07 17:27:37 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Fix RW io mode
+ In RW, allocator can be null, max_buffers can be zero, and we need not
+ to wait while the queue is empty since there is no queue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740633
+
+2014-12-03 16:40:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Cleanup uneeded check and cases
+ There is nothing in between the break and the "done:" anymore, plus
+ USERPTR and DMABUF_IMPORT case is exactly the same.
+
+2014-12-03 17:07:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2pool: Fix CREATE_BUFS support for capture
+ This patch fixes CREATE_BUFS support for capture devices. Initially we
+ would only try and allocate more buffers when the copy threshold
+ is reached. When the threshold was not set (needed) it would never
+ happen. Another problem is that on capture side, acquire returns
+ filled buffer, hence need to pool. We need to set a special flag to
+ force allocation to happen.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741134
+
+2014-12-03 16:27:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix CREATE_BUF probing
+ Current for every memory type we where probing MMAP CREATE_BUFS ioct.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741134
+
+2014-11-18 16:52:40 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: set framerate 0/1 when duration is not known
+ https://bugzilla.gnome.org/show_bug.cgi?id=740130
+
+2014-12-04 17:25:55 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: More fixes for reverse playback
+ When seeking or finding the previous keyframe, do
+ comparisons against targets and segments using composition time
+ to correctly decide which sample times match.
+
+2014-12-03 11:12:55 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Use an empty iterator in iterate_internal_link when no links
+ We used to setup an iterator with 1 GValue set with a NULL object
+ pointer which is not the normal way to do that. Instead we should make
+ sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.
+
+2014-12-03 13:20:57 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Handle seeks past EOS as a seek to the end
+ Fix reverse playback of every frame by making seeks past/to EOS
+ find the last segment and start there.
+
+2014-12-02 15:33:25 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpmpadepay.c:
+ rtpmpadepay: Relax caps to allow any clock-rate
+ Some Wowza setups seem to send an invalid non-90000 clock-rate.
+
+2014-12-01 21:04:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't use GST_CLOCK_TIME_NONE in non GstClockTime variables
+ Use -1 instead as those are gint64/guint64 variables and not GstClockTime
+
+2014-11-07 17:06:49 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.h:
+ v4l2allocator: fix gst_v4l2_allocator_stop prototype
+ gst_v4l2_allocator_stop returns a GstV4l2Return, not a gboolean.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739792
+
+2014-11-07 16:41:52 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: unref pool when v4l2_allocator_new() fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=739791
+
+2014-11-30 17:52:47 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/v4l2_calls.h:
+ v4l2: Remove last include to linux/videodev2.h
+ We now use and update our internal copy so we no longer have to ifdef
+ the entire code for features and defines that where added over the
+ years.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740905
+
+2014-08-24 13:38:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: implement seeking in fragmented mp4 files in pull mode based on the mfra table
+
+2014-11-29 15:25:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: use track fragment decoding time (tfdt) in parse_trun() for interpolation
+ As fallback if we don't have any existing samples
+ as reference point yet.
+ Based on patch by David Corvoysier <david.corvoysier@orange.com>
+
+2014-11-29 14:37:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: parse mfra random access box for fragmented mp4 files
+ If it's present, and we operate in pull mode.
+
+2014-08-15 14:58:26 +0200 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: stop parsing headers for fragmented mp4s at the first moof
+ Currently during header parsing, we scan through the entire file
+ and skip every moof+mdat chunk for fragmented mp4s, which makes
+ start-up incredibly slow. Instead, just stop at the first moof
+ chunk when have a moov, and start exposing the streams, so we
+ can go and start handling the moofs for real.
+
+2014-11-29 13:59:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: add interactive gdkpixbufoverlay test
+ Just need to fix the coordinate system now so
+ that negative offsets are actually negative
+ and not flipped to position things from the
+ opposite border.
+
+2014-11-29 13:53:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
+ gdkpixbufoverlay: add "pixbuf" property
+ So we can set a GdkPixbuf directly instead of
+ reading it from an image file on the file system.
+
+2014-11-29 13:23:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/Makefile.am:
+ * ext/gdk_pixbuf/pixbufscale.c:
+ * ext/gdk_pixbuf/pixbufscale.h:
+ gdkpixbuf: remove pixbufscale code that was never ported
+ Don't think we'll need this again.
+
+2014-11-29 18:35:42 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtprtxreceive: Use offset when copying header
+ The header is not always at the start of the packet, so we need to compute
+ the offset first.
+
+2014-11-28 13:12:46 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/taglib/gstapev2mux.cc:
+ apev2mux: write APE tags at end for wavpack files
+ http://www.wavpack.com/file_format.txt:
+ "Both the APEv2 tags and/or ID3v1 tags must come at the end of the
+ WavPack file, with the ID3v1 coming last if both are present."
+ WavPack files that contain APEv2 tags at the beginning of the files
+ are unplayable on players that use FFmpeg (like VLC) and most other
+ software (except Banshee). Players that use libwavpack directly can
+ play the files because it skips the tags, but does not recognize the
+ tag data at that location.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711437
+
+2014-11-28 10:41:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-segment-seeks.c:
+ tests: add interactive test for gapless playback using SEGMENT seeks
+ Not working too well yet, there are glitches even with WAV or FLAC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=692368
+
+2014-11-26 10:33:09 +0300 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstaspectratiocrop.h:
+ aspectratiocrop: Handle resolution changes properly
+ When an caps-event is received, we must immediately change the crop
+ to videocrop correctly changed caps-event dimension, otherwise the
+ videocrop will first use the previous value of the crop that when
+ resizing video to a smaller resolution may cause an error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740671
+
+2014-11-27 17:10:53 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-27 11:20:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/test-accurate-seek.c:
+ test: use gst_util_uint64_scale_round() for timestamp to sample calculation
+
+2014-11-27 11:16:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-accurate-seek.c:
+ tests: add interactive test for accurate seeking
+ For some audio formats.
+ https://bugzilla.gnome.org/show_bug.cgi?id=655276
+
+2014-11-26 16:04:26 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ isomp4: Check presence of mfhd in moof
+ The 'mfhd' atom is mandatory in 'moof'. We can later on check whether
+ the fragment number properly increases
+
+2014-11-26 15:59:36 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ isomp4: Fix mfro and tfra atom dumping
+ mfro was skipping the version/flags
+ tfra had wrong byte_reader return value checks
+
+2014-11-26 15:58:26 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ * gst/isomp4/qtdemux_dump.h:
+ * gst/isomp4/qtdemux_types.c:
+ isomp4: Add mfhd atom dumping
+
+2014-11-27 00:15:02 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Handle empty segments when seeking in reverse play.
+ Empty segments in an edit list have a media_start time of -1,
+ as they don't actually play any media. Allow for that when
+ aligning to the reference stream in reverse play.
+
+2014-11-24 10:36:54 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ Revert "v4l2allocator: Remove unused variable"
+ This reverts commit ad4480d53408a4d97ab531174ef37f258f3253c0.
+
+2014-11-24 10:36:30 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ Revert "v4l2: move vb_queue probing from allocator to v4l2object"
+ This reverts commit ec6b8b84af719d828ddd91c724e715c0b4a556bc.
+
+2014-11-24 10:33:29 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ Revert "v4l2object: allow to automatic selection of dmabuf"
+ This reverts commit e6c2ad5571e5dedb212287efe238eb450032cd4f.
+
+2014-11-23 16:34:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * REQUIREMENTS:
+ REQUIREMENTS: update a little
+ People actually look at that it seems.
+
+2014-11-23 16:22:12 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/icydemux/Makefile.am:
+ icydemux: does not need to link against zlib
+
+2014-11-22 21:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * ext/speex/gstspeexdec.h:
+ * ext/speex/gstspeexenc.h:
+ speex: remove support for ancient speex versions
+
+2014-11-21 11:21:18 +0100 Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: log connection events at info level
+ https://bugzilla.gnome.org/show_bug.cgi?id=739305
+
+2014-10-20 13:00:37 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: ensure rtx_retry_period >= 0
+ https://bugzilla.gnome.org/show_bug.cgi?id=739344
+
+2014-11-21 11:44:24 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Remove unused variable
+ this was introduced by commit ec6b8b
+ https://bugzilla.gnome.org/show_bug.cgi?id=699382
+
+2014-11-16 12:34:17 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Handle corrupted buffer with empty payload
+ This allow skipping buffer flagged with ERROR that has no payload.
+ This is typical behaviour when a recovererable error occured during
+ capture in the driver, but that no valid data was ever written into that
+ buffer. This patch also translate V4L2_BUF_FLAG_ERROR into
+ GST_BUFFER_FLAG_CORRUPTED. Hence decoding error produce
+ by decoder due to missing frames will now be correctly marked. Finally,
+ this fixes a buffer leak when EOS is reached.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740040
+
+2014-11-21 16:36:15 +0100 Benjamin Gaignard <benjamin.gaignard@linaro.org>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: allow to automatic selection of dmabuf
+ If the v4l2 queue support dmabuf select this buffer pool mode
+ and update the query with allocator.
+ This patch only concern exporting dmabuf and not importing dmabuf
+ fd from downstream element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=699382
+
+2014-11-21 16:13:05 +0100 Benjamin Gaignard <benjamin.gaignard@linaro.org>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: move vb_queue probing from allocator to v4l2object
+ The goal is to make those information available in v4l2_object
+ to be able later to select the best allocation method for the pool
+ https://bugzilla.gnome.org/show_bug.cgi?id=699382
+
+2014-11-20 22:42:59 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: Fix up new_jitterbuffer signal prototype
+
+2014-11-20 20:19:25 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Document how to control per-SSRC retransmission
+
+2014-11-20 20:18:45 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * docs/design/design-rtpretransmission.txt:
+ doc: Trivial spelling and consistency update
+
+2014-11-20 13:14:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: put 0-byte at the end of events
+ Put a 0-byte at the end of the event string. Does not break ABI because
+ old depayloaders will skip the 0 byte (which is included in the length).
+ Expect a 0-byte at the end of the event string or a ; for old
+ payloaders.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=737591
+
+2014-11-20 12:40:28 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ rtpgstdepay: avoid buffer overread.
+ Check that a caps event string is 0 terminated and the event string is
+ terminated with a ; to avoid buffer overreads.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591
+
+2014-11-20 10:45:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: don't limit max video resolution to 4096x4096
+ MAX isn't entirely correct as upper limit either,
+ it should really be MAXUINT32, but it's unlikely
+ to be a problem in the near future.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740407
+
+2014-11-19 15:06:00 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix leak for mikey base64 decoded key-mgmt
+ https://bugzilla.gnome.org/show_bug.cgi?id=740392
+
+2014-11-20 09:01:38 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videofilter/gstvideobalance.c:
+ videobalance: fix unhandled format in passthrough
+ In passthrough we can handle all formats.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740387
+
+2014-11-19 16:12:38 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Restrict resyncing to TS regressions
+ The behavior of resyncing video and audio indepen-
+ dently can cause A/V desyncs. Lets restrict resyncs
+ to jumps backward for now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736397
+
+2014-11-17 23:16:03 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixer2.h:
+ videomixer: fix up QoS handling for live sources
+ Only attempt adaptive drop when we are not live
+ https://bugzilla.gnome.org/show_bug.cgi?id=739996
+
+2014-11-10 22:34:39 +0100 Henning Heinold <henning@itconsulting-heinold.de>
+
+ * tests/examples/rtp/client-PCMA.py:
+ * tests/examples/rtp/server-alsasrc-PCMA.py:
+ examples: port python rtp PCMA client/server tests to 1.0
+ https://bugzilla.gnome.org/show_bug.cgi?id=739930
+
+2014-06-04 12:11:10 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: set the channel positions using the appropriate API
+ This avoids _set_format setting the unpositioned flag when passed
+ NULL as channel positions, as it would not be cleared when setting
+ actual channel positions later.
+
+2014-11-01 22:39:41 +0100 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vpx: mark arnr-type properties as deprecated and set them to no-op
+ ARNR type control in libvpx has been deprecated so this commit mark the
+ vp8enc and vp9enc associated properties as deprecated and change their
+ behavior to just display a warning message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739476
+
+2014-11-10 13:16:01 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpmanager: Trivial typo fix
+
+2014-11-09 11:04:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Use G_DEFINE_TYPE() to register the pad instead of manually registering it
+
+2014-11-06 15:37:28 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: make GstMatroskamuxPad get_type() function thread-safe
+ https://bugzilla.gnome.org/show_bug.cgi?id=739722
+
+2014-11-07 16:11:24 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: fix error message if allocator is already active
+ https://bugzilla.gnome.org/show_bug.cgi?id=739789
+
+2014-11-06 21:21:40 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Improve buffer validation
+ Improve buffer validation by making sure each memory are the right
+ one and that each memory is writable. This fixes tearing issues in
+ case downstream uses gst_buffer_make_writable() or other type
+ of GstBuffer copy where memory are only reffed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739754
+
+2014-11-06 21:38:43 +0100 Josep Torra <n770galaxy@gmail.com>
+
+ * gst/rtsp/Makefile.am:
+ rtsp: fix build in gst-uninstalled setup
+
+2014-10-29 18:44:43 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/imagefreeze/gstimagefreeze.h:
+ imagefreeze: Handle seqnums
+ https://bugzilla.gnome.org/show_bug.cgi?id=739366
+
+2014-11-04 08:18:41 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/libpng/gstpngdec.c:
+ * ext/libpng/gstpngdec.h:
+ pngdec: change parse logic
+ Right now in parse logic the signature is checked every time the parse function
+ is called, and the whole data is the scanned each and every time, even though the
+ data is scanned in the previous instance. Changing the logic such that, we skip
+ the bytes which are already scanned in the previous instances of parse. This
+ helps in avoiding multiple scan of already scanned data/signature.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737708
+
+2014-11-03 15:26:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer2: reverse order of params for converter
+
+2014-11-03 11:44:28 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: fix typo in flags
+ https://bugzilla.gnome.org/show_bug.cgi?id=739549
+
+2014-11-02 23:33:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2src: fix a couple of minor leaks
+
+2014-11-02 19:42:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: post QoS messages when dropping frames due to QoS
+
+2014-11-02 19:29:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ * gst/goom/gstgoom.h:
+ goom: post QoS messages when dropping frames due to QoS
+
+2014-11-02 19:02:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: tweak writing app tag string a little
+
+2014-11-02 16:51:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/level/gstlevel.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ Sprinkle some G_PARAM_DEPRECATED and #ifndef GST_REMOVE_DEPRECATED
+
+2014-11-02 16:58:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/level.c:
+ tests: don't use deprecated property in level unit test
+
+2014-11-02 13:06:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property
+ Properties are so much more useful if you can actually set
+ and get their values.
+
+2014-10-30 17:41:19 +0000 Simon Farnsworth <simon.farnsworth@onelan.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: Clean up interlace support
+ Rather than try and guess interlace support as part of checking supported
+ sizes, look for interlace support specifically in its own function.
+ As a cleanup, use V4L2_FIELD_ANY when probing sizes, which should result in
+ the driver doing the right thing.
+ With my capture setup, this gets me the following sample caps:
+ For 1080i resolution:
+ video/x-raw, format=(string)YUY2, width=(int)1920, height=(int)1080, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)interleaved, framerate=(fraction){ 25/1, 30/1 }
+ For 720p resolution:
+ video/x-raw, format=(string)YUY2, width=(int)1280, height=(int)720, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, framerate=(fraction){ 50/1, 60/1 }
+ For 576i/p resolution (both possible at the point of query):
+ video/x-raw, format=(string)YUY2, width=(int)720, height=(int)576, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string){ progressive, interleaved }, framerate=(fraction){ 25/1, 50/1 }
+ This, in turn, makes 576i work correctly; with the old code,
+ the caps would be interlace-mode=progressive for interlaced video.
+ https://bugzilla.gnome.org/show_bug.cgi?id=726194
+
+2014-11-01 12:18:02 +0100 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * ext/vpx/gstvp8utils.h:
+ vpx: remove compatibility defines
+ We are guaranteed to have VPX_IMG_FMT_I420, VPX_PLANE_Y,
+ VPX_PLANE_U and VPX_PLANE_V as we require libvpx > 1.1.0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739476
+
+2014-11-01 15:33:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * ext/wavpack/gstwavpackcommon.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * ext/wavpack/gstwavpackenc.c:
+ wavpack: remove support for ancient API version
+
+2014-11-01 10:14:31 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8: Use VP8 encoding name
+ Both Firefox and Chrome uses VP8 as the encoding in their SDP.
+ Adding this now defacto standard name removes the need for special
+ case in SDP parsing code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2014-11-01 11:59:26 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpmp2tpay.c:
+ rtpmp2tpay: fix up template caps so we can output the default pt 33
+ Add fixed payload type for mp2t to template caps as well, so
+ our output caps match the advertised default pt. Fixes a
+ regression from 1.2.
+ There's still something wrong with caps negotiation though,
+ rtpmp2tpay payload=96 ! fakesink will not output caps with
+ payload=96.
+
+2014-10-30 15:37:36 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: mikey related memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739430
+
+2014-06-10 10:04:07 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/speex/gstspeexenc.c:
+ * ext/speex/gstspeexenc.h:
+ speexenc: update output segment stop time to match clipped samples
+ This will let oggmux generate a granpos on the last page that properly
+ represents the clipped samples at the end of the stream.
+
+2014-06-10 10:59:13 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflacenc.h:
+ flacenc: update output segment stop time to match clipped samples
+ This will let oggmux generate a granpos on the last page that properly
+ represents the clipped samples at the end of the stream.
+
+2014-10-07 15:29:33 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: cleanly handle streamon failure for output device
+ On streamon failure, the queued buffer is not released from the
+ bufferpool class point of view because it is queued to the driver and
+ the flush logic is not performed since we are not in streaming state.
+ It causes the v4l2 bufferpool to always return that stop method failed
+ and to leak v4l2 objects and buffers.
+ This commit solve this by performing the flush logic in error case, ie
+ flushing the allocator and restoring queued buffer state to non-queued.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738102
+
+2014-10-08 10:31:21 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: implement dispose method
+ Unref objects in dispose method rather than in finalize in order to
+ prevent circular reference.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738102
+
+2014-10-08 10:35:14 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: check that allocator is non null when stopping pool
+ Otherwise, we could dereference NULL allocator when the stop method is
+ called by the GstBufferPool's finalize method.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738102
+
+2014-10-09 12:15:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Implement unlock/unlock_stop
+ This will prevent deadlocks, but will also properly flush the pool and allocator
+ when going to READY state. It should also fix issues reported on mailing list
+ when seeking is performed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738152
+
+2014-10-28 21:32:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pulse/pulsedeviceprovider.h:
+ * sys/v4l2/gstv4l2deviceprovider.h:
+ * sys/v4l2/gstv4l2tuner.h:
+ pulse, v4l2: add missing G_END_DECLS in some places
+
+2014-10-27 17:57:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-27 11:08:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/aacparse.c:
+ aacparse: Fix unit test now that we always have profile/level in the caps
+
+2014-10-26 14:55:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Parallelise 'make check-valgrind'
+ Some of the RTP unit tests are very flaky and will
+ fail more often with the CPU maxed out fully. Those
+ tests need to be fixed in any case though, they also
+ fail on slower machines and also occasionally with
+ normal 'make check'.
+
+2014-10-26 11:47:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Always set profile/level on the caps
+ We have the information already, so why not use it?
+
+2014-10-25 12:36:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: fix crash on some 32-bit systems
+ Make sure to pass right number of bits to gst_structure_new()
+ which is a vararg function.
+ Fixes elements/rtpaux unit test on ppc32.
+
+2014-10-25 00:56:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rgvolume.c:
+ tests: fix rgvolume test on big-endian systems
+
+2014-10-25 00:53:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/mulawdec.c:
+ * tests/check/elements/mulawenc.c:
+ tests: fix mulawdec/mulawenc test for big endian systems
+
+2014-10-24 23:48:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/interleave/interleave.c:
+ interleave: intersect result with filter caps in caps query
+ Fixes crash in audiotestsrc because of an unsupported format
+ getting negotiated on big-endian systems with
+ audiotestsrc ! interleave ! audioconvert ! wavenc
+
+2014-10-23 15:46:13 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ * ext/pulse/pulsedeviceprovider.h:
+ pulse: remove some unused typedefs
+
+2014-10-22 15:28:44 +0200 Ananda <ananda@latelier23.com>
+
+ * ext/speex/gstspeexdec.c:
+ * ext/speex/gstspeexenc.c:
+ speex: Fix segfault when resetting the codecs multiple times
+ https://bugzilla.gnome.org/show_bug.cgi?id=738793
+
+2014-10-22 22:50:54 +0530 Arun Raghavan <arun@accosted.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Temporarily disable stream status posting
+ We need a mechanism in PulseAudio to allow running code outside the
+ mainloop lock. Then we'd be able to post to the bus (taking the
+ GST_OBJECT_LOCK), without worrying about locking order with the mainloop
+ lock, which is the current cause of deadlocks while trying to post the
+ stream status messages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736071
+
+2014-10-22 15:04:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: limit the retry frequency
+ When the RTT and jitter are very low (such as on a local network), the
+ calculated retransmission timeout is very small. Set some sensible lower
+ boundary to the timeout by adding a new property. We use the packet
+ spacing as a lower boundary by default.
+
+2014-10-22 13:40:58 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ gstrtpjitterbuffer: add "rtx-min-delay" property
+ This property is useful to set a min time to wait before sending a
+ retransmission event.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378
+
+2014-10-22 13:29:48 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Refactor code
+ Refactor some code dealing with calculating various timeouts.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=735378
+
+2014-10-10 19:50:06 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: fix Early Feedback Transmission
+ In early retransmission we are allowed to schedule 1 regular RTCP packet
+ at an earlier time. When we do that, we need to set allow_early to FALSE
+ and ignore/drop (or merge) all future requests for early transmission.
+ We now first check if we can schedule an early RTCP and if we can,
+ actually prepare the data for the next RTCP interval.
+ After we send the next regular RTCP after the early RTCP, we set
+ allow_early to TRUE again to allow more early requests.
+ Remove the condition for the immediate feedback for now.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
+
+2014-10-21 13:01:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:10:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: make debug line less confusing
+
+2014-10-21 12:58:13 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 36388a1 to a8c8939
+
+2014-07-02 17:50:35 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ jitterbuffer: rework resync handling
+ Add a need-resync state, this is when we need to try to lock on to a
+ time/RTPtime pair.
+ Always check the RTP timestamps and if they go backwards, mark ourselves
+ as need-resync.
+ Only resync when need-resync is TRUE and we have a valid time. Otherwise
+ we keep the old values. This avoids locking on to an invalid time and
+ causing us to timestamp everything with -1.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
+
+2014-10-03 17:28:06 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: set full stream caps on internal src TCP pads
+ Set the complete stream caps on the TCP internal src pads. Otherwise,
+ ptdemux will not properly detect the caps change.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737868
+
+2014-10-17 22:23:27 +0200 Sjoerd Simons <sjoerd@luon.net>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: Don't set PROXY_CAPS flag on the src pad
+ rtpmux behaves like a funnel in that it forwards whatever upstream is
+ sending buffers. So setting proxy caps doesn't make sense as the
+ upstream don't have to have compatible caps, thus resulting in an empty
+ caps set as a result of a caps query. Instead set fixed caps just
+ as funnel does.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738722
+
+2014-10-20 11:57:38 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/videobox/gstvideobox.c:
+ videobox: critical error when element properties set as max/min
+ left, right, top, bottom can be set from range of -2147483648 to 2147483647
+ when i launch the videobox element with that values, it gives a critical error
+ (gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
+ This happens because min cannot be equal to max.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738838
+
+2014-10-15 17:45:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ Revert "rtp: add h265 RTP payloader + depayloader"
+ This reverts commit d06ba9051f904a7eb482c07a97a1827169158663.
+ This breaks the build, as it depends on parser API in -bad.
+
+2014-10-15 17:34:50 +0200 Jurgen Slowack <jurgen.slowack@barco.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtp: add h265 RTP payloader + depayloader
+
+2014-10-05 21:24:27 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavenc/gstwavenc.h:
+ wavenc: Support RF64 format
+ https://bugzilla.gnome.org/show_bug.cgi?id=725145
+
+2014-10-11 11:18:42 +1100 David Sansome <me@davidsansome.com>
+
+ * gst/equalizer/gstiirequalizer.c:
+ equalizer: Don't call iirequalizer's transform_ip in passthrough mode
+ It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737886
+
+2014-10-10 18:30:07 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsource: Rename seqnum-base to seqnum-offset in caps
+ This was modified back in 1.0 in GstRtpBasePayload
+
+2014-10-10 18:11:19 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ * tests/check/elements/dtmf.c:
+ rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-10-10 17:30:24 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-10-06 14:23:22 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/filters.c:
+ goom2k1: removing block of code that does nothing
+ The loop in zoomFilterSetResolution is meant to change the values in the
+ zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
+ but no conditions that change the value of decc are ever met and the array is
+ filled with zero for each element. Which is the initial state of the
+ array before the loop begins.
+ The loop does nothing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728353
+
+2014-10-04 17:17:13 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: don't log all clock_rate changes as warnings.
+ We never initialize clock_rate explicitly, therefore it is 0 by default. The
+ parameter is a uint32 and the only caller ensure that it is >0, therefore it
+ won't become -1 ever.
+
+2014-10-02 14:26:08 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Fix lifetime of stream headers and queued buffers
+ Stream headers are updated whenever ::set_caps is called, so we can't assume
+ they'll be valid before the message body is written out. We *can* assume that
+ for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those.
+ Also, add some debug logging for stream header interactions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737771
+
+2014-10-02 03:26:22 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix memory leak when prepending ADTS headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=737761
+
+2014-09-23 10:48:09 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * gst/interleave/interleave.c:
+ * gst/interleave/interleave.h:
+ interleave: interleave samples following the Default Channel Ordering
+ In order to have a full mapping between channel positions in the audio
+ stream and loudspeaker positions, the channel-mask alone is not enough:
+ the channels must be interleaved following some Default Channel Ordering
+ as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.
+ As a Default Channel Ordering use the one implied by
+ GstAudioChannelPosition which follows the ordering defined in SMPTE
+ 2036-2-2008[2].
+ NOTE that the relative order in the Top Layer is not exactly the same as
+ the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
+ using so may channels are already aware of such discrepancies.
+ [1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
+ [2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf
+ Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
+
+2014-10-02 10:10:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavenc/gstwavenc.c:
+ wavenc: Send CAPS event after the pad was activated
+ Otherwise the CAPS event will be dropped and we never configure any caps at
+ all, leading to weird behaviour in many situations. Especially header
+ rewriting is not going to work if a capsfilter is after wavenc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737735
+
+2014-10-01 23:12:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Add some more useful debug logging
+
+2014-10-01 23:05:03 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Free queued buffers in ::reset
+ ::render sets a new callback for writing out new buffers only if there aren't
+ already buffers queued for writing with a previously-scheduled callback.
+ However, if the previously-scheduled callback is interrupted by a state change
+ (either manually or due to an error) and there are still buffers in the queue,
+ restarting the pipeline will result in buffers being queued forever, and no
+ callbacks will ever be scheduled, and no buffers will be written out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737739
+
+2014-10-01 17:29:29 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Actually use the correct GstVideoInfo for conversion
+
+2014-10-01 17:24:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Revert the last commit and handle resolutions differences properly
+ This is about converting the format, not about converting any widths and
+ heights. Subclasses are expected to handler different resolutions themselves,
+ like the videomixers already do properly.
+
+2014-10-01 17:12:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: GstVideoConverter currently can't rescale and will assert
+ Leads to ugly assertions instead of properly erroring out:
+ CRITICAL **: gst_video_converter_new: assertion 'in_info->width == out_info->width' failed
+
+2014-09-30 11:35:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp8enc/vp9enc: Protect the encoder with a mutex in all situations
+
+2014-09-30 11:31:43 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: Allow caps renegotiation
+ https://bugzilla.gnome.org/show_bug.cgi?id=726329
+
+2014-09-30 11:28:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: finish() and drain() should return a GstFlowReturn
+
+2014-03-14 12:59:02 +0100 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Allow caps renegotiation
+ https://bugzilla.gnome.org/show_bug.cgi?id=726329
+
+2014-09-29 11:49:45 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: set colorspace for output devices
+ When the v4l2 device is an output device, the application shall set the
+ colorspace. So map GStreamer colorimetry info to V4L2 colorspace and set
+ on set_format. In case we have no colorimetry information, we try to
+ guess it according to pixel format and video size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737579
+
+2014-09-29 22:48:16 +0530 Arun Raghavan <arun@accosted.net>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesrc.c:
+ pulse: Add some documentation about threading and synchronisation
+ This gives a quick introduction to how the pulsesink/pulsesrc code
+ interacts with the pa_threaded_mainloop that we start up to communicate
+ with the server.
+
+2014-09-29 20:18:08 +0530 Arun Raghavan <arun@accosted.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Make emitting stream status messages synchronous
+ The stream status messages are emitted in the PA mainloop thread, which
+ means the mainloop lock is taken, followed by the Gst object lock (by
+ gst_element_post_message()). In all other locations, the order of
+ locking is reversed (this is unavoidable in a bunch of cases where the
+ object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get
+ control to take the mainloop lock).
+ The only way to guarantee that the defer callback for stream status
+ messages doesn't deadlock is to either stop posting those messages, or
+ make sure that the message emission is completed before we proceed to
+ any point that might take the object lock before the mainloop lock
+ (which is what we do after this patch).
+ https://bugzilla.gnome.org/show_bug.cgi?id=736071
+
+2014-09-16 12:12:49 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * gst/wavenc/gstwavenc.c:
+ wavenc: print channel masks in hexadecimal
+
+2014-09-27 16:01:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.h:
+ v4l2: remove redundant struct declaration
+
+2014-09-26 13:46:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix compiler warnings
+ gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
+ 'GstRTSPResult' [-Werror,-Wenum-conversion]
+ res = gst_sdp_message_new (&sdp);
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
+ gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
+ 'GstRTSPResult' [-Werror,-Wenum-conversion]
+ res = gst_sdp_message_parse_uri (uri, sdp);
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+2014-09-25 15:01:14 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: make demuxer reusable
+ Remove pads from flow combiner and reset last
+ flow return to FLOW_OK by resetting the flow combiner.
+ This prevents FLOW_FLUSHING when trying to re-use the
+ demuxer after setting it back to NULL/READY state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737359
+
+2014-09-24 16:46:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/gstcms.c:
+ * gst/videomixer/gstcms.h:
+ * gst/videomixer/videoconvert.c:
+ * gst/videomixer/videoconvert.h:
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixer2pad.h:
+ * gst/videomixer/videomixerorc-dist.c:
+ * gst/videomixer/videomixerorc-dist.h:
+ * gst/videomixer/videomixerorc.orc:
+ videomixer: use video library code instead of copy
+
+2014-09-18 16:39:19 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ audioparsers: Added index check before using the index
+ https://bugzilla.gnome.org/show_bug.cgi?id=736878
+
+2014-09-23 23:33:37 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Do not infer DTS on buffers from sparse streams.
+ DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
+ This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-18 17:08:37 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/goom/ifs.c:
+ goom: Clarified precedence between % and ?
+ https://bugzilla.gnome.org/show_bug.cgi?id=736887
+
+2014-09-18 17:59:31 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtsp: clarify expression so operator precedence is clear
+ https://bugzilla.gnome.org/show_bug.cgi?id=736903
+
+2014-09-18 16:04:03 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * ext/libpng/gstpngdec.c:
+ * gst/alpha/gstalpha.c:
+ * gst/audiofx/audiodynamic.c:
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/gstscaletempo.c:
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavimux.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/rtpsession.c:
+ Miscellaneous minor cleanups
+ Fix redundant variables and assignments,
+ and unreachable breaks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736875
+ https://bugzilla.gnome.org/show_bug.cgi?id=736876
+ https://bugzilla.gnome.org/show_bug.cgi?id=736879
+ https://bugzilla.gnome.org/show_bug.cgi?id=736880
+ https://bugzilla.gnome.org/show_bug.cgi?id=736881
+ https://bugzilla.gnome.org/show_bug.cgi?id=736888
+ https://bugzilla.gnome.org/show_bug.cgi?id=736890
+ https://bugzilla.gnome.org/show_bug.cgi?id=736892
+ https://bugzilla.gnome.org/show_bug.cgi?id=736893
+ https://bugzilla.gnome.org/show_bug.cgi?id=736894
+
+2014-09-24 00:12:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videobox/gstvideobox.c:
+ videobox: remove duplicate assignments
+ https://bugzilla.gnome.org/show_bug.cgi?id=736897
+
+2014-09-23 22:55:48 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Only calculate with durations != -1
+
+2014-09-23 19:08:48 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: collect pad for sparse stream should be created with lock set to false
+ Avoids waiting for buffers from sparse streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-23 19:07:25 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: fix subtitle buffer duration and strip null termination
+ Strip the \0 off the subtitle as we already know the size and also remember
+ to set the duration as buffer copying doesn't do it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-23 19:06:18 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: move subtitle layer above video and set alternate group
+ layer -1 is above video, that is 0
+ And having all subtitles in alternate group 2 means that only one
+ should be selected at a time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-23 09:47:31 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ check/soup: Temporarily disable G_ENABLE_DIAGNOSTIC
+ The SOUP_SERVER_PORT property has been deprecated in recent libsoup
+ versions.
+
+2014-09-23 09:43:05 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ check/soup: Define minimum version required
+ To avoid deprecation warnings
+
+2014-09-19 19:14:28 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Handle mp4a without ESDS atom
+ https://bugzilla.gnome.org/show_bug.cgi?id=736986
+
+2014-09-22 16:15:27 +0200 Linus Svensson <linussn@axis.com>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Fix build problem without XFIXES
+
+2014-09-19 14:34:13 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ dtmf: Removed unused structure members
+ https://bugzilla.gnome.org/show_bug.cgi?id=736883
+
+2014-09-11 13:48:44 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4: fix wrong DAR calculation for PAR <= 1
+ CID #1226452
+ https://bugzilla.gnome.org/show_bug.cgi?id=736396
+
+2014-09-18 16:59:52 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/flv/gstflvdemux.c:
+ flv: Removed unreachable break statements
+ https://bugzilla.gnome.org/show_bug.cgi?id=736884
+
+2014-09-17 16:37:11 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: do not leak encsink pad in error case
+ https://bugzilla.gnome.org/show_bug.cgi?id=736807
+
+2014-09-17 16:23:21 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: do not leak new stream event
+ https://bugzilla.gnome.org/show_bug.cgi?id=736805
+
+2014-09-15 09:08:18 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/y4m/gsty4mencode.c:
+ * gst/y4m/gsty4mencode.h:
+ y4menc: port y4menc to use GstVideoEncoder base class
+ https://bugzilla.gnome.org/show_bug.cgi?id=735085
+
+2014-09-17 13:55:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: OSStatus is not a fourcc, so don't print it as one...
+
+2014-09-16 14:26:08 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: do not leak uid after parsing TOC event
+ https://bugzilla.gnome.org/show_bug.cgi?id=736739
+
+2014-09-16 22:47:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: Declare some more required caps fields in the sink template caps
+ Now only missing are width and height, which are expressed as strings
+ for RTP... so we can't put them into the template caps.
+
+2014-09-16 16:46:07 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.h:
+ gdkpixbufdec: modify wrong packetized mode logic
+ packetized mode is being set when framerate is being set
+ which is not correct. Changing the same by checking the
+ input segement format. If input segment is in TIME it is
+ Packetized, and if it is in BYTES it is not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2014-09-16 11:26:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Remove unused variable and use correct decoder variable name
+
+2014-09-16 11:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: Remove unused variable
+
+2014-09-16 13:24:15 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpeggdec: modify wrong packetized mode logic
+ packetized mode is being set when framerate is being set
+ which is not correct. Changing the same by checking the
+ input segement format. If input segment is in TIME it is
+ Packetized, and if it is in BYTES it is not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2014-09-16 13:23:16 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: modify wrong packetized mode logic
+ packetized mode is being set when framerate is being set
+ which is not correct. Changing the same by checking the
+ input segement format. If input segment is in TIME it is
+ Packetized, and if it is in BYTES it is not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2014-09-15 14:39:41 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/gstximagesrc.h:
+ * sys/ximage/ximageutil.c:
+ * sys/ximage/ximageutil.h:
+ ximagesrc: Remove unused screen-num property
+ The screen number can be still specified as part of the display-name
+ property (e.g. for screen 1 of display 0 use display-name=":0.1").
+ https://bugzilla.gnome.org/show_bug.cgi?id=736122
+
+2014-09-04 16:10:51 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Draw the cursor only when it is active in the capturing region
+ Use XQueryPointer to check that the pointer is actually active inside
+ the capturing region.
+ This prevents drawing the cursor when the pointer is partially outside
+ of the captured region but not active inside the region; in particular
+ this avoids drawing the "window resize" cursor shapes to the captured
+ image when the mouse pointer crosses a window border.
+ NOTE that this is not only an optimization, this also happen to fix
+ a serious problem in multi-screen setups.
+ Because XFixes gives no information of what screen the pointer is on,
+ ximagesrc was always drawing the cursor on the captured screen even if
+ the mouse pointer was on another screen.
+ For example, when capturing from screen 1 (i.e. display-name=":0.1") the
+ cursor was drawn in the captured image even when the mouse pointer was
+ actually on screen 0, which is wrong and visually confusing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=690646
+
+2014-09-05 11:33:31 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Fix drawing the cursor when it is outside the capturing region
+ When the cursor is partially or totally out of the capturing region on
+ the top side or on the left side, it gets drawn fully inside of the
+ region with its coordinates rounded up to the left or to the top border.
+ This is immediately noticeable when using the xid property to capture
+ a specific window.
+ To fix the issue, allow negative cx and cx coordinates when checking the
+ boundaries before drawing the cursor.
+ NOTE that the boundaries checking calculations still allows the cursor
+ to be drawn when it is only partially outside of the capturing region,
+ but this makes sense and gives a more pleasing visual behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=690646
+
+2014-09-05 00:15:30 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/gstximagesrc.h:
+ ximagesrc: Fix the destination coordinates of the cursor
+ XFixes provides the cursor coordinates relative to the root window, this
+ is not taken into account when using the xid property to capture
+ a specific window, the result is that the cursor gets drawn at the wrong
+ position.
+ In order to fix this consider the window location when calculating the
+ cursor position in the destination image.
+ https://bugzilla.gnome.org/show_bug.cgi?id=690646
+
+2014-09-15 14:51:24 +0200 Peter Korsgaard <peter@korsgaard.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: O_CLOEXEC needs _GNU_SOURCE
+ Similar to 94f3d6fc / bz 709423
+ On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
+ defined when _GNU_SOURCE is specified, so do so.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736670
+
+2014-09-15 18:11:37 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/debugutils/gstcapssetter.c:
+ capssetter: update to 1.0 transform_caps sematics
+ In 1.0, we pass the complete caps to transform_caps to allow for better
+ optimizations. Make this function actually work on non-simple caps
+ instead of just ignoring the configured filter caps.
+
+2014-09-08 14:06:00 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavenc/gstwavenc.h:
+ wavenc: use WAVE_FORMAT_EXTENSIBLE for more than 2 channels
+ https://bugzilla.gnome.org/show_bug.cgi?id=733444
+
+2014-09-12 15:06:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Fix parsing of adtl chunks
+ We have to skip 12 bytes of data for the chunk, and the data size
+ passed to the sub-chunk parsing functions should have 4 bytes less
+ than the data size.
+ Also when parsing the sub-chunks, check if we actually have enough
+ data to read instead of just crashing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736266
+
+2014-09-12 10:55:23 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/udp/gstudpsrc.c:
+ udp: include string.h for memcmp and memset
+ https://bugzilla.gnome.org//show_bug.cgi?id=736528
+
+2014-09-12 13:36:18 +0530 Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: don't bitwise OR the same flag twice
+ https://bugzilla.gnome.org//show_bug.cgi?id=736543
+
+2014-09-12 10:35:36 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: handle real audio 28_8
+ Fixes duplicate check for 14_4.
+ https://bugzilla.gnome.org//show_bug.cgi?id=736543
+
+2014-09-11 14:46:09 +0530 Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: don't OR the same flag twice
+ https://bugzilla.gnome.org/show_bug.cgi?id=736462
+
+2014-09-11 12:52:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: If the server reports "Accept-Ranges: none" don't try range requests
+
+2014-09-10 09:50:45 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Unref pool after usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=736384
+
+2014-09-09 19:03:50 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: Don't rank it for now
+ This will prevent the converter to be picked automatically in case
+ someone implement dynamic converter selection support. I'd like this
+ to be ranked only for known device, as it's hard to be sure a device is
+ a converter suited for general purpose. Re-negotiation is also needed
+ before we can rank it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733607
+
+2014-09-05 08:29:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2: Detect bad drivers timestamps
+ Even though the UVC driver do a great deal of effort to prevent bad
+ timestamp to be sent to userspace, there still exist UVC hardware that
+ are so buggy that the timestamp endup nearly random. This code detect
+ and ignore timestamp from these drivers, making these camera usable.
+ This has been tested on both invalid and valid cameras, making sure it
+ does not trigger for valid cameras.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732910
+
+2014-08-29 17:09:30 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Workaround driver that don't support REQBUFS(0)
+ There is still around 18 drivers not yet ported to videobuf2. These driver
+ don't support freeing buffetrs through REQBUFS(0) hence for these the
+ memory type probing fails. In order to gain back our previous behaviour in
+ presence of these, we implement a workaround that assuming MMAP is
+ supported. Note that an allocator is only created for device with
+ STREAMING support in the device capabilities. In such case one of MMAP,
+ USERPTR and DMABUF is required. Though DMABUF came afterward, so is
+ not an option and in practice none of these drivers will only do USERPTR.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735660
+ Also-by: Hans de Goede <hdegoede@redhat.com>
+
+2014-09-04 15:11:40 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Merge min_buffers_for* variable into one
+ Reuse the same min_buffers variable for both capture and output, this
+ reduce the length of lines and make the code more readable.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736072
+
+2014-09-04 18:35:46 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: set min_latency for output device according to required minimum number of buffers
+ Since we can get the minimum number of buffers needed by an output
+ device to work, use it to set min_latency which will determine how many
+ buffers are queued.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736072
+
+2014-09-09 16:10:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/udpsrc.c:
+ tests: udpsrc: add check to make sure multiple memory chunks are used
+
+2014-09-09 15:55:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/udpsrc.c:
+ tests: udpsrc: wait for buffers with GCond instead of sleeping
+ Avoids half-second sleep for no reason.
+
+2014-09-09 15:31:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/udpsrc.c:
+ tests: udpsrc: split out socket setup
+
+2014-09-09 13:46:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: more efficient memory handling
+ Drop use of g_socket_get_available_bytes() which is
+ not useful on all systems (where it returns the size
+ of the entire buffer not that of the next pending
+ packet), and is yet another syscall and apparently
+ very inefficient on Windows in the UDP case.
+ Instead, when reading UDP packets, use the more featureful
+ g_socket_receive_message() call that allows to read into
+ scattered memory, and allocate one memory chunk which is
+ likely to be large enough for a packet, while also providing
+ a larger allocated memory chunk just in case the packet
+ is larger than expected. If the received data fits into the
+ first chunk, we'll just add that to the buffer we return
+ and re-use the fallback buffer for next time, otherwise we
+ add both chunks to the buffer.
+ This reduces memory waste more reliably on systems where
+ get_available_bytes() doesn't work properly.
+ In a multimedia streaming scenario, incoming UDP packets
+ are almost never fragmented and thus almost always smaller
+ than the MTU size, which is also why we don't try to do
+ something smarter with more fallback memory chunks of
+ different sizes. The fallback scenario is just for when
+ someone built a broken sender pipeline (not using a
+ payloader or somesuch)
+ https://bugzilla.gnome.org/show_bug.cgi?id=610364
+
+2014-09-09 12:15:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: rework memory allocation bits and ensure we always have two chunks of memories to read into
+ First chunk is the likely/expected buffer size, second is as
+ fallback in case the packet is larger in the end.
+ Next step: actually use these.
+
+2014-09-09 09:42:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: track max packet size and save allocator negotiated by GstBaseSrc
+
+2014-09-08 16:15:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: fix example command line
+
+2014-09-07 12:46:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix crash with certain videos
+ This is a regression from 1.2 caused by the port
+ to the pad flow combiner.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736192
+
+2014-09-04 16:21:20 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.h:
+ matroska-demux: Don't handle parse errors at the end of file as an error
+ But only if they happen after the Matroska segment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735833
+
+2014-09-04 12:14:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Include redirection target in error messages
+ Just giving the original URI can give the false impression that e.g.
+ that one failed host name resolution, while actually the redirection target
+ did.
+
+2014-09-02 11:13:44 +0400 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Fix synchronization if dynamically changing the FPS
+ https://bugzilla.gnome.org/show_bug.cgi?id=735859
+
+2014-09-02 13:52:43 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: Check if input caps are the same and create output caps from video info
+ This makes sure that also properties like the pixel-aspect-ratio are the same
+ between both streams and that the output caps contain all fields necessary for
+ complete video caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735804
+
+2014-09-02 17:22:07 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: replace with gst_buffer_copy
+ gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.
+ replacing the same with gst_buffer_copy as the functionality is same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735880
+
+2014-09-03 23:06:53 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: mark jpeg and png as parsed so avdec_mjpeg can be used too
+ https://bugzilla.gnome.org/show_bug.cgi?id=735971
+
+2014-09-03 11:46:13 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ gdkpixbufdec: free query after use
+ In gst_gdk_pixbuf_dec_setup_pool(), query is being allocated using
+ gst_query_new_allocation(), but the same is not unreferenced
+ hence calling gst_query_unref() after usage of query.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735950
+
+2014-09-03 23:46:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Silence some warnings for normal file contents
+
+2014-09-01 09:56:02 +0200 Nicolas Huet <nicolas.huet@parrot.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Fix parsing issue when the buffer does not have a complete ADTS/LOAS frame
+ https://bugzilla.gnome.org/show_bug.cgi?id=735520
+
+2014-09-02 09:09:49 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: Get input width/height from the codec instead of the input caps
+ They are reported properly by libvpx if the correct struct members are used.
+ This also fixes handling of resolution changes without input caps changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719359
+
+2013-10-22 18:49:22 +0100 Tom Greenwood <tcdgreenwood@hotmail.com>
+
+ * ext/vpx/gstvp8dec.c:
+ vp8dec: Fix for handling resolution changes when decoding VP8
+ If the resolution changes in the bitstream without the input caps changing we
+ would previously output corrupted video or crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719359
+
+2014-09-02 00:55:17 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: Fix segfault when a new caps is received
+ Remember to unref the output caps when a new caps event is received
+ as it should generate a new one based on the new caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734266
+
+2014-09-02 00:54:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/vp8dec.c:
+ tests: vp8dec: add test for caps renegotiation
+ Check that vp8dec can properly accept a new caps when upstream
+ changes it
+ https://bugzilla.gnome.org/show_bug.cgi?id=734266
+
+2014-08-05 10:34:39 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * ext/vpx/gstvp8dec.c:
+ vp8dec: Reset output and input states when changing format
+ https://bugzilla.gnome.org/show_bug.cgi?id=734266
+
+2014-09-01 16:39:23 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Don't call gst_caps_unref() on template caps when already unreferenced
+ Adding an extra condition while calling gst_caps_unref (templ)
+ and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
+ gst_caps_copy (caps) in line 177, since the functionality is same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735795
+
+2014-08-29 12:01:27 +0200 Hans de Goede <hdegoede@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: get_nearest_size: Fix "Unsupported field type" errors
+ Most V4L2 ioctls like try_fmt will adjust input fields to match what the
+ hardware can do rather then returning -EINVAL. As is docmented here:
+ http://linuxtv.org/downloads/v4l-dvb-apis/vidioc-g-fmt.html
+ EINVAL is only returned if the buffer type field is invalid or not supported.
+ So upon requesting V4L2_FIELD_NONE devices which can only do interlaced
+ mode will change the field value to e.g. V4L2_FIELD_BOTTOM as only returning
+ half the lines is the closest they can do to progressive modes.
+ In essence this means that we've failed to get a (usable) progessive mode
+ and should fall back to interlaced mode.
+ This commit adds a check for having gotten a usable field value after the first
+ try_fmt, to force fallback to interlaced mode even if the try_fmt succeeded,
+ thereby fixing get_nearest_size failing on these devices.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735660
+
+2014-08-29 10:57:20 +0200 Hans de Goede <hdegoede@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: get_nearest_size: Always reinit all struct fields on retry
+ They may have been modified by the ioctl even if it failed. This also makes
+ the S_FMT fallback path try progressive first, making it consistent with the
+ preferred TRY_FMT path.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735660
+
+2014-08-29 11:55:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Store size of data tag in a 64 bit integer locally too
+ Otherwise we will clip the DS64 value of RF64 files to 32 bits again.
+
+2014-08-29 11:53:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Use 64 bit scaling functions now that fact is a 64 bit integer
+
+2014-08-27 18:55:18 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/wavparse/gstwavparse.c:
+ * gst/wavparse/gstwavparse.h:
+ wavparse: support rf64 format
+ https://bugzilla.gnome.org/show_bug.cgi?id=735627
+
+2014-08-28 13:48:50 -0600 Jason Litzinger <jlitzinger@control4.com>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: Ensure caps before pad added.
+ This stores the stream-start, sets caps, and then adds the pad,
+ which ensures that the caps are set for the "pad-added" callback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735626
+
+2014-08-28 15:03:50 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Fallback to PTS if DTS is missing
+ Fixing a regression introduce when fixing:
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-08-28 16:13:29 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Remove impossible error condition
+ We return EOS after the first buffer, and GstPad will make sure now that we
+ won't get any other buffer afterwards until a flush happens. No need to check
+ for it ourselves.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735581
+
+2014-08-28 13:53:23 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ gdkpixbufdec: EOS and NOT_LINKED are no errors in general
+ Don't post an error message for them but let upstream handle
+ anything accordingly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735564
+
+2014-08-27 21:07:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Correctly offset timestamp
+ The previous method would break AV sync in the case audio or video
+ didn't start at the same point in running time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-08-27 20:56:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Save dts from buffer
+ We no longer set dts in muxed buffer. This would lead to encoding tags
+ with timestamp 0 instead of the timestamp of previous buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-07-28 20:58:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Ensure Timestamp starts at 0
+ FLV documentation stipulates that timestamp must start at zero.
+ In order to respect this rule, keep the first timestamp around
+ and offset the timestamp from this value. This allow for longer
+ recording time in presence of timestamp that does not start
+ at 0 already.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-06-06 23:17:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ * gst/flv/gstflvmux.c:
+ flv: Tag timestamp are DTS not PTS
+ The tags in FLV are DTS. In audio cases, and for many video format this makes
+ no difference, but for AVC with B-Frames, PTS need to be computed from
+ composition timestamp CTS, with PTS = DTS + CTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-08-07 21:58:14 -0400 Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Allow rtp caps without clock-rate
+ The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734322
+
+2014-08-18 14:05:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: avoid crashing on dash streams
+ DASH/fragmented moov might have no samples as those are carried
+ in moof fragments. Avoid crashing or failing the stream because
+ of that.
+
+2014-08-18 10:33:48 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ * tests/examples/spectrum/demo-audiotest.c:
+ * tests/examples/spectrum/demo-osssrc.c:
+ examples: use 'post-messages' property instead of deprecated 'message' property
+ https://bugzilla.gnome.org/show_bug.cgi?id=734979
+
+2014-08-18 11:45:54 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/udp/gstudpsrc.c:
+ udp: fix udpsrc documentation
+ udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
+ been removed. This patch replaces those references to socket and close-socket
+ respectively.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734987
+
+2014-08-15 10:09:56 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Make the default timescale 1/1800 second
+ The old default timescale of 1 millisecond produces irrational
+ numbers for a lot of framerate/audio-packet-duration multiples.
+ 1/1800 is a nicer number, as it tends to produce better fractions
+ and therefore slightly higher accuracy overall
+
+2014-08-15 01:17:27 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska: Use gst_video_guess_framerate() function
+ Remove local framerate guessing function in favour of
+ the new gst_video_guess_framerate() function.
+
+2014-08-15 01:12:20 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/Makefile.am:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Improve framerate calculation/guessing
+ Change the way the output framerate is calculated
+ to ignore the first sample (which is sometimes truncated
+ in my testing) and use the new gst_video_guess_framerate()
+ function to recognise common standard framerates better.
+ Remove the code that was sorting the first 20 sample
+ durations and then ignoring the result.
+
+2014-08-14 16:36:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Use the best width/height/etc if downstream can handle that
+ Before it was always using whatever downstream preferred, while
+ the code and documentation claimed something different.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727180
+
+2014-08-14 11:29:00 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Avoid double free of VideoConvert
+ https://bugzilla.gnome.org/show_bug.cgi?id=734764
+
+2014-08-13 11:58:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix indentation
+
+2014-08-13 11:54:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: un-break duration querying
+ Commit 2b9493b5 broke this in two ways: a) we should only
+ pass duration queries in TIME format upstream (or at least
+ not those in DEFAULT or BYTE format), and b) we mustn't
+ overwrite the default value of 'res' from TRUE to FALSE
+ and not set it again later. This led to bogus durations
+ being reported for FLV playback from file, because TIME
+ queries would fail (as 'res' had been set to FALSE) and
+ parsers then do a BYTE query as fallback and try to
+ guesstimate something in return, which of course goes
+ horribly wrong since the BYTE size returned is for the
+ muxed file.
+
+2014-08-13 13:23:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videofilter/gstvideobalance.c:
+ videobalance: Allow any raw caps in passthrough mode, not just the ones we handle
+
+2014-08-13 13:04:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videofilter/gstvideobalance.c:
+ videobalance: Allow ANY capsfeatures, but only in passthrough mode
+ When changing the properties to not be in passthrough mode anymore,
+ we will only accept caps we can process ourselves, potentially causing
+ a not-negotiated error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720345
+
+2014-08-12 11:34:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ docs: update for git
+
+2014-08-12 11:33:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: build ximagesrc again when checks succeed
+ Third time lucky, hopefully.
+
+2014-08-11 09:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: fix x11 checks to be non-fatal again
+ Must pass an action-if-not-found argument to
+ PKG_CHECK_MODULES or it will error out when
+ it can't find the module requested. Also fix
+ AC_CHECK_LIB usage, extra libs argument was
+ in the wrong place.
+
+2014-08-07 17:12:38 +0300 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: forward DISCONT from upstream to the output streams
+ This makes sense in DASH reverse playback, where the upstream dashdemux
+ will download DASH segments in reverse order, but push their buffers
+ forward to qtdemux and mark each segment start as DISCONT. This needs
+ to be forwarded downstream to the parser/decoder, otherwise it won't work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734443
+
+2014-08-10 18:55:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: use pkg-config to detect x11 and simplify checks
+ AC_PATH_XTRA macro unnecessarily pulls in libSM and libICE.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731047
+
+2014-08-10 12:30:07 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: adjust test data to avoid NAL chopping
+ ... and correspondingly unexpected buffer sizes.
+
+2014-08-09 14:22:42 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * ext/speex/gstspeexenc.c:
+ speexenc: Improve annotation of internal function
+ https://bugzilla.gnome.org/show_bug.cgi?id=734542
+
+2014-08-08 12:54:30 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/shapewipe/gstshapewipe.c:
+ * tests/examples/shapewipe/shapewipe-example.c:
+ shapewipe: Unref caps and element after usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=734478
+
+2014-08-09 20:47:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: improve debug logging of fourccs
+ If we can't show ASCII, at least show them
+ in big endian order.
+
+2014-08-09 20:46:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for 'wma ' mapping as found in some ismv files
+ e.g. To_The_Limit_720_2962.ismv
+
+2014-08-09 18:31:20 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for 'vc-1' mapping as found in some ismv files
+ e.g. To_The_Limit_720_2962.ismv
+
+2014-08-07 16:34:36 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ rtph263ppay: Unref pad template caps after use
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435
+
+2014-08-08 12:36:01 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Unref allowed caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734474
+
+2014-08-08 12:40:49 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Unref pad template caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734475
+
+2014-08-08 12:44:09 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/debugutils/gstnavseek.c:
+ navseek: Unref peer pad after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734476
+
+2014-08-08 12:29:52 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Unref pad template caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473
+
+2014-08-05 11:47:39 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: append packetization mode parameter to SDP
+ Append packetization-mode parameter to SDP description.
+ Packetization mode signals the properties of an RTP payload type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733556
+
+2014-08-08 03:58:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux.c:
+ isomp4/qtmux: Write correct file duration when gaps exist.
+ When writing out a trak with an edit list, make sure the
+ overall file duration is also updated to reflect the
+ lengthening of the stream.
+ Add some more debug to qtdemux to warn about streams that
+ are longer than the file and get truncated.
+
+2014-08-04 15:39:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Push the correct segment in TCP mode when seeking
+
+2014-08-03 12:33:32 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: unbreak au aligned byte-stream payloading
+
+2014-07-22 13:24:09 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: append profile-level-id to SDP
+ Append profile-level-id to SDP if available.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733539
+
+2014-07-31 18:47:49 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-07-31 09:53:53 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: Fix warning about missing return value
+
+2014-07-24 15:28:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/ximageutil.c:
+ * sys/ximage/ximageutil.h:
+ ximagesrc: Add missing return value to Buffer dispose function
+ Depending ont he build, the method could return FALSE, hence never
+ free the buffers, or already TRUE and lead to a crash:
+ Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=733695
+
+2014-07-28 16:49:16 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/interleave/interleave.c:
+ * tests/check/elements/interleave.c:
+ interleave: set output caps layout to interleaved
+ Set output caps layout independently from input caps layout which can
+ be either non-interleaved or interleaved.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733866
+
+2014-07-26 12:06:39 -0300 Thiago Santos <ts.santos@osg.sisa.samsung.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: clear gcond
+
+2014-07-25 14:30:33 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ Revert "v4l2bufferpool: Workaround elements not requesting any buffers"
+ This was a tempory workaround, we should fix the encoders that do not
+ negotatiate the amount of buffers they need.
+ This reverts commit d03bcba3db15d06dbdea6b776a6f28ed2f03272a.
+
+2014-07-08 14:31:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't share own pool if min exceed V4L2 capacity
+ If the minimum required buffer exceed V4L2 capacity, don't share down
+ pool. This allow support very high latency, like with x264enc default
+ encoding settings.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732288
+
+2014-07-25 17:42:20 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: query minimum required buffers for output
+ Some v4l2 devices could require a minimum buffers different from default
+ values. Rather than blindly propose a pool with min-buffers set to the
+ default value, it ask the device using control ioctl.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733750
+
+2014-07-23 18:40:10 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: use directly 'obj' instead of 'v4l2sink->v4l2object'
+ https://bugzilla.gnome.org/show_bug.cgi?id=733616
+
+2014-07-23 18:39:50 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2: set debug messages according to device type and IO mode
+ https://bugzilla.gnome.org/show_bug.cgi?id=733616
+
+2014-05-24 19:02:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Remove is_active checks
+ These checks are no longer required with recent change to the bufferpool. This
+ should allow changing the configuartion, hence the way forward renegotiation
+ support.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728268
+
+2014-07-21 18:11:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_lang.c:
+ qtdemux: fix language code parsing for 3-letter codes starting with 'a'
+ And handle special value for 'unspecified' explicitly.
+ https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap4/qtff4.html
+
+2014-07-08 02:18:27 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * ext/jpeg/gstjpegenc.c:
+ jpegenc: Add support for encoding from NV21 and NV12
+ https://bugzilla.gnome.org/show_bug.cgi?id=732870
+
+2014-07-19 18:04:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.4.0 ===
+
+2014-07-19 17:20:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.4.0
+
+2014-07-19 16:35:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2014-07-19 12:32:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: Update translations
+
+2014-07-19 11:30:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videobox/gstvideobox.c:
+ videobox: Don't overwrite the first component with the alpha value for BGRx
+ Instead leave the x component unset when filling the borders.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733380
+
+2014-07-16 17:18:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Properly report in the CAPS query that we can convert ADTS<->RAW
+ https://bugzilla.gnome.org/show_bug.cgi?id=733190
+
+2014-07-13 16:05:56 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/replaygain/gstrgvolume.c:
+ rgvolume: Avoid taking unnecessary refs
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
+
+2014-07-13 16:04:23 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtpmanager/gstrtpdtmfmux.c:
+ rtpdtmfmux: Avoid taking an unnecessary ref
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
+
+2014-07-15 16:59:06 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES
+ https://bugzilla.gnome.org/show_bug.cgi?id=733208
+
+2014-07-11 13:35:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Fix copy threshold implementation
+ We cannot allocate new buffer in acquire, otherwise the base class
+ is not aware and get confused. Instead, copy in _process(). This leads
+ to crash on finalize.
+ Fixes regression, see https://bugzilla.gnome.org/show_bug.cgi?id=732912
+
=== release 1.3.91 ===
-2014-07-11 Sebastian Dröge <slomo@coaxion.net>
+2014-07-11 11:38:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.3.91
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.3.91
+
+2014-07-11 10:58:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
2014-07-10 18:11:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* ext/pulse/Makefile.am:
* ext/pulse/plugin.c:
- * ext/pulse/pulsedevicemonitor.c:
- * ext/pulse/pulsedevicemonitor.h:
* ext/pulse/pulsedeviceprovider.c:
* ext/pulse/pulsedeviceprovider.h:
* sys/v4l2/Makefile.am:
* sys/v4l2/gstv4l2.c:
- * sys/v4l2/gstv4l2devicemonitor.c:
- * sys/v4l2/gstv4l2devicemonitor.h:
* sys/v4l2/gstv4l2deviceprovider.c:
* sys/v4l2/gstv4l2deviceprovider.h:
Rename GstDeviceMonitor to GstDeviceProvider
2013-09-10 16:44:53 -0600 Seán de Búrca <leftmostcat@gmail.com>
- * gst/videomixer/blendorc-dist.c:
- * gst/videomixer/blendorc-dist.h:
* gst/videomixer/videomixerorc-dist.c:
* gst/videomixer/videomixerorc-dist.h:
videomixer: fix build if orc is not installed
* sys/osxaudio/gstosxaudiosrc.h:
* sys/osxaudio/gstosxcoreaudiocommon.c:
* sys/osxaudio/gstosxcoreaudiocommon.h:
- * sys/osxaudio/gstosxringbuffer.c:
- * sys/osxaudio/gstosxringbuffer.h:
osxaudio: port to v.1.0
2013-04-16 19:29:48 -0400 Olivier Crête <olivier.crete@collabora.com>
2012-09-16 15:33:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vp8/GstVP8Enc.prs:
- * ext/vp8/Makefile.am:
- * ext/vp8/gstvp8dec.c:
- * ext/vp8/gstvp8dec.h:
- * ext/vp8/gstvp8enc.c:
- * ext/vp8/gstvp8enc.h:
- * ext/vp8/gstvp8utils.c:
- * ext/vp8/gstvp8utils.h:
- * ext/vp8/plugin.c:
* ext/vpx/GstVP8Enc.prs:
* ext/vpx/Makefile.am:
* ext/vpx/gstvp8dec.c:
* configure.ac:
* ext/gdk_pixbuf/Makefile.am:
- * ext/gdk_pixbuf/gstgdkpixbuf.c:
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
* ext/gdk_pixbuf/gstgdkpixbufdec.c:
* ext/gdk_pixbuf/gstgdkpixbufdec.h:
* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
- * docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpmanager.c:
* ext/soup/gstsoup.c:
* ext/soup/gstsouphttpclientsink.c:
* ext/soup/gstsouphttpclientsink.h:
- * ext/soup/gstsouphttpsink.c:
- * ext/soup/gstsouphttpsink.h:
soup: rename souphttpsink to souphttpclientsink
To avoid confusion, and because we might want a server
sink at some point too.
* sys/v4l2/gstv4l2sink.c:
* sys/v4l2/gstv4l2videooverlay.c:
* sys/v4l2/gstv4l2videooverlay.h:
- * sys/v4l2/gstv4l2xoverlay.c:
- * sys/v4l2/gstv4l2xoverlay.h:
v4l2: update for GstXOverlay => GstVideoOverlay rename
2011-08-07 12:23:26 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/inspect/plugin-isomp4.xml:
- * docs/plugins/inspect/plugin-quicktime.xml:
* gst-plugins-good.spec.in:
* gst/isomp4/LEGAL:
* gst/isomp4/Makefile.am:
* gst/isomp4/qtdemux_types.c:
* gst/isomp4/qtdemux_types.h:
* gst/isomp4/qtpalette.h:
- * gst/quicktime/LEGAL:
- * gst/quicktime/Makefile.am:
- * gst/quicktime/atoms.c:
- * gst/quicktime/atoms.h:
- * gst/quicktime/atomsrecovery.c:
- * gst/quicktime/atomsrecovery.h:
- * gst/quicktime/descriptors.c:
- * gst/quicktime/descriptors.h:
- * gst/quicktime/fourcc.h:
- * gst/quicktime/ftypcc.h:
- * gst/quicktime/gstqtmoovrecover.c:
- * gst/quicktime/gstqtmoovrecover.h:
- * gst/quicktime/gstqtmux-doc.c:
- * gst/quicktime/gstqtmux-doc.h:
- * gst/quicktime/gstqtmux.c:
- * gst/quicktime/gstqtmux.h:
- * gst/quicktime/gstqtmuxmap.c:
- * gst/quicktime/gstqtmuxmap.h:
- * gst/quicktime/gstrtpxqtdepay.c:
- * gst/quicktime/gstrtpxqtdepay.h:
- * gst/quicktime/properties.c:
- * gst/quicktime/properties.h:
- * gst/quicktime/qtatomparser.h:
- * gst/quicktime/qtdemux.c:
- * gst/quicktime/qtdemux.h:
- * gst/quicktime/qtdemux.vcproj:
- * gst/quicktime/qtdemux_dump.c:
- * gst/quicktime/qtdemux_dump.h:
- * gst/quicktime/qtdemux_fourcc.h:
- * gst/quicktime/qtdemux_lang.c:
- * gst/quicktime/qtdemux_lang.h:
- * gst/quicktime/qtdemux_types.c:
- * gst/quicktime/qtdemux_types.h:
- * gst/quicktime/qtpalette.h:
- * gst/quicktime/quicktime.c:
* po/POTFILES.in:
quicktime: rename plugin to isomp4
https://bugzilla.gnome.org/show_bug.cgi?id=648004
* gst/dtmf/Makefile.am:
* gst/dtmf/gstdtmfcommon.h:
* gst/dtmf/gstdtmfsrc.c:
- * gst/dtmf/gstrtpdtmfcommon.h:
* gst/dtmf/gstrtpdtmfdepay.c:
* gst/dtmf/gstrtpdtmfdepay.h:
* gst/dtmf/gstrtpdtmfsrc.c:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/inspect/plugin-quicktime.xml:
- * gst/qtdemux/LEGAL:
- * gst/qtdemux/Makefile.am:
- * gst/qtdemux/gstrtpxqtdepay.c:
- * gst/qtdemux/gstrtpxqtdepay.h:
- * gst/qtdemux/qtatomparser.h:
- * gst/qtdemux/qtdemux.c:
- * gst/qtdemux/qtdemux.h:
- * gst/qtdemux/qtdemux.vcproj:
- * gst/qtdemux/qtdemux_dump.c:
- * gst/qtdemux/qtdemux_dump.h:
- * gst/qtdemux/qtdemux_fourcc.h:
- * gst/qtdemux/qtdemux_lang.c:
- * gst/qtdemux/qtdemux_lang.h:
- * gst/qtdemux/qtdemux_types.c:
- * gst/qtdemux/qtdemux_types.h:
- * gst/qtdemux/qtpalette.h:
- * gst/qtdemux/quicktime.c:
* gst/quicktime/LEGAL:
* gst/quicktime/Makefile.am:
* gst/quicktime/gstrtpxqtdepay.c:
2009-11-28 18:13:31 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/aacparse/Makefile.am:
- * gst/aacparse/gstaacparse.c:
- * gst/aacparse/gstaacparse.h:
- * gst/aacparse/gstbaseparse.c:
- * gst/aacparse/gstbaseparse.h:
- * gst/aacparse/plugin.c:
* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
2009-05-15 17:11:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
- * tests/check/audiotestsrc.flac:
* tests/check/elements/id3demux.c:
* tests/check/elements/souphttpsrc.c:
* tests/check/pipelines/flacdec.c:
- * tests/check/test-cert.pem:
- * tests/check/test-key.pem:
* tests/files/Makefile.am:
* tests/files/audiotestsrc.flac:
* tests/files/test-cert.pem:
* gst/deinterlace/tvtime/weavebff.c:
* gst/deinterlace/tvtime/weavetff.c:
* gst/deinterlace/tvtime/x86-64_macros.inc:
- * gst/deinterlace2/Makefile.am:
- * gst/deinterlace2/gstdeinterlace2.c:
- * gst/deinterlace2/gstdeinterlace2.h:
- * gst/deinterlace2/tvtime/greedy.c:
- * gst/deinterlace2/tvtime/greedyh.asm:
- * gst/deinterlace2/tvtime/greedyh.c:
- * gst/deinterlace2/tvtime/greedyhmacros.h:
- * gst/deinterlace2/tvtime/linear.c:
- * gst/deinterlace2/tvtime/linearblend.c:
- * gst/deinterlace2/tvtime/mmx.h:
- * gst/deinterlace2/tvtime/plugins.h:
- * gst/deinterlace2/tvtime/scalerbob.c:
- * gst/deinterlace2/tvtime/sse.h:
- * gst/deinterlace2/tvtime/tomsmocomp.c:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoop0A.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA8.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA2.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA6.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH2.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVA.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVAH.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
- * gst/deinterlace2/tvtime/vfir.c:
- * gst/deinterlace2/tvtime/weave.c:
- * gst/deinterlace2/tvtime/weavebff.c:
- * gst/deinterlace2/tvtime/weavetff.c:
- * gst/deinterlace2/tvtime/x86-64_macros.inc:
Moved 'deinterlace2' from -bad to -good
And rename it to deinterlace.
* configure.ac:
* docs/plugins/Makefile.am:
- * gst/debug/Makefile.am:
- * gst/debug/breakmydata.c:
- * gst/debug/debug.vcproj:
- * gst/debug/efence.c:
- * gst/debug/efence.h:
- * gst/debug/efence.vcproj:
- * gst/debug/gstdebug.c:
- * gst/debug/gstnavigationtest.c:
- * gst/debug/gstnavigationtest.h:
- * gst/debug/gstnavseek.c:
- * gst/debug/gstnavseek.h:
- * gst/debug/gstpushfilesrc.c:
- * gst/debug/gstpushfilesrc.h:
- * gst/debug/gsttaginject.c:
- * gst/debug/gsttaginject.h:
- * gst/debug/navigationtest.vcproj:
- * gst/debug/negotiation.c:
- * gst/debug/progressreport.c:
- * gst/debug/progressreport.h:
- * gst/debug/rndbuffersize.c:
- * gst/debug/testplugin.c:
- * gst/debug/tests.c:
- * gst/debug/tests.h:
* gst/debugutils/Makefile.am:
* gst/debugutils/breakmydata.c:
* gst/debugutils/debug.vcproj:
2009-02-24 14:55:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gconf/Makefile.am:
- * ext/gconf/gconf.c:
- * ext/gconf/gconf.h:
* ext/gconf/gstgconf.c:
* ext/gconf/gstgconf.h:
* ext/gconf/gstgconfelements.h:
* gst/audiofx/audioecho.c:
* gst/audiofx/audioecho.h:
* gst/audiofx/audiofx.c:
- * gst/audiofx/audioreverb.c:
- * gst/audiofx/audioreverb.h:
* tests/check/Makefile.am:
* tests/check/elements/audioecho.c:
- * tests/check/elements/audioreverb.c:
Rename audioreverb to audioecho. Fixes bug #568395.
The element can add an echo and a simple reverb effect to
an audio stream but for a real reverb filter it would need
Original commit message from CVS:
add some files
+2001-12-17 18:37:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ building up speed
+ Original commit message from CVS:
+ building up speed
+
projects / platform / upstream / gst-plugins-good.git / blobdiff