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=== release 1.9.90 ===

2016-09-30  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.9.90

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

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=== release 1.9.2 ===

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2016-09-01 12:27:15 +0300  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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
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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

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=== release 1.9.1 ===

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2016-07-06 13:06:44 +0300  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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
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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

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=== release 1.8.0 ===

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2016-03-24 12:27:33 +0200  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* 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
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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

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=== release 1.7.91 ===

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2016-03-15 12:04:39 +0200  Sebastian Dröge <sebastian@centricular.com>