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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
Sebastian Dröge's avatar
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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
Sebastian Dröge's avatar
Sebastian Dröge committed
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