ChangeLog

=== release 1.12.1 ===

2017-06-20  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.12.1

2017-06-20 11:08:32 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/bg.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/fr.po:
	* po/hr.po:
	* po/hu.po:
	* po/nb.po:
	* po/pl.po:
	* po/ru.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2017-06-13 17:40:19 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Drop allocation queries
	  They can cause us to deadlock, while we're waiting for a new frame and
	  upstream is waiting for the allocation query to be answered before
	  sending a frame
	  https://bugzilla.gnome.org/show_bug.cgi?id=783753

2017-06-15 10:40:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Use a mutex for protecting against concurrent send/receives
	  We currently send data to the RTSP connection from multiple threads:
	  whenever a command is to be handled and whenever RTCP is generated. This
	  can cause data corruption or worse if both happen at the same time.
	  As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive()
	  calls with a mutex. While this means that we hold a mutex during the IO
	  operation, this is not actually a problem as the IO operation can be
	  interrupted (gst_rtsp_connection_flush()) at any time and is blocking by
	  itself anyway.

2017-06-15 11:50:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: Un-merge the last two stsc entries after serializing
	  The last entry will most likely get new samples added to it in "robust"
	  muxing mode, changing the samples_per_chunk and thus making it wrong to
	  keep the last two entries merged. It will run into an assertion later
	  when adding a new sample to the chunk.
	  Thanks to gdiener@cardinalpeak.com for the analysis of the bug and
	  proposal for a solution.

2017-06-14 00:09:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Actually clip to upstream size instead of size of the data chunk
	  There might be other chunks after the data chunk, so clipping the chunk
	  size with the data size can lead to a negative number and all following
	  calculations go wrong and cause crashes or worse.
	  This was introduced in 3ac119bbe2c360e28c087cf3852ea769d611b120.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783760

2017-05-30 22:23:10 +0200  Juan Navarro <juan.navarro@gmx.es>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: print value of unknown RTCP Payload Type
	  This adds printing the actual value of any unknown RTCP PT
	  to the already existing WARNING log message.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783248

2017-06-02 11:30:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix caps leak

2017-05-24 11:33:05 +0530  vijay <vijay.palaniswamy@in.bosch.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse : Fix, Caps were not set while reusing aacparse
	  While reusing aacparse caps were not set.This fix enables aacparse to reuse in same pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=783027

2017-05-16 12:56:15 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Do not check timecode data for mp4 container
	  Timecode trak is only supported for mov right now, not for mp4. That
	  code would otherwise create an invalid trak if the muxed video contained
	  timecode metadata.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782684

2017-05-10 15:58:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Lateness is in QT timescale, diff in GstClockTime
	  Print the right one in debug output to get meaningful numbers.

2017-05-09 11:41:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Set fb->priv to NULL after freeing just in case
	  https://bugzilla.gnome.org/show_bug.cgi?id=782359

2017-05-08 15:22:00 +0000  Dustin Spicuzza <dustin@virtualroadside.com>

	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: Use GstClock API instead of Sleep() for waiting
	  It's more accurate and allows cancellation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=773681

2017-05-08 15:05:45 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvp9dec.c:
	  vpx: fix build against older libvpx versions
	  Such as 1.3.0 as on raspbian.

2017-05-03 23:23:10 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix corner case causing large CPU usage
	  We were unnecessarily looping/goto-ing repeatedly when we had exactly
	  the amount of data as the free space, and also when the free space was
	  too small. This, as it turns out, is a very common scenario with
	  Directsound on Windows.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=773681
	  We have to do polling here because the event notification API that
	  Directsound exposes cannot be used with live playback since all events
	  must be registered in advance with the capture buffer, you cannot
	  add/remove them once playback has begun. Directsoundsrc had the same
	  problem.
	  See also: https://bugzilla.gnome.org/show_bug.cgi?id=781249

=== release 1.12.0 ===

2017-05-04 15:38:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.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:
	* meson.build:
	  Release 1.12.0

2017-05-04 15:07:27 +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/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-05-04 13:47:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2017-05-02 10:32:30 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix crash on mss stream caused by invalid stsd entry access
	  Since mss has no moov, default stsd entry should be created with media-caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=782042

=== release 1.11.91 ===

2017-04-27 17:29:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.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:
	* meson.build:
	  Release 1.11.91

2017-04-27 15:58:47 +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/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2017-04-27 15:28:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/LINGUAS:
	* po/el.po:
	* po/fur.po:
	  po: Update translations

2017-04-27 12:56:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't crash in debug output if stream==NULL
	  That case is correctly handled below but not in the debug output.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781270

2017-04-25 17:11:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't perform seeks with inconsistent seek values
	  If gst_segment_do_seek() fails, we shouldn't try seeking on that
	  resulting segment but just error out. Crashes further down the line
	  otherwise.

2017-04-24 20:27:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From 60aeef6 to 48a5d85

2017-04-24 17:31:04 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: add test for rtph264depay avc/byte-stream output
	  Make sure avc output doesn't contain SPS/PPS inline, but
	  byte-stream output does.

2017-04-24 17:29:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: don't insert SPS/PPS inline for AVC output
	  SPS/PPS are in the caps in this case and shouldn't be in
	  the stream data.

2017-04-21 19:09:14 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Chain up to the parent class' provide_clock() implementation
	  If no clock was provided directly by rtspsrc. This behaviour was removed
	  by f8013487c91a6ffc552a4b25aa1a70f0bd5377f8 and results in rtspsrc not
	  providing the system clock via the rtpjitterbuffer.
	  As a result, if another element like an audio sink, provides a clock,
	  the pipeline would select that (when going to PAUSED/PLAYING again later).
	  Audio clocks usually don't progress in PAUSED, and thus our live source
	  won't be able to use the clock to produce data, making the sink never
	  preroll and everything is stuck.

2017-04-20 11:22:15 +0200  Jürgen Sachs <juergen.sachs@metz-ce.de>

	* gst/isomp4/qtdemux.c:
	  qtdemux: reset sample_description_id to default
	  Fixes stream where sample_description_id is specified in the tfhd
	  https://bugzilla.gnome.org/show_bug.cgi?id=778337

2017-04-20 13:16:24 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't use an explicit name for requesting audio pads
	  ... unless the muxer uses the same audio pad template name as
	  splitmuxsink. We can't request a pad called "audio_0" on a muxer that
	  wants pads to be "sink_%d".

2017-02-23 09:31:36 +0900  ChangBok Chae <changbok.chea@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: remove duplicated segment initialization
	  It's also done in gst_flv_demux_cleanup().
	  https://bugzilla.gnome.org/show_bug.cgi?id=779106

2017-04-20 20:17:35 +1000  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Correctly catch FLUSH events in probes
	  https://bugzilla.gnome.org/show_bug.cgi?id=767498

2017-04-19 12:28:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  Revert "rtpbin: pipeline gets an EOS when any rtpsources byes"
	  This reverts commit eeea2a7fe88a17b15318d5b6ae6e190b2f777030.
	  It breaks EOS in some sender pipelines, see
	  https://bugzilla.gnome.org/show_bug.cgi?id=773218#c20

2017-04-14 17:01:49 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Reset adapter in more discontinuity cases
	  In push mode we process as much as possible in the adapter. When we receive
	  a DISCONT buffer which we can't match to an actual sample (based on the existing
	  sample table) and there is still data remaining in the incoming adapter,there is
	  one of two cases happening:
	  1) We are doing reverse playback, in which case we should flush out all pending
	  data
	  2) We have leftover data from the previous incoming buffer... which we can't do
	  anything about.
	  For the second case, make sure we flush out the remaining data so that we can start
	  parsing again from scratch.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781319

2017-04-14 10:56:41 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Use GST_ELEMENT_ERROR_WITH_DETAILS
	  Allows the application to know the exact status code that was returned
	  by the server in a programmatic fashion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781304

2017-04-16 18:47:56 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix leak on QtDemuxStreamStsdEntry
	  Fix unit test failure
	  https://bugzilla.gnome.org/show_bug.cgi?id=781362

2017-04-14 13:38:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix timescale of timecode tracks
	  They should have ideally the same timescale of the video track, which we
	  can't guarantee here as in theory timecode configuration and video
	  framerate could be different. However we should set a correct timescale
	  based on the framerate given in the timecode configuration, and not just
	  use the framerate numerator.

2017-04-13 13:25:06 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Properly reset demuxer when all streams are EOS
	  Make sure offset and neededbytes are properly resetted when all
	  streams are EOS in push-mode.
	  Avoids cases when some data might still be pushed by upstream (because
	  it didn't yet see the resulting GST_FLOW_EOS yet) and qtdemux gets
	  completely lost.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781266

2017-04-13 08:00:30 +0200  Edward Hervey <edward@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Make more usage of error macro
	  And make sure we actually use the provided soup_msg argument in the macro

2017-04-12 18:46:53 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/meson.build:
	  meson: Print message when disabling taglib on MSVC

2017-04-12 13:26:59 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't forget to update pad->last_buf
	  buf is the current pad->last_buf value. If ever it gets copied/unreffed,
	  we need to make sure to write back the new  pointer to the last_buf
	  variable.
	  Fixes using wrong pointer values in the case of decrasing DTS value

2017-04-12 11:33:05 +0200  Edward Hervey <edward@centricular.com>

	* tests/check/elements/.gitignore:
	  tests: Add vp9enc to gitignore

2017-04-11 13:41:48 +0200  Jürgen Sachs <juergen.sachs@metz-ce.de>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix: sample description index override in tfhd not evaluated
	  https://bugzilla.gnome.org/show_bug.cgi?id=778337

2017-04-12 11:03:24 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add out-of-bound check
	  Make sure we don't read invalid memory

2016-04-27 12:17:37 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: move parsing of tkhd out of stsd entry loop
	  It needs only to be read once.

2016-04-07 12:23:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check for a different stsd entry before pushing a sample
	  Before pushing a sample, check if there was a change in the current
	  stsd entry. This patch also assumes that the first stsd entry is
	  used as default for the first sample. It might cause an uneeded
	  caps renegotiation when this isn't the case.

2016-04-06 12:55:18 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: parse all stsd entries
	  stsd can have multiple format entries, parse them all.
	  This is required to play DVB DASH profile that uses multiple entries
	  to identify the different available bitrates/options on dash streams
	  The stream format-specific data is not stored into QtDemuxStreamStsdEntry

2016-04-05 14:34:00 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: rework stsd sample entries access
	  Instead of using the stsd as a base pointer, use the actual stsd
	  entry as the stsd can have multiple entries. This is rarely used
	  for file playback but is a possible profile with in DVB DASH specs.
	  This still doesn't support stsd with multiple entries but makes it
	  easier to do so.

2016-04-05 18:00:10 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: get stsd child by index instead of type
	  There might be multiple children with the same type

2017-04-07 16:33:18 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/rtprtx.c:
	  tests/check/rtprtx: add checks for rtprtxqueue's max-size-{time,packets} properties
	  https://bugzilla.gnome.org/show_bug.cgi?id=780867

2017-04-04 17:33:31 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtprtxqueue: implement handling of the max-size-time property
	  https://bugzilla.gnome.org/show_bug.cgi?id=780867

2017-04-10 23:49:06 +0100  Tim-Philipp Müller <tim@centricular.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From 39ac2f5 to 60aeef6

2017-04-10 08:56:00 +0000  Todor Tomov <todor.tomov@linaro.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2object: Copy timestamp when importing buffers
	  This is needed for V4L2_OUTPUT interface, and is harmless of
	  V4L2_CAPTURE interfaces. This will fix timestamp in cases like:
	  v4l2src io-mode=dmabuf ! v4l2videoNenc output-io-mode=dmabuf-import !  ...
	  Same apply for userptr.
	  https://bugzilla.gnome.org/show_bug.cgi?id=781119

2017-04-10 15:55:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix last_dts tracking for raw audio and similar formats
	  Accumulate the durations directly and don't scale yet another time by
	  the number of samples.

2017-04-07 10:48:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* tests/check/elements/splitmux.c:
	  tests: fix leak in splitmux test
	  https://bugzilla.gnome.org/show_bug.cgi?id=781025

2017-04-07 15:29:43 +0800  Lyon Wang <lyon.wang@nxp.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: Scale GAP event timestamp and duration like for buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=781008

2017-02-17 10:01:08 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videodec.h:
	  v4l2dec: Fix race when going from PAUSED to READY
	  Running `gst-validate-launcher -t validate.file.playback.change_state_intensive.vorbis_vp8_1_webm`
	  on odroid XU4 (s5p-mfc v4l2 driver) often leads to:
	  ERROR:../subprojects/gst-plugins-good/sys/v4l2/gstv4l2videodec.c:215:gst_v4l2_video_dec_stop: assertion failed: (g_atomic_int_get (&self->processing) == FALSE)
	  This happens when the following race happens:
	  - T0: Main thread
	  - T1: Upstream streaming thread
	  - T2. v4l2dec processing thread)
	  [The decoder is in PAUSED state]
	  T0. The validate scenario runs `Executing (36/40) set-state: state=null repeat=40`
	  T1- The decoder handles a frame
	  T2- A decoded frame is push downstream
	  T2- Downstream returns FLUSHING as it is already flushing changing state
	  T2- The decoder stops its processing thread and sets `->processing = FALSE`
	  T1- The decoder handles another frame
	  T1- `->process` is FALSE so the decoder restarts its streaming thread
	  T0- In v4l2dec-> stop the processing thread is stopped
	  NOTE: At this point the processing thread loop never started.
	  T0- assertion failed: (g_atomic_int_get (&self->processing) == FALSE)
	  Here I am removing the whole ->processing logic to base it all on the
	  GstTask state to avoid duplicating the knowledge.
	  https://bugzilla.gnome.org/show_bug.cgi?id=778830

=== release 1.11.90 ===

2017-04-07 16:31:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2.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:
	* meson.build:
	  Release 1.11.90

2017-04-07 15:18:11 +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

2017-04-07 15:06:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	  po: Update translations

2017-04-06 12:01:00 +0200  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: streamline and improve AudioSpecificConfig parsing
	  AudioSpecifigConfig is used in a variety of AAC streams but was
	  being parsed differently. Instead, make everyone use the same parsing.
	  * Remove unused 'bits' field (it was always set to 0 if present)
	  * Add proper GAConfig parsing (to know the  number of samples per frame
	  if present).
	  Fixes wrong rate/channels configuration in streams coming from qtdemux
	  https://bugzilla.gnome.org/show_bug.cgi?id=780966

2017-04-05 09:46:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Fix 32bit only printf format
	  The previous patch was using %llu for 64bits printf, which is 32bit
	  specific. We also trace the latency in time human readable form now.

2016-03-16 16:22:48 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: set streamparm for outputs that support it
	  Without a specified framerate from the sink, the decoder frame interval
	  should be set using the framerate of the encoded video stream.
	  Therefore, the v4l2object should be able to change the framerate on the
	  output if the V4L2 device accepts it.
	  This is also necessary for mem2mem encoders so that their bitrate
	  calculation code may work correctly and they may report the correct
	  frame duration on the capture queue.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779466

2016-03-16 16:24:55 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: only set latency if the frame duration is valid
	  If the duration of the v4l2object is GST_CLOCK_TIME_NONE, because the
	  sink did not specify a framerate in the caps and the driver accepts the
	  framerate, the decoder element uses GST_CLOCK_TIME_NONE to calculate and
	  set the element latency.
	  While this is a bug of the capture driver, the decoder element should
	  not use the invalid duration to calculate a latency, but print a warning
	  instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=779466

2016-11-23 12:17:55 -0500  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2sink.c:
	  v4l2sink: Block in preroll_wait on unlock
	  The correct behaviour of anything stuck in the ->render() function
	  between ->unlock() and ->unlock_stop() is to call
	  gst_base_sink_wait_preroll() and only return an error if this returns an
	  error, otherwise, it must continue where it left off!
	  https://bugzilla.gnome.org/show_bug.cgi?id=774945

2017-04-05 15:55:20 +1000  Jan Schmidt <jan@centricular.com>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Add warnings for unsupported frame formats
	  At least output an element warning on the bus when we
	  encounter a frame format GStreamer doesn't currently support.

2017-04-04 17:55:13 +0200  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Handle Parametric Stereo with HE-AAC(v2)
	  According to ISO/IEC:14496-2:2009 , in the case of HE-AACv2 (audioObjecType
	  29) parametric stereo is used (a single mono track is used and then
	  transformations are applied to it to provide a stereo output).
	  We therefore report two channels in the case where there is one reported
	  in the audioChannelConfiguration.
	  Fixes the various issues where a demuxer would report two channels, but
	  then the parser would say there's only one channel, and then the decoder
	  would output two channels.

2017-04-04 15:22:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Simplify buffer refcounting in add_buffer() and remove unneeded NULL checks

2017-04-04 15:08:33 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Select the best pad based on the cached last_buf if any
	  last_buf is the one we're going to write next, not buf. As such we
	  should check timestamps against that one if there is one to select the
	  earliest pad.
	  Also remember the currently selected pad in the very beginning when
	  storing the first last_buf.
	  This both solves some edge cases where not the correct next pad was
	  selected corresponding to the target interleave.

2017-04-04 15:07:40 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Error out immediately if a timecode is to be written but downstream return not-OK

2017-04-03 11:34:49 +0200  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Update variables before early exit
	  This is an update of d78d5896272d78df41e696fac929e7dfb3bb3dfa
	  We still exit as early as possible in case of non-ok/non-unlinked combined
	  flow, but we first make sure that we update the internal position variables.
	  This ensures that if upstreams "ignores" the flow return (and carries on pushing),
	  we don't end up processing data with completely bogus variables/positions.

2017-03-24 00:11:13 +1300  Douglas Bagnall <douglas@halo.gen.nz>

	* gst/interleave/interleave.c:
	* gst/interleave/interleave.h:
	  interleave: avoid using uninitialised ordering_map
	  If self->channel_positions == NULL (which seems unlikely),
	  self->default_channels_ordering_map will be used unintialised.
	  We avoid that by keeping track of the channel_mask, which is set when
	  the ordering map is initialised.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780331

2017-03-23 23:56:31 +1300  Douglas Bagnall <douglas@halo.gen.nz>

	* gst/interleave/interleave.c:
	  interleave: don't overflow channel map with >64 channels
	  When there are more than 64 channels, we don't want to exceed the
	  bounds of the ordering_map buffer, and in these cases we don't want to
	  rempa at all. Here we avoid doing that.
	  https://bugzilla.gnome.org/show_bug.cgi?id=780331

2017-03-28 14:23:16 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: Use get_pkgconfig_variable instead of calling pkg-config ourself
	  It is avalaible in meson 0.36 which is now are requirement

2017-03-28 14:22:41 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* pkgconfig/gstreamer-plugins-good.pc.in:
	* pkgconfig/meson.build:
	  pkgconfig: Do not ever build an installed .pc file

2017-03-28 11:15:53 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* tests/check/meson.build:
	  meson: test: Fix environment object usage

2017-03-28 11:14:47 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>

	* meson.build:
	* pkgconfig/gstreamer-plugins-good.pc.in:
	* pkgconfig/meson.build:
	  pkgconfig: Generate the pkg-config with meson too

2017-03-27 21:52:00 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: In gap mode, consider the mdat offset when calculating the remaining mdat size
	  The mdat generally does not start at offset 0, we have to include the
	  size of the moof and whatever else was in front of the mdat.

2017-03-27 11:43:31 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/atomsrecovery.c:
	  atomsrecovery: Error out when fseek() fails instead of silently ignoring
	  CID 1403262

2017-03-23 22:13:05 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>