Move gtk plugin from -bad

https://bugzilla.gnome.org/show_bug.cgi?id=754094
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..62aa724
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,67 @@
+aclocal.m4
+autom4te.cache
+config.h
+config.h.in
+config.h-new
+config.log
+config.status
+config.guess
+config.sub
+config.rpath
+configure
+libtool
+stamp-h
+stamp-h.in
+stamp-h1
+gst-element-check-*.m4
+ltmain.sh
+missing
+mkinstalldirs
+compile
+install-sh
+depcomp
+autoregen.sh
+ABOUT-NLS
+/INSTALL
+_stdint.h
+gst-plugins-good-*.tar.*
+.dirstamp
+
+/m4
+
+.deps
+.libs
+*.lo
+*.la
+*.o
+Makefile.in
+Makefile
+*~
+*.swp
+*.gc??
+
+/m4
+
+/ext/dv/smpte_test
+/gst/multifile/test-splitmux-part-reader
+/gst/deinterlace/tvtime.h
+
+tmp-orc.c
+*orc.h
+
+/tests/examples/jack/jack_client
+
+Build
+*.user
+*.suo
+*.ipch
+*.sdf
+*.opensdf
+*.DS_Store
+
+/test-driver
+*.log
+*.trs
+
+/build
+/subprojects
diff --git a/.gitmodules b/.gitmodules
new file mode 100644
index 0000000..1e46cf5
--- /dev/null
+++ b/.gitmodules
@@ -0,0 +1,3 @@
+[submodule "common"]
+        path = common
+        url = https://anongit.freedesktop.org/git/gstreamer/common.git
diff --git a/AUTHORS b/AUTHORS
new file mode 100644
index 0000000..5cef5a3
--- /dev/null
+++ b/AUTHORS
@@ -0,0 +1,21 @@
+Erik Walthinsen <omega@temple-baptist.com>
+Matt Howell <mhowell@users.sourceforge.net>
+Brent Bradburn <bbradburn@users.sourceforge.net>
+Wim Taymans <wim.taymans@chello.be>
+Richard Boulton <richard@tartarus.org>
+Zaheer Abbas Merali <zaheerabbas at merali dot org>
+David I. Lehn <dlehn@users.sourceforge.net>
+Chris Emerson <chris@tartarus.org>
+Jens Thiele <karme@unforgettable.com>
+Thomas Nyberg <thomas@codefactory.se>
+Bastien Nocera <hadess@hadess.net>
+Christian Fredrik Kalager Schaller <Uraeus@linuxrising.org>
+Thomas Vander Stichele <thomas@apestaart.org>
+Andy Wingo <wingo@pobox.com>
+Cameron Hutchison <camh@xdna.net>
+David Schleef <ds@schleef.org>
+Benjamin Otte <in7y118@public.uni-hamburg.de>
+Ronald Bultje <rbultje@ronald.bitfreak.net>
+Julien MOUTTE <julien@moutte.net>
+Jan Schmidt <thaytan@mad.scientist.com>
+Arwed v. Merkatz <v.merkatz@gmx.net>
diff --git a/COPYING b/COPYING
new file mode 100644
index 0000000..8add30a
--- /dev/null
+++ b/COPYING
@@ -0,0 +1,504 @@
+		  GNU LESSER GENERAL PUBLIC LICENSE
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diff --git a/ChangeLog b/ChangeLog
new file mode 100644
index 0000000..1414846
--- /dev/null
+++ b/ChangeLog
@@ -0,0 +1,130505 @@
+=== release 1.12.0 ===
+
+2017-05-04  Sebastian Dröge <slomo@coaxion.net>
+
+	* configure.ac:
+	  releasing 1.12.0
+
+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>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Also add videometa if there is padding to the right and bottom
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780478
+
+2017-03-21 12:54:27 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	  rtpmux: fix output segment and buffer DTS to correspond to the flattened PTS
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780347
+
+2017-03-23 17:53:19 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: Remove some unused variables
+
+2017-03-23 15:01:16 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Remove a couple of unneeded levels of indentation
+
+2017-03-22 18:18:40 +0000  Enrique Ocaña González <eocanha@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: distinguish TFDT with value 0 from no TFDT at all
+	  TFDTs with time 0 are being ignored since commit 1fc3d42f. They're
+	  mistaken with the case of not having TFDT, but those two cases
+	  must be distinguished in some way.
+	  This patch passes an extra boolean flag when the TFDT is present.
+	  This is now the condition being evaluated, instead of checking for
+	  0 time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780410
+
+2017-03-22 19:15:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Reset current chunk after writing out timecode
+	  If we have multiple tracks with timecodes, or it's not the first track
+	  that has timecodes, or not the first buffer, we already started a chunk
+	  for media data. We now need to "close" that chunk because we wrote data
+	  for the timecode track and a new chunk has to be started for the
+	  original track the next time it has data.
+
+2017-03-22 18:52:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: Do timecode handling per track, not per muxer instance
+	  There could be multiple video tracks with timecodes.
+
+2017-03-22 00:38:51 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/matroska/matroska-demux.c:
+	  qtdemux: matroskademux: Ignore repeated seek events
+	  Similar to what was done in adaptivedemux, ignore seek
+	  events we've already handled - such as when they are received
+	  on every srcpad of files with lots of streams.
+
+2017-03-21 14:55:32 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  dashdemux: Update mdatleft from overall mdatsize and offset when observing a gap
+	  Otherwise mdatleft will have a value calculated from the initial
+	  mdatsize minus the parts of the stream that we saw, which is not
+	  including all the parts of the stream that might've been skipped.
+
+2017-03-20 17:03:32 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	  docs: update two references to the removed 'mad' plugin
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776140
+
+2017-03-20 12:03:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxqueue.c:
+	  rtprtxqueue: add basic documentation and example pipelines
+	  Mostly explaining the difference between rtprtxqueue and rtprtxsend.
+
+2017-03-17 20:58:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/meson.build:
+	  v4l2: Fix meson plugin shared object name
+	  It didn't match between AutoMake and Meson, and the Meson name
+	  didn't math the plugin name (video4linux2).
+
+2017-03-16 18:20:54 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtprtxreceive: fix example pipelines and improve the documentation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771383
+
+2017-03-17 14:10:40 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstflacparse.h:
+	  flacparse: fix playback if sample number does not start at 0
+	  This reverts commit 29b807685d3c962bbe8afe351c5dca97d59eb5e0, while
+	  fixing the original breaking tests/check/pipelines/flacdec.
+
+2017-03-17 11:30:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstflacparse.h:
+	  Revert "flacparse: fix playback if sample number does not start at 0"
+	  This breaks gst-validate on the build server (though not locally),
+	  and a unit test, and I can't run unit tests right now for some
+	  unrelated reason.
+	  This reverts commit 0747b56f8e7f4731d67f8d13a4bdc453dde0fdf7.
+
+2017-03-16 17:44:41 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: print the correct variable in debug statement
+	  This debug statement is meant to print the time since the last (early)
+	  RTCP transmission, not the last regular RTCP transmission (which also
+	  happens to be set a few lines above to current_time, so the debug output
+	  is just confusing)
+
+2017-03-16 17:42:27 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: convert LOG message to TRACE
+	  This is printed too often (for every chained buffer!) and just clutters the logs.
+
+2017-03-16 14:58:45 +0100  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: fix warning message
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780105
+
+2017-03-16 13:54:54 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstflacparse.h:
+	  flacparse: fix playback if sample number does not start at 0
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777738
+
+2017-03-15 18:58:55 +0100  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	  rtpsource: get clock-rate from pt if needed to generate SR
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780105
+
+2017-03-16 13:52:48 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Include GStreamer souphttpsrc version in default User-Agent string
+
+2017-03-16 00:41:44 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: fix crash with empty sprops-parameters
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780040
+
+2017-03-11 21:20:40 -0800  Thiago Santos <thiagossantos@gmail.com>
+
+	* gst/isomp4/atomsrecovery.c:
+	* gst/isomp4/atomsrecovery.h:
+	  atomsrecovery: also handle extra atoms after 'mdia' in a 'trak'
+	  Take into account the atoms at the end of the 'trak' atom when
+	  recovering it. So that its size (already computed and added in the trak
+	  size) isn't making offsets wrong.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771478
+
+2017-03-11 12:56:33 -0800  Thiago Santos <thiagossantos@gmail.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: avoid fallthrough to moovrecovery failure section
+	  Return before that to preserve our successfull results, otherwise no
+	  moov recovery information would be written
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771478
+
+2017-03-11 12:27:28 -0800  Thiago Santos <thiagossantos@gmail.com>
+
+	* gst/isomp4/atomsrecovery.c:
+	  atomsrecovery: expect more atom types at the headers
+	  Skip more atoms at the header until it finds the 'mdat' to continue the
+	  moov recovery
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771478
+
+2017-03-14 16:42:25 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* Makefile.am:
+	* configure.ac:
+	* tests/examples/Makefile.am:
+	* tests/examples/pulse/.gitignore:
+	* tests/examples/pulse/Makefile.am:
+	* tests/examples/pulse/pulse.c:
+	  pulse example: Remove
+	  That example only tested the property probe interface, which has been removed.
+	  The same kind of thing can now be done with the generic gst-device-monitor tool.
+
+2017-03-14 16:38:02 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2: Remove unused macro
+
+2017-03-14 16:35:25 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2: Remove unused definitions
+
+2017-03-14 10:10:19 +0100  Emeric Grange <egrange@gopro.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_types.c:
+	  qtmux: add CineForm support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780024
+
+2017-03-14 15:09:44 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Only create new chunks if we have more than a single stream
+	  There's no point in creating multiple chunks otherwise, it only wastes
+	  some bytes for storing the chunk offsets.
+
+2017-03-14 10:09:46 +0100  Emeric Grange <egrange@gopro.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: add S16L support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=780022
+
+2017-03-14 15:48:08 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/check/elements/splitmux.c:
+	  splitmux test: Use passed first/last timestamps
+	  Don't hard-code the expected timestamp range, use the
+	  values the caller is passing in.
+
+2017-03-12 11:42:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* Makefile.am:
+	* docs/plugins/inspect/plugin-soup.xml:
+	  Add old plugin names to cruft list
+	  This will help fixing uninstalled setup. Also fix missing path
+	  correction in one of the plugin xml.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779344
+
+2016-12-15 12:38:40 +0100  Michael Dutka <mail@michael-dutka.de>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph265depay.c:
+	  rtph264depay, rtph265depay: remove stray g_debug()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779858
+
+2017-03-10 11:24:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: init fourcc
+	  Initialize the fourcc to 0 so that we can detect failure later.
+
+2017-03-08 22:50:52 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/level.c:
+	* tests/check/elements/rglimiter.c:
+	  tests: Add missing LDADD for libm in tests using math.h
+	  Also, remove the math.h include for the one that just prentend to need
+	  it.
+
+2017-03-08 22:15:46 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	  Fix shout2 plugin doc generation
+	  In the previous patch, we also renamed shout2send to shout2, so it does
+	  not clash with it's feature. Though we forgot to rename it in the doc
+	  reference. This patch also add a cruft detection on the xml that made me
+	  miss this error.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779344
+
+2017-03-04 10:52:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-shout2.xml:
+	* ext/pulse/Makefile.am:
+	* ext/pulse/meson.build:
+	* ext/shout2/gstshout2.c:
+	* ext/soup/Makefile.am:
+	* ext/soup/meson.build:
+	* sys/oss4/Makefile.am:
+	  Fix plugin filenames to match plugin names
+	  - libgstpulse.so becomes libgstpulseaudio.so
+	  - libgstsouphttpsrc.so becomes libgstsoup.so
+	  - libgstoss4audio.so becomes libgstoss4.so
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779344
+
+2017-03-08 16:01:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	  qtmux: Free EDTS instead of just clearing it and setting it to NULL
+
+2017-03-08 15:27:32 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Fix some memory leaks related to timecode tracks
+
+2017-03-04 00:34:44 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/check/elements/splitmux.c:
+	  splitmux: Add unit test for reverse playback
+	  Ensure that reverse playback works and generates the range
+	  of timestamps (0-3s) we expect, in monotonically descending order.
+
+2017-02-28 11:50:45 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Fix reverse playback
+	  Fix the check for whether the start time of the segment has
+	  been reached when playing in reverse. Otherwise, playback
+	  stops after reaching the start of any file part, instead of
+	  continuing until all parts within the segment have played
+
+2017-02-22 03:01:31 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Don't lose crypto info on a new moof
+	  We parse the next moof in advance of having pushed
+	  all samples from the previous one in some cases, and
+	  we'll still need the crypto info from the previous
+	  fragment so keep around any unused crypto info entries
+	  when adding new ones
+
+2017-02-27 13:55:58 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Update modification times when sending the moov
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779422
+
+2017-03-01 16:11:47 -0800  Michael Smith <mlrsmith@gmail.com>
+
+	* gst/audioparsers/gstsbcparse.h:
+	  sbcparse: Fix up values for allocation enumeration.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779389
+
+2017-02-28 13:10:50 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtprtxreceive: fix potential leak of old, unassociated, association requests
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722560
+
+2017-02-28 15:47:23 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Don't increment -1 / unset indices
+	  CID 1398545
+
+2017-02-28 15:20:31 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Protect against NULL pointer dereference for streams without caps
+	  CID 1363332
+
+2017-02-28 12:57:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: Free mac on errors
+	  CID 1212149
+
+2017-02-28 12:45:24 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbispay: Add missing break to for loop
+
+2017-02-28 11:02:54 +0100  Edward Hervey <edward@centricular.com>
+
+	* tests/check/Makefile.am:
+	  check: Fix splitmux test CFLAGS
+	  Needs to know where the gstapp headers are
+
+2017-02-27 21:02:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix compilation with gcc 7
+	  qtdemux.c: In function ‘qtdemux_parse_samples’:
+	  qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’ instead [-Werror=int-in-bool-context]
+	  if (stream->samples_per_frame * stream->bytes_per_frame) {
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
+
+2017-02-27 21:01:23 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstmpegaudioparse.c:
+	  mpegaudioparse: Fix compilation with gcc 7
+	  gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’:
+	  gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
+	  memset (mp3parse->xing_seek_table_inverse, 0, 256);
+	  ^~~~~~
+	  gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_handle_first_frame’:
+	  gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to number of elements without multiplication by element size [-Werror=memset-elt-size]
+	  memset (mp3parse->xing_seek_table_inverse, 0, 256);
+	  ^~~~~~
+
+2017-02-27 19:31:39 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbispay: When getting new headers, replace the old version of them
+	  This prevents storing an infinite amount of e.g. comment headers if they
+	  come without a new initialization header in front of them. There can
+	  only be one header of each type.
+
+2017-02-27 19:25:35 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/rtp-payloading.c:
+	  rtp-payloading: Add new test for Vorbis renegotiation
+	  Check if encoding, payloading, depayloading and decoding works if the
+	  stream configuration (and thus the headers) change.
+
+2017-02-27 19:24:07 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  vorbispay: Only replace headers when receiving a new config header
+	  If we also replace all headers when receiving any possibly following
+	  comments header, we would throw away the config header before being able
+	  to make use of it.
+
+2017-02-23 12:11:15 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/splitmux.c:
+	  tests: splitmux: add unit test for content with sparse streams
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761086
+
+2017-02-22 11:23:19 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxpartreader: ignore sparse streams when calculating the end offset of a part
+	  A sparse stream's ending timestamp can be considerably smaller
+	  than the ending timestamps of the other streams, which can lead
+	  to skipping considerable time from the next part.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761086
+
+2017-02-22 11:21:06 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxpartreader: identify sparse streams
+
+2017-02-25 21:47:03 -0300  Edgard Lima <edgard.lima@gmail.com>
+
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/rtp/gstrtpg726depay.c:
+	* gst/rtp/gstrtpg726depay.h:
+	* gst/rtp/gstrtpg726pay.c:
+	* gst/rtp/gstrtpg726pay.h:
+	* gst/rtp/gstrtppcmadepay.c:
+	* gst/rtp/gstrtppcmadepay.h:
+	* gst/rtp/gstrtppcmapay.c:
+	* gst/rtp/gstrtppcmapay.h:
+	* gst/rtp/gstrtppcmudepay.c:
+	* gst/rtp/gstrtppcmudepay.h:
+	* gst/rtp/gstrtppcmupay.c:
+	* gst/rtp/gstrtppcmupay.h:
+	* gst/rtp/gstrtpspeexdepay.c:
+	* gst/rtp/gstrtpspeexdepay.h:
+	* gst/rtp/gstrtpspeexpay.c:
+	* gst/rtp/gstrtpspeexpay.h:
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	* sys/v4l2/gstv4l2colorbalance.c:
+	* sys/v4l2/gstv4l2colorbalance.h:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2src.h:
+	* sys/v4l2/gstv4l2tuner.c:
+	* sys/v4l2/gstv4l2tuner.h:
+	* sys/v4l2/gstv4l2vidorient.c:
+	* sys/v4l2/gstv4l2vidorient.h:
+	* sys/v4l2/v4l2_calls.c:
+	* sys/v4l2/v4l2_calls.h:
+	  Update Edgard Lima's email
+	  https://bugzilla.gnome.org/show_bug.cgi?id=779230
+
+2017-02-08 13:36:00 +0000  Andrew <nifigase@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  rtpjitterbuffer: Don't always reset PTS to 0 after a gap
+	  In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
+	  timestamps is more than (3 * jbuf->clock_rate) we call
+	  rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
+	  the pipeline (playes, mixers) just skip frames/samples until pts becomes
+	  equal to pts before gap.
+	  In version 1.10.2 and before this checking was bypassed for packets with
+	  "estimated dts", and gaps were handled correctly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778341
+
+2017-02-24 15:59:41 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* meson.build:
+	  meson: Update version
+
+2017-02-24 15:37:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.11.2 ===
+
+2017-02-24 15:07:23 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	  Release 1.11.2
+
+2017-02-24 12:50:21 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2017-02-24 12:44:58 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/el.po:
+	  po: Update translations
+
+2017-02-10 20:50:17 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Extract redirection uri on libsoup's restarted callback
+	  Let libsoup handle redirection automatically.
+	  And then, to figure out redirection uri, extract it on "restarted"
+	  callback which will be fired before soup_session_send() is returned.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778428
+
+2017-01-02 19:29:04 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Update image size when extrapolating
+	  Update the image size according the amount of data we are going to
+	  read/write. This workaround bugs in driver where the sizeimage provided
+	  by TRY/S_FMT represent the buffer length (maximum size) rather then the expected
+	  bytesused (buffer size).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775564
+
+2017-02-17 15:50:32 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: fix typo in _acquire_format() error messages
+	  Fixes:
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778815
+
+2017-02-07 17:27:56 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/matroskamux.c:
+	* tests/check/elements/qtmux.c:
+	  tests: matroskamux, qtmux: don't add codec_data buffers to template caps
+	  streamheader and codec_data buffers fields are only meant to be
+	  in the negotiated caps, not the template caps.
+	  Fixes false-positive leaks of those buffers detected by the leaks
+	  tracer, as template caps are static, and we decided to not include
+	  code in gstreamer core to handle this unusual case of template caps
+	  having buffers in them.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768762
+
+2017-02-09 12:46:54 +0000  Jochen Henneberg <jh@henneberg-systemdesign.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbispay: Update and send out headers when new headers are received
+	  The payloader needs to reset and update the vorbis config data which is
+	  pushed on the network if it receives new headers, or at least, it may
+	  have to do so.
+	  Without this, the stream configuration could change without the
+	  payloader sending the new configuration to the other side.
+
+2017-02-15 14:48:58 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Change files on incompatible caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-15 16:35:01 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Reset ready_for_output on state change
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-15 15:09:06 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Remove unused next_max_out_running_time
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-15 15:07:32 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Remove unused muxed_out_time
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761761
+
+2017-02-17 13:07:05 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  Revert "qtdemux: Always snap to the start of the keyframe"
+	  This reverts commit 107902ec514bd826aa29d2298107e2c091e1c779.
+	  This commit intended to ensure that keyframe seeks land at the
+	  start timestamp of a keyframe, rather than in the middle of one,
+	  but they cause trouble on files with sparse streams, or with
+	  JPEG 'cover art' tracks that have only one or a few JPEG samples
+	  with very long durations.
+	  That's still desirable for doing seamless cutting of videos,
+	  but needs a rethink for implementation.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778690
+
+2017-02-17 01:22:11 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/audiofx/audioecho.c:
+	* gst/audiofx/audioecho.h:
+	  audiofx/echo: added surround-delay and surround-mask
+	  Add a new boolean surround-delay property that makes
+	  audioecho just apply a delay to certain channels to create
+	  a surround effect, rather than an echo on all
+	  channels. This is useful when upmixing from stereo - for example.
+	  Add a surround-mask property to control which channels
+	  are considered surround sound channels when adding a
+	  delay with surround-delay = true
+	  Original patch from Jochen Henneberg <jh@henneberg-systemdesign.com>
+
+2017-02-15 00:13:30 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Use IP_MULTICAST_ALL for filtering IPv4 packets if available
+	  This goes around the inefficient control message based filtering and
+	  does all the filtering kernel-side. Unfortunately this is Linux-only and
+	  there is no IPv6 variant of it (yet).
+
+2017-02-14 19:53:30 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* Makefile.am:
+	  meson: dist meson build files
+	  Ship meson build files in tarballs, so people who use tarballs
+	  in their builds can start playing with meson already.
+
+2017-02-10 10:53:05 +0100  Søren Juul <zpon.dk@gmail.com>
+
+	* gst/icydemux/gsticydemux.c:
+	* tests/check/elements/icydemux.c:
+	  icydemux: reset tags on empty value
+	  Some radio streams uses StreamTitle='' to reset the title after a
+	  track stopped playing, e.g. while the host talks between tracks or
+	  during news segments.
+	  This change forces an empty tag object to be distributed if
+	  StreamTitle or StreamUrl is received with empty value, thus allowing
+	  downstream elements to get notified about this.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778437
+
+2017-02-13 11:17:25 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Properly notify missing elements
+	  If the srtp elements are not present, post a message on the bus
+	  informing about the missing plugins.
+
+2017-02-10 10:32:57 -0300  Juan Pablo Ugarte <ugarte@endlessm.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: mark singleton caps as "may be leaked" objects.
+	  Set MAY_BE_LEAKED flag on static pads returned by gst_v4l2_object_get_*_caps()
+	  functions. Made functions thread safe by using g_once_init[enter|leave]
+	  funtions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778453
+
+2017-02-09 14:18:30 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: Remove now unused done label
+
+2017-02-09 12:55:32 +0100  Nick Kallen <nickkallen@me.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: do not cache caps
+	  Upstream elements like videoflip can transform caps, such as changing width and height.
+	  When an imagefreeze downstream receives an ACCEPT_CAPS query it will NOW return
+	  all caps that it can accept.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778389
+
+2017-02-09 11:29:43 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Add a comment about how atom_trak_set_elst_entry() works
+
+2014-08-22 09:55:43 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux_dump.c:
+	  qtdemux: demote some log messages to TRACE level
+	  Don't spam debug log with uninteresting stuff.
+
+2017-02-08 17:24:26 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Clear edit lists every time we recalculate them
+	  We recalculate them, so any old information has to be forgotten.
+	  Otherwise we write invalid edit lists when writing headers multiple
+	  times.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778330
+
+2017-02-07 13:10:18 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxsrc: Allow for buffers before the segment when measuring
+	  Used signed calculations when measuring the max_ts of an input
+	  fragment, so as to calculate the correct duration and offset
+	  when buffers have timestamps preceding their segment
+
+2017-02-02 12:55:25 +0100  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	  rtpsession: relate received FIRs and PLIs to source
+	  This is needed in order to:
+	  - Avoid ignoring requests for different media sources.
+	  - Add SSRC field in the GstForceKeyUnit event.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=778013
+
+2017-01-30 20:20:08 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: sanity check number of segments in edit list
+	  Fixes crash with fuzzed file.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777940
+
+2017-01-02 22:16:39 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Skip seeking query if upstream format is time
+	  Don't need to querying byte-format seeking for time-format
+	  upstream case
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776715
+
+2016-12-01 12:47:08 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Use upstream's StreamFlags if there are
+	  When multiple demuxer's are used, upstream might want to indicate
+	  default streams using GST_STREAM_FLAG_{SELECT, UNSELECT}
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775440
+
+2017-01-27 16:14:16 +0200  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/isomp4/atoms.c:
+	  qtmux: Timecode track fixes for STSD entry
+	  The n_frames field (frames per second) should follow the nominal frame
+	  rate for drop-frame timecodes.
+	  Also, the trak's timescale (and duration, accordingly) should follow the
+	  STSD entry's timescale and frame duration (fps_n and fps_d accordingly),
+	  not the other way around.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777832
+
+2017-01-19 11:08:11 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: retry request on early termination from the server
+	  Fix a regression introduced by commit 183695c61a54f1 (refactor to use
+	  Soup's sync API). The code previously attempted to reconnect when the
+	  server closed the connection early, for example when the stream was put
+	  in pause for some time.
+	  Reintroduce this feature by checking if EOS is received before the
+	  expected content size is downloaded. In this case, do the request
+	  starting at the previous read position.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776720
+
+2017-01-10 09:40:56 -0700  Matt Staples <staples255@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: find_stream_by_channel should ignore unconfigured streams
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777101
+
+2017-01-25 18:43:00 +0000  Brendan Shanks <brendan.shanks@teradek.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Fix debug typo and remove misleading warning
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777362
+
+2017-01-25 20:56:24 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/examples/rtp/client-PCMA.c:
+	  rtp: Remove unused variable in example
+	  client-PCMA.c:84:22: warning: unused variable 'isrc' [-Wunused-variable]
+	  GObject *session, *isrc, *osrc;
+	  ^
+
+2017-01-09 11:32:35 +0530  Rahul Bedarkar <rahul.bedarkar@imgtec.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: check for not NULL before clearing adapter
+	  In case wavparse receives a manually injected FLUSH_STOP event
+	  while operating in pull mode we get criticals because we'd try
+	  to clear a NULL adapter.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777123
+
+2017-01-24 19:23:44 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* tests/check/meson.build:
+	  meson: Properly use ':' for defining keywords
+
+2017-01-17 16:41:58 +0100  Jean-Christophe Trotin <jean-christophe.trotin@st.com>
+
+	* sys/v4l2/gstv4l2allocator.c:
+	  v4l2allocator: reference memory before the buffer is queued
+	  In gst_v4l2_allocator_qbuf(), the memory is referenced after the
+	  buffer is queued. Once queued (VIDIOC_QBUF), the buffer might be handled
+	  by the V4L2 driver (e.g. decoded) and dequeued (gst_v4l2_allocator_dqbuf),
+	  through a different thread, before the memory is referenced (gst_memory_ref).
+	  In this case, in gst_v4l2_allocator_dqbuf(), the memory is unreferenced
+	  (gst_memory_unref) before having been referenced: the memory refcount
+	  reaches 0, and the memory is freed.
+	  So, to avoid this crossing case, in gst_v4l2_allocator_qbuf(), the
+	  memory shall be referenced before the buffer is queued.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777399
+
+2017-01-24 17:59:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	  qtmux: Only write 4 byte zero padding to the Video Sample Description in MOV
+	  For MP4 this is not defined, and it actually breaks things for MSE in
+	  Chrome if we do this. For MOV this is required by some broken software
+	  but the official specification says it's optional:
+	  https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFChap3/qtff3.html
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777540
+
+2017-01-02 13:42:04 +0100  Santiago Carot-Nemesio <scarot@twilio.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpstats.h:
+	  rtpstats: Keep number of nacks sent/received per source
+	  Currently, the nack packets sent or received are kept at session level,
+	  which makes it impossible to distinguish how many of these packages were
+	  sent/received per ssrc when several sources are in the same session. This
+	  patch is aligned with the https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776714
+
+2016-12-08 15:59:33 +0100  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: Fix handling of config-interval
+	  Insert VPS/SPS/PPS before the first NAL unit containing an I-frame in an
+	  access unit only. If an access unit consists of several such NAL units
+	  (tiles) VPS/SPS/PPS should only be inserted before the first of them so
+	  that parameters are only updated between frames.
+	  Do not insert VPS/SPS/PPS before P-frames when config-interval is -1.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775817
+
+2017-01-19 12:29:44 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: report a useful error message when soup_session_send fails
+	  This helps to understand cases where libsoup doesn't set the message
+	  status code after running soup_session_send.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-19 11:05:00 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: properly check that seek range was respected
+	  This check must be done only when we are sure the request was
+	  successfully sent. soup_session_send() might fail without setting the
+	  status code. In this case status code is 0 so we would only catch the
+	  error after the seek range check. In this case we would report an error
+	  saying that the seek range was not respected, instead of reporting the
+	  underlying error that triggered the soup_session_send() failure.
+	  https://bugzilla.gnome.org/attachment.cgi?bugid=777222
+
+2017-01-09 21:04:51 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
+	  gdkpixbufoverlay: add a positioning coefficient pair
+	  ... so as to allow one clearly defined (absolute) positioning mode
+	  that can cater for a variety of absolute but also relative positioning
+	  with respect to edge or center.
+
+2017-01-21 20:48:22 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	  gdkpixbufoverlay: update composition in _before_transform
+	  ... since we need to determine passthrough mode for buffer preparation before
+	  calling into _transform_ip.
+
+2017-01-07 20:11:13 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	  gdkpixbufoverlay: handle setting NULL gdkpixbuf
+	  ... which is a clearer way to clear any current overlay, other than
+	  fiddling with alpha or positioning properties to make it virtually go away.
+
+2017-01-20 17:16:10 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Stop reading a ncdt sub-tag if it goes behind the surrounding tag
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777532
+
+2017-01-20 07:58:26 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Fix various out of bounds reads when parsing ncdt tags
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777500
+
+2017-01-19 13:46:58 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Increment current stts index whenever we finished one stts entry
+	  Otherwise we could read more chunks than there are available, doing an
+	  out of bounds read and potentially crash.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777469
+
+2017-01-19 13:25:53 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  Revert "qtdemux: Increment current stts index in all code paths after reading one chunk"
+	  This reverts commit 99d5d7570d0b53dad3bc8eb653b1320ee422aace. It broke
+	  playback of various valid files.
+
+2017-01-19 07:52:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Increment current stts index in all code paths after reading one chunk
+	  Otherwise we could read more chunks than there are available, doing an
+	  out of bounds read and potentially crash.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777469
+
+2017-01-19 08:37:37 +0100  Edward Hervey <edward@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Initialize return variable
+	  In the normal use-case we would end up with ret being unitialized
+	  causing havoc.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-13 12:27:40 +0000  David Warman <dwarman@manglebit.org>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: avoid XMP tag parsing fatal error.
+	  qtdemux_handle_xmp_taglist() requires a writable taglist,
+	  but qtdemux->tag_list can become non-writable, specifically
+	  after sending global tags (qtdemux.c:958), which adds a
+	  second reference.  Ensure the list is made writable before
+	  calling (make_writable will copy the list if necessary).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766177
+
+2016-05-31 13:17:45 -0300  Thiago Santos <thiagossantos@gmail.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: rework taglist handling
+	  Keep taglist around during element existance to avoid having to
+	  create it at different places before usage. Makes code simpler to handle.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766177
+
+2017-01-16 11:58:02 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: make flow return values handling clearer
+	  The flow return values was stored in the element before because the
+	  result had to be set from callbacks. This is not the case anymore, we
+	  can return the flow result directly from functions, making the code
+	  easier to understand.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-13 16:40:43 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: properly track redirections
+	  The current code configures libsoup to handle redirections
+	  transparently, without informing the caller, thus preventing the element
+	  to record the redirect code and location uri.
+	  Fix this by always setting the SOUP_MESSAGE_NO_REDIRECT, preventing
+	  libsoup from handling the redirection. When we receive a redirection
+	  request and libsoup can safely handle it, return a custom error which
+	  triggers a retry with the new URI.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777222
+
+2017-01-17 10:53:39 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: add 4444 and 4444xq variants to video/x-prores pad template caps
+	  They are handled since commit 7b565475bf551c53b8eed46f7086f3b372f1f6c4
+	  (qt: Add support for ProRes 4444 XQ).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777377
+
+2017-01-17 10:48:57 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/matroska/ebml-read.c:
+	  matroska: Quiet a WARN when parsing push mode
+	  This warning was noisy when returning EOS, which is
+	  just used to indicate more data is needed from upstream.
+
+2017-01-16 14:50:22 +0100  Georg Lippitsch <glippitsch@toolsonair.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't write Sync Sample Atom for ProRes
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777331
+
+2015-01-28 08:58:26 +0100  Enrico Jorns <ejo@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/v4l2_calls.h:
+	  v4l2: Remove usage and definition of LOG_CAPS macro
+	  Unlike former definitions of LOG_CAPS, the current implementation simply
+	  expands to GST_DEBUG_OBJECT. The LOG_CAPS macro is rarely used and most
+	  uses duplicate already existing GST_DEBUG_OBJECT lines. Therefore, the
+	  caps are often printed twice which unnecessarily clutters the debug log.
+	  Replace LOG_CAPS calls with GST_DEBUG_OBJECT, remove LOG_CAPS calls, and
+	  delete the definition of LOG_CAPS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776899
+
+2017-01-16 15:40:43 +0100  Jean-Christophe Trotin <jean-christophe.trotin@st.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: remove duplicated line of code
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777330
+
+2017-01-16 15:17:15 +0100  Jean-Christophe Trotin <jean-christophe.trotin@st.com>
+
+	* sys/v4l2/gstv4l2allocator.c:
+	  v4l2allocator: fix memory type in allocator probe
+	  The buffer memory type provided to the VIDIOC_CREATE_BUFS ioctl shall
+	  be set with the value ("memory") given as input parameter of the
+	  gst_v4l2_allocator_probe() function.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777327
+
+2017-01-14 15:27:19 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: fix other icon counter check
+	  It's never going to be 0 if we first increment and then check.
+
+2017-01-14 15:16:53 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: boldly assume that first 'covr' image is the front cover
+
+2017-01-14 15:09:07 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: extract cover art images into GST_TAG_IMAGE not PREVIEW_IMAGE
+	  These are usually much bigger than icon size and required by
+	  iTunes to be certain fairly large sizes. In qtmux it is also
+	  the IMAGE tags which we write out as 'covr' atoms.
+
+2017-01-14 15:05:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: also set PICTURE tag width and height if available
+
+2017-01-14 14:58:52 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: fix encoder init error with some GST_TAG_PREVIEW_IMAGEs
+	  The encoder fails to initialise when we try to set GST_TAG_PREVIEW_IMAGEs
+	  sent to use by qtdemux from iTunes-generated m4a files. We should
+	  not just blindly translate the PREVIEW tag to file icon image types,
+	  but check if the specific conditions required are met (i.e. image
+	  type 1 must be a 32x32 PNG icon, and what we're getting is 500x500).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776962
+
+2017-01-13 12:39:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  meson: bump version
+
+2016-12-22 17:40:40 +0200  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Add option for timecode-based split
+	  If this option is given, it will calculate the next split point based on
+	  timecode difference.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774209
+
+2017-01-13 00:01:06 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't reset request pad numbering across uses
+	  When reset, don't restart request pad numberings, as
+	  request pads can survive across state changes. Only
+	  restart at 0 if all request pads are handed back first.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777174
+
+2017-01-11 18:52:28 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+	* gst/rtpmanager/gstrtprtxqueue.c:
+	* gst/rtpmanager/gstrtprtxqueue.h:
+	  rtxqueue: Expose basic statistics as properties.
+	  Statistics about the total number of retransmission requests
+	  and the actual number of retransmitted packets can be helpful
+	  at application-level.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777182
+
+2017-01-12 17:45:35 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: simplify video/x-h264 caps handling
+	  'stream-format' and 'alignment' are defined in pad template caps so
+	  there is no need to check them again here. Also remove bitrate parsing from
+	  caps as bitrate in caps doesn't make sense but from tags, which is
+	  actually the case.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777181
+
+2016-12-08 17:02:22 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: add basic HEVC/H.265 muxing support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736752
+
+2017-01-11 18:29:05 +0100  Georg Lippitsch <glippitsch@toolsonair.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Calculate clean aperture size
+	  Calculate clean aperture dimensions by first guessing
+	  display aspect ratio based on pixel aspect ratio and
+	  frame size.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777100
+
+2017-01-10 18:19:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/qtdemux_types.c:
+	  qtmux: Write tapt atom for MOV files if PAR not 1/1
+	  Needed for QuickTime 7 to properly play files.
+	  Also write the clap atom for MOV files always, not only when ProRes is
+	  used as a video codec. It's mandatory for MOV.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777100
+
+2017-01-12 16:32:45 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.11.1 ===
+
+2017-01-12 15:31:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	  Release 1.11.1
+
+2017-01-12 14:38:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2017-01-12 14:36:22 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/el.po:
+	* po/hr.po:
+	* po/id.po:
+	* po/zh_CN.po:
+	  po: Update translations
+
+2017-01-11 17:53:32 -0800  Andre McCurdy <armccurdy@gmail.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: free seqh after calling qtdemux_parse_svq3_stsd_data()
+	  The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to
+	  be freed by the caller after use.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777157
+	  Signed-off-by: Andre McCurdy <armccurdy@gmail.com>
+
+2017-01-10 16:01:35 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	  isomp4: Don't spam debug log with knonw/padding atoms
+	  Only output WARNING messages for atoms we don't know how to handle
+	  instead of for padding/known atoms we don't need to do any processing
+	  on
+	  https://bugzilla.gnome.org/show_bug.cgi?id=777095
+
+2017-01-09 19:05:10 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtph263depay.c:
+	* gst/rtp/gstrtpsbcdepay.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  Fix indentation
+
+2017-01-09 19:04:04 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  tests: rtpjitterbuffer: fix compiler warning due to c99-ism
+	  rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode
+
+2016-11-11 14:31:03 +1100  Matthew Waters <matthew@centricular.com>
+
+	* gst/autodetect/gstautodetect.c:
+	  autodetect: bring the element state down after success
+	  Otherwise some messages that are emitted by the element on NULL->READY
+	  will not reach the application.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764947
+
+2017-01-08 01:13:32 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Write tfdt atom into fragmented files.
+	  The DASH spec requires that tfdt atoms be present, so
+	  write one out. ISO/IEC 23009-1:2014 6.3.4.2
+	  https://bugzilla.gnome.org/show_bug.cgi?id=708221
+
+2017-01-07 23:55:42 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Don't reset output timestamps when no tfdt
+	  If a fragmented stream doesn't have a tfdt, don't
+	  reset the output timestamps at each fragment boundary
+	  by erroneously using the default value of 0. Introduced
+	  by commit 69fc48
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754230
+
+2016-12-16 16:51:48 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* ext/vpx/meson.build:
+	* gst/equalizer/meson.build:
+	* gst/isomp4/meson.build:
+	* meson.build:
+	  meson: Install presets files
+
+2017-01-03 10:12:30 +0530  Garima Gaur <garima.g@samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: fix some caps leaks
+	  https://bugzilla.gnome.org//show_bug.cgi?id=776789
+
+2016-12-22 17:34:08 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Return a bin with a "location" property as a sink
+	  Splitmuxsink might be called with a custom bin as a sink. If it has a
+	  "location" property, it can be used.
+
+2016-11-18 22:42:18 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmux: Rewrite buffer collection and scheduling
+	  Majorly change the way that splitmuxsink collects
+	  incoming data and sends it to the output, so that it
+	  makes all decisions about when / where to split files
+	  on the input side.
+	  Use separate queues for each stream, so they can be
+	  grown individually and kept as small as possible.
+	  This removes raciness I observed where sometimes
+	  some data would end up put in a different output file
+	  over multiple runs with the same input.
+	  Also fixes hangs with input queues getting full
+	  and causing muxing to stall out.
+
+2016-11-17 23:40:27 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	* tests/check/elements/splitmux.c:
+	  splitmuxsink: Add format-location-full signal
+	  Add a new signal for formatting the filename, which receives
+	  a GstSample containing the first buffer from the reference
+	  stream that will be muxed into that file.
+	  Useful for creating filenames that are based on the
+	  running time or other attributes of the buffer.
+	  To make it work, opening of files and setting filenames is
+	  now deferred until there is some data to write to it,
+	  which also requires some changes to how async state changes
+	  and gap events are handled.
+
+2016-12-31 01:54:01 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Always snap to the start of the keyframe
+	  When performing a key-unit seek, always snap to the start ts
+	  of the keyframe buffer we landed on so that the keyframe is
+	  entirely within the resulting outgoing segment. That seems
+	  the most sensible result, since the user requested snapping
+	  to the keyframe position.
+
+2016-12-31 01:48:04 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Omit cslg_shift when snapping seeks
+	  Segments times and seek requests are stored and handled
+	  in raw 'PTS' time, without the cslg_shift - which only applies
+	  to outgoing samples. Omit the cslg_shift portion when
+	  extracting PTS to compare for internal seek snaps.
+	  If the cslg_shift is included, then keyframe+snap-before seeks
+	  generate a segment start/stop time that already includes the
+	  cslg_shift, and it's then added a 2nd time, causing the
+	  first buffer(s) to have timestamps that are out of segment.
+
+2016-12-30 22:31:38 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	  qtmux: Remove bogus check in atom_stsc_add_new_entry()
+	  Remove an old check from atom_stsc_add_new_entry() that
+	  extends the last entry in the STSC if the samples per chunk
+	  matches, as the new interleave merging logic requires that
+	  the final entry by updateable. There's already code
+	  below which simply merges the final entry into the previous
+	  one when needed, so rely on that instead.
+	  Fixes asserts like:
+	  ERROR:atoms.c:2940:atom_stsc_update_entry: assertion failed:
+	  (atom_array_index (&stsc->entries, len - 1).first_chunk == first_chunk)
+
+2016-04-24 21:38:51 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix key_time in gst_qtdemux_adjust_seek()
+	  time in segment should be PTS based (not DTS).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765498
+
+2016-12-28 22:49:27 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	* gst/multifile/gstsplitmuxpartreader.h:
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Pass seek flags when activating.
+	  Pass all seek flags when activating a part
+	  based on a seek, so that SNAP flags are preserved.
+
+2016-11-26 01:13:19 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmux: Fix a small race in the splitmuxsrc
+	  Make sure the state of the parser is set to
+	  collecting streams before chaining up to the
+	  parent change_state() method, to close a
+	  small window that can cause playback to
+	  never commence.
+
+2017-01-02 15:06:33 +0100  Edward Hervey <edward@centricular.com>
+
+	* tests/check/elements/amrparse.c:
+	  check: Remove dead code
+
+2016-12-31 09:52:25 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/multifile/gstmultifilesink.c:
+	* gst/multifile/gstmultifilesink.h:
+	  multifilesink: refactor max_files handling a bit
+	  Use GQueue instead of a GSList so we don't have to traverse
+	  the whole list to append something every time. And it also
+	  keeps track of the number of items in it for us.
+	  Add a function to add filenames to the list of old files and
+	  use it in more places, so that memory doesn't build up in
+	  other modes either if no max_files limit is specified.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766991
+
+2016-05-29 17:21:47 +0100  Ursula Maplehurst <ursula@kangatronix.co.uk>
+
+	* gst/multifile/gstmultifilesink.c:
+	  multifilesink: don't leak memory when no max-files limit is set
+	  Technically we weren't leaking the memory, just storing it internally
+	  and never using it until the element is freed. But we'd still use more
+	  and more memory over time, so this is not good over longer periods
+	  of time. Only keep track of files if there's actually a limit set,
+	  so that we will prune the list from time to time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766991
+
+2016-12-29 12:39:20 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: adjust segment stop for KEY_UNIT negative rate seeking
+
+2016-12-29 12:25:35 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: implement pull mode SNAP flag seeking
+
+2016-12-29 11:26:33 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: tweak KEY_UNIT SNAP seek handling
+	  Previously, seeking to position y where y is (strictly) within a keyframe
+	  would seek to that keyframe both with SNAP_BEFORE and SNAP_AFTER,
+	  where the latter is now adjusted to really snap to the next keyframe.
+
+2016-12-28 13:23:11 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: correctly perform pull mode KEY_UNIT seeking
+	  Rather amazingly (and equally unnoticed), keyunit seeking resulted in segments
+	  where start != time (which is bogus for simple avi timeline).  So, properly
+	  adjust the segment (start) rather than fiddling with segment time (only).
+
+2016-12-28 13:04:54 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: restore considering of pull mode KEY_UNIT seeking
+	  ... by using the original seek event's flags rather than the corresponding
+	  segment flags, which do not have such counterpart flags (and
+	  do no longer have them covertly sneaking in nowadays).
+
+2015-05-08 12:44:01 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: only drop actual streamheader buffers with xiph codecs
+	  With Xiph codecs the stream header buffers are both in the caps and are
+	  usually also at the beginning of each input stream, but it's perfectly
+	  possible that the input stream does not have the stream header buffers
+	  inline in the data. Matroskamux would drop the first N buffers assuming
+	  they're stream headers, but this meant it would drop actual payload data
+	  when the stream didn't contain the stream headers inline. Fix this by
+	  only dropping leading buffers if they're flagged as stream headers. This
+	  fixes issues with streams that are being tapped into after streaming
+	  has started.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=749098
+
+2016-12-21 17:43:58 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* tests/check/elements/matroskamux.c:
+	  matroskamux: adjust unit test to modified behaviour
+	  Now matroskamux mark all packets of audio-only streams as keyframes so
+	  in test_block_group after pushing the test audio data 4 buffers are produced
+	  and not more 2. The last buffer is the original data and must match with what
+	  pushed. The remaining ones are matroskamux headers
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754696
+
+2016-05-30 01:15:31 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: mark all packets of audio-only streams as keyframes
+	  This helps with streaming audio-only streams via multifdsink,
+	  tcpserversink and such.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754696
+
+2015-03-28 18:15:36 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: add G722 audio support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746574
+
+2016-12-13 11:11:07 +0900  Wonchul Lee <wonchul.lee@collabora.com>
+
+	* gst/udp/gstudpsrc.c:
+	  updsrc: Add to join multiple multicast interfaces
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776030
+
+2015-03-25 13:51:30 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpklvdepay.c:
+	  rtpklvdepay: add the SPARSE flag to the outgoing stream-start event
+
+2016-12-14 14:37:45 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpmanager: place content before Since-version API marker
+	  Avoids confusing the parser
+
+2016-12-14 14:16:53 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2: fix 404 in package origin
+
+2016-12-14 21:45:15 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Check if we have enough data available when parsing edit lists
+	  Also consume the data entry by entry to get complicated indexing out of
+	  the code.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776107
+
+2016-12-14 19:15:03 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't check size in a non-list value
+	  After commit 1ea9735a I see these error while using the webcam
+	  integrated in my laptop:
+	  GStreamer-CRITICAL **: gst_value_list_get_size: assertion 'GST_VALUE_HOLDS_LIST (value)' failed
+	  The issue is gst_v4l2src_value_simplify() was doing its job of
+	  generating a single value, rather than the original list. That why,
+	  when getting the list size, a critical warning was raised.
+	  This patch takes advantage of the compiler optimizations to verify
+	  first if the list was simplified, thus use it directly, otherwise,
+	  if it is a list, verify its size.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=776106
+
+2016-12-14 10:39:12 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  tests/jitterbuffer: Major refactoring and cleanups
+	  * Changed PCMU->TEST for common macros
+	  * Changed verify-functions (lost & rtx) into macros.
+	  * Remove option to add marker-bit for test-buffers (not used anywhere)
+	  * Add new push_test_buffer function that makes sure there are correlation
+	  between dts and the time on the clock. (classic test-mistake)
+	  * Established a generic starting-point for tests with the
+	  construct_deterministic_initial_state function and use it where
+	  applicable, which removes lots of "boilerplate" everywhere.
+	  * Add basic lost-event test
+	  * Remove as much "magic constants" as possible.
+	  * Remove 3 tests that no longer are testing anything that others don't,
+	  and was completely unmaintainable.
+	  * Remove unnecessary use of the testclock
+	  * Verify each test is testing what it actually says it does (and modify
+	  where it doesn't)
+	  In general, make the tests much smaller, better, more maintainable and
+	  readable.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774409
+
+2016-12-14 09:54:11 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* .gitignore:
+	* Makefile.am:
+	* configure.ac:
+	* gst-plugins-good.spec.in:
+	  Remove generated .spec file
+	  Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-14 10:15:10 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Check that the XiTh size is big enough
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-09 20:27:53 +0900  Heekyoung Seo <heekyoung.seo@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Check node length of video sample description
+	  Add check for node length of video sample description and its fields and
+	  for the XiTh atom.
+	  Also unify the code a bit.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-08 18:50:52 +0900  Heekyoung Seo <heekyoung.seo@lge.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Enable xvid/mp2 codec support
+	  Add support for xvid video and mp2 audio, add m2v1 fourcc.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-13 22:32:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvp9depay.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	* tests/check/elements/rtprtx.c:
+	* tests/check/elements/vp9enc.c:
+	  gst: Don't declare variables inside the for loop header
+	  This is a C99 feature.
+
+2016-12-11 13:27:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/gstscaletempo.c:
+	  scaletempo: Ensure to reinit buffers whenever they were not allocated yet
+	  That is, whenever we go through start/stop we have to ensure that on the
+	  next opportunity the buffers are reallocated again. Otherwise the
+	  buffers might be NULL because the element was reused with the same
+	  configuration as before (i.e. set_caps() wouldn't have reinited the
+	  buffers).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775898
+
+2016-12-10 12:52:18 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/design/Makefile.am:
+	* docs/design/design-rtpauxiliary.txt:
+	* docs/design/design-rtpcollision.txt:
+	* docs/design/design-rtpretransmission.txt:
+	  docs: design: remove, moved to gst-docs
+
+2016-12-09 17:17:35 -0300  Thibault Saunier <tsaunier@gnome.org>
+
+	* meson.build:
+	  meson: Support building without Gst debug
+
+2016-12-09 17:55:39 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	* gst/flx/gstflxdec.h:
+	  flxdec: Only send SEGMENT events after CAPS
+	  I.e., don't just forward the event but delay it if we don't have caps on
+	  the srcpad yet.
+
+2016-12-09 17:49:40 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: Unref and unmap buffers in all code paths as needed
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775888
+
+2016-12-06 17:42:31 +0530  Arun Raghavan <arun@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't set empty interlace-mode list
+	  If for some reason we fail to probe formats (all try_fmt calls fail, for
+	  example), this is not a critical error, but we end up with an empty list
+	  of interlace modes. This causes all subsequent negotiation to fail.
+	  This patch fixes interlace-mode setting to be skipped if we failed to
+	  detect any.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775702
+
+2016-12-07 17:22:22 +0530  Garima Gaur <garima.g@samsung.com>
+
+	* gst/monoscope/gstmonoscope.c:
+	  monoscope: Unref allocation query after finished with it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775752
+
+2016-12-06 07:48:47 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: Allocate 0-initialized memory for the decoded frame
+	  Otherwise we might leak arbitrary information from the uninitialized
+	  memory if not every pixel is written.
+	  https://scarybeastsecurity.blogspot.gr/2016/12/1days-0days-pocs-more-gstreamer-flic.html
+
+2016-12-05 07:57:19 -0700  Matt Staples <staples255@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix session cleanup when handling redirect on PLAY
+	  Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
+	  removing code from gst_rtspsrc_send that changed the state varable upon
+	  encountering a redirect. Better to let the redirect handlers in
+	  gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
+	  state-dependent cleanup.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775543
+
+2016-09-07 16:10:27 +0300  Aleix Conchillo Flaque <aleix@oblong.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: always send teardown request
+	  Allow CMD_CLOSE to cancel all commands not only CMD_PAUSE
+	  and ignore CMD_WAIT while closing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748360
+
+2016-12-03 08:19:27 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* README:
+	* common:
+	  Automatic update of common submodule
+	  From f980fd9 to 39ac2f5
+
+2016-12-01 17:08:09 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  jitterbuffer: Don't leak duplicate items
+	  When providing items with a seqnum, there is a (very small) probability
+	  that an element with the same seqnum already exists. Don't forget
+	  to free that item if it wasn't inserted.
+	  And avoid returning undefined values when dealing with duplicate items
+
+2016-12-01 11:23:02 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Sanitize unknown codec caps
+	  We might have non-printable characters in the unknown fourcc, replace
+	  them with '_', in the same way we do it for unknown tags.
+
+2016-12-01 20:04:28 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Free vprp chunk also if it existed but we made no use of it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775479
+
+2016-12-01 17:38:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroskademux: Fix memory leak when parsing attachments
+	  gst_tag_image_data_to_image_sample() does not take ownership of the
+	  passed memory, so don't set it to NULL to allow us to free it later.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775472
+
+2016-12-01 14:56:18 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroskademux: Unify zlib/bzip2 decompress loops with the ones from qtdemux
+	  Especially, simplify the code a bit.
+
+2016-12-01 14:41:48 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Increase inflate buffer in bigger steps
+	  1024 bytes is quite small, let's do 4096 bytes (or one page).
+	  Also remove redundant if, we're always in that case when getting here.
+
+2016-12-01 14:30:49 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Ensure that size of the pasp atom is as much as we need
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:30:10 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Free compressed moov node and it's corresponding decompressed data
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:29:21 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Check size of compressed MOOV header against available data
+	  And actually read the size of the cmvd atom from the right position.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:27:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix zlib inflate loop
+	  Handle errors cleanly, deallocate all memory and return the actual size
+	  of the inflated data.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 13:38:16 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: Make sure we have enough data in the codec_data to be able to parse it
+	  Also error out cleanly if mapping the buffer failed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775450
+
+2016-12-01 13:32:22 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix out of bounds read in tag parsing code
+	  We can't simply assume that the length of the tag value as given
+	  inside the stream is correct but should also check against the amount of
+	  data we have actually available.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775451
+
+2016-12-01 15:06:06 +0530  Garima Gaur <garima.g@samsung.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtpsbcdepay.c:
+	  rtp: Fix some memory leaks in usage of gst_pad_get_current_caps()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775071
+
+2016-11-30 17:56:02 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Read interlacing information from 'fiel' atom
+	  Read interlacing and TFF/BFF information from the 'fiel' atom and pass it
+	  into the caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775414
+
+2016-11-29 13:55:40 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix compiler warning
+	  qtdemux.c: In function ‘qtdemux_parse_trak’:
+	  qtdemux.c:10184:38: error: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 9 has type ‘gint {aka const int}’ [-Werror=format=]
+	  GST_DEBUG_OBJECT (qtdemux, "Found jpeg: len %u, need %lu", len,
+	  ^
+
+2016-11-28 13:45:24 -0800  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Change off_t type to gint
+	  off_t is a signed integer type provided by sys/types.h on posix systems.
+	  Replace with gint for building on non-posix systems (like windows).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775287
+
+2016-11-22 21:00:25 -0800  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* meson.build:
+	  meson: add libm to has_function checks
+	  The functions from math.h may be implemented in libm.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774876
+
+2016-10-27 23:02:37 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* ext/meson.build:
+	  Revert "meson: dv plugin now works on MSVC"
+	  This reverts commit 05a89613feff70cff416367f5aa807a1d5c68b63.
+	  Let's not put in stuff that needs unreleased Meson. This can go in
+	  for the next cycle.
+
+2016-11-28 13:51:41 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Ensure that tags are valid UTF-8 before adding them to the taglist
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775219
+
+2016-11-28 12:22:49 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/multipart/multipartdemux.c:
+	  multipartdemux: Post an error message on the bus if we got EOS without having added any pads
+
+2016-11-28 12:00:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Handle non-UTF8 headers and error reasons more gracefully
+	  Especially don't put them into GstStructures in one way or another, just
+	  ignore them or error out cleanly depending on the importance of their
+	  content.
+
+2016-11-28 09:30:25 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvrawpay.c:
+	  vrawpay: Error out cleanly if mapping the video frame fails
+	  Instead of later dereferencing NULL and crashing.
+
+2016-11-27 11:14:13 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: Update statistics before pushing
+	  If an element queries the number of retransmission buffers pushed
+	  *while* the push is still taking place (and before the object lock
+	  is taken just after) it would end up with the wrong statistic
+	  being reported.
+	  Increment it just before the push, avoids races when getting statistics
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768723
+
+2016-11-26 11:20:51 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* .gitmodules:
+	  common: use https protocol for common submodule
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-07-28 18:51:24 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  gstv4l2bufferpool: lock flush_stop against regular qbuf
+	  These can be called from different threads and both manipulate the
+	  pool->buffers array. Lock them properly and let flush_stop move the
+	  array contents into a temporary array on the stack to avoid having
+	  to call release_buffer under the object lock.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775015
+
+2016-11-24 14:25:22 +0100  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  gstv4l2bufferpool: remove critical error message when process is called on an inactive pool
+	  If the pool is inactive, it is guaranteed to also be flushing, so the
+	  following check will return GST_FLOW_FLUSHING anyway.
+	  This can happen if a v4l2src is blocking on DQBUF in create and is sent
+	  an EOS event on another thread. In that case the pool is set to
+	  flushing/inactive without locking, the v4l2src is unblocked, and may
+	  call pool_process with a valid buffer on the already inactive pool.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775014
+
+2016-11-24 14:41:52 +0100  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: release buffer if create fails
+	  gst_base_src_get_range does not expect a buffer to be returned in
+	  the error case, so we are leaking a reference here if create fails.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=775014
+
+2016-11-23 18:34:04 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Handle create_session() returning NULL in bundle code
+	  CID 1394492.
+
+2016-11-22 16:42:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Make sure to only change DTS of writable buffers
+	  And trivial cleanup
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 16:42:26 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Error out much earlier if we don't have a valid PTS
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 16:18:41 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Only use buffer durations if they are actually valid
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 15:59:19 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Revert commits that set DTS and duration on buffers unconditionally
+	  39f7e52266fde3b3c035e22cbcbb2bb1fa207b17 was setting the buffer duration
+	  to 0 if is not valid, under the assumption that this is "the last"
+	  buffer and no others are coming next. This is wrong, last_buf is the
+	  previous buffer and not the very last one.
+	  4e3c13c87c258c9c95e2217d32ab314d12b5fffc was setting DTS to 0 if there
+	  was none. This will set DTS to 0 for all e.g. audio streams, completely
+	  messing up calculations if streams don't start at 0.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 15:58:37 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Only write "gap" edit list if there is a non-zero gap
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-23 07:09:06 +1100  Matthew Waters <matthew@centricular.com>
+
+	* gst/flx/flx_color.c:
+	* gst/flx/flx_fmt.h:
+	* gst/flx/gstflxdec.c:
+	* gst/flx/gstflxdec.h:
+	  flxdec: rewrite logic based on GstByteReader/Writer
+	  Solves overreading/writing the given arrays and will error out if the
+	  streams asks to do that.
+	  Also does more error checking that the stream is valid and won't
+	  overrun any allocated arrays.  Also mitigate integer overflow errors
+	  calculating allocation sizes.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774859
+
+2016-11-23 11:20:49 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: Don't unref() parent in the chain function
+	  We don't own the reference here, it is owned by the caller and given to
+	  us for the scope of this function. Leftover mistake from 0.10 porting.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774897
+
+2016-11-22 20:33:29 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/vpx/gstvpxdec.c:
+	  vpxdec: libvpx's release buffer is sometimes called with fb->priv==NULL
+	  Don't assert on this but just ignore these cases.
+
+2016-11-22 20:24:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Fix cluster searching if we search multiple times in one chunk
+	  After finding a cluster id in the byte reader, we skip ahead the reader
+	  position by one further byte to be able to continue searching from there
+	  inside the same chunk if the cluster candidate was a false positive.
+	  We have to accomodate for that additional byte when resuming the search,
+	  otherwise all following pulls are off-by-one for every resume and we run
+	  into an assertion.
+
+2016-11-22 20:01:20 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-ids.c:
+	  matroska: Add size checks to the parsing of FLAC headers
+
+2016-11-22 23:46:00 +1100  Matthew Waters <matthew@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: fix some warnings comparing unsigned < 0
+	  bf43f44fcfada5ec4a3ce60cb374340486fe9fac was comparing an unsigned
+	  expression to be < 0 which was always false.
+	  gstflxdec.c: In function ‘flx_decode_brun’:
+	  gstflxdec.c:322:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
+	  if ((glong) row - count < 0) {
+	  ^
+	  gstflxdec.c:332:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
+	  if ((glong) row - count < 0) {
+	  ^
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774834
+
+2016-11-21 16:17:31 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: Enable up to 16 unpositioned raw audio channels
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774789
+
+2016-11-22 19:05:00 +1100  Matthew Waters <matthew@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: add some write bounds checking
+	  Without checking the bounds of the frame we are writing into, we can
+	  write off the end of the destination buffer.
+	  https://scarybeastsecurity.blogspot.dk/2016/11/0day-exploit-advancing-exploitation.html
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774834
+
+2016-11-21 15:25:23 +0000  David Evans <bbcrddave@gmail.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Be sure not to read off end of FLAC dfLa box
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773712
+
+2016-11-21 11:48:58 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: add support for skipping invalid data in push mode
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-21 11:48:29 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-parse.c:
+	* gst/matroska/matroska-read-common.c:
+	* gst/matroska/matroska-read-common.h:
+	  matroskaparse: add support for skipping invalid data
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-18 17:00:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Move to new helper function to parse authentication responses
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-20 14:12:16 +0100  christophecvr <stefansat@telenet.be>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix wrong compiler warning with gcc 6.2
+	  | ../../../git/gst/isomp4/qtdemux.c: In function 'qtdemux_parse_tree':
+	  | ../../../git/gst/isomp4/qtdemux.c:10224:24: error: 'size' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+	  |                  offset += size;
+	  |                         ^~
+	  | ../../../git/gst/isomp4/qtdemux.c:10197:25: note: 'size' was declared here
+	  |                  guint32 size, tag;
+	  |                          ^~~~
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774747
+
+2016-11-20 16:15:07 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* Makefile.am:
+	* configure.ac:
+	* win32/MANIFEST:
+	* win32/common/config.h:
+	  win32: remove copies of generated headers
+
+2016-11-20 13:14:08 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavidemux.h:
+	  avidemux: Ensure that raw video have properly aligned buffers
+	  That is, aligned to to 32 bytes for video. Fixes crashes if the raw
+	  buffers are passed to SIMD processing functions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774428
+
+2016-11-20 13:08:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Ensure that raw audio and video have properly aligned buffers
+	  That is, aligned to the basic type for audio and to 32 bytes for video.
+	  Fixes crashes if the raw buffers are passed to SIMD processing functions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774428
+
+2016-11-14 14:44:11 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Always write edit lists for the tracks to give a more accurate duration
+	  Always write an edit list for the whole track. In general this is not
+	  necessary except for the case of having a gap or DTS adjustment but
+	  it allows to give the whole track's duration in the usually more
+	  accurate media timescale.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774403
+
+2016-11-18 22:45:45 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Remove useless return variable
+	  qtdemux_expose_streams() returns flow error immediately, if there is an error.
+	  So, the variable for the flow return is not needed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774674
+
+2016-11-17 13:59:48 +0000  David Evans <bbcrddave@gmail.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_dump.c:
+	* gst/isomp4/qtdemux_dump.h:
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: Add support for FLAC encapsulated in ISOBMFF
+	  As defined by
+	  https://git.xiph.org/?p=flac.git;a=blob_plain;f=doc/isoflac.txt
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773712
+
+2016-11-17 19:59:53 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	  rtpmux: Mark pad as needing reconfiguration again if it failed
+	  And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:59:26 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/monoscope/gstmonoscope.c:
+	  monoscope: Mark pad as needing reconfiguration again if it failed
+	  And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:58:52 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Mark pad as needing reconfiguration again if reconfiguration failed
+	  And consider negotiation failures on flushing pads as FLUSHING, not as
+	  NOT_NEGOTIATED.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:56:23 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdec.c:
+	  dvdec: Fix handling of negotiation failures
+	  Return NOT_NEGOTIATED if sending the caps event fails, or FLUSHING if
+	  the pad was flushing at that point.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 17:16:26 -0800  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* meson.build:
+	  meson: add_global_arguments -> add_project_arguments
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774656
+
+2016-11-16 10:53:51 +0530  Vinod Kesti <vinodkesti@yahoo.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: pad request fails for flvmux
+	  splitmuxsink requests pad from element using pad template like "video_%u", "audio_%u" and "sink_%d". This is true for most of the muxers.
+	  But splitmuxsink not able to request pad to flvmux as flvmux has "audio" and "video" as pad templates.
+	  fix: splitmuxsink should fallback to "audio" and  "video" when template not found.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774507
+
+2016-11-17 10:24:28 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-parse.c:
+	  matroskaparse: Add remaining relevant parts from a3a55305 to the parser
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-16 22:39:01 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-parse.c:
+	  matroskaparse: ignore parsing errors at the end of the file
+	  This is the same change as a3a55305 for the parser.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-16 08:56:34 +0100  Philippe Normand <philn@igalia.com>
+
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/rtpbundle.c:
+	* tests/check/meson.build:
+	* tests/examples/rtp/.gitignore:
+	* tests/examples/rtp/Makefile.am:
+	* tests/examples/rtp/client-rtpbundle.c:
+	* tests/examples/rtp/server-rtpbundle.c:
+	  rtpbin: receive bundle support
+	  A new signal named on-bundled-ssrc is provided and can be
+	  used by the application to redirect a stream to a different
+	  GstRtpSession or to keep the RTX stream grouped within the
+	  GstRtpSession of the same media type.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772740
+
+2016-11-15 16:52:39 +0530  Vinod Kesti <vinodkesti@yahoo.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: assertion while converting ADTS stream to RAW
+	  aacparse resizes input buffer while converting ADTS stream to RAW,
+	  During buffer resize buffer write permission is not checked.
+	  This throws gst_buffer_is_writable assertion and leads to AV sync issue some times.
+	  It is corrected by making buffer writeable using gst_buffer_make_writable
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774129
+
+2016-11-15 21:17:51 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Don't modify upstream TIME segment
+	  TIME segment implies that stream/running time is being handled by upstream.
+	  So, we shouldn't override it without any clue.
+	  This patch is for fixing seek in DASH streaming.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774196
+
+2016-11-14 22:33:27 +0530  Arun Raghavan <arun@osg.samsung.com>
+
+	* config.h.meson:
+	  meson: Add define for v4l2-probe config option
+
+2016-11-14 17:37:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/interleave/deinterleave.c:
+	  deinterleave: Reset caps accumulator to ANY when resyncing the adapter, not EMPTY
+	  The accumulator is filled by intersecting with all the pad caps, as such
+	  it must be initialized with ANY (like it is before the iteration is
+	  started) and not to EMPTY.
+	  Fixes the CAPS query always returning EMPTY caps when resyncing happened
+	  during the query, e.g. because pads were added/removed.
+
+2016-11-14 12:13:14 +0100  Petr Kulhavy <brain@jikos.cz>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: remove redundant saddr unref
+	  The g_object_unref (saddr) before receiving message seems to be redundant as it
+	  is done just before jumping to retry
+	  Though not directly related, part of
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772841
+
+2016-11-12 23:34:23 +0100  Petr Kulhavy <brain@jikos.cz>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: receive control messages only in multicast
+	  Control messages are used only in multicast mode - to detect if the destination
+	  address is not ours and possibly drop the packet. However in non-multicast
+	  modes the messages are still allocated and freed even if not used. Therefore
+	  request control messages from g_socket_receive_message() only in multicast
+	  mode.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772841
+
+2016-11-11 10:45:01 -0800  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* gst/matroska/matroska-mux.c:
+	  Use intermediate guint when handling GstVideoMultiviewFlags
+	  The underlying integer type of the enum GstVideoMultiviewFlags is
+	  implementation defined and may not have the same size as guint.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-11-11 10:44:18 -0800  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* gst/multifile/gstsplitfilesrc.c:
+	  splitfilesrc: update uri_get_type to match the prototype in GstURIHandlerInterface
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-10-26 22:37:34 -0700  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* meson.build:
+	  meson: don't add_global_arguments when being built as a subproject
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773568
+
+2016-10-21 15:49:36 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstflacparse.h:
+	  flacparse: fix header rewriting being ignored
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727802
+
+2016-11-09 06:25:27 +0000  Sean DuBois <sean@siobud.com>
+
+	* gst/flv/gstflvmux.c:
+	* gst/flv/gstflvmux.h:
+	  flvmux: Add metadatacreator property
+	  Allow users to set metadatacreator value in the meta packet
+	  https://bugzilla.gnome.org/show_bug.cgi?id=774131
+
+2016-11-01 19:56:36 +0200  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Use first buffer TS as mux start time
+	  Do not use last buffer TS + buffer duration because buffer duration
+	  might be inaccurate, especially for frame rates like 30fps where a
+	  rounding error is observed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773785
+
+2016-11-03 15:03:59 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: fix timer-reuse bug
+	  When doing rtx, the jitterbuffer will always add an rtx-timer for the next
+	  sequence number.
+	  In the case of the packet corresponding to that sequence number arriving,
+	  that same timer will be reused, and simply moved on to wait for the
+	  following sequence number etc.
+	  Once an rtx-timer expires (after all retries), it will be rescheduled as
+	  a lost-timer instead for the same sequence number.
+	  Now, if this particular sequence-number now arrives (after the timer has
+	  become a lost-timer), the reuse mechanism *should* now set a new
+	  rtx-timer for the next sequence number, but the bug is that it does
+	  not change the timer-type, and hence schedules a lost-timer for that
+	  following sequence number, with the result that you will have a very
+	  early lost-event for a packet that might still arrive, and you will
+	  never be able to send any rtx for this packet.
+	  Found by Erlend Graff - erlend@pexip.com
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773891
+
+2016-10-09 15:59:05 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: fix lost-event using dts instead of pts
+	  The lost-event was using a different time-domain (dts) than the outgoing
+	  buffers (pts). Given certain network-conditions these two would become
+	  sufficiently different and the lost-event contained timestamp/duration
+	  that was really wrong. As an example GstAudioDecoder could produce
+	  a stream that jumps back and forth in time after receiving a lost-event.
+	  The previous behavior calculated the pts (based on the rtptime) inside the
+	  rtp_jitter_buffer_insert function, but now this functionality has been
+	  refactored into a new function rtp_jitter_buffer_calculate_pts that is
+	  called much earlier in the _chain function to make pts available to
+	  various calculations that wrongly used dts previously
+	  (like the lost-event).
+	  There are however two calculations where using dts is the right thing to
+	  do: calculating the receive-jitter and the rtx-round-trip-time, where the
+	  arrival time of the buffer from the network is the right metric
+	  (and is what dts in fact is today).
+	  The patch also adds two tests regarding B-frames or the
+	  “rtptime-going-backwards”-scenario, as there were some concerns that this
+	  patch might break this behavior (which the tests shows it does not).
+
+2016-11-03 16:33:53 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: fix bug in reschedule_timer
+	  The new timeout is always going to be (timeout + delay), however, the
+	  old behavior compared the current timeout to just (timeout), basically
+	  being (delay) off.
+	  This would happen if rtx-delay == rtx-retry-timeout, with the result that
+	  a second rtx attempt for any buffers would be scheduled immediately instead
+	  of after rtx-delay ms.
+	  Simply calculate (new_timeout = timeout + delay) and then use that instead.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773905
+
+2016-11-03 13:27:51 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/wavparse.c:
+	* tests/files/Makefile.am:
+	* tests/files/audiotestsrc.wav:
+	  tests: wavparse: add test for processing an actual .wav file
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773861
+
+2016-11-03 12:34:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Don't set caps to NULL after setting them on the srcpad
+	  We would like to check later on EOS if we found a known stream type or
+	  not, to possibly post an error message.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773861
+
+2016-11-02 14:33:28 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't deref NULL pads in debug output
+	  That tends to crash.
+
+2016-11-02 11:46:07 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  isomp4: Don't use gst_video_colorimetry_to_string_full()
+	  The API was reverted. Just use the plain
+	  gst_video_colorimetry_to_string() function.
+
+2016-11-02 11:00:13 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Fix GObject warnings on shutdown.
+	  Commit 83e718 added a pad template to splitmux request
+	  pads, which means that GstElement now releases the pads on
+	  dispose, but after having removed all elements in the bin
+	  and unlinked them. Make sure we can handle cleanup in that case
+	  without throwing assertions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773784
+
+2016-11-02 02:25:51 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	* gst/multifile/gstsplitmuxsrc.h:
+	  splitmuxsrc: Store seek seqnum and send it on EOS / segment events.
+	  GES relies on the EOS event having the seqnum of the seek that
+	  caused it.
+
+2016-11-02 02:25:00 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Forward a not-linked error on the bus
+	  Handle not-linked as for other fatal errors and post it
+	  onto the bus so the app knows
+
+2016-11-01 21:00:15 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix compiler warning
+	  qtdemux.c: In function ‘qtdemux_parse_tree’:
+	  qtdemux.c:10139:16: error: ‘color_table_id’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
+	  if (color_table_id != 0) {
+	  ^
+	  qtdemux.c:10121:19: note: ‘color_table_id’ was declared here
+	  guint16 color_table_id;
+	  ^~~~~~~~~~~~~~
+
+2016-10-20 17:40:59 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Use a default interleave of 250ms for all codecs
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-10-19 14:33:33 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Use a default interleave when ProRes is used
+	  The ProRes guidelines suggest an interleave of 0.5s is common, but
+	  specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
+	  be used per chunk.
+	  It might also make sense to use similar numbers in general.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-10-19 14:25:28 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: Allow configuring the interleave size in bytes/time
+	  Previously we were switching from one chunk to another on every single
+	  buffer. This wastes some space in the headers and, depending on the
+	  software, might depend in more reads (e.g. if the software is reading
+	  multiple samples in one go if they're in the same chunk).
+	  The ProRes guidelines suggest an interleave of 0.5s is common, but
+	  specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
+	  be used per chunk. This will be handled in a follow-up commit.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-09-30 18:22:27 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Set compressor name, horizontal/vertical resolution and depth for ProRes
+	  This is also required by some software to handle ProRes files.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 18:05:38 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/qtdemux.c:
+	  qt: Add support for ProRes 4444 XQ
+	  And also 4444 in the muxer.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 17:58:37 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/qtdemux_types.c:
+	  qtmux: Write 'clap' atom for ProRes
+	  It's required for ProRes to work with other software.
+	  It is also in the MP4 standard, but inventing values here seems a bit
+	  tricky for the general case and it does not really give any extra
+	  information.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 09:55:58 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Read colorimetry information from colr atom if available
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772181
+
+2016-09-29 21:56:18 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Always write colr atom with the colorimetry information
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772181
+
+2016-09-29 18:16:18 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Fix writing of the 'fiel' extension atom
+	  This was also wrong for JPEG2000. Also write it for all MOV files and
+	  JPEG2000, not only for ProRes.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-29 17:40:23 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	  qtmux: Write 4 bytes of zeroes at the end of the sample description extensions
+	  This is working around some broken software.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-28 20:55:24 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	  atoms: 'pasp' atom is also part of MP4, write it always
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-07-11 19:30:12 +0300  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Write additional atoms for prores video
+	  These required atoms are: colorimetry, field information, spatial/temporal
+	  quality, and vendor.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2014-06-16 17:20:32 +0200  Stian Selnes <stian.selnes@gmail.com>
+
+	* gst/rtp/gstrtph263depay.c:
+	  rtph263depay: Don't drop mode b packets with picture start code
+	  Some buggy payloaders, e.g. rtph263pay, may use mode B for packets
+	  that starts with a picture (or GOB) start code although it's not
+	  allowed. Let's be nice and not drop these packets/frames.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773516
+
+2016-06-22 13:59:35 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtp/gstrtph263ppay.c:
+	* tests/check/elements/rtph263.c:
+	  rtph263ppay: Fix caps leak
+	  Fix leaking caps when downstream has not-fixed caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773515
+
+2016-10-26 16:42:19 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: Fix indentation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773514
+
+2016-10-18 11:35:58 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: Use GST_TRACE_OBJECT for logging bitstream parsing
+	  Bump the bitstream parsing to TRACE log level so it doesn't flood the
+	  output when trying to read the more useful DEBUG and LOG messages.
+	  Also use GST_DEBUG_OBJECT instead of GST_DEBUG in various places
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773514
+
+2016-10-18 11:09:10 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: Fix leak for B-fragments
+	  Altough commits 6a16be7, 64f9d08 and 0c7e3a8 fixed some issues they
+	  introduced others. This patch fixes the leak of one macroblock for every
+	  B fragment.
+	  Macroblock structures must not be freed immediately after finding the
+	  boundaries as they are stored and used later. However the inital dummy
+	  structure (used for finding the first boundary) must be freed.
+	  CID #1212156
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773512
+
+2016-10-20 13:14:13 +0200  Alejandro G. Castro <alex@igalia.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpbin: avoid generating errors when rtcp messages are empty and check the queue is not empty
+	  Add a check to verify all the output buffers were empty for the
+	  session in a timout and log an error.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773269
+
+2016-10-26 13:21:29 +0200  Alejandro G. Castro <alex@igalia.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  rtpbin: pipeline gets an EOS when any rtpsources byes
+	  Instead of sending EOS when a source byes we have to wait for
+	  all the sources to be gone, which means they already sent BYE and
+	  were removed from the session. We now handle the EOS in the rtcp
+	  loop checking the amount of sources in the session.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773218
+
+2016-10-21 17:31:00 +0000  Matt Staples <staples255@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Also handle redirect on PLAY
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772610
+
+2016-08-30 10:24:43 +0200  Petr Kulhavy <brain@jikos.cz>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: allow missing control attribute in case of a single stream
+	  Improve RFC2326 - chapter C.3 compatibility:
+	  In case just a single stream is specified in SDP and the control attribute
+	  is missing do not drop the stream but rather assume "a=control:*"
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770568
+
+2016-10-08 18:11:17 +0200  William Manley <will@williammanley.net>
+
+	* sys/v4l2/gstv4l2allocator.c:
+	  v4l2: Warn, don't assert if v4l gives us a buffer with a too large size
+	  I've seen problems where the `bytesused` field of `v4l2_buffer` would be
+	  a silly number causing the later call to:
+	  gst_memory_resize (group->mem[i], 0, group->planes[i].bytesused);
+	  to result in this error to be printed:
+	  (pulsevideo:11): GStreamer-CRITICAL **: gst_memory_resize: assertion 'size + mem->offset + offset <= mem->maxsize' failed
+	  besides causing who-knows what other problems.
+	  We make the assumption that this buffer has still been dequeued correctly
+	  so just clamp to a valid size so downstream elements won't end up in
+	  undefined behaviour.
+	  The invalid `v4l2_buffer` I saw from my capture device was:
+	  buffer = {
+	  index = 0,
+	  type = 1,
+	  bytesused = 534748928, // <- Invalid
+	  flags = 8260, // V4L2_BUF_FLAG_TIMESTAMP_MONOTONIC | V4L2_BUF_FLAG_ERROR | V4L2_BUF_FLAG_DONE
+	  field = 01330, // <- Invalid
+	  timestamp = {
+	  tv_sec = 0,
+	  tv_usec = 0
+	  },
+	  timecode = {
+	  type = 0,
+	  flags = 0,
+	  frames = 0 '\000',
+	  seconds = 0 '\000',
+	  minutes = 0 '\000',
+	  hours = 0 '\000',
+	  userbits = "\000\000\000"
+	  },
+	  sequence = 0,
+	  memory = 2,
+	  m = {
+	  offset = 3537219584,
+	  userptr = 140706665836544, // Could be nonsense, not sure
+	  planes = 0x7ff8d2d5b000,
+	  fd = -757747712
+	  },
+	  length = 2764800,
+	  reserved2 = 0,
+	  reserved = 0
+	  }
+	  This is from gdb with my own annotations added.
+	  This was with gst-plugins-good 1.8.1, a Magewell XI100DUSB-HDMI video
+	  capture device and kernel 3.13 using a dodgy HDMI cable which is great at
+	  breaking HDMI capture devices.  I'm using io-mode=userptr and have built
+	  gst-plugins-good without libv4l.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769765
+
+2016-10-20 20:41:07 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Use a better default value for the movie header timescale
+	  Take the maximum video timescale, or if no video track is present the
+	  previous value of 1800.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769041
+
+2016-10-20 20:07:19 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Be more clever with the default video track timescale
+	  Use the number of milliframes per second for integral and drop-frame
+	  framerates, as suggested by the QT file format specification and other
+	  places. We already did that for integral framerates before, but not for
+	  drop-frame framerates. This now keeps precision better.
+	  For all other framerates, check if it's close to a well-known framerate
+	  and use that instead.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769041
+
+2016-10-10 13:00:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: extract interlaced information from jpeg video
+	  This information is hidden in a small chunk of data.
+	  Format found at https://developer.apple.com/standards/qtff-2001.pdf,
+	  page 92, "Video Sample Description", under table 3.1.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767771
+
+2016-10-26 12:46:28 +0530  Jagadish <jagadishkamathk@gmail.com>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	  gdkpixbufoverlay: Fixing x and y offset computation
+	  While computing the x and y offsets, it's the video resolution and
+	  resized overlay resolution to be used instead of actual overlay image
+	  resoltuion. Due to this, the overlay image used to get wrongly overlayed
+	  in undesired location
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757292
+
+2016-11-01 18:09:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  meson: update version
+
+2016-10-24 16:56:31 +0000  Enrique Ocaña González <eocanha@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Use the tfdt decode time on byte streams when it's significantly different than the time in the last sample
+	  We consider there's a sifnificant difference when it's larger than on second
+	  or than half the duration of the last processed fragment in case the latter is
+	  larger.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754230
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.10.0 ===
+
+2016-11-01 17:57:44 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.10.0
+
+2016-11-01 17:47:31 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2016-11-01 17:41:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/el.po:
+	  po: Update translations
+
+2016-10-27 12:01:55 +0200  Tobias Schneider <tobias.schneider@voiceinterconnect.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: fix extra-controls leak
+	  Gst struct v4l2object->extra_controls is created if user sets appropriate
+	  option but it is not freed on destruction of v4l2object.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773580
+
+2016-10-31 18:00:07 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  Revert "souphttpsrc: reduce reading latency by using non-blocking read"
+	  This reverts commit 8816764112408766889c8b680a3af51115df4bf5.
+	  It causes issues with the timeouts, and causes connections to be closed
+	  without actual reason. Needs further investigation.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773509
+
+2016-10-31 09:00:49 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Don't try to add srcpad if we don't know valid caps yet
+	  Otherwise we'll run into an assertion on specially crafted files.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773643
+
+2016-10-27 11:23:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* meson.build:
+	  meson: Remove uselessly duplicated dep checks
+	  These checks are done inside the meson.build files for each plugin.
+
+2016-10-27 11:22:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* ext/meson.build:
+	  meson: dv plugin now works on MSVC
+	  Needs a Meson patch to filter out the useless -lpthread
+	  https://github.com/mesonbuild/meson/pull/962
+
+2016-10-27 14:03:48 +0200  Branko Subasic <branko@axis.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: allow resolutions above 4096
+	  Modify the caps string to allow width and height greater than 4096.
+	  There is no need to restrict it since the matroska format allows the
+	  width and height values to be up to eight bytes long.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773582
+
+2016-10-23 17:23:10 -0700  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Check for G_PLATFORM_WIN32 for presence of ipi_spec_dest
+	  G_OS_WIN32 is only set when not building with cygwin, but
+	  ipi_spec_dest is missing both with and without cygwin.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-26 08:51:40 +0200  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: reset read_position when reading fails
+	  souphttpsrc maintains two variables for the position:
+	  * 'request_position' is where we want to be
+	  * 'read_position' is where we are
+	  During Normal operations both are updated in sync when data arrives. A seek
+	  changes 'request_position' but not 'read_position'.
+	  When the two positions get out of sync, then a new request is send and the
+	  'Range' header is adjusted to the current 'request_position'.
+	  Without this patch, if reading fails, then the source is destroyed. This
+	  triggers a new request, but the range remains unchanged. As a result, the
+	  old range is used and old data will be read.
+	  Changing the 'read_position' to -1 makes it explicitly different from
+	  'request_position' and as a result the 'Range' header is updated correctly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773509
+
+2016-10-25 08:54:34 -0700  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* meson.build:
+	  meson: Don't depend on gstreamer-check-1.0 on windows
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-25 15:24:20 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: reset connection info to non-flushing when closing
+	  This solves a hanging mainloop in following scenario:
+	  * connect to source
+	  * network/server drops
+	  * pipeline set to NULL (and connection to flushing as part)
+	  * pipeline set to PAUSED/PLAYING (connection to non-flushing, but not recorded)
+	  * [connecting still not possible]
+	  * pipeline set to NULL => mainloop hangs (since no actual flushing is done)
+
+2016-10-26 14:32:48 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Only allow one video request pad
+	  The pacing of the overall muxing is controlled
+	  by the video GOPs arriving, so we can only handle
+	  1 video stream, and the request pad is named accordingly.
+	  Ignore a request for a 2nd video pad if there's already
+	  an active one.
+
+2016-10-26 11:59:32 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Take ownership of floating refs
+	  sink the floating ref when handed a muxer or sink to use so
+	  we clearly take ownership.
+
+2016-10-25 14:51:52 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Set child elements to NULL when removing.
+	  Make sure that elements are in the NULL state when removing.
+	  Fixes critical warnings when errors occur early on in starting up.
+
+2016-10-25 14:50:53 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Set pad template on request sink pads
+	  Ensure that the ghost pad returned as a request pad
+	  has the template that was requested
+
+2016-10-25 10:50:47 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* meson.build:
+	* tests/check/meson.build:
+	  Revert "meson: move gstreamer-check-1.0 dependency to tests/check"
+	  This reverts commit 46632694662b96fddb848a1f2091a215b28a2d35.
+	  Does not actually work. See:
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773114#c31
+
+2016-06-08 11:24:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Assume PTS is DTS when PTS is missing
+	  This fixes issue for encoders that only sets the DTS. We assume that
+	  there was no re-ordering when that happens.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762207
+
+2016-10-24 00:34:15 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/meson.build:
+	  meson: fix build outside of gst-all
+
+2016-10-21 00:42:54 -0700  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* sys/directsound/meson.build:
+	  meson: directsound: Add ole32 library dependency
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-21 00:42:18 -0700  Scott D Phillips <scott.d.phillips@intel.com>
+
+	* meson.build:
+	* tests/check/meson.build:
+	  meson: move gstreamer-check-1.0 dependency to tests/check
+	  https://bugzilla.gnome.org/show_bug.cgi?id=773114
+
+2016-10-20 22:08:14 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/videomixer.c:
+	  tests: videomixer: disable racy flush_start_flush_stop test
+	  It's been broken for years, and it's unlikely it will ever
+	  be fixed for collectpads/videomixer now that there's compositor
+	  which works fine. So let's disable it, since all it does
+	  is that it creates noise that distracts from other failures.
+	  Also see the corresponding adder bug as it failed in the same way:
+	  https://bugzilla.gnome.org/show_bug.cgi?id=708891
+
+2016-10-09 16:56:10 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: Fix souphttpsrc tests without CK_FORK=no
+	  It seems that the forked processes all attempt to handle the listening
+	  socket from the server, and only one has to shutdown the socket to break
+	  the server completely.
+	  Create a new server inside each test to avoid this.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772656
+
+2016-10-09 15:23:51 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+	* tests/check/elements/level.c:
+	  tests: Fix level test in CK_FORK=no mode
+	  The tests accumulate buffers in GstCheck's buffers list, and the list is
+	  not (consistently) reset between tests. Do that and remove the now
+	  conflicting unrefs for outbuffers.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772644
+
+2016-10-07 13:04:27 +0530  Gaurav Gupta <g.gupta@samsung.com>
+
+	* sys/waveform/gstwaveformsink.c:
+	  waveformsink: Fix Memory leak using GST_PTR_FORMAT
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772497
+
+2016-10-18 12:23:42 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/monoscope/meson.build:
+	  meson: Add missing gstaudio dep to monoscope
+	  In file included from ../subprojects/gst-plugins-good/gst/monoscope/gstmonoscope.c:42:0:
+	  ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
+	  #include <gst/audio/audio-enumtypes.h>
+	  ^
+	  compilation terminated.
+	  https://ci.gstreamer.net/job/GStreamer-master-meson/271/console
+
+2016-10-16 02:18:22 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/multifile/meson.build:
+	  meson: Add missing pbutils dependency to multifile
+	  Found via the Jenkins CI:
+	  FAILED: subprojects/gst-plugins-good/gst/multifile/gstmultifile@sha/gstsplitmuxsink.c.o
+	  [...]
+	  In file included from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.h:24:0,
+	  from ../subprojects/gst-plugins-good/gst/multifile/gstsplitmuxsink.c:59:
+	  ../subprojects/gst-plugins-base/gst-libs/gst/pbutils/pbutils.h:30:43: fatal error: gst/pbutils/pbutils-enumtypes.h: No such file or directory
+	  #include <gst/pbutils/pbutils-enumtypes.h>
+	  ^
+	  compilation terminated.
+	  https://ci.gstreamer.net/job/GStreamer-master-meson/263/console
+
+2016-10-15 22:11:08 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* meson.build:
+	  meson: Don't set c_std to gnu99
+	  Use the default for each compiler on every platform instead. This
+	  improves our compatibility with compilers that don't have gnu99 as
+	  a c_std.
+
+2016-10-04 18:04:11 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* meson.build:
+	* tests/check/getpluginsdir:
+	* tests/check/meson.build:
+	  meson: Make use of new environment object and set plugin path to builddir
+	  Workaround source_root being the root directory of all projects in the subproject
+	  case and remove now unneeded getpluginsdir
+	  Bump meson requirement to 0.35
+
+2016-10-06 11:15:54 +0530  Gaurav Gupta <g.gupta@samsung.com>
+
+	* tests/examples/rtp/client-rtpaux.c:
+	  tests: Fix memory leak in test rtpaux test
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772496
+
+2016-10-03 11:27:54 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: Forward latency queries to upstream
+	  Without this, latency queries to imagefreeze will fail.
+
+2016-09-30 11:35:39 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* hooks/pre-commit.hook:
+	* meson.build:
+	* tests/check/getpluginsdir:
+	  meson: Setup pre commit hook and fix getpluginsdir for standalone case
+
+2016-09-29 04:55:14 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Handle stop point from segment
+	  If the seek stop point (or start, during reverse play)
+	  was within the segment we just finished, go EOS immediately
+	  instead of proceeding through all other parts and sending
+	  0 length seeks to them.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-29 03:21:26 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Drop lock shutting down pads
+	  Avoid a sporadic deadlock on shutdown by dropping
+	  the splitmux lock around pad shutdown
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-29 02:47:36 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxsrc: Fix extra unref handling queries
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-29 04:50:25 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	* gst/multifile/gstsplitmuxpartreader.h:
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Avoid stall when parts get out of sync
+	  When one part moves ahead of the others - due to excessive
+	  downstream queueing, or really small input files - then
+	  we can end up activating parts more than once. That can lead to
+	  effects like shutting down pad tasks prematurely.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772138
+
+2016-09-30 11:41:19 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  meson: update version
+
+=== release 1.9.90 ===
+
+2016-09-30 13:02:19 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.9.90
+
+2016-09-30 12:17:26 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2016-09-30 11:43:54 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/el.po:
+	  po: Update translations
+
+2016-09-30 13:22:32 +0530  Arun Raghavan <arun@osg.samsung.com>
+
+	* tests/check/pipelines/tagschecking.c:
+	  tests: Fix tagschecking failure due to missing PTS
+	  qtmux now needs the PTS (commit a993883b7), so let's make sure we
+	  produce one with our buffers.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772228
+
+2016-09-28 23:03:58 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE
+	  Just error out if there is no valid PTS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=772143
+
+2016-09-29 17:37:28 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: Add JPEG2000 ihdr atom to the list of known ones
+	  Otherwise qtdemux is always going to complain about it being unknown.
+
+2016-09-29 10:19:56 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Always write the default frame duration for VP8/9 too
+	  The WebM spec allows this now, and it allows us to guess a framerate.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=772141 and
+	  also https://bugzilla.gnome.org/show_bug.cgi?id=654379
+
+2016-09-27 15:26:19 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph265depay.c:
+	  rtph26[45]depay: Don't handle NALs inside STAP units twice
+	  They've already been handled before pushing them into the adapter.
+
+2016-09-27 12:39:12 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/meson.build:
+	  meson: tests: fix vp8 availability checks
+	  Those variables are not defined if vp8 was not found.
+
+2016-09-27 10:23:38 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/multifile/gstmultifilesink.c:
+	  Revert "multifilesink: streamline the file-switch code a bit"
+	  This reverts commit f1ceaab02f3f557e23b77b14771a575788f92bb4.
+	  This broke atomic file writes in "buffer" mode. It did make
+	  sure that any streamheaders are prepended to each file in
+	  buffer mode as well, but that's not really needed in practice,
+	  whereas atomic file writes are, so let's restore the status
+	  quo ante for now since this was primarily a code cleanup anyway,
+	  and if anyone needs to streamheaders in buffer mode too they
+	  can make a patch to implement that differently. Re-implementing
+	  the atomic writes in the element also seems way too much work.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766990
+
+2016-09-27 10:22:57 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/multifile/gstmultifilesink.c:
+	  Revert "multifilesink: close file on write error with next-file mode is set to buffer"
+	  This reverts commit 84e441d2685cf223d348a95be0c5ba693bbf6624.
+	  This will no longer be needed once we revert f1ceaab02.
+
+2016-09-26 13:22:29 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* tests/check/meson.build:
+	  meson: Add gst-plugins-base plugins directories to be used by tests
+
+2016-09-26 14:30:00 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/vpx/meson.build:
+	* meson.build:
+	* tests/check/getpluginsdir:
+	* tests/check/meson.build:
+	  meson: add unit tests
+	  Only works properly in an installed setup currently, most
+	  likely won't work with a subprojects setup yet.
+
+2016-09-24 09:36:24 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	* po/meson.build:
+	  meson: hook up translations
+
+2016-09-08 17:30:41 +0530  Arun Raghavan <arun@arunraghavan.net>
+
+	* ext/pulse/pulsesrc.c:
+	  pulsesrc: Don't negotiate to less than two segments
+	  GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
+	  sure that if our buffer parameters are such that the maxlength is not at
+	  least 2x fragsize, we still request the ringbuffer to keep that much
+	  space so it continues to work.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770446
+
+2016-09-24 23:22:01 +0530  Arun Raghavan <arun@arunraghavan.net>
+
+	* gst/rtp/gstrtpsbcpay.c:
+	* gst/rtp/gstrtpsbcpay.h:
+	  rtpsbcpay: Fix timestamping
+	  We were just picking the timestamp of the last buffer pushed into our
+	  adapter before we had enough data to push out.
+	  This fixes things to figure out how large each frame is and what
+	  duration it covers, so we can set both the timestamp and duration
+	  correctly.
+	  Also adds some DISCONT handling.
+
+2016-07-12 18:14:52 +0200  Georg Lippitsch <glippitsch@toolsonair.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Fix fourcc for ProRes Proxy
+	  This is apco, according to
+	  https://wiki.multimedia.cx/index.php?title=Apple_ProRes
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-18 20:55:31 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/vpx/meson.build:
+	  meson: fix build with vpx 1.3.x
+	  vpx >= 1.4.0 is optional
+
+2016-09-15 18:19:35 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Use new bin suppressed flags API for managing the element flags
+
+2016-09-15 09:52:31 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/jack/gstjackaudioclient.c:
+	* gst/rtp/dboolhuff.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/videofilter/gstvideoflip.c:
+	  ext, gst: fix indentation
+
+2016-09-15 09:52:17 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/flvmux.c:
+	* tests/check/elements/rtph263.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	* tests/check/elements/rtpsession.c:
+	* tests/check/elements/rtpvp9.c:
+	  tests: fix indentation
+
+2016-08-11 11:04:22 -0600  Thomas Bluemel <tbluemel@control4.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
+	  Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one
+	  definitely lost packets is encountered.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769757
+
+2016-08-11 23:07:44 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: improved rtx-rtt averaging
+	  The basic idea is this:
+	  1. For *larger* rtx-rtt, weigh a new measurement as before
+	  2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
+	  3. For very large measurements, consider them "outliers"
+	  and count them a lot less
+	  The idea being that reducing the rtx-rtt is much more harmful then
+	  increasing it, since we don't want to be underestimating the rtt of the
+	  network, and when using this number to estimate the latency you need for
+	  you jitterbuffer, you would rather want it to be a bit larger then a bit
+	  smaller, potentially losing rtx-packets. The "outlier-detector" is there
+	  to prevent a single skewed measurement to affect the outcome too much.
+	  On wireless networks, these are surprisingly common.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-05 12:51:59 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
+	  Assuming equidistant packet spacing when that's not true leads to more
+	  loss than necessary in the case of reordering and jitter. Typically this
+	  is true for video where one frame often consists of multiple packets
+	  with the same rtp timestamp. In this case it's better to assume that the
+	  missing packets have the same timestamp as the last received packet, so
+	  that the scheduled lost timer does not time out too early causing the
+	  packets to be considered lost even though they may arrive in time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-27 10:39:50 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Don't request rtx if 'now' is past retry period
+	  There is no need to schedule another EXPECTED timer if we're already
+	  past the retry period. Under normal operation this won't happen, but if
+	  there are more timers than the jitterbuffer is able to process in
+	  real-time, scheduling more timers will just make the situation worse.
+	  Instead, consider this packet as lost and move on. This scenario can
+	  occur with high loss rate, low rtt and high configured latency.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-26 18:01:48 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix lost duration when gap after lost timer
+	  This patch fixes an issue with the estimated gap duration when there is
+	  a gap immediately after a lost timer has been processed. Previously
+	  there was a discrepancy beteen the gap in seqnum and gap in dts which
+	  would cause wrong calculated duration. The issue would only be seen with
+	  retranmission enabled since when it's disabled lost timers are only
+	  created when a packet is received and the actual gap length and last dts
+	  is known.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-19 01:11:58 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Expose rtx-deadline as a property
+	  The default -1 gives the old behavior.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-11 12:02:19 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Improved expected-timer handling when gap > 0
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-11 11:51:50 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Major improvements for RTX stats
+	  Stats should also be collected for unsuccessful packets.
+	  rtx-rtt is very important for determining the necessary configured
+	  latency on the jitterbuffer. It's especially important to be able to
+	  increase the latency when retransmitted packets arrive too late and are
+	  considered lost. This patch includes these late packets in the
+	  calculation of the various rtx stats, making them more correct and
+	  useful.
+	  Also in the case where the original packet arrives after a NACK is sent,
+	  the received RTX packet should update the stats since it provides useful
+	  information about RTT.
+	  The RTT is only updated if and only if all requested retranmissions are
+	  received. That way the RTT is guaranteed to make sense. If not we don't
+	  know which request the packet is a response to and the RTT may be bogus.
+	  A consequence of this patch is that RTT is not updated for a request
+	  when one of the RTX packets for that seqnum is lost, but that since
+	  measured RTT will be more accurate.
+	  The implementation store the RTX information from the timed out timers
+	  and use this when the retransmitted packet arrives. For performance
+	  these timers are stored separately from the "normal" timers in order to
+	  not impact performance (see attached performance test).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-08-11 11:02:44 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Add and expose more stats and increase testing of it
+	  Add num-pushed and num-lost.
+	  Expose num-late, num-duplicates and avg-jitter.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-07 10:20:02 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtxreceive: Set buffer flag for retransmitted packets
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-07-09 23:47:41 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Option to disable rtx-delay-reorder
+	  When disabled we can save some iterations over timers.
+	  There is probably an argument for rtx-delay-reorder to exist, but
+	  for normal operations, handling jitter (reordering) is something a
+	  jitterbuffer should do, and this variable feels like functionality that
+	  is not "in-sync" with what the jitterbuffer is trying to achieve.
+	  Example: You have 50ms jitter on your network, and are receiving
+	  audio packets with 10ms durations. An audio packet should not be
+	  considered late until its rtx-timeout has expired (and hence a rtx-event
+	  is sent), but with rtx-delay-reorder, events will be sent pretty much
+	  all the time due to the jitter on the network.
+	  Point being: The jitterbuffer should adapt its size to the measured network
+	  jitter, and then rtx-delay-reorder needs to adapt as well, or simply
+	  get out of the way and let the other (better) rtx-mechanisms do their job.
+	  Also change find_timer to only use seqnum as an argument, since there
+	  will only ever be one timer per seqnum at any given time. In the
+	  one case where the type matters, the caller simply checks the type.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769768
+
+2016-09-14 09:58:41 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: Fix double free from coverity
+	  CID #1372887
+
+2016-09-14 09:58:37 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: Indent as per gst-indent
+
+2016-09-14 11:30:41 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  configure: Depend on gstreamer 1.9.2.1
+
+2016-09-14 10:17:02 +0900  Wonchul Lee <wonchul.lee@collabora.com>
+
+	* gst/autodetect/gstautodetect.c:
+	  autodetect: Use gst_bin_set_suppressed_flags() API
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771395
+
+2016-09-09 15:36:12 +0200  Thomas Scheuermann <Thomas.Scheuermann@barco.com>
+
+	* ext/jack/gstjackaudioclient.c:
+	  jack: Fix pipeline hang when jack changes sample rate or buffer size
+	  If jackd changes the buffer size or sample rate, jackaudiosink hangs
+	  and can't be stopped. This also happens if jack is configured as slave
+	  and a gstreamer pipeline is started on the slave machine while the jack
+	  master isn't running yet. If the the jack master is started it changes
+	  the buffer size / sample rate and jackaudiosink can't be stopped.
+	  This fix calls jack_shutdown_cb when jack_sample_rate_cb or
+	  jack_buffer_size_cb is called.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771272
+
+2016-09-12 20:08:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Fix field ordering for reverse playback
+	  And actually calculate the field duration instead of a frame duration so
+	  that we can properly timestamp output frames in fields=all mode.
+	  This is probably still broken for reverse playback in telecine mode.
+
+2016-09-12 09:02:00 +0000  Thomas Klausner <tk@giga.or.at>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Fix compilation on NetBSD
+	  https://bugzilla.gnome.org/show_bug.cgi?id=771278
+
+2016-09-10 20:51:10 +1000  Jan Schmidt <jan@centricular.com>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From b18d820 to f980fd9
+
+2016-09-09 14:02:25 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: offset is irrelevant when no crypto info
+	  Cause later it will try to use the crypto info array to get an index and
+	  attach on of the positions as buffer's crypto info.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770951
+
+2016-09-10 09:53:57 +1000  Jan Schmidt <jan@centricular.com>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From f49c55e to b18d820
+
+2016-09-07 15:33:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/osxaudio/Makefile.am:
+	  osxaudio: Distribute device provider files
+	  Those where missing the the dev release tarballs for 1.9.2 which
+	  prevented building from tarball on OSX platform
+
+2016-09-06 09:49:39 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix crash with no cenc aux offset
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770951
+
+2016-09-05 09:39:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: parse a bit more of the humongous LOAS data
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:39:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: make it clear when a potential LOAS frame is not one
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:38:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: add a few comments to anchor parsing to the spec
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:37:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstaacparse.h:
+	  aacparse: improve channel/rate handling
+	  Keep track of the last parsed channels/rate fields so they can be
+	  used even if the element was not yet configured.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:35:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: fix varlength number reading as per spec
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-09-05 09:35:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: strip uneeded static arrays slack
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769278
+
+2016-07-18 19:18:58 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpmp4adepay.c:
+	* gst/rtp/gstrtpmp4adepay.h:
+	  rtpmp4adepay: Only declare a stream to be framed once a marker bit has been seen
+	  This may cause a few packets to be processed by the parser, but it's
+	  better than never pushing out buffers from a slightly broken stream
+	  where no marker bits are set.
+
+2016-09-06 14:25:42 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: Fix timestamping in reverse playback mode
+	  This is only supported right now if after a demuxer that supports reverse
+	  playback, e.g. with DV container inside AVI container.
+
+2016-09-05 12:23:54 -0300  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* meson.build:
+	  meson: Bump version to 1.9.2
+
+2015-06-26 20:13:17 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+	* gst/isomp4/GstQTMux.prs:
+	* gst/isomp4/Makefile.am:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Implement the preset interface.
+	  + And provide a "youtube" preset, which based on
+	  https://support.google.com/youtube/answer/1722171 sets
+	  faststart to True.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751559
+
+2016-09-01 12:27:35 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.9.2 ===
+
+2016-09-01 12:27:15 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.9.2
+
+2016-09-01 11:23:33 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  po: Update translations
+
+2016-09-01 10:59:51 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/examples/equalizer/demo.c:
+	* tests/examples/spectrum/demo-audiotest.c:
+	* tests/examples/spectrum/demo-osssrc.c:
+	  tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS
+	  We use gdk_cairo_create() which is deprecated since 3.22.
+
+2016-08-31 05:50:44 +1000  Jan Schmidt <jan@centricular.com>
+
+	* sys/osxvideo/Makefile.am:
+	* sys/osxvideo/cocoawindow.h:
+	* sys/osxvideo/osxvideosink.h:
+	  osxvideo: Remove QuickTime references.
+	  QuickTime.h is no longer available on OS X 10.12 (Sierra),
+	  and both the header and the framework seem unnecessary
+	  for compilation - at least as of 10.11 (El Capitan).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770526
+
+2016-08-19 11:11:03 -0700  Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+	* ext/dv/gstdvdemux.c:
+	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
+	* gst/avi/gstavidemux.c:
+	* gst/debugutils/rndbuffersize.c:
+	* gst/flv/gstflvdemux.c:
+	* gst/imagefreeze/gstimagefreeze.c:
+	* gst/isomp4/qtdemux.c:
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-parse.c:
+	* gst/multifile/gstsplitmuxsrc.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/wavparse/gstwavparse.c:
+	  Use the new API to post flow ERROR messages on the bus
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770158
+
+2016-08-26 21:32:07 +0200  Josep Torra <n770galaxy@gmail.com>
+
+	* tests/check/elements/.gitignore:
+	  gitignore: ignore qtdemux, rtph261 and rtpvp9 tests
+
+2016-08-26 21:22:16 +0200  Josep Torra <n770galaxy@gmail.com>
+
+	* tests/check/Makefile.am:
+	  tests: use GST_NET_LIBS instead of hardcoded -lgstnet
+	  Fixes build in OSX when running 'make check' in gst-uninstalled.
+
+2016-08-26 21:14:47 +0200  Josep Torra <n770galaxy@gmail.com>
+
+	* tests/check/elements/rtp-payloading.c:
+	  tests: remove a wrong 'const' specifier
+	  Fixes "error: duplicate 'const' declaration specifier"
+
+2016-08-26 21:11:59 +0200  Josep Torra <n770galaxy@gmail.com>
+
+	* configure.ac:
+	* tests/check/Makefile.am:
+	  build: silence error about pthread for 'make check' in osx
+	  Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2016-08-26 20:31:10 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/Makefile.am:
+	  vp9enc: Fix build of unit test by letting it link to libgstvideo
+
+2016-08-26 12:06:35 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpmux.h:
+	  Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems"
+	  This broke API, so we need a better solution!
+	  This reverts commit c7579d31a6e9d788e94b83258309063d0aae481e.
+
+2016-06-08 15:06:28 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtpvp9depay.c:
+	* tests/check/Makefile.am:
+	* tests/check/elements/rtpvp9.c:
+	  rtpvp9depay: Support flexible mode
+
+2016-06-06 17:03:36 +0200  Stian Selnes <stian@pexip.com>
+
+	* ext/vpx/gstvp9enc.c:
+	* tests/check/Makefile.am:
+	* tests/check/elements/vp9enc.c:
+	  vp9enc: Fix leak of vpx_image_t
+
+2016-05-06 13:33:22 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph263pdepay.c:
+	* tests/check/elements/rtph263.c:
+	  rtph263pdepay: Don't try to push empty frame
+	  If the result of depayloading is an empty frame, just drop it. This is
+	  likely the result of a buggy payloader.
+
+2016-05-06 16:06:53 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpmux.h:
+	  rtpmux: fix PROP_TIMESTAMP_OFFSET range problems
+	  It could not set the offset for the full guint32 range.
+
+2016-05-06 09:44:42 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: introduce max-streams property
+	  To be able to cap the number of allowed streams for one session.
+	  This is useful for preventing DoS attacks, where a sender can change
+	  SSRC for every buffer, effectively bringing rtpbin to a halt.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770292
+
+2016-03-31 00:10:49 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: reordered packets are very normal, and should not be a warning
+
+2016-02-05 14:19:25 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: degrade g_warning to GST_ERROR
+	  So we don't blow up while investigating
+
+2016-02-04 14:16:40 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph263pdepay.c:
+	* tests/check/elements/rtph263.c:
+	  rtph263pdepay: Fix picture header for non-writable payload
+	  Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
+	  the payload. In this case the payload modifications will not affect the
+	  rtp buffer. So instead of modifying the payload buffer directly we
+	  should modify the buffer that actually gets pushed on the adapter.
+
+2015-11-19 11:50:47 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph261depay.c:
+	* tests/check/Makefile.am:
+	* tests/check/elements/rtph261.c:
+	  rtph261depay: Fix check of valid payload length
+	  Packets with no H.261 payload should be dropped to avoid invalid
+	  write/reads.
+
+2015-11-09 10:06:21 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	* tests/check/elements/rtph263.c:
+	  rtph263pay: Fix double free, invalid reads and leak
+
+2014-06-30 15:43:58 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: sanity check RTT before ignoring PLI/FIR
+
+2014-06-30 15:07:45 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: handle sdes messages with non-utf8 more gracefully
+
+2014-06-17 08:52:50 +0200  Stian Selnes <stian.selnes@gmail.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: change log level on bitstream parsing messages
+
+2016-07-07 11:13:18 +0200  Mikhail Fludkov <misha@pexip.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests/rtprtx: refactor the tests to use gstharness
+	  The functionality of all the tests was kept exactly the same. Some tests
+	  were renamed:
+	  test_push_forward_seq -> test_rtxsend_rtxreceive
+	  test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
+	  test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
+	  test_rtxreceive_data_reconstruction was testing that retransmitted
+	  buffer produced by rtxsend was correctly transformed to the original
+	  buffer by rtxreceive. Now we are checking for this in all the tests
+	  where both rtxsend & rtxreceive are involved. That's why the test was
+	  removed.
+
+2016-08-25 15:52:36 +0200  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: Set RTP marker bit
+	  Set the RTP marker bit on the last RTP packet of an H.265 access unit.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770394
+
+2016-07-26 19:39:58 +0200  Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+	* gst/videofilter/gstvideoflip.c:
+	* gst/videofilter/gstvideoflip.h:
+	  videoflip: added GstVideoDirection interface
+	  It implements now this interface with its video-direction
+	  property. Values are changed to GstVideoOrientationMethod but they have
+	  the same value than the originals.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768687
+
+2015-11-06 10:39:16 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  gstrtpsession: refactor duplicate code into a function
+	  Less code, easier to read, more consistent.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770293
+
+2016-08-23 17:06:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: fix typo in max-misorder-time property name
+
+2016-08-22 00:05:52 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: fix printf format compiler warning in debug message
+	  On 32-bit x86: gstsplitmuxsink.c:966:31: warning: format ‘%u’ expects
+	  argument of type ‘unsigned int’, but argument 9 has type
+	  ‘guint64 {aka long long unsigned int}’
+
+2016-08-12 21:12:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* .gitignore:
+	* config.h.meson:
+	* ext/cairo/meson.build:
+	* ext/dv/meson.build:
+	* ext/flac/meson.build:
+	* ext/gdk_pixbuf/meson.build:
+	* ext/jack/meson.build:
+	* ext/jpeg/meson.build:
+	* ext/libpng/meson.build:
+	* ext/meson.build:
+	* ext/pulse/meson.build:
+	* ext/shout2/meson.build:
+	* ext/soup/meson.build:
+	* ext/speex/meson.build:
+	* ext/taglib/meson.build:
+	* ext/vpx/meson.build:
+	* ext/wavpack/meson.build:
+	* gst/alpha/meson.build:
+	* gst/apetag/meson.build:
+	* gst/audiofx/meson.build:
+	* gst/audioparsers/meson.build:
+	* gst/auparse/meson.build:
+	* gst/autodetect/meson.build:
+	* gst/avi/meson.build:
+	* gst/cutter/meson.build:
+	* gst/debugutils/meson.build:
+	* gst/deinterlace/meson.build:
+	* gst/dtmf/meson.build:
+	* gst/effectv/meson.build:
+	* gst/equalizer/meson.build:
+	* gst/flv/meson.build:
+	* gst/flx/meson.build:
+	* gst/goom/meson.build:
+	* gst/goom2k1/meson.build:
+	* gst/icydemux/meson.build:
+	* gst/id3demux/meson.build:
+	* gst/imagefreeze/meson.build:
+	* gst/interleave/meson.build:
+	* gst/isomp4/meson.build:
+	* gst/law/meson.build:
+	* gst/level/meson.build:
+	* gst/matroska/meson.build:
+	* gst/meson.build:
+	* gst/monoscope/meson.build:
+	* gst/multifile/meson.build:
+	* gst/multipart/meson.build:
+	* gst/replaygain/meson.build:
+	* gst/rtp/meson.build:
+	* gst/rtpmanager/meson.build:
+	* gst/rtsp/meson.build:
+	* gst/shapewipe/meson.build:
+	* gst/smpte/meson.build:
+	* gst/spectrum/meson.build:
+	* gst/udp/meson.build:
+	* gst/videobox/meson.build:
+	* gst/videocrop/meson.build:
+	* gst/videofilter/meson.build:
+	* gst/videomixer/meson.build:
+	* gst/wavenc/meson.build:
+	* gst/wavparse/meson.build:
+	* gst/y4m/meson.build:
+	* meson.build:
+	* meson_options.txt:
+	* sys/directsound/meson.build:
+	* sys/meson.build:
+	* sys/v4l2/meson.build:
+	* sys/ximage/meson.build:
+	* tests/check/meson.build:
+	* tests/meson.build:
+	  Add support for Meson as alternative/parallel build system
+	  https://github.com/mesonbuild/meson
+	  With contributions from:
+	  Tim-Philipp Müller <tim@centricular.com>
+	  Jussi Pakkanen <jpakkane@gmail.com> (original port)
+	  Highlights of the features provided are:
+	  * Faster builds on Linux (~40-50% faster)
+	  * The ability to build with MSVC on Windows
+	  * Generate Visual Studio project files
+	  * Generate XCode project files
+	  * Much faster builds on Windows (on-par with Linux)
+	  * Seriously fast configure and building on embedded
+	  ... and many more. For more details see:
+	  http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
+	  http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
+	  Building with Meson should work on both Linux and Windows, but may
+	  need a few more tweaks on other operating systems.
+
+2016-08-20 16:59:30 +0800  Jie Jiang <jiangjie@nudt.edu.cn>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  Fixed splitmuxsink 32-bit overflow bug
+	  Extend the byte tracking counters to 64-bit on
+	  all platforms, instead of using gsize, which overflows
+	  after 4GB.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=770019
+
+2016-08-19 17:18:16 +0300  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/isomp4/atoms.c:
+	  isomp4: Fix coverity warning
+	  If atom_copy_data fails to write anything, return 0
+	  CID #1371458
+
+2016-04-09 07:51:03 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	* sys/v4l2/v4l2-utils.c:
+	  v4l2: consistently check #ifdef HAVE_GUDEV instead of #if
+	  Both work with autotools but they definitely don't mean the same thing, cause
+	  problems with other build systems, and are bad form. Existence should always be
+	  checked with #ifdef or #if defined.
+
+2016-04-19 10:53:05 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	* sys/directsound/gstdirectsoundsink.h:
+	  directsound: port away from old DirectX API
+	  D3DX has been deprecated for the last 4 years and latest versions of
+	  Windows no longer ship headers for it. This is fine as long as you're
+	  building with Cerbero's Wine-based DirectX headers, but sucks if you
+	  want to build against the actual Windows SDK.
+	  We were just using it to get error strings anyway, so just use the
+	  generic error string API.
+
+2016-08-18 12:02:01 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  Revert "flacparse: Add maximum bitrate tag"
+	  This reverts commit c703ab69f526092bb26cce41ca691a896c8383d8.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769392
+
+2016-08-18 09:57:51 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations
+	  Need to set max-misorder-time and max-dropout-time to 0 so the
+	  jitterbuffer does not base them on packet rate calculations.
+	  If it does, out gap is big enough to be considered a new stream and
+	  we wait for a few consecutive packets just to be sure
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2016-08-09 12:55:59 +0300  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Add option to split at exactly max-size-time
+	  Will try to request a keyframe from the encoder to be sent at the target
+	  running time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769664
+
+2016-08-09 20:16:16 +0300  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Allow time and bytes to reach their respective thresholds
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769664
+
+2016-08-17 09:49:04 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Allow mimetypes with properties as long as they're application/sdp
+	  Some servers add properties like charset, e.g.
+	  application/sdp; charset=utf8
+	  Ideally we should also parse the charset and do conversion of all messages,
+	  but that's for a later time.
+
+2016-06-24 16:32:37 +0300  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: Added support for writing timecode track
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767950
+
+2016-08-11 16:32:21 -0600  Thomas Bluemel <tbluemel@control4.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: Initialize bytes_sent field.
+	  This fixes endpoints not receiving any data intermittently.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769773
+
+2016-08-10 11:45:13 -0600  Thomas Bluemel <tbluemel@control4.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpstats.c:
+	  rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2016-08-10 11:26:17 -0600  Thomas Bluemel <tbluemel@control4.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Don't warn for duplicate packets
+	  This is a normal scenario and should not be a warning.  This can
+	  happen frequently when re-transmits of lost packets are enabled.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762208
+
+2016-08-08 13:49:19 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmux: Fix typo converting to running time.
+	  Use the correct collected timestamp.
+
+2016-08-08 02:53:48 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  Revert "splitmuxsink: Use GstBin async-handling instead of our own."
+	  This reverts commit fa008f271a52f82dededc28bd81b020ca7939b47.
+	  async-handling in GstBin causes the pipeline to spin at 100%
+	  CPU as the top-level pipeline tries to change that state
+	  to PLAYING constantly. This is a workaround for a core
+	  problem, essentially, but an improvement in this case for now.
+
+2016-08-08 00:56:38 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmux: Recheck state after unlocking mutex.
+	  After dropping the splitmux lock, re-check the state,
+	  don't just fall through and sleep unconditionally,
+	  as we may have already missed the wakeup.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769514
+
+2016-08-03 03:32:07 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Don't stop and error on EOS flow return
+	  Don't immediately halt on EOS flow return from downstream
+	  due to out of segment. Let the demuxer handle it and send
+	  EOS.
+
+2016-08-04 00:36:28 -0300  Thiago Santos <thiagossantos@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: avoid unref of null buffer
+	  The current 'l' pointer will be NULL when the loop
+	  is interrupted with a 'break' statement. Need to have
+	  it advance to the next list item before interrupting.
+
+2016-08-02 14:01:14 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>
+
+	* gst/wavparse/Makefile.am:
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Add tags for container format and bitrate for uncompressed PCM
+	  The PCM bitrate is added to help downstream elements (like uridecodebin)
+	  figure out a proper network buffer size
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769390
+
+2016-08-01 18:52:26 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: Add maximum bitrate tag
+	  https://bugzilla.gnome.org/show_bug.cgi?id=769392
+
+2016-07-28 17:58:16 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: When receiving a DISCONT buffer that does not point to a sample, remember the offset
+	  And don't just reset everything. This makes sure that we can continue to
+	  handle data in the following scenario:
+	  moov: discont
+	  moof: discont
+	  mdat: continuous
+	  Previously this would fail because the offset would be the accumulated offset
+	  from moov and moof at the mdat position, while the buffer offset might be
+	  something completely different.
+
+2016-07-25 13:34:02 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpbvpay.c:
+	* gst/rtp/gstrtpceltpay.c:
+	* gst/rtp/gstrtpg722pay.c:
+	* gst/rtp/gstrtph263ppay.c:
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtpilbcpay.c:
+	  rtp: Filter with the filter caps in the payloader's getcaps
+
+2016-03-03 11:35:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: include http-status-code in error message details
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763038
+
+2016-07-25 18:20:03 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Fix debug statement signedness.
+	  The ts variable is a GstClockTime, don't print it
+	  as a GstClockTimeDiff.
+
+2016-07-17 22:41:02 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Handle negative running time
+	  Use signed clock times for running time everywhere
+	  so that we handle negative running times without
+	  going haywire, similar to what queue and multiqueue
+	  do these days.
+
+2016-07-18 00:12:55 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Drop lock when sending dummy event
+	  When pushing the dummy event into the multiqueue,
+	  drop the splitmux lock or else we might deadlock.
+
+2016-06-30 01:56:41 +1000  Jan Schmidt <thaytan@noraisin.net>
+
+	* gst/rtp/gstrtph264pay.c:
+	  rtph264pay: Intersect with filter caps in getcaps function.
+	  Always intersect with the filter caps in the getcaps function
+	  to make sure we return a subset of what was requested.
+	  Other payloaders also have this problem and need fixing
+	  in future commits.
+
+2016-07-12 17:30:56 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/qtdemux.c:
+	  tests: qtdemux: fix element and pad leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768739
+
+2016-07-12 16:45:36 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/audiofirfilter.c:
+	* tests/check/elements/audioiirfilter.c:
+	* tests/check/elements/rtp-payloading.c:
+	* tests/check/elements/videobox.c:
+	* tests/check/pipelines/effectv.c:
+	  tests: fix bus leaks
+	  gst_bus_add_signal_watch() takes a ref on the bus which should be
+	  released using gst_bus_remove_signal_watch().
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768739
+
+2016-07-14 03:07:11 +0800  Ting-Wei Lan <lantw@src.gnome.org>
+
+	* configure.ac:
+	  configure: Call AG_GST_PKG_CONFIG_PATH to set GST_PKG_CONFIG_PATH
+	  GST_PKG_CONFIG_PATH is used in docs/plugins directory, so
+	  AG_GST_PKG_CONFIG_PATH must be called to set it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768787
+
+2016-07-12 07:39:58 +0200  Edward Hervey <edward@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Don't drop final bytes of a range request
+	  At the end of a range request, we don't want to return GST_FLOW_EOS otherwise
+	  the last bytes we just read will be dropped by basesrc.
+	  Instead just return GST_FLOW_OK (which was set just before) and let basesrc
+	  handle the fact we are at the end of the segment.
+
+2016-07-11 18:30:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	  v4l2provider: Fix device type detection
+	  The type detection would lead to assertion as it would try
+	  to create a device without having found any type for it. It
+	  also didn't detect MPLANE devices properly.
+
+2016-07-11 18:29:01 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't assert when used by the monitor
+	  The monitor sets the object->element object as a GstObject. This
+	  works for debug traces, but will assert for ELEMENT_ERROR. This
+	  was the only case where that could happen. Add a check for that.
+
+2016-07-11 17:38:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Indent very long line
+
+2016-07-12 00:42:02 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: At the end of a range request, read another time to finalize the request
+	  If we're at the end of a range request, read again to let libsoup
+	  finalize the request. This allows to reuse the connection again later,
+	  otherwise we would have to cancel the message and close the connection.
+
+2016-07-11 21:13:47 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From f363b32 to f49c55e
+
+2016-07-11 19:57:18 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Fix keep-alive handling
+	  We have to get rid of the message on EOS when the complete stream is read to
+	  remember that we successfully finished handling this specific message.
+	  Otherwise we will cancel it later and close the connection instead of reusing
+	  it at a later time.
+	  It might also make sense to reuse connections if a non-200 response is
+	  received. As long as there was no connection error, the HTTP connection should
+	  be re-usable.
+
+2016-07-11 12:05:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	  Also enable V4L2 probe on aarch64 (aka ARM 64bit)
+
+2016-07-11 11:59:19 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* tests/examples/rtp/client-PCMA.c:
+	  rtp example: Fix leak
+	  Also stop fetching the internal source as this
+	  functionality has been broken.
+
+2016-07-08 14:58:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	  Enable v4l2 probe on Linux/ARM
+	  Most of those have V4L2 drivers these days enabling it make sure that it
+	  this code is enabled in major distribution, hence that HW accelerated
+	  decoder/encoder can be used on platforms that support it. The probes are
+	  slightly increasing the first init of gstreamer library, though the
+	  result is cached in the registry for later use.
+
+2016-07-11 09:46:49 +0200  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	* tests/check/elements/rtp-payloading.c:
+	  rtph265pay: Accept array_completeness=1
+	  When parsing NAL unit type in codec_data, check the 6bits of
+	  NAL_unit_type only and do not require the array_completeness bit to be
+	  0, since the default and mandatory value of array_completeness is 1 for
+	  hvc1.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768653
+
+2016-07-10 21:35:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Also copy device_caps in gst_v4l2_dup
+	  This fixes regression where M2M error out saying they have no output
+	  format (the V4L2 CAPTURE side).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-10 21:30:27 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Use correct in6_pktinfo struct instead of in_pktinfo
+	  Fixes the build on FreeBSD, which does not have the latter.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768623
+
+2016-07-08 17:28:19 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: fix multiplanar capture
+	  After switching to using V4L2_CAP_DEVICE_CAPS we lost support for
+	  multiplanar device types. After some research, it looks like
+	  vcap.capabilities treated the multiplanar flag of output and capture
+	  devices equally, but not the new device_caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-08 14:56:30 +0200  Mats Lindestam <matslm@axis.com>
+
+	* gst/multipart/multipartmux.c:
+	* gst/multipart/multipartmux.h:
+	  multipartmux: Use PTS and DTS instead of timestamp
+	  And pass-through both of them.
+	  Based on a patch by Göran Jönsson <goranjn@axis.com>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767900
+
+2016-06-30 14:40:40 +0200  Thomas Scheuermann <Thomas.Scheuermann@barco.com>
+
+	* ext/jack/gstjackaudioclient.c:
+	  jack: don't wait for callbacks if the jack server shut down
+	  Otherwise we'll wait forever.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747275
+
+2016-06-23 15:30:19 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Let upstream events go through upstream
+	  There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+	  Some elements might want to have that information.
+
+2016-06-23 15:22:56 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Let upstream events go through upstream
+	  There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+	  Some elements might want to have that information.
+
+2016-06-23 15:17:36 +0200  Edward Hervey <edward@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: Let upstream events go through upstream
+	  There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+	  Some elements might want to have that information.
+	  Also remove downstream-only CAPS event handling and minimize code
+
+2016-07-07 23:53:54 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2.c:
+	  v4l2: fix v4l2 probe build error
+	  A typo in gst_v4l2_probe_and_register() caused a build error when building
+	  with --enable-v4l2-probe. Fixing it.
+	  gstv4l2.c: In function 'gst_v4l2_probe_and_register':
+	  gstv4l2.c:150:25: error: 'struct v4l2_capability' has no member named 'capabilitites'
+	  device_caps = vcap.capabilitites;
+
+2016-07-01 22:53:33 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: use gst_caps_intersect_full in negotiate()
+	  Instead of reimplementing the GST_CAPS_INTERSECT_FIRST
+	  interection mode.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-02 01:56:07 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2radio.c:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: use opened device caps instead of physical device ones
+	  The same physical device can export multiple devices. In
+	  this case, the capabilities field now contains a union of
+	  all caps available from all exported V4L2 devices alongside
+	  a V4L2_CAP_DEVICE_CAPS flag that should be used to decide
+	  what capabilities to consider. In our case, we need the
+	  ones from the exported device we are using.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-07 18:24:59 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Remove suspicious checks for pads being active and linked
+	  We should add all pads, no matter if they are linked or active or not at this
+	  point. Skipping some that are not will cause different behaviour than with
+	  other muxers.
+
+2016-07-07 18:23:07 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Error out if we start writing data with some pads not having a codec id yet
+	  This can only happen if a) upstream somehow gets around the CAPS event failing
+	  or b) there never being any CAPS event.
+	  The following code assumes that all pads have a codec-id.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768509
+
+2016-07-07 18:14:43 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Consistently use gst_matroska_mux_set_codec_id() for setting the codec id
+
+2016-07-04 09:50:11 +0200  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtph265pay.h:
+	* tests/check/elements/rtp-payloading.c:
+	  rtph265pay/depay: Sync against RFC 7798
+	  Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
+	  sprop-parameter-sets.
+	  rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
+	  handles profile-id, tier-flag and level-id in caps query.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-07-06 09:25:00 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+	* gst/flv/gstflvdemux.c:
+	* gst/flv/gstflvdemux.h:
+	  flvdemux: Push nominal bitrate tags
+	  Add per-stream tag lists, which are used to send nominal
+	  bitrate tags. When remuxing FLV => FLV, this now passes
+	  through the upstream bitrate.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768440
+
+2016-07-06 09:24:49 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+	* gst/flv/gstflvdemux.c:
+	* gst/flv/gstflvdemux.h:
+	  flvdemux: Refactor metadata tag handling
+	  The FLV header cannot be trusted to indicate video or
+	  audio presence, as the comments already mention. Don't
+	  delay pushing tags waiting for streams that might never
+	  appear.
+	  Tags are now pushed immediately after they change:
+	  - After parsing an onMetaData script object
+	  - After negotiating caps on a pad
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768440
+
+2016-07-06 12:44:10 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix AAC codec_data values
+	  As seen in the parent switch for object_type_id, the 4 possible values are
+	  0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values.
+	  Looks like it was a typo making them decimal instead of hexadecimal.
+	  CID 1363328
+
+2016-07-06 13:51:03 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:06:44 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.9.1
+
+2016-07-06 11:46:26 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2016-07-06 11:22:53 +0300  Steven Hoving <sh@bigbrother.nl>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix error messages to first convert to doubles before division
+
+2016-07-06 10:18:30 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/da.po:
+	* po/hr.po:
+	* po/pt_BR.po:
+	* po/sk.po:
+	  po: Update translations
+
+2016-07-05 21:11:35 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
+	  There's a small window for a race condition otherwise.
+
+2016-07-04 17:45:40 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/qtmux.c:
+	  qtmux: Use complete AAC caps with codec_data in the tests
+
+2016-07-04 16:58:38 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: Reject raw AAC if no codec_data is found in the caps
+	  If necessary, a demuxer will have to invent something here but this is only a
+	  problem with non-conformant files anyway.
+
+2016-07-04 16:55:32 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Invent AAC codec_data if none is present
+	  Without, raw AAC can't be handled and we have some information available in
+	  the decoder that most likely allows us to decode the stream in one way or
+	  another. This is the same code already used by matroskademux for the same
+	  reasons, and ffmpeg/vlc play such files just fine too by guesswork.
+
+2016-07-04 14:54:13 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Reject raw AAC caps without codec_data
+	  The resulting file is not going to be playable without guesswork and raw caps
+	  should always have codec_data.
+
+2016-05-10 15:48:49 +0200  Edward Hervey <edward@centricular.com>
+
+	  qtdemux: Handle upstream GAP in push-mode/time segment
+	  This is to handle cases where upstream handles the fragmented streaming in TIME
+	  segments and sends us data with gaps within fragments. This would happen when dealing
+	  with trick-modes.
+	  When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
+	  it must obey the following rules:
+	  * The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
+	  * The buffers containing the first sample after a gap:
+	  * MUST start at the beginning of a sample,
+	  * MUST have the DISCONT flag set,
+	  * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767354
+
+2016-07-01 11:54:57 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/v4l2/v4l2-utils.c:
+	  v4l2: fix potential double-free of error debug string
+	  gst_v4l2_clear_error() doesn't work like g_clear_error(), it
+	  doesn't NULLify the pointer, so set freed debug string to NULL
+	  so it doesn't get freed again if gst_v4l2_clear_error() is
+	  called twice on the error.
+	  CID 1362901
+
+2016-07-01 10:05:00 +0000  Brad Lackey <blackey@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Don't disable UDP protocols on redirecting
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768232
+
+2016-07-01 17:28:17 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Push caps only when it was updated
+	  Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps
+	  event per moof without consideration of duplication.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768268
+
+2016-06-30 15:01:46 +0200  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: fix invalid memory access
+	  10 bytes was allocated for stream_format but size of "byte-stream" is
+	  more. Use g_strdup() instead.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-29 23:31:20 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2: Use a non-timer GstPoll
+	  Otherwise set_flushing() will have undefined semantics and nowadays causes a
+	  g_critical() to warn about that.
+
+2016-06-19 02:08:25 -0300  Thiago Santos <thiagossantos@gmail.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: dynamically adjust blocksize
+	  Update the blocksize depending on how much is obtained from a read
+	  of the input stream. This avoids doing too many reads in small chunks
+	  when larger amounts of data are available and also prevents using
+	  a very large memory area to read a small chunk of data.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767833
+
+2016-06-28 16:44:50 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo
+
+2016-06-28 15:15:14 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS
+
+2016-06-28 15:08:04 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Move #includes around to a) work around broken glibc header and b) Windows
+
+2016-06-28 14:25:03 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Fix compilation on Windows and *BSD/OSX
+
+2016-06-23 20:21:59 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Filter out multicast packets that are not for our multicast address
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767980
+
+2016-06-28 10:57:27 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
+	  If we consider the RTSP state, what can happen is that it is PLAYING but the
+	  element already asynchronously tried to PAUSE and it just did not happen yet.
+	  We would then override this setting to PAUSED (while the element actually is
+	  in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
+	  to produce packets while the sinks are all PAUSED, piling up thousands of
+	  packets in the rtpjitterbuffer and other elements and finally failing.
+
+2016-06-27 09:20:35 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: Add comment about H263/MPEG4P2 being non-standard for FLV
+	  They are however supported by ffmpeg and apparently used out there.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768006
+
+2016-06-24 14:48:53 +0300  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: Add support for H263 and MPEG4 part2
+	  https://bugzilla.gnome.org/show_bug.cgi?id=768006
+
+2016-06-21 17:10:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	  Update plugins doc
+	  This is partly automated using "make update" in docs/plugins, but also
+	  required manual merge. Additionally, missing plugins and elements have
+	  been added.
+
+2016-06-21 17:51:38 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/splitmux.c:
+	  tests: splitmux: skip tests if theora or ogg plugins are not available
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767861
+
+2016-06-21 11:46:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From ac2f647 to f363b32
+
+2016-06-21 07:40:42 -0400  Aaron Boxer <boxerab@gmail.com>
+
+	* gst/rtp/gstrtpj2kpay.c:
+	  gstrtpj2kpay: use tile bit and tile number to determine if there are multiple tiles in packet
+	  Now we don't have to rely on a special value for the tile number.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767817
+
+2016-06-21 09:34:56 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtpj2kpay: fix compiler warning on OS/X
+	  gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767817
+
+2016-06-21 09:34:37 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/gst-plugins-good-plugins.prerequisites:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	  docs: update
+
+2016-05-16 17:31:58 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/capssetter.c:
+	* tests/check/elements/icydemux.c:
+	* tests/check/elements/jpegenc.c:
+	* tests/check/elements/level.c:
+	* tests/check/elements/multifile.c:
+	* tests/check/elements/qtmux.c:
+	* tests/check/elements/rtprtx.c:
+	* tests/check/elements/udpsrc.c:
+	  fix buffer leaks in tests
+	  Need to call gst_check_drop_buffers() to release the buffers exchanged
+	  during the test.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-17 12:52:43 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/interleave.c:
+	  interleave: fix message leaks in test
+	  Flush the bus when cleaning up so pending messages are destroyed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-17 12:58:06 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/videomixer.c:
+	  videomixer: fix event leaks in test
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-13 15:12:22 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/deinterleave.c:
+	  deinterleave: fix leaks
+	  - Flush the bus so messages aren't leaked
+	  - Fix pad leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-06-17 15:29:16 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	  rtph264pay: Deprecated sprop-parameter-set property
+	  This is supposed to be either in the codec_data (avc stream format) or inside
+	  the stream, and we extract it from there. It should not be set from a
+	  property as it's stream specific.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767789
+
+2016-06-17 12:16:32 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: make all srtp encoder properties explicit
+	  The Session Data Protocol doesn't allow specifying a cipher for the
+	  SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
+	  "aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
+	  an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
+	  cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767799
+
+2016-06-17 19:59:13 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/soup/gstsoup.c:
+	  soup: work around frequent deadlocks in GLib type initialisation
+	  .. by registering the types from the plugin init function. This
+	  seems to help, but we'll see if it's enough (might need similar
+	  things elsewhere).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=693911
+	  https://bugzilla.gnome.org/show_bug.cgi?id=674885
+
+2016-06-17 16:08:08 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: The prores variant is stored in the variant field, not format
+	  And the caps in the sink pad template already used variant (only).
+
+2016-06-17 13:00:48 +0200  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtph265pay.h:
+	  rtph265pay: Remove sprop-parameter-sets property
+	  There is no valid use case when this property is needed since the values
+	  must be in either codec_data or buffer data.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-10 16:17:26 +0200  Jonas Holmberg <jonashg@axis.com>
+
+	* docs/plugins/scanobj-build.stamp:
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: Read NALU type the same way everywhere
+	  Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the
+	  same way as in other places.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-17 13:58:33 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  rtpjitterbuffer: fix RTPJitterBufferMode documentation
+	  Documentation lacks '@' before each enum values and there was an extra
+	  line after symbol section which confuses GTK-Doc parser.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767788
+
+2016-05-23 10:18:48 +0200  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: take the lock when changing stats
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-06-15 11:19:43 +0200  Jürgen Slowack <jurgen.slowack@barco.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection
+	  Fixes sps/pps/vps insertion via the config-interval property.
+	  https://bugzilla.gnome.org//show_bug.cgi?id=767680
+
+2016-06-11 12:16:03 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/pipelines/simple-launch-lines.c:
+	  simple-launch-lines: Use correct JPEG2000 caps
+
+2016-06-10 13:43:09 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: fix indentation
+
+2016-06-10 13:42:01 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: fix date parsing when there are trailing spaces
+	  Fixes parsing of "Thu May 11 15:57:46 2006 ".
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767496
+
+2016-05-13 15:08:24 -0400  Aaron Boxer <boxerab@gmail.com>
+
+	* gst/rtp/gstrtpj2kcommon.h:
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kpay.c:
+	  gstrtpj2k: set sampling field required by RFC
+	  This field is now required in the sink caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766236
+
+2016-06-09 09:30:48 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: Fix unref assertion failure
+	  Fix unref assertion failure
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767424
+
+2016-05-14 14:46:17 +0200  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Work with non-TIME segments
+	  With non-time segments, it now assumes that the arrival time of packets
+	  is not relevant and that only the RTP timestamp matter and it produces
+	  an output segment start at running time 0.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766438
+
+2016-06-07 20:53:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/libpng/gstpngdec.c:
+	  pngdec: Wait for segment event before checking it
+	  The heuristic to choose between packetise or not was changed to use the
+	  segment format. The problem is that this change is reading the segment
+	  during the caps event handling. The segment event will only be sent
+	  after. That prevented the decoder to go in packetize mode, and avoid
+	  useless parsing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2016-06-06 17:00:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: Wait for segment event before checking it
+	  The heuristic to choose between packetise or not was change to use the
+	  segment format. The problem is that this change is reading the segment
+	  during the caps event handling. The segment event will only be sent
+	  after. That prevented the decoder to go in packetize mode, and avoid
+	  useless parsing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2016-06-07 16:42:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: Keep part of the input buffer
+	  Instead of completely getting rid of the input buffer, copy
+	  the metadata, the flags and the timestamp into an empty buffer.
+	  This way the decoder base class can copy that information again
+	  to the output buffer.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758424
+
+2016-06-07 16:41:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: Coding style fixes
+
+2016-06-07 16:09:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Coding style fixes
+
+2016-06-07 16:04:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2transform.c:
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2: Add an error return to _try/_set_format
+	  This way one can easily ignore errors. Previously, error were always
+	  posted ont he bus.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766172
+
+2016-06-07 16:01:55 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/v4l2-utils.c:
+	* sys/v4l2/v4l2-utils.h:
+	  v4l2-util: Introduce GstV4l2Error
+	  This is to allow returning an error that can easily be sent as
+	  message to the application if the element needs it. Using this
+	  also allow ignoring errors.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766172
+
+2016-06-07 12:41:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: Avoid decide allocation on active pool
+	  v4l2src will renegotiate only if the format have changed. As of now,
+	  it's not possible to change the allocationw without resetting the
+	  camera. To avoid unwanted side effect, simply keep the old allocation
+	  if no renegotiation is taking place. This fixes assertion and possible
+	  failures in USERPTR or DMABUF import mode (when using downstream pools).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754042
+
+2016-04-28 13:44:49 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: Show state name in debugging
+	  Makes it easier to trace what's going on
+
+2016-05-10 15:45:42 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Remove useless variable
+	  That variable is only needed for a debug statement, move it there
+
+2016-05-10 15:10:36 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: Add/Fix comments on the various structure variables
+	  No variables were added/removed. This was just a good excuse to:
+	  * Comment what most variables are used for (and when)
+	  * Order them in such a way as to show first the common variables used
+	  in all cases, followed by those only used in push-mode
+
+2016-05-10 15:07:40 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Remove unused structure
+	  Let's just remove it, been commented for 7+ years :)
+
+2015-09-02 11:48:29 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: use decoder stop command instead of queueing empty buffers
+	  Only if the decoder stop command fails, keep queueing empty buffers to
+	  signal end of stream as before.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733864
+
+2014-12-12 14:31:36 +0100  Peter Seiderer <ps.report@gmx.net>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: add gst_v4l2_decoder_cmd helper
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733864
+
+2016-06-01 20:28:39 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Forward segments directly if we are operating in PUSH mode on fragmented streams
+	  We shouldn't go through segment activation as we will only have a limited
+	  understanding of how the whole stream timeline looks like from the moof. We
+	  only know about the current fragment, while upstream knows about the whole
+	  stream.
+	  This fixes seeking in DASH streams, both for seeks after the current moof and
+	  for seeks into the current moof. The former would fail because the moof ends
+	  and we can't activate any segment, the latter would cause a segment that stops
+	  at the moof end, and no further fragments would be played because we end up
+	  being EOS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-06-06 17:54:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2transform.c:
+	  v4l2transform: Use looser caps for upstream
+	  When we fixate for upstream, try to not introduce new fields when not
+	  needed. This was imported from videoconvert element.
+
+2015-01-28 12:07:58 +0100  Enrico Jorns <ejo@pengutronix.de>
+
+	* sys/v4l2/gstv4l2transform.c:
+	  gstv4l2transform: format fixation for preferring passthrough
+	  * If outgoing format is unfixated, try to set it to input format.
+	  * Call gst_caps_fixate () at end of fixation routine
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766719
+
+2016-05-20 12:49:53 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2transform.c:
+	  v4l2transform: allow to change pixel aspect ratio
+	  Scalers may change width and height independently,
+	  allow to change pixel aspect ratio.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766712
+
+2016-05-20 12:32:25 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2transform.c:
+	  v4l2transform: fix scaling in case of fixed pixel aspect ratio
+	  To change pixel aspect ratio from DAR to PAR, the necessary scaling factor
+	  is DAR/PAR, not DAR*PAR.
+	  For good measure, add debug output similar to the fixed-width and
+	  fixed-height cases.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766711
+
+2016-05-13 16:39:25 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: fill colorimetry in gst_v4l2_object_acquire_format
+	  Instead of relying on the default colorimetry chosen by
+	  gst_video_info_set_format(), set info.colorimetry from the
+	  values returned by G_FMT. This allows decoders to propagate
+	  their input colorimetry downstream.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766383
+
+2016-05-18 10:17:12 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: refactor gst_v4l2_object_get_colorspace to take a v4l2_format parameter
+	  Move the extraction of colorimetry parameters from struct v4l2_format and the
+	  setting of the identity matrix for RGB formats into the function to avoid code
+	  duplication.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766383
+
+2016-05-13 14:58:41 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: use visible size, not coded size, for downstream negotiation filter
+	  gst_v4l2_probe_caps() returns the coded size, not the visible size. Subtract
+	  the known padding from probed caps with the coded size before using them as
+	  filter for caps negotiation with downstream elements.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766382
+
+2016-05-13 14:45:02 +0200  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: use G_SELECTION instead of G_CROP in gst_v4l2_object_acquire_format
+	  The gst_v4l2_object_acquire_format() function is used by v4l2videodec to obtain
+	  the currently set capture format. Since G_FMT returns the coded size, the
+	  visible size needs to be obtained from the compose rectangle in order to
+	  negotiate it with downstream elements. The G_CROP call hasn't worked on mem2mem
+	  capture queues for a long time. Instead use the G_SELECTION call to obtain the
+	  compose rectangle and only fall back to G_CROP for ancient kernels.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766381
+
+2016-01-27 09:57:38 +0100  Andreas Naumann <anaumann@ultratronik.de>
+
+	* sys/v4l2/gstv4l2sink.c:
+	  v4l2sink: Use V4L2_BUF_TYPE_VIDEO_OUTPUT_OVERLAY if driver advertises it.
+	  On modern kernels, the G/S_FMT ioctls will always fail using
+	  V4L2_BUF_TYPE_VIDEO_OVERLAY with VFL_DIR_TX (e.g. real overlay out drivers)
+	  since this is not the intented use (rather rx, according to v4l2 API doc).
+	  Probably this is why the Video Output Overlay interface was created, so if
+	  the driver advertises it we might as well use.
+	  For old kernels (pre 2012) the old way might still work so keeping this for
+	  compatibility.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761165
+
+2016-06-06 18:52:01 +0100  Kieran Bingham <kieran@bingham.xyz>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Use non-deprecated V4L2 type for RGB15
+	  Support for the updated V4L2_PIX_FMT_XRGB555 was added in commit
+	  2538fee2fd8fdb74b05f0a511281bc4707e7cc44 however, when setting the format
+	  for use in v4l2 ioctls, the old deprecated format is still used. Convert
+	  this to the new accepted format type, as the preferred format.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767300
+
+2016-05-04 14:50:32 +0200  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: preserve seek flags
+	  Without this some flags get lost in streaming mode.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767194
+
+2016-06-06 10:47:52 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/Makefile.am:
+	* ext/soup/gstsouphttpclientsink.c:
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  Revert "WIP revert soup"
+	  This reverts commit fdac3a7a231f3848665636cf8122f96103b46e3b.
+	  Was not supposed to be pushed but a local workaround for
+	  https://bugzilla.gnome.org/show_bug.cgi?id=693911#c13
+
+2016-06-03 13:09:35 +0200  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: complete warn log with SSRC
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767195
+
+2016-05-31 15:29:13 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/Makefile.am:
+	* ext/soup/gstsouphttpclientsink.c:
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  WIP revert soup
+
+2016-06-03 13:18:31 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: Unref seek event in any case
+	  It would be leaked if no seek handler was currently set.
+
+2016-06-03 10:49:17 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	* ext/dv/gstdvdemux.h:
+	  dvdemux: Properly set event/message sequence numbers based on the previous seek
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-03 10:36:32 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	* ext/dv/gstdvdemux.h:
+	  dvdemux: Remember if upstream had a time segment and if not properly create time segments
+	  Previously the segment.time was wrong, and the position was not updated
+	  correctly, resulting in seeks in PUSH mode with upstream providing a BYTES
+	  segment to not work at all.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-03 09:54:53 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: Implement SEEKING query so we can actually seek if upstream can't seek in TIME
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 14:19:15 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: Recalculate the frame offsets at the beginning of each BYTE segment and whenever upstream gives us a timestamp
+	  This fixes seeking in DV streams where upstream operates in PUSH mode with a
+	  TIME segment (e.g. avidemux). Without this, we would generate wrong durations
+	  and timestamps after a seek.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 13:53:44 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdemux.c:
+	* ext/dv/gstdvdemux.h:
+	  dvdemux: Pass-through buffer DISCONT flags
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 16:16:45 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpvp9depay.c:
+	  rtpvp9depay: Don't assert on flexible mode packets
+	  Instead just post a warning on the bus for now.
+
+2016-06-02 15:03:17 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: rtpbin: fix caps leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767156
+
+2016-06-02 15:00:01 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* tests/check/elements/amrparse.c:
+	  tests: amrparse: clean up test
+	  - use GST_CHECK_MAIN() to reduce boilerplate
+	  - unref the input caps using a teardown function to prevent leaks
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767156
+
+2016-05-20 15:22:35 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: Ensure DISCONT flag is properly propagated
+	  The output of deinterlace at startup, or when receiving a new DISCONT
+	  buffer, should have the DISCONT flag set on the first buffer.
+
+2016-05-31 21:34:04 +0200  Josep Torra <adn770@gmail.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2src: check for valid size on raw video buffers
+	  Discard buffers that doesn't contain enough data when dealing
+	  with raw video inputs.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767086
+
+2016-05-31 17:10:36 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Use the demuxer segment instead of a new one for MSS streams
+	  Upstream might have told us something about the to be expected segment, so
+	  let's use that information instead of coming up with a [0,-1] segment.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 17:04:32 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Only activate segments and send SEGMENT events if we have streams
+	  But in that case also remove the pending newsegment event, otherwise we would
+	  later send a possibly outdated event.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 16:53:50 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: In PULL mode, nothing is ever going to send us a SEGMENT event
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 16:38:34 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Don't override TIME segments from upstream that we just saw
+	  The point of d8fb7a9c96b108814beeaa0e63f818d4648c7fe9 was to not have any
+	  spurious segments stored for later if we do BYTES->TIME conversion, but
+	  overriding any TIME segments from upstream does not make any sense.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=763165
+	  https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2015-07-16 09:48:46 +0530  Prashant Gotarne <ps.gotarne@samsung.com>
+
+	* gst/multifile/gstmultifilesrc.c:
+	  multifilesrc: set position as offset from start-index
+	  query position in GST_FORMAT_BUFFER returns
+	  offset from start-index rather than index.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752462
+
+2016-05-27 12:49:32 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/pipelines/simple-launch-lines.c:
+	* tests/files/Makefile.am:
+	* tests/files/gradient.j2k:
+	  tests: add unit test for JPEG-2000 rtp payloader leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766870
+
+2016-05-25 17:11:13 +0200  Pierre Lamot <pierre.lamot@openwide.fr>
+
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtpj2kpay: Fix buffer memory leak
+	  Input buffer memory was not unmapped
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766870
+
+2016-05-18 12:12:15 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: fix caps leak
+	  gst_v4l2_object_probe_caps() was taking an extra ref on the returned
+	  caps for no reason.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766610
+
+2016-05-22 20:14:18 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/videocrop/gstvideocrop.c:
+	  videocrop mark crop properties as mutable in playing state
+
+2016-05-20 16:47:35 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: fix buffer leak when flushing
+	  When early returning in gst_soup_http_src_read_buffer() because the
+	  element is FLUSHING, we need to unmap and unref the buffer which was just created.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766718
+
+2016-05-20 11:15:44 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Set seek event seqnum on all SEGMENT events
+	  Some were forgotten.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-20 11:12:44 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavidemux.h:
+	  avidemux: Pass through seek event seqnums in all SEGMENT/EOS events and SEGMENT_DONE messages/events
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-20 10:56:52 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Set seek event seqnum in EOS and SEGMENT_DONE messages/events
+	  Also actually store the seqnum in pull mode seeks.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-17 13:40:38 +0300  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: fix caps leak
+	  The caps returned by gst_pad_get_current_caps() was never unreffed when
+	  not early returning.
+	  Fix a leak with the elements/deinterlace test.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766558
+
+2016-01-25 16:25:51 +0100  Mikhail Fludkov <misha@pexip.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* tests/check/Makefile.am:
+	* tests/check/elements/rtpsession.c:
+	  rtpsession: don't act on suspicious BYE RTCP
+	  Some endpoints (like Tandberg E20) can send BYE packet containing our
+	  internal SSRC. I this case we would detect SSRC collision and get rid
+	  of the source at some point. But because we are still sending packets
+	  with that SSRC the source will be recreated immediately.
+	  This brand new internal source will not have some variables incorrectly
+	  set in its state. For example 'seqnum-base` and `clock-rate` values will be
+	  -1.
+	  The fix is not to act on BYE RTCP if it contains internal or unknown
+	  SSRC.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762219
+
+2015-11-15 14:54:28 +0100  Mikhail Fludkov <misha@pexip.com>
+
+	* tests/check/elements/rtpsession.c:
+	  rtpsession: Add test for locking of the stats signal
+	  Keeping the lock while emitting the stats signal introduces potential
+	  deadlock in those situations when the signal callback wants the access
+	  to rtpsession's properties which also requre the lock.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762216
+
+2016-05-19 15:36:57 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: don't hold object lock whilst pushing out headers
+	  matroskademux would take the GST_OBJECT_LOCK in
+	  - gst_matroska_demux_push_codec_data_all()
+	  - gst_matroska_demux_query()
+	  Some parse element such as FLAC checks upstream seekability, and
+	  there is some use cases that matroska-demux is linked to a parse element
+	  (e.g.,FLAC format) without intermediate elements (e.g., queue).
+	  In this case, matroska-demux never returns from _push_codec_data_all()
+	  because the parser can return only after it receives the response to
+	  the upstream query, but that's not going to happen because it's
+	  deadlocked.
+	  Elements must not hold the object lock whilst pushing out events
+	  or data.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766645
+
+2016-05-19 12:43:01 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: Set sent_buffers and streamheader_buffers to NULL after freeing
+	  Otherwise we might use an already freed list later and crash or worse.
+
+2016-05-18 18:32:57 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: fix Since version for new "loop" property
+
+2016-05-16 16:18:37 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* gst/rtsp/gstrtpdec.c:
+	  rtpdec: fix clock leak
+	  gst_system_clock_obtain() returns a new ref.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766521
+
+2016-05-17 05:33:35 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: add doc blurb with since marker for new "loop" property
+
+2015-11-13 15:52:35 +0100  Dimitrios Katsaros <patcherwork@gmail.com>
+
+	* gst/avi/gstavimux.c:
+	  avimux: add support for png
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758059
+
+2016-05-15 22:07:14 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxsrc: Connect to demux signals before activating
+	  Fix a race in splitmuxsrc by properly connecting to the
+	  demuxer signals we're interested in *before* setting it running.
+
+2016-05-15 13:31:37 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	  docs: Update for git master
+
+2016-05-15 12:16:23 +0200  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpmp4gpay.c:
+	* gst/rtp/gstrtpmp4gpay.h:
+	  rtpmp4gpay: Don't produce timestamps based on byte count
+	  The GST_BUFFER_OFFSET of output buffers returned to GstRtpBasePayload
+	  should reflect the number of "samples" in the unit of the RTP clock in this
+	  buffer. If this is not true, then it shouldn't be set.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761943
+
+2016-05-15 12:24:03 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: Fix strcmp usage
+	  Just use g_strcmp0 which can handle NULL entries
+
+2016-03-04 10:14:00 +0100  Carlos Rafael Giani <dv@pseudoterminal.org>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Use audio/x-unaligned-raw instead of audio/x-raw for L16 data
+	  Directly setting audio/x-raw caps leads to problems when the delivered
+	  data blocks do not align properly at sample boundaries (for example, a
+	  data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
+	  let a parser be autoplugged.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=689460
+
+2016-05-12 11:52:09 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Parsing elst box based on version
+	  segment_duration and media_time should be parsed based on version
+	  of elst box. Specification defines that an elst box with version 1
+	  has uint64 and int64 values for segment_duration and media_time,
+	  respectively.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766301
+
+2016-05-14 12:57:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: check if request was cancelled when sending message
+	  It might be that the request was aborted by the application and
+	  we can return immediatelly
+
+2016-05-14 12:43:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: proxy resolver is on by default
+	  Remove from the session creation parameters
+
+2016-05-14 12:15:48 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/soup/Makefile.am:
+	  soup: update build to warn about newer deprecated functions
+	  We already depend on 2.48
+
+2016-05-14 11:09:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: reduce reading latency by using non-blocking read
+	  Non-blocking read will return the amount of data available without
+	  blocking to wait for the full requested size.
+	  The downside is that now it souphttpsrc needs to have a waiting
+	  mechanism in case there is no data available yet to avoid busy
+	  looping arond the inputstream.
+
+2016-05-15 12:30:50 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Take the lock already when reading the other stats, not just for the hash table
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-14 17:04:57 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/ebml-read.c:
+	  matroska: use math-compat.h for NAN define
+
+2016-05-14 23:39:22 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Use GstBin async-handling instead of our own.
+	  Set the async-handling property on GstBin to let it manage
+	  async-handling instead of the local handling from the previous
+	  commit. Works because of #174a5e in core
+
+2016-05-13 10:17:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: refactor to use Soup's sync API
+	  Replace the async API with the sync API to remove all the extra mainloop
+	  and context handling. Currently it blocks reading until 'blocksize'
+	  bytes are available but that can be improved by using:
+	  https://developer.gnome.org/gio/unstable/GPollableInputStream.html#g-pollable-input-stream-read-nonblocking
+	  https://bugzilla.gnome.org/show_bug.cgi?id=693911
+
+2016-05-14 04:50:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: souphttpsrc: replace deprecated API
+	  Avoid using soup_server_run_async and old get_port() APIs,
+	  replace with me soup_server_listen and get the port through the
+	  URIs list returned from the server.
+
+2016-05-14 12:34:10 +0200  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: Upgrade debug message to error
+	  It causes the entire pipeline to fail, it should be easier to find.
+
+2016-05-14 18:32:52 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Hide internal async state changes.
+	  When switching fragments, hide the async-start/async-done
+	  messages from the parent bin, as otherwise we sometimes (very rarely)
+	  hang in PAUSED instead of returning / continuing to PLAYING
+	  state.
+
+2016-05-13 21:20:28 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Remove stray carriage-return from debug
+
+2016-05-13 16:43:21 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/Makefile.am:
+	  rtp: Ship gstrtpj2kcommon.h file to fix distcheck
+
+2015-04-30 14:43:04 +0200  Jesper Larsen <knorr.jesper@gmail.com>
+
+	* gst/avi/gstavimux.c:
+	  avimux: Do not write index and header if idx is NULL
+	  Fixes criticals with e.g.
+	  videotestsrc num-buffers=1 ! identity drop-probability=1.0 ! avimux ! fakesink
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748700
+
+2016-05-12 08:43:39 -0400  Aaron Boxer <boxerab@gmail.com>
+
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtpj2kpay: manage T tile invalidation bit correctly, update tile id in header correctly.
+	  1. according to RFC, T bit is only set when either the RTP packet only contains the J2K main header, or the packet contains tile parts from multiple tiles. This is now being managed correctly in the code. The second scenario cannot happen with our payloader, since tile headers are always placed in their own RTP packet, and so a packet cannot contain tile parts from multiple tiles.
+	  However, I have added code to track if multiple tile parts are included in a single RTP packet, in case in the future we want to put header and data in same packet.
+	  2. Old code would set the tile id to zero for all J2K packets. This is now set correctly to the appropriate tile id.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-12 08:41:51 -0400  Aaron Boxer <boxerab@gmail.com>
+
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtpj2kpay: manage fragmented headers correctly
+	  J2K main header framentation across multiple RTP packets is now handled correctly
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-11 15:04:26 -0400  Aaron Boxer <boxerab@gmail.com>
+
+	* gst/rtp/gstrtpj2kcommon.h:
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kdepay.h:
+	* gst/rtp/gstrtpj2kpay.c:
+	* gst/rtp/gstrtpj2kpay.h:
+	  rtpj2k: move common code to shared header, code clean up
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-11 15:01:32 -0400  Aaron Boxer <boxerab@gmail.com>
+
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtpj2k: update documentation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-12 14:43:43 +0200  Patricia Muscalu <patricia@axis.com>
+
+	* gst/auparse/gstauparse.c:
+	* gst/auparse/gstauparse.h:
+	  auparse: Fix sticky event misordering warning
+	  Make sure that src pad has caps before sending segment event.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766359
+
+2016-05-11 09:28:13 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Don't notify about stats property changes while taking the session lock
+	  The signal handlers might want to actually get the value of the stats
+	  property, which would take the session lock again and deadlock.
+	  This was introduced by 2e960e70750a0cb7e1117d0c09d08597866a29ee.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-03 13:59:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: improve edts segment handling after seeks in push mode
+	  Properly handle edts segments for push-based operation seeking.
+	  We only support edts that a single segment that has media at the end,
+	  being preceeded by any number of gap segments.
+	  This also allows the qt segment rate to be respected after seeks
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765669
+
+2016-05-03 10:41:06 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: properly activate segment with rate != 1.0
+	  Also use the qt rate to identify the position within a qt segment
+	  to properly translate playback time to qt media time
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765669
+
+2016-05-03 11:45:01 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix stall when receiving already lost packet
+	  When a packet arrives that has already been considered lost as part of a
+	  large gap the "lost timer" for this will be cancelled. If the remaining
+	  packets of this large gap never arrives, there will be missing entries
+	  in the queue and the loop function will keep waiting for these packets
+	  to arrive and never push another packet, effectively stalling the
+	  pipeline.
+	  The proposed fix conciders parts of a large gap definitely lost (since
+	  they are calculated from latency) and ignores the late arrivals.
+	  In practice the issue is rare since large gaps are scheduled immediately,
+	  and for the stall to happen the late arrival needs to be processed
+	  before this times out.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765933
+
+2016-05-05 14:18:21 +0200  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Take session lock when creating stats
+	  The access to the session hash table must happen while the session lock is
+	  taken, otherwise another thread might modify the hash table while we're
+	  creating the stats.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-03 21:17:01 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: update segment when new duration is found
+	  Otherwise the old segment will have a shorter stop time and would
+	  cause the stream to end too early.
+
+2016-05-04 11:37:29 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: dismember activate_segment into 2 parts
+	  One that updates and push a new segment, the other will move the
+	  stream to the new segment starting position
+
+2016-05-04 09:30:27 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdec.c:
+	* ext/dv/gstdvdemux.c:
+	  dv: Use correct pixel-aspect-ratio values
+	  The previous ones resulted in odd display aspect ratios and were different
+	  from the ones used by e.g. ffmpeg. The new ones now result in display aspect
+	  ratios of 4:3 and 16:9.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765946
+
+2015-11-09 17:55:09 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/splitmux.c:
+	  tests: add splitmuxsrc test for new "format-location" signal
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753625
+
+2015-11-09 17:51:12 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: add a format-location signal that allows bypassing the location property
+	  This signal allows a user to directly return a sorted list of
+	  files to be joined, so that they don't have to follow the
+	  filename pattern that the "location" property expects.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753625
+
+2016-05-04 11:15:20 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Fix deadlock case when source reaches EOS
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765072
+
+2016-05-03 22:59:27 -0700  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: simplify and correct header scanning
+	  The wav spec tells that 'fmt' (and 'bext' if present) must come before 'data'.
+	  There is no requirement for 'fmt' to be first. We already had a list of chunks
+	  to skip, but it is easier to just skip any chunk while seeking for 'fmt'.
+	  This fixes reading files generated by ProTools.
+
+2016-04-30 22:15:13 +0900  Hyunjun Ko <zzoon@igalia.com>
+
+	* sys/osxaudio/Makefile.am:
+	* sys/osxaudio/gstosxaudio.c:
+	* sys/osxaudio/gstosxaudiodeviceprovider.c:
+	* sys/osxaudio/gstosxaudiodeviceprovider.h:
+	* sys/osxaudio/gstosxaudiosink.c:
+	* sys/osxaudio/gstosxaudiosink.h:
+	* sys/osxaudio/gstosxaudiosrc.c:
+	* sys/osxaudio/gstosxaudiosrc.h:
+	  osxaudio: Support audio device provider on osx
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753265
+
+2016-05-01 15:09:27 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/avi/gstavimux.c:
+	  avimux: set audio header rate according to calculated bps in stop_file
+	  ... now that set_fields is no longer called there by
+	  e538608b3f90539003de21c1db238f3c9b946e30
+
+2016-04-29 15:04:11 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: Store the segment sequence number in the EOS events and SEGMENT_DONE events/message
+	  Also instead of storing it per stream, store it globally in the demuxer. It's
+	  the same for each stream anyway.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765806
+
+2016-04-11 10:54:38 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Always bind to ANY when address is a multicast address and not only on Windows
+	  For IPv6 addresses, binding to a multicast group does not work on Linux
+	  either. Always bind to ANY and then later join the multicast group.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764679
+
+2016-04-26 17:01:49 +0800  Song Bing <b06498@freescale.com>
+
+	* sys/ximage/ximageutil.c:
+	  ximageutil: shouldn't implement transform if don't support it
+	  shouldn't implement transform if don't support it. Or gst_buffer_copy_into()
+	  will print ERROR log.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765583
+
+2016-04-28 16:24:52 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: Allow MPEG-1 Layer 1 and 2 in addition to 3 in MP4
+	  Via the MPEG-4 Part 3 spec we can support the other layers too.
+	  Also correct the samples per frame calculation for MP3 if it's MPEG-2 or
+	  MPEG-2.5.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765725
+
+2016-04-27 20:46:34 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: Update caps for TCP whenever they change
+	  We only changed them for UDP so far, which caused the wrong seqnum-base and
+	  other information to be passed to rtpjitterbuffer/etc when seeking. This
+	  usually wasn't that much of a problem as the code there is robust enough, but
+	  every now and then it causes us to drop up to 32756 packets before we
+	  continue doing anything meaningful.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 20:33:38 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Ensure to not take caps with the wrong pt for getting the clock-rate
+	  Especially the caps on the pad might be out of date, and the new caps would be
+	  provided for the current pt via the request-pt-map signal.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 18:27:17 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Don't propagate spurious state change returns from internal elements further
+	  We handle them inside rtspsrc and override them in all other cases anyway, so
+	  do the same for "internal" state changes like PAUSED->PAUSED and
+	  PLAYING->PLAYING.
+	  This keeps unexpected NO_PREROLL to confuse state changes in GstBin.
+	  See also https://bugzilla.gnome.org/show_bug.cgi?id=760532
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 14:09:03 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavimux.c:
+	  avimux: Don't override maximum audio chunk size with the scale again just before writing it
+	  set_fields() should only be called in the beginning, otherwise we will never
+	  remember the maximum audio chunk size and write a wrong block align... which
+	  then causes wrong timestamps and other problems.
+
+2016-04-27 13:53:00 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/avi/gstavimux.c:
+	  avimux: Actually store the largest audio chunk size for the VBR case of MP2/MP3
+	  3ea338ce271e1f6a96d2ed49d4472b091f6f8b7e changed avimux to do that, but it
+	  never actually kept track of the max audio chunk for MP3 and MP2. These are
+	  knowing the hdr.scale only after parsing the frames instead of at setcaps
+	  time.
+
+2016-04-25 15:03:14 +0200  Mats Lindestam <matslm@axis.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: Allow setting "socket-v6" without setting "socket" too
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764897
+
+2016-04-22 15:02:16 +0100  Mario Sanchez Prada <mario@endlessm.com>
+
+	* ext/vpx/gstvpxenc.c:
+	  vpxenc: Properly handle frames with too low duration
+	  When a frame's duration is too low, calling gst_util_uint64_scale()
+	  to scale its value can result into it being truncated to zero, which
+	  will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
+	  when trying to encode.
+	  To prevent this from happening, we simply ignore the duration when
+	  encoding if it becomes zero after scaling, logging a warning message.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765391
+
+2016-04-22 15:48:08 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: fix description of linear interlacing method
+
+2016-04-21 14:08:19 -0300  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/flv/gstflvmux.c:
+	  flv: Handle the case where we do not get any CollectData in handle_buffer
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765320
+
+2016-04-11 22:41:20 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Do not use unreliable framerate
+	  timescale/1 is unreliable value for framerate. Due to downstream
+	  element usually use framerate generated by qtdemux, let it be omitted
+	  until the framerate can be reliably calculated.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764733
+
+2016-04-21 12:53:33 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  Revert "qtdemux: expose streams with first moof for fragmented format"
+	  This reverts commit d8bb6687ea251570c331038279a43d448167d6ad.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764733
+
+2016-02-09 17:17:09 +0000  Alex Ashley <bugzilla@ashley-family.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: support seeking of CENC encrypted streams
+	  When playing a stream that has been protected by DASH CENC, playback
+	  will fail if a seek is performed. Qtdemux produces the error "stream
+	  is protected using cenc, but no cenc protection system information
+	  has been found" and playback stops.
+	  The problem is that gst_qtdemux_reset() gets called as part of the
+	  FLUSH during a seek. This function frees the protection_system_ids
+	  array. When gst_qtdemux_configure_protected_caps() is called after the
+	  seek has completed, the protection_system_ids array is empty and
+	  qtdemux is unable to create the correct output caps for the protected
+	  stream.
+	  This commit changes it to only free the protection_system_ids on
+	  hard resets.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761787
+
+2016-04-18 14:33:10 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	* gst/udp/gstudpsrc.h:
+	  udpsrc: add "retrieve-sender-address" property
+	  This allows disabling of sender address retrieval, which might
+	  be useful in certain scenarios, like when the socket is connected,
+	  or the sender address is not of interest (e.g. when receiving an
+	  MPEG-TS stream). Disabling sender address retrieval in those
+	  cases can have minor performance advantages.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=563323
+
+2015-11-26 13:15:06 +0100  Dimitrios Katsaros <patcherwork@gmail.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Change warning handling to break infinite message loop
+	  v4l2src can cause an "infinite message loop" when a base control exposed as a
+	  property is not provided by the device. In these cases, if in the warning message
+	  handling for the bus, the GST_DEBUG_BIN_TO_DOT_FILE* category of functions are used,
+	  the src lookup causes a new warning to be posted on the bus, causing a loop.
+	  This patch changes the warning for these controls so they are not posted on the bus.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758703
+
+2016-04-15 10:44:02 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  spitmuxsink: Avoid creating small file at EOS
+	  When EOS is reached, the current file get closed and the last
+	  GOP in the mq was written in a new file.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765072
+
+2016-04-15 19:59:15 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/gstscaletempo.c:
+	  scaletempo: S16 uses S32 temporary buffers, float/double their own type
+	  Make sure to allocate not only a S16 buffer for S16 but a twice as big one to
+	  hold S32.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=765116
+
+2016-04-16 02:17:26 +1000  Jan Schmidt <jan@centricular.com>
+
+	* ext/pulse/pulsesink.c:
+	  Revert "pulsesink: uncork if needed upon commit"
+	  This reverts commit 0dd46accf6d282ff07065852bd91c85c78af3394.
+	  With some audiosinks, starting the ringbuffer on the first commit
+	  causes audio glitches at startup by starting to output segments
+	  from the ringbuffer before it has been filled / fully prerolled. This
+	  doesn't usually happen with pulsesink because we map the pulseaudio
+	  ringbuffer directly, but we should keep things consistent with
+	  other sinks with regards to startup latency, plus it gives more
+	  headway to avoid glitching, should the initial 2nd segment take
+	  more than 10ms to generate.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=657076
+
+2016-04-15 00:46:56 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: add srtp rollover counters from mikey crypto sessions
+	  The server can send multiple crypto sessions, one for each SSRC with its
+	  own rollover counter. We parse this information and pass it to the SRTP
+	  decoder via the "request-key" signal.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=730540
+
+2016-04-15 14:35:07 +0000  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix debug output when resyncing
+	  Don't output the pointer value of the time() function as a timestamp
+	  by using the correct variable.
+	  Fixes build on Raspberry Pi 3.
+
+2016-04-15 11:36:36 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: If no proxy is set by properties, use the default libsoup proxy resolver
+	  That is, use whatever system settings there might exist. This is the same
+	  behaviour we use in the HTTP source.
+
+2016-04-14 10:01:28 +0100  Julien Isorce <j.isorce@samsung.com>
+
+	* README:
+	* common:
+	  Automatic update of common submodule
+	  From 6f2d209 to ac2f647
+
+2016-04-13 18:45:07 +0100  Damian Ziobro <damian@xmementoit.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Add max_files_number property
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744612
+
+2016-04-13 10:57:03 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: drop reference to videomixer 2
+	  Fix a small grammar mistake on "overlayed" while at it.
+
+2016-04-13 09:57:16 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/ximage/ximageutil.c:
+	  ximage: Initialize all fields in the meta explicitly
+	  The meta is not allocated with all fields initialized to zeroes.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764902
+
+2016-04-12 09:41:00 +0000  Paolo Pettinato <ppettina@cisco.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	  rtpmux: Forward sticky events on buffer lists too, not only on buffers
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764933
+
+2016-04-12 15:01:28 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Drain the field history if the caps are changing
+	  Otherwise we will use fields from the old caps with everything set up for the
+	  new caps, causing crashes and worse.
+	  Also don't do anything if the same caps are set twice.
+
+2016-04-12 15:00:31 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Instead of confusing crashes later, just error out immediately if mapping a video frame fails
+	  This probably still crashes but at least we get some hint about what goes
+	  wrong instead of random behaviour later.
+
+2016-04-12 11:38:51 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: check stream is available in PIFF parser
+	  qtdemux->streams is an array, it will never evaluate to true when comparing
+	  to NULL. Instead we want to check the number of streams to make sure the
+	  stream is available.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753614
+	  CID 1358389
+
+2016-04-12 11:37:36 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  Revert "qtdemux: redundant check in PIFF parser"
+	  This reverts commit 41e10524f3babdd92aac8c8c9d5b9cdf184c2d4e.
+
+2016-04-12 11:05:50 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: redundant check in PIFF parser
+	  qtdemux->streams is an array of size GST_QTDEMUX_MAX_STREAMS, it will never
+	  evaluate to true when comparing to NULL.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753614
+	  CID 1358389
+
+2016-04-12 11:56:08 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: avoid leaking GValues
+	  unset the GValue if we don't use it any more to avoid leaks.
+
+2016-04-12 10:15:39 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation
+	  The head of the queue is the oldest packet (as in lowest seqnum), the tail is
+	  the newest packet. To calculate the fill level, we should calculate tail-head
+	  while considering wraparounds. Not the other way around.
+	  Other code is already doing this in the correct order.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764889
+
+2016-04-11 10:44:56 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/Makefile.am:
+	  rtpmanager: It's GST_LIBS, not GST_LIBS_LIBS
+
+2016-04-11 08:33:17 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix parsing segment duration of empty edit list box
+	  For empty edit list, segment-duration in edit list box should not be
+	  used for segment event.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764870
+
+2016-04-08 13:05:57 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: make timecodescale configurable
+	  In some use cases the default timecodescale will produce blocks with the same timestamp
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764769
+
+2016-04-07 13:01:52 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jiterbuffer: Move assertion to the right location
+	  We shouldn't have "late" lost timers at that point
+
+2016-03-02 14:25:24 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: Speed up lost timeout handling
+	  When downstream blocks, "lost" timers are created to notify the
+	  outgoing thread that packets are lost.
+	  The problem is that for high packet-rate streams, we might end up with
+	  a big list of lost timeouts (had a use-case with ~1000...).
+	  The problem isn't so much the amount of lost timeouts to handle, but
+	  rather the way they were handled. All timers would first be iterated,
+	  then the one selected would be handled ... to re-iterate the list again.
+	  All of this is being done while the jbuf lock is taken, which in some use-cases
+	  would return in holding that lock for 10s... blocking any buffers from
+	  being accepted in input... which would then arrive late ... which would
+	  create plenty of lost timers ... which would cause the same issue.
+	  In order to avoid that situation, handle the lost timers immediately when
+	  iterating the list of pending timers. This modifies the complexity from
+	  a quadratic to a linear complexity.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-03-02 14:23:01 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: Don't create lost events if we don't need them
+	  When "do-lost" is set to FALSE we don't use/send the lost events.
+	  In that case, don't create them to start with :)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-03-02 13:57:07 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: Add tracing of lock usage
+	  Helps with debugging lock usage
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-02-10 19:56:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	  v4l2: Don't leak v4l2 objects and props on probe errors
+
+2016-04-04 17:42:03 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/rtp-payloading.c:
+	  tests: add unit test for jpeg depayloader packet loss handling
+	  Make sure it always outputs something that looks like a valid
+	  JPEG frame, ie. starts with an SOI marker and ends with an EOI
+	  marker.
+
+2016-03-15 03:25:26 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtp/gstrtpjpegdepay.c:
+	  rtpjpegdepay: Don't send invalid frames downstream after packet loss or a DISCONT
+	  After clearing the adapter due to a DISCONT, as might happen when some packet(s)
+	  have been lost, the depayloader was pushing data into the adapter (which had no
+	  header due to the clear), creating a headerless frame out of it, and sending it
+	  downstream. The downstream decoder would then usually ignore it; unless there
+	  were lots of DISCONTs from the jitterbuffer in which case the decoder would reach
+	  its max_errors limit and throw an element error. Now we just discard that data.
+	  It is probaby not worth trying to salvage this data because non-progressive
+	  jpeg does not degrade gracefully and makes the video unwatchable even with
+	  low packet loss such as 3-5%.
+
+2016-01-05 16:15:16 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtpjitterbuffer: Add RFC7273 media clock handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762259
+
+2015-07-10 09:44:15 +0200  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: PIFF box detection and parsing support
+	  The PIFF data is stored in a custom UUID box which is parsed and the
+	  crypto_info of the element is updated accordingly. This allows
+	  downstream decryptors to process and decrypt the protected content.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753614
+
+2016-04-01 12:15:05 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtpvorbisdepay.c:
+	  rtpvorbisdepay: remove dead code
+	  payload_buffer hasn't been assigned a value before the jumps to
+	  switch_failed or packet_short. So the value must be NULL. No need
+	  to unmap and unref.
+	  CID #1316476
+
+2016-03-31 14:57:20 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph263pay.c:
+	  rtph263pay: fix leak
+	  Free memory of current macroblock once it isn't needed so it isn't leaked
+	  by the call of the gst_rtp_h263_pay_B_mbfinder function.
+	  if (!(mac = gst_rtp_h263_pay_B_mbfinder (context, gob, mac, mb))) {
+	  CID 1212156
+
+2016-03-31 02:15:04 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmux: Handle a hang draining out at EOS
+	  Make sure that all data is drained out when the reference pad
+	  goes EOS. Fixes a problem where data that arrives on other
+	  pads after the reference pad finishes can stall forever and
+	  never pass EOS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763711
+
+2016-03-18 15:45:01 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Fix occasional deadlock when ending file with subtitle
+	  Deadlock occurs when splitting files if one stream received no buffer during
+	  the first GOP of the next file. That can happen in that scenario for example:
+	  1) The first GOP of video is collected, it has a duration of 10s.
+	  max_in_running_time is set to 10s.
+	  2) Other streams catchup and we receive the first subtitle buffer at ts=0 and
+	  has a duration of 1min.
+	  3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to
+	  1min. That buffer is blocked in handle_mq_input() because
+	  max_in_running_time is still 10s.
+	  4) Since all in_running_time are now > 10s, max_out_running_time is now set to
+	  10s. That first GOP gets recorded into the file. The muxer pop buffers out
+	  of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the
+	  GstDataQueue is empty.
+	  5) A 2nd GOP of video is collected and has a duration of 10s as well.
+	  max_in_running_time is now 20s. Since subtitle's in_running_time is already
+	  1min, that GOP is already complete.
+	  6) But let's say we overran the max file size, we thus set state to
+	  SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of
+	  previous GOP) arrives in handle_mq_output(), EOS event is sent downstream
+	  instead. But since the subtitle queue is empty, that's never going to
+	  happen. Pipeline is now deadlocked.
+	  To fix this situation we have to:
+	  - Send a dummy event through the queue to wakeup output thread.
+	  - Update out_running_time to at least max_out_running_time so it sends EOS.
+	  - Respect time order, so we set out_running_tim=max_in_running_time because
+	  that's bigger than previous buffer and smaller than next.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763711
+
+2015-11-17 18:17:35 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	* tests/check/elements/rtpsession.c:
+	  rtpsession: Add new signal 'on-app-rtcp'
+	  Similar to the 'on-feedback-rtcp' signal, but emitted for RTCP APP
+	  packets.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762217
+
+2016-03-24 15:57:11 +0900  Minjae Kim <nate.kim@lge.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpmanager: Set to initial value for 'ntpns' in get_current_times()
+	  Initialize "ntpns" variable to -1 as the OE compiler for some reason doesn't
+	  realize that the variable is set in all code paths.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764119
+
+2016-01-31 11:08:38 +1100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpjpegpay.c:
+	  rtpjpegpay: Allow different quantization tables for components 2 and 3
+	  RFC 2435 mentions in section 4.1 that U/V use table number 1, but this seems
+	  just like an example. Some encoders are not following that and there seems to
+	  be no reason to reject their streams.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761345
+
+2016-03-24 19:23:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/gstvpxdec.c:
+	  vpxdec: Use threads on multi-core systems
+	  This is a redo of commit b848c1b6ffd1e508228820a013f94fb445e4777f. The
+	  code was lost when the elements where ported to use a baseclass.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=764169
+
+2016-02-29 23:40:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* tests/check/elements/splitmux.c:
+	  splitmuxsink: only try to create internal sink if it doesn't exist
+	  This allows splitmuxsink to be reused after being put to NULL.
+	  Test included
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762893
+
+2015-10-01 13:41:23 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: probe all colorspace supported by device
+	  A device can support more than one colorspace for a given image
+	  dimension and pixel format. So we have to probe all the supported
+	  colorspace and not only rely on the default one. Otherwise we could end
+	  up with negotiation failure if the caps colorimetry field don't match
+	  the v4l2 device default one even if the v4l2 could support such
+	  colorimetry.
+	  This patch enable probing if colorspace for both capture and output
+	  device. It really makes sense for output device since the colorspace
+	  shall be set by the application and a little less for capture device
+	  which, at the moment, shall provide the colorspace; ie: the v4l2
+	  specification seems to not take into account the fact that a capture
+	  device could do colorspace conversion.
+	  As a side effet, probing takes some times and so sligthly delay v4l2
+	  initialization. Note that this patch only probe colorspace and not all
+	  colorspace, matrix, transfer and range combination to avoid taking too
+	  much time, especially with low-speed devices as full probing do 1782
+	  ioctl.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755937
+
+2016-03-24 16:21:56 +0100  Edward Hervey <edward@centricular.com>
+
+	* tests/check/elements/flvdemux.c:
+	  check: Fix indentation
+
+2016-03-24 16:20:39 +0100  Edward Hervey <edward@centricular.com>
+
+	* tests/check/elements/flvdemux.c:
+	  tests: Remove unused variables
+
+2016-03-16 20:26:16 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/interleave/deinterleave.c:
+	  deinterleave: Return the current caps on the srcpads on caps queries
+	  It's not like we could accept any other caps here. The caps are decided by the
+	  upstream caps event.
+	  Also keep the filter order intact when filtering the results against the
+	  filter caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-24 15:14:23 +0900  Jimmy Ohn <yongjin.ohn@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix qtdemux memory leak in src_convert function
+	  If we don't find the index of the sample correctly in src_convert function,
+	  we have to unref about the qtdemux before returning value.
+	  So, I have modify it about instead pass qtdemux as a parameter into
+	  src_convert function.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763973
+
+2016-03-22 13:15:20 +0900  Jimmy Ohn <yongjin.ohn@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Add check condition for fail case in get_duration function
+	  Currently, get_duration function always return the TRUE even though
+	  it can't be set duration correctly. So, we need to add the else condition
+	  about the fail case. Also, we already set the GST_CLOCK_TIME_NONE
+	  in this function. So I have modify it which is related code in some
+	  function.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763968
+
+2016-03-21 10:11:23 +0900  Jimmy Ohn <yongjin.ohn@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Modify data type of duration in handle_src_query function
+	  Data type of duration need to modify from guint64 to GstClockTime
+	  for consistency in handle_src_query function.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763965
+
+2016-03-18 14:40:58 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* tests/check/elements/deinterlace.c:
+	  deinterlace: Added unit tests for field=auto
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763869
+
+2016-03-17 21:21:02 +0200  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: Added "auto" fields mode
+	  The "auto" fields mode will detect the upstream and downstream framerates and
+	  will decide to deinterlace all or only top fields.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763869
+
+2016-03-16 20:17:55 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/gstflvdemux.c:
+	* tests/check/elements/flvdemux.c:
+	  flvdemux: don't emit pad-added until caps are ready
+	  In other words, gst_pad_get_current_caps should never return NULL
+	  in a pad-added callback from the demuxer.
+	  Added tests for the two special cases with AAC and H.264 where this
+	  would happen every time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763780
+
+2016-03-04 10:30:12 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/aalib/gstaasink.c:
+	* ext/cairo/gstcairooverlay.c:
+	* ext/dv/gstdvdec.c:
+	* ext/dv/gstdvdemux.c:
+	* ext/flac/gstflacdec.c:
+	* ext/flac/gstflacenc.c:
+	* ext/flac/gstflactag.c:
+	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
+	* ext/jack/gstjackaudiosink.c:
+	* ext/jack/gstjackaudiosrc.c:
+	* ext/jpeg/gstjpegdec.c:
+	* ext/jpeg/gstjpegenc.c:
+	* ext/jpeg/gstsmokedec.c:
+	* ext/jpeg/gstsmokeenc.c:
+	* ext/libcaca/gstcacasink.c:
+	* ext/libpng/gstpngdec.c:
+	* ext/libpng/gstpngenc.c:
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulsesrc.c:
+	* ext/raw1394/gstdv1394src.c:
+	* ext/raw1394/gsthdv1394src.c:
+	* ext/shout2/gstshout2.c:
+	* ext/soup/gstsouphttpclientsink.c:
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/speex/gstspeexdec.c:
+	* ext/speex/gstspeexenc.c:
+	* ext/taglib/gstapev2mux.cc:
+	* ext/taglib/gstid3v2mux.cc:
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp8enc.c:
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvp9enc.c:
+	* ext/wavpack/gstwavpackdec.c:
+	* ext/wavpack/gstwavpackenc.c:
+	* gst/alpha/gstalpha.c:
+	* gst/alpha/gstalphacolor.c:
+	* gst/apetag/gstapedemux.c:
+	* gst/audiofx/audiopanorama.c:
+	* gst/audiofx/gstscaletempo.c:
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	* gst/auparse/gstauparse.c:
+	* gst/autodetect/gstautoaudiosink.c:
+	* gst/autodetect/gstautoaudiosrc.c:
+	* gst/autodetect/gstautovideosink.c:
+	* gst/autodetect/gstautovideosrc.c:
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavimux.c:
+	* gst/avi/gstavisubtitle.c:
+	* gst/cutter/gstcutter.c:
+	* gst/debugutils/breakmydata.c:
+	* gst/debugutils/cpureport.c:
+	* gst/debugutils/gstcapsdebug.c:
+	* gst/debugutils/gstcapssetter.c:
+	* gst/debugutils/gstnavigationtest.c:
+	* gst/debugutils/gstnavseek.c:
+	* gst/debugutils/gstpushfilesrc.c:
+	* gst/debugutils/gsttaginject.c:
+	* gst/debugutils/progressreport.c:
+	* gst/debugutils/rndbuffersize.c:
+	* gst/debugutils/testplugin.c:
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/dtmf/gstdtmfsrc.c:
+	* gst/dtmf/gstrtpdtmfdepay.c:
+	* gst/dtmf/gstrtpdtmfsrc.c:
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gstedge.c:
+	* gst/effectv/gstop.c:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstradioac.c:
+	* gst/effectv/gstrev.c:
+	* gst/effectv/gstripple.c:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gststreak.c:
+	* gst/effectv/gstvertigo.c:
+	* gst/effectv/gstwarp.c:
+	* gst/flv/gstflvdemux.c:
+	* gst/flv/gstflvmux.c:
+	* gst/goom/gstgoom.c:
+	* gst/goom2k1/gstgoom.c:
+	* gst/icydemux/gsticydemux.c:
+	* gst/id3demux/gstid3demux.c:
+	* gst/imagefreeze/gstimagefreeze.c:
+	* gst/interleave/deinterleave.c:
+	* gst/interleave/interleave.c:
+	* gst/isomp4/gstrtpxqtdepay.c:
+	* gst/isomp4/qtdemux.c:
+	* gst/law/alaw-decode.c:
+	* gst/law/alaw-encode.c:
+	* gst/law/mulaw-decode.c:
+	* gst/law/mulaw-encode.c:
+	* gst/level/gstlevel.c:
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/matroska-parse.c:
+	* gst/matroska/webm-mux.c:
+	* gst/monoscope/gstmonoscope.c:
+	* gst/multifile/gstmultifilesink.c:
+	* gst/multifile/gstmultifilesrc.c:
+	* gst/multifile/gstsplitfilesrc.c:
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsrc.c:
+	* gst/multipart/multipartdemux.c:
+	* gst/multipart/multipartmux.c:
+	* gst/replaygain/gstrganalysis.c:
+	* gst/replaygain/gstrglimiter.c:
+	* gst/replaygain/gstrgvolume.c:
+	* gst/rtp/gstasteriskh263.c:
+	* gst/rtp/gstrtpL16depay.c:
+	* gst/rtp/gstrtpL16pay.c:
+	* gst/rtp/gstrtpL24depay.c:
+	* gst/rtp/gstrtpL24pay.c:
+	* gst/rtp/gstrtpac3depay.c:
+	* gst/rtp/gstrtpac3pay.c:
+	* gst/rtp/gstrtpamrdepay.c:
+	* gst/rtp/gstrtpamrpay.c:
+	* gst/rtp/gstrtpbvdepay.c:
+	* gst/rtp/gstrtpbvpay.c:
+	* gst/rtp/gstrtpceltdepay.c:
+	* gst/rtp/gstrtpceltpay.c:
+	* gst/rtp/gstrtpdvdepay.c:
+	* gst/rtp/gstrtpdvpay.c:
+	* gst/rtp/gstrtpg722depay.c:
+	* gst/rtp/gstrtpg722pay.c:
+	* gst/rtp/gstrtpg723depay.c:
+	* gst/rtp/gstrtpg723pay.c:
+	* gst/rtp/gstrtpg726depay.c:
+	* gst/rtp/gstrtpg726pay.c:
+	* gst/rtp/gstrtpg729depay.c:
+	* gst/rtp/gstrtpg729pay.c:
+	* gst/rtp/gstrtpgsmdepay.c:
+	* gst/rtp/gstrtpgsmpay.c:
+	* gst/rtp/gstrtpgstdepay.c:
+	* gst/rtp/gstrtpgstpay.c:
+	* gst/rtp/gstrtph261depay.c:
+	* gst/rtp/gstrtph261pay.c:
+	* gst/rtp/gstrtph263depay.c:
+	* gst/rtp/gstrtph263pay.c:
+	* gst/rtp/gstrtph263pdepay.c:
+	* gst/rtp/gstrtph263ppay.c:
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtpilbcdepay.c:
+	* gst/rtp/gstrtpilbcpay.c:
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kpay.c:
+	* gst/rtp/gstrtpjpegdepay.c:
+	* gst/rtp/gstrtpjpegpay.c:
+	* gst/rtp/gstrtpklvdepay.c:
+	* gst/rtp/gstrtpklvpay.c:
+	* gst/rtp/gstrtpmp1sdepay.c:
+	* gst/rtp/gstrtpmp2tdepay.c:
+	* gst/rtp/gstrtpmp2tpay.c:
+	* gst/rtp/gstrtpmp4adepay.c:
+	* gst/rtp/gstrtpmp4apay.c:
+	* gst/rtp/gstrtpmp4gdepay.c:
+	* gst/rtp/gstrtpmp4gpay.c:
+	* gst/rtp/gstrtpmp4vdepay.c:
+	* gst/rtp/gstrtpmp4vpay.c:
+	* gst/rtp/gstrtpmpadepay.c:
+	* gst/rtp/gstrtpmpapay.c:
+	* gst/rtp/gstrtpmparobustdepay.c:
+	* gst/rtp/gstrtpmpvdepay.c:
+	* gst/rtp/gstrtpmpvpay.c:
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtppcmadepay.c:
+	* gst/rtp/gstrtppcmapay.c:
+	* gst/rtp/gstrtppcmudepay.c:
+	* gst/rtp/gstrtppcmupay.c:
+	* gst/rtp/gstrtpqcelpdepay.c:
+	* gst/rtp/gstrtpqdmdepay.c:
+	* gst/rtp/gstrtpsbcdepay.c:
+	* gst/rtp/gstrtpsbcpay.c:
+	* gst/rtp/gstrtpsirendepay.c:
+	* gst/rtp/gstrtpsirenpay.c:
+	* gst/rtp/gstrtpspeexdepay.c:
+	* gst/rtp/gstrtpspeexpay.c:
+	* gst/rtp/gstrtpstreamdepay.c:
+	* gst/rtp/gstrtpstreampay.c:
+	* gst/rtp/gstrtpsv3vdepay.c:
+	* gst/rtp/gstrtptheoradepay.c:
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtpvorbisdepay.c:
+	* gst/rtp/gstrtpvorbispay.c:
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtp/gstrtpvp9depay.c:
+	* gst/rtp/gstrtpvp9pay.c:
+	* gst/rtp/gstrtpvrawdepay.c:
+	* gst/rtp/gstrtpvrawpay.c:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpdtmfmux.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpptdemux.c:
+	* gst/rtpmanager/gstrtprtxqueue.c:
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	* gst/rtsp/gstrtpdec.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/shapewipe/gstshapewipe.c:
+	* gst/smpte/gstsmpte.c:
+	* gst/smpte/gstsmptealpha.c:
+	* gst/udp/gstdynudpsink.c:
+	* gst/udp/gstmultiudpsink.c:
+	* gst/udp/gstudpsrc.c:
+	* gst/videobox/gstvideobox.c:
+	* gst/videocrop/gstaspectratiocrop.c:
+	* gst/videocrop/gstvideocrop.c:
+	* gst/videofilter/gstgamma.c:
+	* gst/videofilter/gstvideobalance.c:
+	* gst/videofilter/gstvideoflip.c:
+	* gst/videofilter/gstvideomedian.c:
+	* gst/videomixer/videomixer2.c:
+	* gst/wavenc/gstwavenc.c:
+	* gst/wavparse/gstwavparse.c:
+	* gst/y4m/gsty4mencode.c:
+	* sys/directsound/gstdirectsoundsink.c:
+	* sys/oss/gstosssink.c:
+	* sys/oss/gstosssrc.c:
+	* sys/osxaudio/gstosxaudiosink.c:
+	* sys/osxaudio/gstosxaudiosrc.c:
+	* sys/osxvideo/osxvideosink.m:
+	* sys/sunaudio/gstsunaudiosink.c:
+	* sys/sunaudio/gstsunaudiosrc.c:
+	* sys/waveform/gstwaveformsink.c:
+	* sys/ximage/gstximagesrc.c:
+	* tests/check/elements/autodetect.c:
+	* tests/check/elements/qtmux.c:
+	  good: use new gst_element_class_add_static_pad_template()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763076
+
+2016-03-04 09:42:44 +0100  David Buchmann <david.buchmann@gmail.com>
+
+	* tests/check/elements/flvmux.c:
+	  flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762207
+
+2015-11-04 14:51:19 +0900  Jihae Yi <jihae.yi@samsung.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: avoid potentially overflowing expression
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757569
+
+2016-03-22 10:43:45 +0900  Jimmy Ohn <yongjin.ohn@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Add the function to get channels and sample rate for AAC
+	  Add aac_get_channels and sample_rate function to get these value for
+	  AAC.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=749110
+
+2016-03-24 13:33:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 12:27:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.8.0
+
+2016-03-24 12:02:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2016-03-16 20:18:41 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/interleave/deinterleave.c:
+	  deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time
+	  Doing queries while holding the object lock is a bit dangerous, and in this
+	  case causes deadlocks.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-17 20:53:27 +0200  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Fix typo to not change the input caps but our filtered caps
+	  Changing the input caps and not using them anymore afterwards is useless, and
+	  it breaks negotiation in pipelines like:
+	  gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
+	  deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
+	  fakesink
+
+=== release 1.7.91 ===
+
+2016-03-15 12:04:39 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.7.91
+
+2016-03-15 11:53:37 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2016-03-15 11:41:22 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/hu.po:
+	* po/sr.po:
+	  po: Update translations
+
+2016-03-15 03:26:14 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/rtpsource.c:
+	  rtpmanager: Some comment and documentation clarifications/fixes
+
+2016-03-13 10:33:13 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  Revert "flacparse: push tags in pre_push_frame"
+	  This reverts commit 4065fcb80a49924b70f0c8fc159dec0ff47943a1.
+	  flacparse should not push tags by itself, the base class is going to do that
+	  while properly merging in upstream tags. It just didn't because of a bug in
+	  the base class, which was hidden by this commit.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763553
+
+2016-02-25 05:17:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtp/dboolhuff.c:
+	* gst/rtp/dboolhuff.h:
+	* gst/rtp/gstrtpsbcpay.c:
+	  win32: Don't use __attribute__ on MSVC
+	  Use MSVC-equivalents for alignment and packing compiler directives when building
+	  on MSVC
+
+2016-02-25 05:16:42 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/matroska/ebml-read.c:
+	  win32: Don't try to include xmath.h on newer Visual Studio
+
+2016-02-25 05:16:09 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/flx/gstflxdec.c:
+	* gst/goom/gstgoom.c:
+	* gst/goom2k1/gstgoom.c:
+	* gst/monoscope/gstmonoscope.c:
+	  gst Factor out endian-order RGB formats
+	  MSVC seems to ignore preprocessor conditionals inside static pad
+	  template macros.
+
+2016-03-08 17:37:17 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	  dirctsoundsink: Setting volume should not unmute
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:57:24 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	  dirctsoundsink: Fix volume reset on unmute
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:03:55 +0100  Alban Bedel <alban.bedel@avionic-design.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: fix capture with bayer formats other than bggr
+	  gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8
+	  for all bayer formats. This is obviously broken if the device use
+	  another ordering. Fix this by properly reading the format parameter.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763318
+
+2016-03-07 10:28:06 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: reset pending segment if we are already pushing one
+	  When upstream is running in bytes in push-mode, qtdemux will
+	  convert seeks from time to bytes and send it upstream. Upstream
+	  element will perform a byte seek and send a byte segment to qtdemux
+	  that will convert it to time and push it downstream.
+	  There is, however, the pending_segment variable that stores a new
+	  segment event to be pushed before the next data. When handling seeks
+	  as mentioned above this variable was being ignored and, if it contained
+	  some segment event, it would override the one resulting from the seek.
+	  This would restore a previous segment and would cause the seek segment
+	  to be discarded downstream.
+	  This patch fixes this issue by unrefing any pending segment as the
+	  seek from upstream should contain the latest one that should be
+	  used, as requested by the application.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763165
+
+2016-03-07 10:27:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: run gst-indent
+	  Otherwise commits will fail with our indent check hook
+
+2016-03-04 15:09:45 +0100  Josep Torra <n770galaxy@gmail.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: fix colorimetry for NV12
+	  Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior.
+	  At #v4l it was described that documentation might be wrong and that
+	  we should trust this macro instead.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762529
+
+2016-03-04 14:07:19 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Fix multicast group joining with provided sockets on Windows
+	  On Windows the socket will be bound to ANY instead of the multicast group,
+	  as binding to a multicast group does not work. Which would mean that we
+	  override src->addr to become ANY and won't automatically join a multicast
+	  group anymore on Windows.
+	  On Linux we would automatically join a multicast group, keep it consistent.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=763093
+
+2016-03-02 13:13:24 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases"
+	  This reverts commit a7fb7b53592d87f7983544debb74d364fc3257ad.
+	  The mutex is taken by the caller, we should keep it locked when returning so
+	  the caller can unlock it again.
+
+2016-03-01 15:01:22 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: push tags in pre_push_frame
+	  Push a tag event before pre-roll if we have tags.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762660
+
+=== release 1.7.90 ===
+
+2016-03-01 18:15:43 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.7.90
+
+2016-03-01 17:03:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/ca.po:
+	* po/da.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/or.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/tr.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2016-03-01 16:53:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/bg.po:
+	* po/cs.po:
+	* po/de.po:
+	* po/fr.po:
+	* po/nl.po:
+	* po/pl.po:
+	* po/ru.po:
+	* po/sv.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	  po: Update translations
+
+2016-03-01 14:14:02 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases
+
+2016-02-29 10:10:24 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: remove impossible condition
+	  It is impossible for a guint to have a negative value, no need to check for
+	  this. Introduced in commit 6861d11c49ea0f30d2432cf4ebf6108bc89897f1
+	  CID 1354509
+
+2016-02-28 10:12:36 +0100  Petr Viktorin <encukou@gmail.com>
+
+	* gst/alpha/gstalpha.c:
+	  alpha: Fix sample pipeline
+	  Use the zorder pad property to make sure the semitransparent
+	  video is on top of the background.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762809
+
+2016-02-28 13:42:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/replaygain/gstrgvolume.c:
+	* tests/check/elements/rgvolume.c:
+	  rgvolume: make tag list writable before modifying it
+	  Making the event itself writable is not enough, it won't make
+	  the actual taglist in the event writable as well. Instead, just
+	  make a copy of the taglist and then create a new tag event from
+	  that if required, replacing the old one. Before we would
+	  inadvertently modify taglists upstream elements might still
+	  be holding on to. Add unit test for this as well.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762793
+
+2016-02-28 13:01:34 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Properly error out if binding the UDP sockets fails
+	  udpsrc is not returning us a socket in that case.
+
+2016-02-27 20:33:32 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/goom/gstgoom.c:
+	  goom: Use goom_set_resolution() instead of recreating the goom instance when the resolution changes
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762765
+
+2016-02-27 20:32:45 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/goom/gstgoom.c:
+	  Revert "goom: Initialize the goom struct only once we know width/height and recreate it if those change"
+	  This reverts commit cc6e102643c1bae928316dca9f34db028fb9a67e.
+
+2016-02-27 20:31:15 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/goom/gstgoom.c:
+	  goom: Initialize the goom struct only once we know width/height and recreate it if those change
+	  Fixes crash when the width and/or height is changing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762765
+
+2016-02-26 12:41:07 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From b64f03f to 6f2d209
+
+2016-02-25 22:54:18 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	  docs: add rtpopusdepay and rtpopuspay to documentation
+
+2016-02-17 15:15:11 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopusdepay.h:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	  rtp: opus: move Opus RTP payloader/depayloader from -bad to -good
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756282
+
+2016-02-17 15:10:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	  Merge branch 'plugin-move-rtp-opus'
+	  Move Opus RTP depayloader/payloader from -bad to -good.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756282
+
+2016-02-25 11:33:13 +0100  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: cenc aux info parsing from mdat support in PULL mode
+	  This is already supported for PUSH mode but was failing in PULL mode.
+	  The aux info is sometimes stored in the mdat before the first sample,
+	  so the loop task needs to pull data stored at that location and
+	  perform the aux info cenc parsing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761700
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2016-02-24 11:28:09 +0100  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: prevent buffer flow if any stream failed to be exposed
+	  In some cases the stream configuration can fail, for instance if the
+	  stream is protected and no decryptor was found. For those situations
+	  the demuxer shouldn't emit any data on the corresponding source pad of
+	  the stream and bail out.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2016-02-24 09:12:03 +0100  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: don't push encrypted buffer without cenc metadata
+	  When the cenc metadata is stored outside of the moof box and the
+	  stream is exposed it is possible that the cenc metadata hasn't been
+	  processed yet while the first buffer is being pushed. When this
+	  happens the buffer can't possibly be decrypted downstream so don't
+	  push it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2015-10-21 16:21:45 +0200  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: read saio aux_info_type as a FOURCC
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756897
+
+2016-02-23 18:27:47 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/dv/gstdvdec.c:
+	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/smpte/gstsmpte.c:
+	  gst: Handle gst_pad_get_current_caps() returning NULL gracefully
+
+2016-02-23 18:12:54 +0200  Dave Craig <dcraig@brightsign.biz>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
+	  Remove calls to gst_pad_has_current_caps() which then go on to call
+	  gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
+	  use gst_pad_get_current_caps() and check for NULL.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2015-12-16 12:40:39 +0000  Dave Craig <dcraig@brightsign.biz>
+
+	* ext/flac/gstflacenc.c:
+	* gst/flv/gstflvmux.c:
+	* gst/imagefreeze/gstimagefreeze.c:
+	* gst/rtp/gstrtph264depay.c:
+	* gst/shapewipe/gstshapewipe.c:
+	* gst/videocrop/gstaspectratiocrop.c:
+	  gst: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
+	  Remove calls to gst_pad_has_current_caps() which then go on to call
+	  gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
+	  use gst_pad_get_current_caps() and check for NULL.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2015-12-16 10:54:17 +0000  Dave Craig <dcraig@brightsign.biz>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: Handle gst_pad_get_current_caps() returning NULL gracefully
+	  This can happen when the pipeline is currently shutting down.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2016-02-23 15:57:18 +0100  Linus Svensson <linussn@axis.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: Don't handle seek until ready
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762542
+
+2016-02-23 15:55:13 +0100  Linus Svensson <linussn@axis.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: Unref seek event
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762542
+
+2016-02-22 11:01:40 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* gst/multifile/gstmultifilesink.c:
+	  multifilesink: close file on write error with next-file mode is set to buffer
+	  If we have an error during fwrite call, file stays open and thus next
+	  incoming buffer will trigger an assert when trying to opening a new
+	  file.
+	  This happens if we do not restart element, file is closed at stop, and
+	  if application handles the returned GST_FLOW_ERROR to keep bin alive.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762434
+
+2016-02-19 23:44:42 +0100  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: don't output empty tags/tag elements
+	  Such files will not play on Android, because of bug in libwebm matroska parsing, which is still present in 6.0.1
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762349
+
+2016-02-04 15:59:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: make up an OpusHead block if possible when missing
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761489
+
+2016-02-04 10:43:15 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: make up an OpusHead block if possible when missing
+	  This block is needed in the Matroska file, but data coming from
+	  RTP may not have one.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761489
+
+2016-02-22 13:53:21 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: make stream-id more readable and order-friendly
+	  ... as streams are so ordered by id by e.g. decodebin
+	  (and as typically already honoured by other demuxers).
+
+2016-02-22 13:25:51 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-ids.h:
+	* gst/matroska/matroska-mux.c:
+	  matroska: remove confusing duplicate track uid field
+
+2016-02-22 14:03:02 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtpvp9pay.c:
+	  rtpvp9pay: add missing break
+	  VP9_PAY_PICTURE_ID_7BITS and VP9_PAY_PICTURE_ID_15BITS are mutually
+	  exclusive options of the picture-id-mode. We can break after the
+	  first case.
+	  1 or 2 bytes need to be added to the header length depending on the
+	  PictureID size.
+	  https://tools.ietf.org/html/draft-uberti-payload-vp9-00#section-4.2
+	  CID 1353479
+
+2016-02-22 09:09:01 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Fix buffer memory leak
+	  buffer being mapped is not being unmapped in some cases
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762420
+
+2015-11-04 10:19:03 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpmanager: Don't warn for duplicate/reordered packets
+	  This is a normal scenario and should not be a warning.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762208
+
+2016-02-21 09:47:43 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/alpha/alpha.vcproj:
+	* gst/auparse/auparse.vcproj:
+	* gst/avi/avi.vcproj:
+	* gst/cutter/cutter.vcproj:
+	* gst/debugutils/debug.vcproj:
+	* gst/debugutils/navigationtest.vcproj:
+	* gst/effectv/effectv.vcproj:
+	* gst/flx/flxdec.vcproj:
+	* gst/goom/goom.vcproj:
+	* gst/goom2k1/goom.vcproj:
+	* gst/interleave/interleave.vcproj:
+	* gst/isomp4/qtdemux.vcproj:
+	* gst/law/alaw.vcproj:
+	* gst/law/mulaw.vcproj:
+	* gst/matroska/matroska.vcproj:
+	* gst/multipart/multipart.vcproj:
+	* gst/rtp/rtp.vcproj:
+	* gst/smpte/smpte.vcproj:
+	* gst/spectrum/spectrum.vcproj:
+	* gst/udp/udp.vcproj:
+	* gst/videobox/videobox.vcproj:
+	* gst/videocrop/videocrop.vcproj:
+	* gst/videofilter/gamma.vcproj:
+	* gst/videofilter/videobalance.vcproj:
+	* gst/videofilter/videofilter.vcproj:
+	* gst/videofilter/videoflip.vcproj:
+	* gst/videomixer/videomixer.vcproj:
+	* gst/wavenc/wavenc.vcproj:
+	* gst/wavparse/wavparse.vcproj:
+	* gst/y4m/y4menc.vcproj:
+	* win32/MANIFEST:
+	* win32/vs6/autogen.dsp:
+	* win32/vs6/gst_plugins_good.dsw:
+	* win32/vs6/libgstalaw.dsp:
+	* win32/vs6/libgstalpha.dsp:
+	* win32/vs6/libgstalphacolor.dsp:
+	* win32/vs6/libgstapetag.dsp:
+	* win32/vs6/libgstaudiofx.dsp:
+	* win32/vs6/libgstauparse.dsp:
+	* win32/vs6/libgstautodetect.dsp:
+	* win32/vs6/libgstavi.dsp:
+	* win32/vs6/libgstcutter.dsp:
+	* win32/vs6/libgstdirectsound.dsp:
+	* win32/vs6/libgsteffectv.dsp:
+	* win32/vs6/libgstflx.dsp:
+	* win32/vs6/libgstgoom.dsp:
+	* win32/vs6/libgsticydemux.dsp:
+	* win32/vs6/libgstid3demux.dsp:
+	* win32/vs6/libgstinterleave.dsp:
+	* win32/vs6/libgstjpeg.dsp:
+	* win32/vs6/libgstlevel.dsp:
+	* win32/vs6/libgstmatroska.dsp:
+	* win32/vs6/libgstmedian.dsp:
+	* win32/vs6/libgstmonoscope.dsp:
+	* win32/vs6/libgstmulaw.dsp:
+	* win32/vs6/libgstmultipart.dsp:
+	* win32/vs6/libgstpng.dsp:
+	* win32/vs6/libgstqtdemux.dsp:
+	* win32/vs6/libgstrtp.dsp:
+	* win32/vs6/libgstrtsp.dsp:
+	* win32/vs6/libgstsmpte.dsp:
+	* win32/vs6/libgstspeex.dsp:
+	* win32/vs6/libgstudp.dsp:
+	* win32/vs6/libgstvideobalance.dsp:
+	* win32/vs6/libgstvideobox.dsp:
+	* win32/vs6/libgstvideocrop.dsp:
+	* win32/vs6/libgstvideoflip.dsp:
+	* win32/vs6/libgstvideomixer.dsp:
+	* win32/vs6/libgstwaveform.dsp:
+	* win32/vs6/libgstwavenc.dsp:
+	* win32/vs6/libgstwavparse.dsp:
+	* win32/vs7/libgstdirectsound.vcproj:
+	* win32/vs8/gst-plugins-good.sln:
+	* win32/vs8/libgst1394.vcproj:
+	* win32/vs8/libgstaasink.vcproj:
+	* win32/vs8/libgstalaw.vcproj:
+	* win32/vs8/libgstalpha.vcproj:
+	* win32/vs8/libgstalphacolor.vcproj:
+	* win32/vs8/libgstannodex.vcproj:
+	* win32/vs8/libgstapetag.vcproj:
+	* win32/vs8/libgstaudiofx.vcproj:
+	* win32/vs8/libgstauparse.vcproj:
+	* win32/vs8/libgstautodetect.vcproj:
+	* win32/vs8/libgstavi.vcproj:
+	* win32/vs8/libgstcacasink.vcproj:
+	* win32/vs8/libgstcdio.vcproj:
+	* win32/vs8/libgstcutter.vcproj:
+	* win32/vs8/libgstdirectsound.vcproj:
+	* win32/vs8/libgstdv.vcproj:
+	* win32/vs8/libgsteffectv.vcproj:
+	* win32/vs8/libgstflac.vcproj:
+	* win32/vs8/libgstflxdec.vcproj:
+	* win32/vs8/libgstgoom.vcproj:
+	* win32/vs8/libgsticydemux.vcproj:
+	* win32/vs8/libgstid3demux.vcproj:
+	* win32/vs8/libgstjpeg.vcproj:
+	* win32/vs8/libgstladspa.vcproj:
+	* win32/vs8/libgstlevel.vcproj:
+	* win32/vs8/libgstmatroska.vcproj:
+	* win32/vs8/libgstmng.vcproj:
+	* win32/vs8/libgstmonoscope.vcproj:
+	* win32/vs8/libgstmulaw.vcproj:
+	* win32/vs8/libgstmultipart.vcproj:
+	* win32/vs8/libgstpng.vcproj:
+	* win32/vs8/libgstrtp.vcproj:
+	* win32/vs8/libgstrtsp.vcproj:
+	* win32/vs8/libgstshout2.vcproj:
+	* win32/vs8/libgstsmpte.vcproj:
+	* win32/vs8/libgstspeex.vcproj:
+	* win32/vs8/libgsttaglib.vcproj:
+	* win32/vs8/libgstudp.vcproj:
+	* win32/vs8/libgstvideobalance.vcproj:
+	* win32/vs8/libgstvideobox.vcproj:
+	* win32/vs8/libgstvideoflip.vcproj:
+	* win32/vs8/libgstvideomixer.vcproj:
+	* win32/vs8/libgstwavenc.vcproj:
+	* win32/vs8/libgstwavparse.vcproj:
+	  win32: remove outdated build cruft
+	  This hasn't been touched for generations, doesn't work,
+	  and is just causing confusion. We also don't want to
+	  maintain these files manually.
+
+2016-02-20 11:51:56 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2: don't use undeclared core debug category symbols
+
+2016-02-06 14:39:05 +0100  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: workaround for files with wrong color_table_id value
+	  Instead of erroring out, just use the default color table.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761637
+
+2016-02-19 15:02:04 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvmux.c:
+	* gst/rtp/gstrtpvp9depay.c:
+	  flvmux, rtpvp9depay: fix indentation
+
+2016-02-19 15:03:04 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2src: fix indentation
+
+2015-12-04 00:46:34 +1100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: plug leak(s) in error-scenario
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762210
+
+2015-12-04 00:46:12 +1100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: fix eos event leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762209
+
+2016-02-19 14:41:07 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/flvdemux.c:
+	* tests/check/elements/flvmux.c:
+	* tests/check/elements/rtph263.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  tests: fix indentation
+
+2016-02-18 16:09:29 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  tests: rtpjitterbuffer: port testharness to GstHarness and cleanup/improve
+	  Probably found a bug as well, in that there are some timestamps in
+	  there that are looking very wrong. (marked with FIXME)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762267
+
+2016-02-18 10:27:19 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  tests: rtpjitterbuffer: test cleanups/improvements
+	  Use fail_unless and friends instead of g_assert
+	  Factor seq-num checking out to separate function
+	  Check more return-values from push and crank and others
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762254
+
+2015-12-03 11:07:05 +0100  Stian Selnes <stian@pexip.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  tests: rtpjitterbuffer: fix leaks in unit test
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762214
+
+2016-02-19 12:38:28 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.7.2 ===
+
+2016-02-19 11:49:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/gst-plugins-good-plugins.prerequisites:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.7.2
+
+2016-02-19 10:31:48 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  po: Update translations
+
+2016-02-18 18:33:13 +0100  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: plug leaks in cenc aux info parsing
+
+2016-02-18 13:43:07 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	  tests: fix spurious souphttpsrc test timouts
+	  Set GSETTINGS_BACKEND=memory, apparently there's something
+	  about fork() and the dconf backend (or whatever else that
+	  drags in or activates) that messes up locking and causes
+	  timeouts due to deadlocks in g_mutex_lock(), since
+	  everything works fine with CK_FORK=no as well.
+
+2016-02-18 11:10:14 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Unmap wavpack header buffer after creating it
+	  Otherwise it will be mapped writable all the time and we can't read from it
+	  anywhere.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762239
+
+2015-12-08 18:49:40 +0100  Stian Selnes <stian@pexip.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Add test for big seqnum gap handling
+	  Make sure that the packets queued when detecting a big gap are pushed
+	  after reset (5 consective seqnums) and not dropped.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762211
+
+2016-02-17 15:03:13 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtputils.h:
+	  rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions
+
+2016-02-09 13:17:00 +0000  Alex Ashley <bugzilla@ashley-family.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: only transform protected caps once
+	  Commit 7873bede3134b15e5066e8d14e54d1f5054d2063
+	  (https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the
+	  behaviour of qtdemux to call gst_qtdemux_configure_stream() for
+	  every new moof.
+	  When playing a protected stream, gst_qtdemux_configure_stream()
+	  calls gst_qtdemux_configure_protected_caps(). The
+	  gst_qtdemux_configure_protected_caps() function takes the original
+	  media format, puts this in a field called "original-media-type"
+	  and then changes the caps to "application/x-cenc".
+	  The gst_qtdemux_configure_protected_caps() did not handle the case
+	  of being called multiple times, causing it to incorrectly set the
+	  caps. The second call was causing the caps to be set to:
+	  application/x-cenc, original-media-type"application/x-cenc"
+	  This commit makes gst_qtdemux_configure_protected_caps() check that
+	  the caps have already been transformed, so that it only gets
+	  changed once.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761769
+
+2015-11-03 14:50:53 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	  opus: Add proper support for multichannel audio
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-06-30 13:51:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	  opus: Copy metadata in the (de)payloader, but only the relevant ones
+	  The payloader didn't copy anything so far, the depayloader copied every
+	  possible meta. Let's make it consistent and just copy all metas without tags or
+	  with only the audio tag.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-05-04 11:23:16 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	  opusdepay: Set multistream=FALSE on the Opus caps
+	  The RTP Opus mapping only allows mono/stereo, and not multistream Opus
+	  streams.
+
+2015-03-24 13:57:54 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: Forward stereo preferences from caps upstream
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-24 13:56:21 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: Set the number of channels to 2 as per RFC draft
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-23 12:24:55 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	  opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-02-19 14:30:10 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: default encoding name to OPUS
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 14:05:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: make caps writable before truncating them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-05 10:27:51 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: negotiate the encoding name
+	  Chrome uses a different encoding name that gstreamer.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2014-11-01 10:10:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopus: Use OPUS encoding name
+	  Both Firefox and Chrome uses OPUS as the encoding in their SDP.
+	  Adding this now defacto standard name remove the need for special
+	  case in SDP parsing code.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2013-01-31 12:30:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  opuspay: fix timestamps
+	  Copy timestamps to payloaded buffer.
+	  Avoid input buffer memory leak.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929
+
+2012-11-03 20:38:00 +0000  Tim-Philipp Müller <tim@centricular.net>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopusdepay.h:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	  Fix FSF address
+	  https://bugzilla.gnome.org/show_bug.cgi?id=687520
+
+2012-10-22 12:08:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  opuspay: remove pointless caps serialization
+	  Remove the caps serialization in the rtp caps. the spec nor the receiver
+	  does anything with it.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547
+
+2012-10-17 17:34:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	  Use gst_element_class_set_static_metadata()
+	  where possible. Avoids some string copies. Also re-indent
+	  some stuff. Also some indent fixes here and there.
+
+2012-09-20 18:41:24 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: Allocate the rtp buffer correctly
+	  Use the right functions to allocate the rtp buffer
+
+2012-09-14 17:08:49 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopuspay.c:
+	  replace gst_element_class_set_details_simple with gst_element_class_set_metadata
+
+2012-03-07 17:14:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  opus: port to updated 0.11
+
+2011-12-30 11:41:17 +0100  Edward Hervey <edward.hervey@collabora.co.uk>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopusdepay.h:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	  Merge remote-tracking branch 'origin/master' into 0.11-premerge
+	  Conflicts:
+	  docs/libs/Makefile.am
+	  ext/kate/gstkatetiger.c
+	  ext/opus/gstopusdec.c
+	  ext/xvid/gstxvidenc.c
+	  gst-libs/gst/basecamerabinsrc/Makefile.am
+	  gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
+	  gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
+	  gst-libs/gst/video/gstbasevideocodec.c
+	  gst-libs/gst/video/gstbasevideocodec.h
+	  gst-libs/gst/video/gstbasevideodecoder.c
+	  gst-libs/gst/video/gstbasevideoencoder.c
+	  gst/asfmux/gstasfmux.c
+	  gst/audiovisualizers/gstwavescope.c
+	  gst/camerabin2/gstcamerabin2.c
+	  gst/debugutils/gstcompare.c
+	  gst/frei0r/gstfrei0rmixer.c
+	  gst/mpegpsmux/mpegpsmux.c
+	  gst/mpegtsmux/mpegtsmux.c
+	  gst/mxf/mxfmux.c
+	  gst/videomeasure/gstvideomeasure_ssim.c
+	  gst/videoparsers/gsth264parse.c
+	  gst/videoparsers/gstmpeg4videoparse.c
+
+2011-12-09 17:25:41 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  opusenc: add upstream negotiation for multistream ability
+	  This will help elements that cannot deal with multistream,
+	  such as the RTP payloader.
+	  The caps now do not include a "streams" field anymore, but
+	  a "multistream" boolean, since we have no real use for knowing
+	  the exact amount of streams.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
+2011-12-07 15:13:11 -0200  Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopusdepay.h:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	  Adding opus RTP payloader/depayloader element
+	  Adding OPUS RTP module based on the current draft:
+	  http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt
+	  https://bugzilla.gnome.org/show_bug.cgi?id=664817
+
+2016-02-17 13:26:02 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtputils.c:
+	* gst/rtp/gstrtputils.h:
+	  rtp: h264/h265: avoid duplication of read_golomb()
+	  There is no need to have two identical implementations of the read_golomb
+	  function.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-17 14:37:44 +0100  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Simple implementation of TRICKMODE_KEY_UNITS
+	  When the trickmode key-units flag is set on the segment, simply skip
+	  any sample on a video stream that isn't a keyframe
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762185
+
+2015-08-21 14:15:18 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: send GAP events for lagging audio and video streams too
+	  Send GAP events for non-subtitle streams too if they lag too much
+	  behind, but use a higher threshold than for subtitles.
+	  This helps with fixing prerolling with a file where one of the
+	  audio streams only has data starting from 19s onwards. It's not
+	  a complete fix yet, it also requires changes elsewhere, such as
+	  in baseparse, to make sure caps are propagated.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=614460
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753899
+
+2015-12-23 19:54:13 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpvp9depay.c:
+	* gst/rtp/gstrtpvp9depay.h:
+	* gst/rtp/gstrtpvp9pay.c:
+	* gst/rtp/gstrtpvp9pay.h:
+	  rtpvp9pay: rtpvp9depay: Initial implementation of draft 01
+	  Quick and dirty implementation of an RTP payloader and depayloader
+	  for VP9. In particalur it assumes no spatial or temporal layering,
+	  non-flexible mode, and some other bits and pieces.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754773
+
+2016-02-16 09:02:30 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Fix string memory leak
+	  codec_name is not being freed in all conditions leading to memory leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762117
+
+2015-12-10 12:15:52 +0100  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: add "get-session" signal
+	  This gets the GstRTPSession element, as compared to the RTPSession object
+	  that is returned by get-internal-session.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759293
+
+2016-02-16 00:19:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	  rtp: h265: hook up move RTP H.265 payloader/depayloader to build
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-16 00:14:27 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265depay.h:
+	* gst/rtp/gstrtph265pay.c:
+	  rtp: h265: use common meta utility functions
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-05 18:18:31 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtph265depay.h:
+	* gst/rtp/gstrtph265pay.h:
+	* gst/rtp/gstrtph265types.h:
+	  rtp: h265: remove codecparser dependency from h265 payloader/depayloader
+	  Looks like it just uses the NAL enums and nothing else from
+	  the codecparsers, and that's the only reason it had to be
+	  moved from -good to -bad when it was originally added. We
+	  can probably keep those NAL enums up to date enough, so let's
+	  remove the codecparser dependency so it can be moved back into
+	  -good.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-16 00:24:58 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	  Merge branch 'plugin-move-rtp-h265'
+	  Move RTP H.265 payloader/depayloader from -bad to -good.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-05 15:34:51 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265depay.h:
+	  gstrtph265depay: keep consistency with rtph264depay
+	  Use gst_rtp_drop_meta() and the same function prototype for
+	  gst_rtp_copy_meta() to keep consistency with the RTP elements in
+	  gst-plugins-good
+
+2016-02-05 13:56:34 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: fix termination of access unit
+	  Only consider the access unit complete when the next-occurring VCL NAL unit
+	  has the first bit after its NAL unit header equal to 1.
+
+2016-01-15 16:10:02 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: fix unneeded sub-buffer creation
+	  We create a sub-buffer just to copy over its metas and then throw it
+	  away immediately, just use the original input buffer directly.
+
+2016-01-15 15:56:59 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: add "send VPS/SPS/PPS with every key frame" mode
+	  It's not enough to have timeout or event based VPS/SPS/PPS information
+	  sent in RTP packets. There are some scenarios when key frames may appear
+	  more frequently than once a second, in which case the minimum timeout
+	  for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
+	  It might also be desirable in general to make sure the VPS/SPS/PPS is
+	  available with every keyframe (packet loss aside), so receivers can
+	  actually pick up decoding immediately from the first keyframe if
+	  VPS/SPS/PPS is not signaled out of band.
+	  This commit adds the possibility to send VPS/SPS/PPS with every key frame.
+	  This mode can be enabled by setting "config-interval" property to -1. In
+	  this case the payloader will add VPS, SPS and PPS before every key (IDR)
+	  frame.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2016-01-15 15:19:41 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtph265pay.h:
+	  rtph265pay: change config-interval property type from uint to int
+	  This way we can use -1 as special value, which is nicer than MAXUINT.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-08-15 16:22:20 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: make sure we call handle_nal for each NAL
+	  Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure
+	  we correctly extract the SPS and PPS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=730999
+
+2015-08-15 14:45:34 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: Copy metadata in the payloader, but only the relevant ones
+	  The payloader didn't copy anything so far, the depayloader copied every
+	  possible meta. Let's make it consistent and just copy all metas without
+	  tags or with only the video tag.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-15 11:41:40 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-15 11:30:36 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: fix potential crash when shutting down
+	  A race condition in the state change function may cause buffers to be
+	  unreffed while they are still used by the streaming thread in
+	  gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
+	  parent class first in the state change function to make sure streaming
+	  has stopped and only then free those buffers.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741381
+
+2015-08-14 15:08:08 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtph265pay: fix buffer leak when using SPS/PPS
+	  Fixes a buffer leak that would occur if the pipeline was shutdown while a
+	  SPS/PPS header was being created.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741271
+
+2015-08-14 11:49:51 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265depay.h:
+	  rtph265depay: copy metadata in the depayloader, but only the relevant ones
+	  The payloader didn't copy anything so far, the depayloader copied every
+	  possible meta. Let's make it consistent and just copy all metas without
+	  tags or with only the video tag.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-12 17:54:52 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: checking if depay has sps/pps nals before insertion
+	  Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 17:22:42 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: only update the srcpad caps if something else than the codec_data changed
+	  h264parse and gstrtph264depay do the same, let's keep the behaviour
+	  consistent. As we now include the codec_data inside the stream, this causes
+	  less caps renegotiation.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 16:43:48 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: PPS replaces old PPS if it has the same id
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 16:11:00 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: Insert SPS/PPS NALs into the stream
+	  rtph264depay does the same and this fixes decoding of some streams with 32
+	  SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
+	  but the field in the codec_data for the number of SPS or PPS is only 5
+	  (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+	  This looks like a mistake in the part of the spect about the codec_data.
+
+2015-08-12 15:49:50 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: implement process_rtp_packet() vfunc
+	  For more optimised RTP packet handling: means we don't need to map the
+	  input buffer again but can just re-use the mapping the base class has
+	  already done.
+	  Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 15:14:50 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
+	  Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code.
+
+2015-08-12 14:59:53 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtph265depay: prevent trying to get 0 bytes from adapter
+	  This causes an assertion and would lead to getting a NULL instead
+	  of a buffer. Without proper checking this would easily lead to a
+	  segfault.
+	  Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199
+
+2015-07-29 17:29:28 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtp: remove dead assignment
+	  Value set to ret will be overwritten at least once at the end of the while
+	  loop, removing assignment.
+
+2015-04-24 16:48:23 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  remove unused enum items PROP_LAST
+	  This were probably added to the enums due to cargo cult programming and are
+	  unused.
+
+2015-03-06 14:54:41 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtp: donl_present variable unused
+	  donl_present is not implemented, yet the value is set and checked a few times.
+	  Cleaning this.
+	  CID #1249687
+
+2015-01-08 15:36:04 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/rtp/gstrtph265pay.c:
+	  rtp: value truncated too short creates dead code
+	  type is truncated to 0-31 with "& 0x1f", but right after that it is checks if
+	  the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and
+	  GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will
+	  never be True if the value is maximum 31 after the truncation.
+	  The intention of the code was to truncate to 0-63.
+
+2015-01-08 15:27:44 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtp: fix nal unit type check
+	  After further investigation the previous commit is wrong. The code intended to
+	  check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c
+	  does. Type 40 would not be complete.
+
+2015-01-08 13:47:09 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	  rtp: fix dead code and check for impossible values
+	  nal_type is the index for a GstH265NalUnitType enum. There are two types of dead
+	  code here:
+	  First, after checking if nal_type is >= 39 there are two OR conditionals that
+	  check if the value is in ranges higher than that number, so if nal_type >= 39
+	  falls in the True branch those other conditions aren't checked and if it falls
+	  in the False branch and they are checked, they will always also be False. They
+	  are redundant.
+	  Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41
+	  should never be True.
+	  Removing this redundant checks.
+	  CID 1249684
+
+2014-10-16 10:34:01 +0200  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265depay.h:
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtph265pay.h:
+	  rtp: add h265 RTP payloader + depayloader
+
+2016-02-15 11:51:46 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* tests/check/elements/rtpmux.c:
+	  tests: rtpmux: Fix element memory leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=762057
+
+2016-02-12 20:57:29 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/monoscope/monoscope.c:
+	  monoscope: rework the scaling code
+	  The running average was wrong and the resulting scaling factor was only held in
+	  place using the CLAMP. In addtion we are now convering quickly to volume
+	  changes.
+	  FInally now with this change, we can change the resolution defines and
+	  everythign adjusts.
+
+2016-01-28 17:00:55 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/monoscope/convolve.c:
+	* gst/monoscope/monoscope.c:
+	* gst/monoscope/monoscope.h:
+	  monoscope: use constants in the drawing code
+	  Make all the drawing ops be based on the constants. This way we can change
+	  the fixed size at least at compile time.
+
+2016-01-28 09:51:17 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/monoscope/gstmonoscope.c:
+	  monoscope: replace hardcoded values by constants
+	  This at least establishes the relationship.
+
+2016-01-28 09:43:12 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/monoscope/convolve.c:
+	* gst/monoscope/convolve.h:
+	* gst/monoscope/monoscope.c:
+	* gst/monoscope/monoscope.h:
+	  monoscpe: make the convolver use dynamic memory
+	  Replace all #defines with members and initialize the convolver with a parameter.
+
+2016-01-28 08:56:44 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/monoscope/README:
+	  monoscope: update README
+	  We can already create multiple instances.
+
+2016-01-28 08:53:35 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/monoscope/convolve.c:
+	* gst/monoscope/monoscope.c:
+	  monoscope: code cleanup
+	  Use constants more often. Cleanup comments and add more to explain how things
+	  work.
+
+2016-02-08 23:41:32 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: remove check for impossible condition
+	  Commit bd27a1f30b4458f2edee53c76dd07fb35904b61d added a few error handling
+	  memory management checks. These check srccaps to see if it needs to be
+	  unreferenced before returning, in the case of invalid_caps this goto jump
+	  always happens before srccaps is set, so it will always be NULL in this
+	  error label.
+	  CID #1352035
+
+2016-02-08 12:48:46 +0100  Piotr Drąg <piotrdrag@gmail.com>
+
+	* po/POTFILES.in:
+	  po: update POTFILES
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761705
+
+2016-02-08 15:31:55 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2allocator.c:
+	  v4l2allocator: Fix spelling of reenqueueing
+	  To match commit 7d7074cef0272cd5155098bfc2bda6849dd89267. I love the idea
+	  of aiming for the maximum number of consecutive vowels.
+
+2016-02-08 10:17:49 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2allocator.c:
+	  v4l2allocator: Fix spelling of queueing
+	  Didn't know which one to choose between queuing and queueing, so I picked
+	  the one with the biggest amount of vowels in a row ;-P (both are
+	  acceptable apparently)
+
+2016-02-07 15:02:35 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: Don't pass the same data over and over
+	  We already pass the entire frame to the decoder. If the decoder ask for
+	  more data, don't pass the same data again as this leads to infinit loop.
+	  Instead, simply fail the fill function to signal the problem with that
+	  frame. It will then be skipped properly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761670
+
+2016-02-08 00:10:33 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/lzo.c:
+	  matroska: get rid of _stdint.h include
+
+2016-02-05 20:00:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/Makefile.am:
+	  tests: extend the AM_TESTS_ENVIRONMENT from check.mak
+	  To get the CK_DEFAULT_TIMEOUT defined for all tests
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761472
+
+2016-02-05 18:04:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From 86e4663 to b64f03f
+
+2016-01-30 18:43:30 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpjpegpay.c:
+	  rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers
+	  For APP/JPG markers the size is following and we have to skip that. This is
+	  not really a problem unless the marker contains e.g. a preview JPEG or
+	  something else that we might interprete as another marker.
+
+2016-01-26 22:37:30 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix framerate calculation for fragmented format
+	  qtdemux calculates framerate using duration and the number of sample.
+	  In case of fragmented mp4 format, however, the number of sample can
+	  be figure out after parsing every moof box. Because qtdemux does not
+	  parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
+	  framerate calculation.
+	  This patch will triger gst_qtdemux_configure_stream() for every new moof.
+	  Then, framerate will be calculated by using duration and n_samples of the moof.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760774
+
+2016-01-28 22:36:23 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: handling zero segment-duration edit list
+	  Based on document ISO_IEC_14496-12, edit list box can have
+	  segment duration as zero. It does not imply that media_start equals to
+	  media_stop. But, it just indicates a sample which should be presented
+	  at the first. This patch derives segment duration using media_time
+	  and duration of file. And set derived duration to segment-duration.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760781
+
+2016-01-28 21:36:54 +0900  Seungha Yang <sh.yang@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: expose streams with first moof for fragmented format
+	  In case of push mode, qtdemux expose streams after got moov box.
+	  We can not guarantee that a moov box has sample data such as sample duration
+	  and the number of sample in stbl box for fragmented format case.
+	  So, if a moov has no sample data, streams will not be exposed until get the first moof.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760779
+
+2016-01-27 18:48:17 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS
+
+2016-01-27 18:44:23 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps
+	  Prevents double-negotiation during startup and in some other cases.
+
+2016-01-27 16:43:22 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/deinterlace.c:
+	  deinterlace: Add negotiation unit tests for all 4 modes
+	  These now check the output caps based on the input caps and a following
+	  capsfilter and make sure the caps are exactly as expected.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760995
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 17:39:20 +0100  Vivia Nikolaidou <vivia@toolsonair.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Do passthrough in auto mode if downstream only supports interlaced
+	  If the following conditions are met:
+	  1) upstream and downstream caps are compatible
+	  2) upstream is interlaced
+	  3) downstream doesn't support progressive mode
+	  then deinterlace will just do passthrough instead of failing to link.
+	  This is done with the following scenario in mind:
+	  videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
+	  name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
+	  queue ! deinterlace name=dein_desktop ! autovideosink
+	  In this case, dein_src will do the deinterlacing. However,
+	  videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
+	  name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
+	  queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue !
+	  "video/x-raw,interlace-mode=interleaved" ! fakesink
+	  In this case, caps auto-negotiation will make dein_file and dein_desktop do
+	  the deinterlacing, while dein_src will be passthrough.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760995
+
+2016-01-26 18:05:51 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: Add mode=auto-strict
+	  In this mode we will passthrough all progressive caps but interlaced caps must be
+	  caps where we actually support deinterlacing.
+	  This is the only difference between auto and auto-strict, auto would
+	  passthrough all unsupported interlaced caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 17:50:30 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Implement reconfiguration a bit better
+	  And e.g. consider reconfiguration caused by RECONFIGURE events too.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 11:57:09 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Rewrite caps negotiation
+	  Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind
+	  of caps were last set, and e.g. if we last had interlaced caps or not. That's
+	  just broken.
+	  Also previously the handling of non-sysmem caps features was rather random and
+	  unusuable.
+	  Now the behaviour is the following, depending on the mode property:
+	  1) mode=disabled
+	  Completely do passthrough of everything
+	  2) mode=interlaced
+	  Only accept formats we can actually deinterlace, and accept interlaced
+	  and progressive content and always run the deinterlacer and output
+	  progressive content
+	  3) mode=auto (i.e. playbin)
+	  Accept all progressive formats as passthrough, accept all formats that we
+	  can deinterlace ourselves (which we do then), but also accept everything
+	  else for which we then just passthrough. In auto mode, deinterlacing is best
+	  effort: If we can, we deinterlace, if we can't we just output interlaced
+	  content.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720388
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760553
+
+2016-01-26 11:34:40 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Remove unused, obsolete bufferalloc code
+
+2016-01-26 18:50:38 +0100  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: use A_AAC instead of A_AAC/MPEGx/y
+	  Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761144
+
+2016-01-25 17:21:24 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/rtp/gstrtph261pay.c:
+	  gst: Fix unintialized variable warnings
+	  While cross-compiling with Linaro GCC 5.1-2015.08, it complained
+	  about a couple unitialized variables.
+	  This patch initializes them to zero.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761094
+
+2016-01-25 15:03:23 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxsrc: print potentially negative offset with a sign
+
+2016-01-21 17:41:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Re-add colorimetry field for RGB formats
+	  This time, check if it's an RGB format and sets the transformation
+	  matrix to identity. The rest of the colorimetry information is
+	  meaningfull and shall be kept.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759624
+
+2016-01-22 10:03:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: fix sRGB colorspace definition
+	  V4l2 can also use the sRGB colorspace for YUV formats and thus needs a
+	  default matrix.
+
+2016-01-21 15:29:46 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/debugutils/gsttaginject.c:
+	  taginject: fix sample pipeline in docs
+	  https://bugzilla.gnome.org/show_bug.cgi?id=679571
+
+2016-01-21 10:49:44 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Add adobe colorspace support
+	  Use the new primaries and transfer function for Adobe RGB.
+	  Explicitly list the colorimetry instead of using the default GStreamer
+	  ones. The defaults for BT2020, for example, do not match.
+	  Explicitly set the matrix of SRGB to RGB.
+
+2016-01-20 13:41:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/vpx/gstvp8enc.c:
+	  vp8enc: Ensure that we always have valid frame user data before using it
+	  Otherwise we're going to dereference NULL pointers.
+
+2016-01-20 10:02:48 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/vpx/gstvpxdec.c:
+	  vpxdec: Unref frame in all code paths of handle_frame()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-19 22:49:20 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* ext/vpx/gstvpxenc.c:
+	  vpxenc: Unref frame on ERROR
+	  All code paths for handle_frame() must somehow take ownership of the frame, be
+	  it by actually unreffing, forwarding the frame elsewhere or storing it for
+	  later.
+	  http://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-20 18:20:43 +1100  Jan Schmidt <jan@centricular.com>
+
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	  v4l2: Don't free props structure twice.
+	  gst_v4l2_device_provider_probe_device() frees the passed props
+	  structure, don't free it again in the caller.
+
+2016-01-19 15:15:35 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Cleanup uneeded return statement
+
+2016-01-19 15:14:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't set colorimetry for non YUV formats
+	  Setting colormetry in caps for RGB have no meaning, but worst it
+	  confuses the converters downstream.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759624
+
+2016-01-19 13:01:17 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpchannels.c:
+	* gst/rtp/gstrtpchannels.h:
+	  rtp: fix compiler warnings with gcc-6
+	  In file included from gstrtpL16depay.h:27:0,
+	  from gstrtp.c:73:
+	  gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
+	  static const GstRTPChannelOrder channel_orders[] =
+
+2016-01-19 14:57:03 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Don't play anything after the end of the data chunk even when seeking
+	  Especially in push mode we would completely ignore the size of the data chunk
+	  when not stop position is given for the seek. Instead make sure that the end
+	  offset is at most the end of the data chunk if known.
+	  Without this we would output anything after the data chunk, possibly causing
+	  loud noises if the media file is followed by an INFO chunk or an ID3 tag.
+
+2016-01-19 14:55:57 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
+	  We use that to signal "infinity", taking the difference between that and some
+	  other value is not going to give us any useful result for the end offsets of
+	  segments.
+
+2016-01-18 11:30:45 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
+	  This reverts commit 271501f6576de4d141e7c2f618e28b9e3b1e5b38.
+	  It wasn't meant to be pushed yet as the commit message indicates.
+
+2016-01-12 14:01:21 -0800  Aleix Conchillo Flaqué <aconchillo@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
+	  We check the stream profile and use the proper RTCP caps:
+	  application/x-srtcp if we are using a secure profile and
+	  application/x-rtcp otherwise.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760556
+
+2016-01-05 16:15:16 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  WIP: rtpjitterbuffer: Add RFC7273 media clock handling
+
+2016-01-15 11:36:35 +0000  Thibault Saunier <tsaunier@gnome.org>
+
+	* ext/vpx/gstvpxenc.c:
+	  vp8enc: Return FLOW_ERROR when an error accures
+	  FALSE would mean FLOW_OK
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-15 03:57:45 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* sys/osxaudio/gstosxcoreaudiohal.c:
+	  osxaudio: break as soon as the device is found
+	  No need to loop further if there's no side-effects for it
+
+2016-01-15 03:56:49 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* sys/osxaudio/gstosxaudioringbuffer.c:
+	* sys/osxaudio/gstosxcoreaudiohal.c:
+	  osxaudio: Fix error handling when selecting/opening devices
+	  Post an element error when the CoreAudio device cannot be selected or opened.
+	  Also ensure that we post a GST_ERROR with more detail.
+
+2016-01-13 23:40:20 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: When flushing on EOS, don't process more data than the "data" size
+	  Even if we have more data queued up when flushing than the size of the data
+	  chunk, don't process and output it. If the data size is known, this likely
+	  contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
+	  outputting them as if they were data is going to cause unexpected behaviour
+	  and unpleasant audio noises.
+
+2014-08-29 15:40:23 +0200  Antonio Ospite <ao2@ao2.it>
+
+	* tests/check/pipelines/wavenc.c:
+	  tests: fix a thinko in the wavenc example
+	  The code is supposed to follow somehow what the comment above says, that
+	  is to have one channel with a wave of freq 440 and the other channel
+	  with a wave of freq 880, but an off by one error results in frequencies
+	  of 0 and 440.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735673
+
+2014-08-29 15:07:58 +0200  Antonio Ospite <ao2@ao2.it>
+
+	* gst/interleave/interleave.c:
+	  interleave: Fix the example by setting channel-masks in the sink pads
+	  The current example does not work, it fails with:
+	  ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
+	  gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
+	  streaming task paused, reason not-negotiated (-4)
+	  This is because negotiation with wavenc gets messed up by the missing
+	  channel positions configuration.
+	  The proper way to define the channel layout when using the interleave
+	  element in code would be to set the channel-positions property, but
+	  gst-launch-1.0 does not know how to deal with arrays; so the example
+	  pipeline works around the issue by setting the channel-masks in the sink
+	  pads.
+	  Also fix a repetition in the deinterleave example description
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735673
+
+2016-01-11 16:29:55 +0000  Tim Sheridan <tim.sheridan@imgtec.com>
+
+	* gst/audioparsers/gstsbcparse.c:
+	  sbcparse: Fix frame length calculation
+	  SBC frame length calculation wasn't being rounded up to the nearest byte
+	  (as specified in the A2DP 1.0 specification, section 12.9). This could
+	  cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
+	  calculated frame lengths.
+	  Incorrect frame length calculation causes frame coalescing to fail, as
+	  subsequent frames in the stream aren't found in the expected locations.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742446
+
+2016-01-10 22:54:12 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: demote warning on wrong reserved value to fixme
+	  We are likely just parsing a backward-compatible stream we
+	  don't fully support.
+
+2016-01-08 16:27:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: simplify caps selection
+	  The downstream caps query with a filter alraedy gives us the possible
+	  intersection so there is no need to check it again with downstream
+	  if it is supported. Just try to set it directly.
+
+2016-01-07 20:42:41 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: fix unnecessary sub-buffer creation
+	  We create a sub-buffer just to copy over its metas and then
+	  throw it away immediately, just use the original input buffer
+	  directly.
+
+2016-01-07 20:38:27 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpdvdepay.c:
+	  rtpdvdepay: fix unnecessary sub-buffer creation
+	  We create a sub-buffer just to copy over its metas and then
+	  throw it away immediately, just use the original input buffer
+	  directly.
+
+2016-01-07 20:34:05 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpamrdepay.c:
+	  rtpamrdepay: fix unnecessary sub-buffer creation
+	  We create a sub-buffer just to copy over its metas and then
+	  throw it away immediately, just use the original input buffer
+	  directly.
+
+2016-01-07 20:27:29 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpvrawdepay.c:
+	  rtpvrawdepay: fix major memory leak and performance issue
+	  We call gst_rtp_buffer_get_payload() which creates a sub-buffer
+	  of each input buffer, just to copy over metas, and then leak it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760289
+
+2016-01-08 15:32:47 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rganalysis.c:
+	  rganalysis: Fix compiler warnings in the unit test
+	  elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined
+	  [-Werror,-Wshift-negative-value]
+	  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
+	  ~~ ^
+	  elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined
+	  [-Werror,-Wshift-negative-value]
+	  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
+	  ~~ ^
+	  elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined
+	  [-Werror,-Wshift-negative-value]
+	  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
+	  ~~ ^
+
+2016-01-05 18:13:06 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: don't map buffer multiple times when parsing
+
+2016-01-07 18:20:30 +0200  Steven Hoving <sh@bigbrother.nl>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroska: Store subtitle stream count in the correct variable
+	  And don't override the video stream count instead.
+
+2016-01-05 18:59:06 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/equalizer/gstiirequalizernbands.c:
+	  equalizer: The child-proxy API is GObject based in 1.x
+	  Not GstObject anymore.
+
+2015-05-21 17:41:12 +0200  Pablo Anton <pablo.anton@vodalys-labs.com>
+
+	* sys/v4l2/gstv4l2transform.c:
+	  v4l2-*: Configuring output pool correctly for using drivers min_buffer if present.
+	  Signed-off-by: Pablo Anton <pablo.anton@vodalys-labs.com>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755736
+
+2015-12-31 15:46:31 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: add debug msg on CRC mismatch while validating frame header
+
+2015-12-31 16:00:49 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: drop unneeded braces at _parse_frame() exit
+	  Additionally, drop redundant comment & line break
+
+2015-12-31 15:55:18 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: minor grammar correction
+
+2015-12-31 15:34:57 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: update URLs on pointers to online spec
+
+2015-12-31 14:40:15 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: make buffer DTS setting explicitly unconditional
+	  We are setting it to PTS regardless of block_strategy
+
+2015-12-31 14:21:40 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: add actual invalid block type to warning
+	  For someone that read the spec is clear the only *invalid*
+	  data block type is 127. For the rest, its useful information.
+	  Additionally. values 7-126 are currently reserved by the
+	  spec so the situation might change in the future.
+
+2015-12-31 14:12:36 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: use shift instead of mask & comp
+	  We are only interested on the first bit of the first
+	  byte of the metadata block header to figure out whether
+	  is marked as the last one. The shift makes it quite
+	  clearer.
+
+2015-12-31 12:52:13 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: warn on wishful parsing of weird headers
+	  If we get anything from 7 to 126 as type when parsing
+	  a metadata block header, we are likely dealing with a
+	  FLAC stream version we don't fully understand. Issue
+	  a warning if so.
+	  Document function assumptions regarding the passed-on
+	  type while at this.
+
+2015-12-31 11:33:45 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: show meaningful info on frame CRC check
+	  As CRCs are calculated for the comparition already, we
+	  might as well (cheaply) inform the user how the numbers
+	  differ if a missmatched pair is found.
+	  While at it:
+	  Rephrase candidate-frame message to make more sense
+
+2015-12-31 02:40:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: drop remaining trailing whitespace
+
+2015-12-31 02:15:06 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: drop superflous else clauses
+
+2015-12-31 01:09:51 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: factor out buffer time and offset resetting
+	  Avoids multiple occurrences of the same resetting pattern
+
+2015-12-31 00:54:48 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: move block handling by type out of _parse_frame()
+
+2015-10-07 18:51:25 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: replace duplicated codes to call new base sdp apis
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745880
+
+2015-12-30 12:16:56 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: drop redundant return statement on _header_is_valid()
+	  Fix the rather vague error message while at it.
+
+2015-12-30 01:56:26 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: rework gst_flac_parse_frame_is_valid()
+	  drop unnecessary nesting looking for end of frame
+
+2015-12-30 00:37:04 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: factor out context clearing routine
+
+2015-12-29 18:05:56 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Guard against no codec data in prores caps creation
+	  CID 1346532
+
+2015-12-29 17:58:38 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/vpx/gstvpxdec.c:
+	  vpxdec: Initialize buffer variable to NULL
+	  False positive but trivial to fix and possibly causing compiler warnings at
+	  some point in the future too.
+	  CID 1346535
+
+2015-07-27 15:53:26 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	  v4l2deviceprovider: add properties to the device
+	  Add properties to the device with exactly the same keys and sematics
+	  as what pulseaudio uses as property keys.
+	  Also handle the case when a device is probed manually and not through gudev.
+	  https://bugzilla.gnome.org//show_bug.cgi?id=759780
+
+2015-12-25 11:41:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/gstscaletempo.c:
+	  scaletempo: Free the various buffers in GstBaseTransform::stop()
+	  Previously we leaked them completely, but as they're specific to the caps
+	  freeing them in stop() instead of finalize() makes most sense.
+
+2015-12-24 15:28:06 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.7.1 ===
+
+2015-12-24 14:16:21 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.7.1
+
+2015-12-24 13:19:24 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2015-12-24 12:22:32 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/cs.po:
+	* po/de.po:
+	* po/el.po:
+	* po/hu.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/pl.po:
+	* po/ru.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	  po: Update translations
+
+2015-12-21 09:57:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: drop flushes from our own offset seek
+	  Prevents downstream from receiving flushes for a seek only in
+	  upstream. Those seeks are only to start reading from the right
+	  offset when skipping or returning to qt atoms.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758928
+
+2015-11-11 16:53:19 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Always set the channel mask for PCM streams
+	  Just use the gst_audio_channel_get_fallback_mask function for now as
+	  the specification is too complicated and nobody implements it.
+
+2015-12-21 11:37:26 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	  directsoundsink: Fix sleep for buffer-time lower than 200000
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748680
+
+2015-12-21 12:31:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  configure: Use -Bsymbolic-functions if available
+	  While this is more useful for libraries, some of our plugins with multiple
+	  files and some internal API can also benefit from this.
+
+2015-12-18 15:34:52 +0000  William Manley <will@williammanley.net>
+
+	* gst/debugutils/progressreport.c:
+	* gst/debugutils/progressreport.h:
+	  progressreport: add support for using format=buffers with do-query=false
+	  This is useful for investigating and debugging pipelines which are
+	  producing buffers at a slower/faster rate than you would expect.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759635
+
+2015-12-18 15:49:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Update formats table
+	  This change add all the new RGB based format. Those format removes the
+	  ambiguity with the ALPHA channel. Some other missing multiplanar format
+	  has been added with some additional cleanup.
+
+2015-12-18 05:17:15 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't write invalid edit list start time.
+	  Avoid writing a negative number as a large positive
+	  integer in an edit list when the first_ts is smaller
+	  than the first_dts - which can happen when the first
+	  packet received has a PTS but no DTS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759615
+
+2015-12-04 23:16:45 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Only update running time when it increases.
+	  Don't increment running time from every buffer. The correct
+	  logic to only increment when running time advances is a
+	  little further down, so delete this left-over line.
+
+2015-11-18 11:01:20 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: Implement prores support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758258
+
+2015-11-18 16:20:38 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.h:
+	  matroska-demux: Play ProRes video streams
+	  Generate video/x-prores caps for ProRes video streams.
+	  Every frame needs an 8 byte header prepended, as described in
+	  http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout
+	  so do that in a post-processing callback.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758258
+
+2015-12-18 10:18:09 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* ext/dv/gstdvdec.h:
+	  dvdec: Remove unused fields
+	  Remove unused fields frame_len and space
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759614
+
+2015-12-17 16:03:04 +0100  Vincent Dehors <vincent.dehors@openwide.fr>
+
+	* gst/rtp/gstrtpj2kdepay.c:
+	  rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758943
+
+2015-12-16 11:43:58 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* ext/raw1394/gstdv1394src.c:
+	* ext/raw1394/gsthdv1394src.c:
+	  dv1394: log error if failed to set socket status flag
+	  Log an error message if failed to set write or read socket as
+	  non-blocking.
+	  CID 1139608
+	  CID 1139609
+
+2015-12-15 17:10:00 +0000  Dave Craig <davecraig@unbalancedaudio.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	  audioparsers: Check for NULL return value of gst_pad_get_current_caps()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759503
+
+2015-12-16 09:35:53 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	  docs: update to git
+
+2015-12-15 14:27:22 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/Makefile.am:
+	  vpx: Add missing headers in Makefile.am
+	  This fixes distcheck.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-09-24 12:57:00 +0530  Prashant Gotarne <ps.gotarne@samsung.com>
+
+	* ext/vpx/Makefile.am:
+	* ext/vpx/gstvp8enc.c:
+	* ext/vpx/gstvp8enc.h:
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvp9enc.h:
+	* ext/vpx/gstvpxenc.c:
+	* ext/vpx/gstvpxenc.h:
+	  vpx: created common baseclass GstVPXEnc
+	  GstVP8Enc and GstVP9Enc has almost 80% code in common.
+	  created common baseclass GstVPXEnc for GstVP8Enc and GstVP9Enc
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-12-15 12:57:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvpxdec.c:
+	* ext/vpx/gstvpxdec.h:
+	  vpxdec: Remove unneeded add video_meta
+	  This also remove copies for VP8, which was not correctly in place
+	  in previous related patch.
+
+2015-12-15 09:49:24 +0530  Prashant Gotarne <ps.gotarne@samsung.com>
+
+	* ext/vpx/Makefile.am:
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp8dec.h:
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvp9dec.h:
+	* ext/vpx/gstvpxdec.c:
+	* ext/vpx/gstvpxdec.h:
+	  vpx: created common base class GstVPXdec for vpx decoders
+	  Base class for the vp8dec and vp9dec.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-06-10 09:17:08 -0400  Xavier Claessens <xavier.claessens@collabora.com>
+
+	* configure.ac:
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: Add GTlsInteraction property
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750709
+
+2015-12-14 09:05:06 -0500  Evan Callaway <evan.callaway@ipconfigure.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Retry connection if tunneling needs authentication
+	  Leverage response from gst_rtsp_connection_connect_with_response to
+	  determine if the connection should be retried using authentication.  If
+	  so, add the appropriate authentication headers based upon the response
+	  and retry the connection.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=749596
+
+2015-12-14 14:19:05 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: check port-range format
+	  The string could exist but with a wrong format, in that case we still want
+	  to reset the values of client_port_range.min and max like we do if there is
+	  no string.
+	  CID 1139593
+
+2015-12-14 14:55:12 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	  directsoundsink: Check device property and fail if device can't be found
+	  Don't use default if a specific device is set but it can't be found.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759452
+
+2015-12-14 14:15:00 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	  directsoundsink: Fix handling of the mute property
+	  - set mute value at startup
+	  - correct set and get mute functions
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2015-12-11 11:23:13 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* sys/directsound/gstdirectsoundsink.c:
+	  directsoundsink: Check the return value of GetStatus() too to decide if there was an error
+	  If GetStatus() fails, the status itself won't be very meaningful but we also
+	  have to look at its return value. This fixes blocking pipelines when removing
+	  sound devices or during other errors, where we wouldn't notice the error and
+	  then wait forever.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734098
+
+2015-12-10 17:41:46 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	  isomp4: remove unused parameters in build_*_extension
+	  AtomTRAK parameter is not used by build_mov_alac_extension(),
+	  build_jp2h_extension(), or build_mov_alac_extension()  and can be
+	  removed.
+
+2015-12-10 15:11:07 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  isomp4: replace variable only used once
+	  Replace has_shift variable with value since it is only use once.
+
+2015-12-09 12:24:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix packet dropping after a big discont
+	  We would queue 5 consective packets before considering a reset and a proper
+	  discont here. Instead of expecting the next output packet to have the current
+	  seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
+	  going to drop all queued up packets.
+
+2015-12-09 11:49:02 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/interleave/interleave.h:
+	  interleave: Remove unsed field
+	  Remove unused field collect_event in interleave.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759226
+
+2015-12-07 16:33:14 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Stop pushing data as soon as possible in push-mode
+	  When working in push-mode, we attempt to push out everything currently
+	  buffered in the adapter.
+	  This has two pitfalls:
+	  * We could stop earlier (the moment we get a non-ok or non-not-linked)
+	  * We return the last combined flow return, which might be completely
+	  different from the previous combined flow return
+
+2015-12-07 09:08:09 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From b319909 to 86e4663
+
+2015-12-07 14:41:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Add a warning if an empty RTCP packet is tried to be sent
+	  https://bugzilla.gnome.org/show_bug.cgi?id=759119
+
+2015-11-30 19:20:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp8dec.h:
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvp9dec.h:
+	  vpxdec: Use GstMemory to avoid copies
+	  With the VPX decoders it's not simple to use downstream buffer pool,
+	  because we don't know the image size and alignment when buffers get
+	  allocated. We can though use GstAllocator (for downstream, or the system
+	  allocator) to avoid a copy before pushing if downstream supports
+	  GstVideoMeta. This would still cause a copy for sink that requires
+	  specialized memory and does not have a GstAllocator for that, though
+	  it will greatly improve performance for sink like glimagesink and
+	  cluttersink. To avoid allocating for every buffer, we also use a
+	  internal buffer pool.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745372
+
+2015-11-30 08:42:35 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: Avoid over-skipping when checking LOAS config
+	  There might be multiple LOAS config in a row in a full frame. The first
+	  one might be a multi-layer config (which we can't properly parse yet)...
+	  but then followed by a valid (single-layer) one.
+	  The code was previously skipping whole frames (instead of just the LOAS
+	  config we failed to read) resulting in multiple frames (seen up to 6s in
+	  some situation) being dropped before finally getting the configuration.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758826
+
+2015-11-25 17:08:56 +0100  Edward Hervey <edward@centricular.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Properly set SPARSE stream flags for subpicture/subtitle
+	  And while we're at it, also detect 'DXSA' as being a variant fourcc
+	  of 'DXSB' for XSUB
+
+2015-11-30 21:23:52 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: souphttpsrc: grammar fix
+
+2015-11-30 21:01:17 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: souphttpsrc: switch shoutcast stream provider
+	  Fixes failing ICY test. Previous provider has
+	  streaming disabled outside UK.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758114
+
+2015-11-18 16:10:11 +0100  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* gst/avi/gstavimux.c:
+	  avimux: don't crash if we never got audio caps before stopping
+	  auds.blockalign is set once the first caps arrive. If
+	  gst_avi_mux_stop_file() is called before this happens then auds.blockalign
+	  is zero and gst_avi_mux_audsink_set_fields() cause a crash:
+	  [...]
+	  avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign;
+	  [...]
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758912
+
+2015-12-01 18:20:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: don't block when resurecting a buffer
+	  When we are resurecting a buffer, don't block. instead let us copy a
+	  buffer.
+
+2015-12-01 00:30:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: remove extra variable to improve readability
+	  Makes it easier to see that the event is being replaced/unrefed
+
+2015-12-01 00:22:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: respect seqnum in seek events
+	  Propagate the original seek seqnum to events originated from
+	  seeking to make sure they have the same value
+
+2015-12-01 00:03:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: flush upstream when seeking in pull mode
+	  Makes sure upstream will unblock and return the thread so that
+	  seeking can continue
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758861
+
+2015-11-27 09:27:29 +0100  Anton Bondarenko <antonbo@axis.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	  rtph264pay: add "send SPS/PPS with every key frame" mode
+	  It's not enough to have timeout or event based SPS/PPS information sent
+	  in RTP packets. There are some scenarios when key frames may appear
+	  more frequently than once a second, in which case the minimum timeout
+	  for "config-interval" of 1 second for sending SPS/PPS is not sufficient.
+	  It might also be desirable in general to make sure the SPS/PPS is
+	  available with every keyframe (packet loss aside), so receivers can
+	  actually pick up decoding immediately from the first keyframe if
+	  SPS/PPS is not signaled out of band.
+	  This patch adds the possibility to send SPS/PPS with every key frame. This
+	  mode can be enabled by setting "config-interval" property to -1. In this
+	  case the payloader will add SPS and PPS before every key (IDR) frame.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-11-27 09:03:51 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph264pay.h:
+	* tests/check/elements/rtp-payloading.c:
+	  rtph264pay: change config-interval property type from uint to int
+	  This way we can use -1 as special value, which is nicer than MAXUINT.
+	  This is backwards compatible even with the GValue API, as shown by
+	  a unit test.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-11-26 21:46:11 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: add support for Opus
+	  Add support for demuxing Opus encapsulated in MP4 files, based on the
+	  following spec: https://www.opus-codec.org/docs/opus_in_isobmff.html
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742643
+
+2015-11-25 22:48:32 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: use macro for codec_name
+	  Use _codec() macro instead of duplicating code.
+
+2015-03-25 16:32:55 +0100  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2: videodec: choose format from caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733827
+
+2015-03-27 15:02:33 +0100  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2: add gst_v4l2_object_probe_caps
+	  Add a variant of gst_v4l2_object_get_caps that bypasses the probed_caps cache.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733827
+
+2015-11-19 17:20:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2.c:
+	  v4l2-probe: Skip devices without supported formats
+
+2015-11-13 12:35:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	* sys/v4l2/gstv4l2.c:
+	  v4l2: Track /dev/video* to triggered required probe
+	  If something in /dev/video* get added, removed or replaced, we need to
+	  probe the devices again in order to ensure the dynamic devices are up to
+	  date.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758085
+
+2015-11-25 14:51:40 +1100  Alessandro Decina <alessandro.d@gmail.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpmanager: rtpsession: don't send empty RTCP packets
+	  generate_rtcp can produce empty packets when reduced size RTCP is turned on.
+	  Skip them since it doesn't make sense to push them and they cause errors with
+	  elements that expect RTCP packets to contain data (like srtpenc).
+
+2015-11-24 10:57:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: restore the segment on case of soft reset
+	  When seeking back to restore the mdat position a flush is pushed
+	  through and it resets downstream segment information. Make sure
+	  that after the flush (that does a soft reset) a segment will
+	  be pushed again
+	  Fixes regressions spotted at
+	  https://ci.gstreamer.net/job/GStreamer-master-validate/2100/
+
+2015-11-20 12:44:22 +0000  Graham Leggett <minfrin@sharp.fm>
+
+	* gst/multifile/gstmultifilesink.c:
+	  multifilesink: fix spelling of variable
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758390
+
+2015-11-20 11:05:51 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: unite duplicate FourCC
+	  Unite in fourcc.h the FourCCs that are used twice or more in qtdemux
+
+2015-11-19 15:33:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2transform.c:
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2: Fix capture/output-io-mode properties
+	  There was some miss-match in the implementation. This makes it
+	  concistent, though functionally it worked, except the video decoder
+	  output-io-mode getter.
+
+2015-11-19 19:48:06 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	  atoms: remove unused argument of build_mov_wave_extension()
+	  AtomTrak * trak argument of build_move_wave_extension() isn't used.
+	  Removing it.
+
+2015-11-19 19:28:20 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: remove duplicate FourCC
+	  Use the available FourCCs in fourcc.h instead of duplicating them.
+
+2015-11-19 18:36:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	  isomp4: centralize all FourCC
+	  10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c
+	  already exist in fourcc.h. Don't duplicate these and use them directly.
+	  Plus moving 6 to fourcc.h, to centralize them all.
+
+2015-11-19 17:32:12 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/matroska/webm-mux.c:
+	  matroska/webmmux: fix outdated example launch lines
+	  Update gst-launch-0.10 lines to gst-launch-1.0
+
+2015-11-16 13:26:50 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  isomp4: add support for Opus in mp4mpux
+	  Add support for muxing MP4 files containing Opus. Based on the spec
+	  detailed here:
+	  https://www.opus-codec.org/docs/opus_in_isobmff.html
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742643
+
+2015-11-18 19:10:56 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Replace tabs with spaces
+
+2015-11-18 19:07:53 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers
+	  This fixes seeking if the first entries in the samples table are negative. The
+	  binary search would always fail on this as the array would not be sorted if
+	  interpreting the negative numbers as huge positive numbers. This caused us to
+	  always output buffers from the beginning after a seek instead of close to the
+	  seek position.
+	  Also add a case to the comparison function for equality.
+
+2015-11-18 16:01:48 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: remove duplicate check
+	  We want 1 or 2 streamheaders, the check  if (bufarr->len != 1 &&
+	  bufarr->len != 2) is enough. Not need to check if bufarr->len is <= 0 or
+	  > 255.
+
+2015-11-18 14:48:36 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: Fix error leak and handle error
+	  g_thread_try_new allows for possiblity of failures. In case it fails,
+	  error is not handled and leaked.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758260
+
+2015-11-15 17:16:29 -0800  Josep Torra <n770galaxy@gmail.com>
+
+	* gst/rtp/gstrtpgstdepay.c:
+	  rtpgstdepay: Properly handle backward compat for event deserialization
+	  Actual code is checking for a NULL terminator and a ';' terminator,
+	  for backward compat, in a chained way that cause all events being rejected.
+	  The proper condition is to reject the events when terminator isn't
+	  in ['\0', ';'] set.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758151
+
+2015-11-15 17:11:02 -0800  Josep Torra <n770galaxy@gmail.com>
+
+	* tests/check/elements/rtp-payloading.c:
+	  tests: rtp-payloading: Test for handling of custom events in rtpgst
+	  Add a simple test that checks proper serialization/deserialization
+	  of custom events with rtpgstpay and rtpgstdepay.
+
+2015-11-16 16:23:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp9dec.c:
+	  vpxdec: Use threads on multi-core systems
+	  This adds an automatic mode to the threads property of vpxdec in order to
+	  use as many threads as there is CPU on the platform. This brings back
+	  GStreamer VPX decoding performance closer to what is achieved by other
+	  players, including Chromium.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758195
+
+2015-11-16 10:58:32 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: only send initial gaps for non-fragmented streams
+	  It would be unusual to have the header segment with an 'edts' atom
+	  indicating gaps at the beginning when handling fragmented streams.
+	  The header usually doesn't contain any timestamping information, this
+	  should come from the playlist/manifest and the segments with media
+	  in those scenarios.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758171
+
+2015-11-17 09:41:34 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  Revert "Revert "qtdemux: respect qt segments in push-mode for empty starts""
+	  This reverts commit d842ff288a9d01214a046becbfd9cbff3a4acea0.
+	  This was reverted by accident
+
+2015-11-17 12:39:05 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	* gst/udp/gstudpsrc.h:
+	  udpsrc: Add "loop" property for enabling/disabling multicast loopback
+	  On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it
+	  is a setting for the receiver socket. As such we will need it on udpsrc too to
+	  allow filtering out our own multicast packets.
+
+2015-11-16 13:52:05 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  Revert "qtdemux: respect qt segments in push-mode for empty starts"
+	  This reverts commit 142d8e2d23e5602e7382977af1043d621625f8c8.
+
+2015-11-16 16:56:04 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix string memory leak
+	  The string got using g_strdup_printf will be allocated memory
+	  and should be freed after use.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758161
+
+2015-11-14 21:51:11 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2/object: remove unnecessary NULL check before g_free()
+
+2015-11-14 21:45:29 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* sys/oss/gstosssrc.c:
+	  osssrc: remove unnecessary NULL check before g_free()
+
+2015-11-14 21:43:24 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* sys/sunaudio/gstsunaudiosrc.c:
+	  sunaudiosrc: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:36:30 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:31:08 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:26:21 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroska/read-common: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:43:10 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	  isomp4/atoms: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:35:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	  rtp/theorapay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:33:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtp/vorbispay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:31:34 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/rtp/gstrtpjpegpay.c:
+	  rtp/jpegpay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:27:04 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/rtp/gstrtpgstpay.c:
+	  rtpgstpay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:22:09 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:14:25 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: remove unnecessary NULL check before g_free()
+
+2015-11-14 20:09:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/effectv/gstop.c:
+	  effectv/optv: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:05:03 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/effectv/gstshagadelic.c:
+	  effectv/shagadelictv: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:01:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/effectv/gstripple.c:
+	  effectv/ripple: remove unnecessary NULL checks before g_free()
+
+2015-11-14 19:56:57 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/effectv/gstradioac.c:
+	  effectv/radioac: remove unnecessary NULL checks before g_free()
+
+2015-11-14 19:52:12 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/effectv/gststreak.c:
+	  effectv/streak: remove unnecessary NULL check before g_free()
+
+2015-11-14 17:04:55 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2: remove unnecessary NULL checks before g_free()
+
+2015-11-14 16:57:13 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* ext/vpx/gstvp9enc.c:
+	  vp9enc: remove unnecessary NULL check before g_free()
+
+2015-11-14 16:54:42 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* ext/vpx/gstvp8enc.c:
+	  vp8enc: remove unnecessary NULL check before g_free()
+
+2015-11-14 16:20:33 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: remove unnecessary NULL checks before g_free()
+
+2015-11-13 13:34:02 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: add support of NV16, NV61 and NV24 formats
+	  Mapped respectively to V4L2_PIX_FMT_NV16/V4L2_PIX_FMT_NV16M,
+	  V4L2_PIX_FMT_NV61,V4L2_PIX_FMT_NV61M and V4L2_PIX_FMT_NV24 v4l2 formats.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758058
+
+2015-11-11 14:10:53 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	  splitmuxpartreader: Fix GCond leak
+	  inactive_cond is not being cleared resulting in memory leak.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757924
+
+2015-08-06 12:44:20 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: fix output state memory leak
+	  When jpeg_finish_decompress is called, output state reference is being created.
+	  But if there is any failures in finishing decompress, it jumps to setjmp,
+	  and at that point state was not referenced. Resulting in leak of output state.
+	  Hence adding another setjmp after output state is referenced.
+	  Similarly adding another setjmp to unmap the frame in case error happens before
+	  finish_decompress
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753087
+
+2015-08-10 11:23:45 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: respect qt segments in push-mode for empty starts
+	  In push-mode it is hard to support qt segments overall but it is
+	  possible to support when the file isn't heavily edited but just contain
+	  a segment to indicate a gap at the beginning. This also allows properly
+	  timestamping data that has negative DTS in push-mode.
+	  It is relevant to support those for 2 scenarios:
+	  1) fragmented streaming
+	  2) HTTP playback of 'regular' mp4
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753484
+
+2015-11-05 18:39:33 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* ext/pulse/pulsedeviceprovider.c:
+	  pulse: Don't leak caps and structures in the device provider
+
+2015-11-04 19:01:20 +0530  Arun Raghavan <arun@centricular.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpmanager: Document properties that are expressed in bits per second
+	  This changed in 928cd110bcea5d143cab3ea747991851d52ecbad and
+	  73c0c2920f9aca96982a4de0c20b3417aa148b81 but was not documented.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-11-04 18:51:32 +0530  Arun Raghavan <arun@centricular.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	  rtpmanager: Trivial gst-indent fixes
+
+2015-08-12 13:35:40 +0200  Philippe Normand <philn@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: support for cenc auxiliary info parsing outside of moof box
+	  When the cenc aux info index is out of moof boundaries, keep track of
+	  it and parse the beginning of the mdat box, before the first sample.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755614
+
+2015-11-03 20:33:10 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Use codecutils helpers for creating Opus caps
+	  Also fix up codec data with values from the container.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-11-03 14:51:48 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: There is no multistream field for Opus anymore
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-11-03 12:42:52 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/webm-mux.c:
+	  matroska/webmmux: Support Opus in webmmux and VP9 in matroskamux
+	  https://bugzilla.gnome.org/show_bug.cgi?id=729950
+
+2015-11-03 12:40:15 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Parse and handle CodecDelay, SeekPreroll and DiscardPadding
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 12:18:19 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-ids.h:
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Write CodecDelay, DiscardPadding and SeekPreroll for Opus
+	  And also adjust timestamps and durations according to the codec delay, both
+	  should include it for whatever reason.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 11:49:54 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Opus headers are not in-band
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 22:01:07 +0530  Arun Raghavan <git@arunraghavan.net>
+
+	* sys/v4l2/gstv4l2.c:
+	  v4l2: Set O_CLOEXEC on the device fd
+	  This is needed to make sure that child processes don't inherit the video
+	  device fd which can cause problems with some drivers.
+
+2015-11-03 14:46:30 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpmanager: switch G_GINT64_FORMAT for GST_STIME_ARGS
+	  No need to use G_GINT64_FORMAT for potentially negative values of
+	  GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
+	  Plus it creates more readable values in the logs.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757480
+
+2015-11-03 14:26:29 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff
+	  No need to manually handle negative values of diff, GST_STIME_ARGS does
+	  exactly this.
+
+2015-11-02 16:53:15 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: use GST_STIME_ARGS for GstClockTimeDiff
+	  No need to manually handle negative values of diff, GST_STIME_ARGS does
+	  exactly this.
+
+2015-11-02 16:43:46 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: use GST_STIME_ARGS for GstClockTimeDiff
+	  No need to manually handle negative values of diff, GST_STIME_ARGS is
+	  available for this.
+
+2015-10-30 10:05:37 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/audiofx/audiochebband.c:
+	  audiochebband: Fix typo in example pipeline
+	  Fix typo in example pipeline.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757340
+
+2015-10-28 23:47:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	  v4l2: fix double-unref in the v4l2 device provider
+
+2015-10-27 10:48:00 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-ids.c:
+	  matroskamux: don't drop JPEG frames that only have PTS but no DTS set
+	  For the MS/VfW codec ids, we want to write DTS timestamps instead
+	  of PTS because that's what everyone else seems to do (and it's also
+	  how it is in AVI). So for those input formats we use the buffer DTS
+	  instead of the PTS. However, if there's no DTS set but only the PTS
+	  then just take the PTS instead of dropping the input buffer. This
+	  is useful especially for I-frame only codecs like JPEG and huffyuv,
+	  but should also be fine as fallback in general.
+	  Fixes regression with input JPEG frames that only have PTS set on them.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756967
+
+2015-10-24 23:57:38 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/splitmux.c:
+	  tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753622
+
+2015-10-24 23:57:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: do not destroy the multiqueue & muxer when going to NULL
+	  Instead, delay it until all request pads have been released. This is
+	  because the release_pad() vfunc requires the multiqueue and muxer to
+	  be there in order to release their request pads as well. If those
+	  elements are destroyed earlier, release_pad() does not work, no
+	  pads are released and some resources are leaked.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753622
+
+2015-10-20 15:28:10 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Read buffer timestamp *after* actually setting it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756809
+
+2015-10-24 17:14:07 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/gstscaletempo.c:
+	* gst/audiofx/gstscaletempo.h:
+	  scaletempo: Fix handling of rate < 0
+	  We have to reverse all samples in a buffer before processing them to properly
+	  have continuous data from one buffer to another. As a result we will have a
+	  negative applied rate and a rate of 1.0.
+	  Also make sure that input buffers are correctly clipped to the segment,
+	  otherwise our calculations are going to go wrong.
+	  Also copy over the segment event's sequence number to the output segment while
+	  we're at it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=757033
+
+2015-10-19 18:04:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: break as soon as non-interlaced if found
+	  It looks for a non-interlaced entry on the filter caps, break
+	  as soon as one is found to avoid wasting cpu
+
+2015-10-19 17:50:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: implement accept-caps
+	  Implement accept-caps handler to avoid doing a full caps query
+	  downstream to handle it.
+	  This commit implements accept-caps as a simplification of the _getcaps
+	  function, so it exposes the same limitations that getcaps would.
+	  For example, not accepting renegotiation to caps with capsfeatures when
+	  it was last configured to a caps that it has to deinterlace.
+
+2015-10-19 17:06:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/deinterlace.c:
+	  tests: deinterlace: fix small typo in comment
+
+2015-10-26 00:41:28 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/files/Makefile.am:
+	  check: Dist splitvideo0[012].ogg test files.
+
+2015-10-23 20:16:17 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/gstscaletempo.c:
+	* gst/audiofx/gstscaletempo.h:
+	  scaletempo: Add support for F64
+
+2015-10-22 17:40:38 -0700  Mischa Spiegelmock <mspiegelmock@gmail.com>
+
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* gst/multipart/multipartdemux.c:
+	* gst/rtp/README:
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/udp/gstudpsrc.c:
+	  docs: Minor fixes in various places
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756996
+
+2015-10-21 17:43:31 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/goom/plugin_info.c:
+	  goom: remove compiler trick
+	  After commit 2cb6cfed22166b262ae50cb58f3ff11dd8ba91f9 there is no need to
+	  trick the compiler anymore about the usage of variable cpuFlavour.
+
+2015-10-21 14:35:02 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From b99800a to b319909
+
+2015-10-21 17:41:38 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/audiofx/audiofxbaseiirfilter.h:
+	  audiofx: remove unused variable
+	  Remove unsued variable have_coeffs in audiofxbaseiirfilter
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756905
+
+2015-10-20 17:29:42 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Use new GST_ENABLE_EXTRA_CHECKS #define
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756870
+
+2015-10-21 14:25:55 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* README:
+	* common:
+	  Automatic update of common submodule
+	  From 9aed1d7 to b99800a
+
+2015-10-21 11:53:09 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: relax creation time parsing
+	  Parse wrong timestamps like we used to write as well,
+	  e.g. 10:9:42, and the hour might be without a leading
+	  zero in any case.
+
+2015-10-21 11:45:35 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: fix indentation
+
+2015-10-21 11:44:50 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: extract both creation date and time
+	  Before we only extracted the date part.
+
+2015-10-21 11:16:01 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: fix writing of creation time
+	  Don't write time as e.g. 11:9:42
+
+2015-10-13 12:42:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtpj2kpay: update fragment offset
+	  It was always being set to 0, making the resulting stream broken
+	  for the receiver
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756422
+
+2015-10-19 15:36:37 +0300  Ryan Hendrickson <ryan.hendrickson@alum.mit.edu>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't unconditionally use strnlen()
+	  It's not available on older OSX and we can as well use memchr() here.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756154
+
+2015-10-19 17:38:32 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/auparse/gstauparse.c:
+	  auparse: Fix event memory leak
+	  Free the event after being handled to prevent memory leak.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756799
+
+2015-10-19 09:14:19 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: unify raw audio caps into a single caps structure
+
+2015-10-14 15:42:50 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: add support for FFV1 coded streams in mov
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752495
+
+2015-10-14 15:53:26 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: EOS immediately if we have an empty seek segment
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748316
+
+2015-10-14 10:43:19 +0300  Stavros Vagionitis <stavrosv@digisoft.tv>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Make non-inclusive segment boundaries inclusive
+	  The problem is that the filesrc and souphttpsrc are behaving
+	  differently regarding the calculation of the segment boundaries. The
+	  filesrc is using a non-inclusive boundaries, while the souphttpsrc
+	  uses inclusive. Currently the hlsdemux calculates the boundaries as
+	  inclusive, so for this reason there is no problem with the souphttpsrc,
+	  but there is an issue in the filesrc.
+	  The GstSegment is non-inclusive, so the proposed solution is to use
+	  non-inclusive boundaries in the hlsdemux in order to be consistent.
+	  Make the change in the hlsdemux, will break the souphttpsrc, which
+	  will expect inclusive boundaries, but the hlsdemux will offer
+	  non-inclusive. This change makes sure that the non-inclusive
+	  boundaries are converted to inclusive.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748316
+
+2015-10-11 22:07:54 +0000  Graham Leggett <minfrin@sharp.fm>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	* ext/soup/gstsouphttpclientsink.h:
+	  souphttpclientsink: Add the retry and retry-delay properties
+	  These allow a failed request to be retried after the given number of seconds
+	  instead of failing the pipeline. Take account of the Retry-After header if
+	  present. Add retries parameter that controls the number of times an HTTP
+	  request will be retried before failing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756318
+
+2015-10-14 12:03:15 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix caps leak
+	  If the QtDemuxStream are re-used they may already have caps which used
+	  to be leaked.
+	  Reproduced using the
+	  validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
+	  scenario.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756561
+
+2015-10-14 09:29:50 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix taglist memory leak
+	  Free the stream and its sub items instead of just the stream
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756544
+
+2015-10-11 12:06:26 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: Allow negotiating to S8 as a raw format but stop making it best choice
+	  Negotiation to audio/x-raw,format=S8 was not possible because S8 does
+	  not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756387
+
+2015-10-11 09:18:40 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: Add prores support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756388
+
+2015-10-12 18:56:32 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	  tests: add GST_PLUGINS_BASE_LIBS for flvdemux check
+	  So it pulls in the right libgsttag-1.0.
+
+2015-10-11 22:27:47 +0100  Julien Isorce <j.isorce@samsung.com>
+
+	* gst/goom/Makefile.am:
+	* gst/goom/gstaudiovisualizer.c:
+	* gst/goom/gstaudiovisualizer.h:
+	* gst/goom/gstgoom.h:
+	* gst/goom2k1/Makefile.am:
+	* gst/goom2k1/gstaudiovisualizer.c:
+	* gst/goom2k1/gstaudiovisualizer.h:
+	* gst/goom2k1/gstgoom.h:
+	  goom/goom2k1: remove obsolete left over files
+	  They now use the new GstAudioVisualizer base class
+	  from gst-plugins-base/gst-libs/gst/pbutils
+	  Also fixed undefined reference to gst_audio_visualizer_get_type
+	  Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-12 10:48:23 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/audioparsers/gstmpegaudioparse.c:
+	  mpegaudioparse: Fix buffer memory leak during failures
+	  mapped buffer is not being unmapped during failures
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756231
+
+2015-10-12 11:18:51 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: Check if soup message is created
+	  If soup message is not created then the same should not be passed
+	  on, which is resulting in segfault. Hence throwing a warning message
+	  and returning
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:15:15 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: Check if location being set is valid
+	  Adding a check in set_property to find if the location uri is valid
+	  and printing warning if not valid.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:09:30 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: Fix memory leaks during failures
+	  freeing streamheader_buffers and sent_buffers during failure cases.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:03:17 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/soup/gstsouphttpclientsink.c:
+	  souphttpclientsink: Replace redundant free_buffer_list function
+	  Removing free_buffer_list and replacing it with already available function
+	  g_list_free_full
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-11 16:40:01 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* tests/check/Makefile.am:
+	  check: Don't forget base CFLAGS for flvdemux check
+	  elements/flvdemux.c:25:25: fatal error: gst/tag/tag.h: No such file or directory
+
+2015-10-11 11:37:51 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/ebml-write.c:
+	* gst/matroska/ebml-write.h:
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/matroska-mux.h:
+	  matroskamux: Create a TIME segment when creating streamable output
+	  Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which
+	  does the same for flvmux.
+
+2015-09-23 13:50:52 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/Makefile.am:
+	* gst/flv/gstflvdemux.c:
+	* tests/check/Makefile.am:
+	* tests/check/elements/flvdemux.c:
+	  flvdemux: output speex vorbiscomment as a GstTagList
+	  This is what speexdec expects.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755478
+
+2015-09-22 22:59:16 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/gstflvmux.c:
+	* tests/check/elements/flvmux.c:
+	  flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
+	  Or else flvdemux don't understand it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754435
+
+2015-09-02 10:44:59 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/gstflvmux.c:
+	* tests/check/elements/flvmux.c:
+	  flvmux: use time segment and copy timestamps when streamable
+	  Add a basic test using speex data to verify timestamping.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754435
+
+2015-09-23 13:14:03 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: speex is also always 16KHz
+	  This is just a cosmetic change for the logs, since the right caps
+	  for Speex is being set elsewhere.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755479
+
+2015-07-14 15:19:44 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	  rtpmanager: Add 'source-stats' to stats and notify
+	  Add statitics from each rtp source to the rtp session property.
+	  'source-stats' is a GValueArray where each element is a GstStructure of
+	  stats for one rtp source.
+	  The availability of new stats is signaled via g_object_notify.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752669
+
+2015-06-05 17:20:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  rtpsession: Implement sending of reduced size RTCP packets
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750456
+
+2015-10-08 15:01:13 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/audiofx/audiodynamic.h:
+	  audiofx: Remove unused variable
+	  Remove unused variable 'degree' in audiodynamic
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756234
+
+2015-10-08 14:44:07 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Fix memory leak for corrupted file
+	  Free brands before overriding them.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756226
+
+2015-10-08 11:44:04 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
+	  gdkpixbufdec: Fix pixbuf_loader leak during failures
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756219
+
+2015-10-07 23:23:45 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Add missing break
+
+2015-10-07 13:03:02 +0200  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	* gst/rtpmanager/rtpstats.c:
+	* gst/rtpmanager/rtpstats.h:
+	  rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-10-07 13:02:12 +0200  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	  rtpmanager: add "max-dropout-time" and "max-misorder-time" props
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-10-07 17:14:57 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Fix date memory leak
+	  When getting date from taglist, the memory should be freed after
+	  using it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756171
+
+2015-10-05 11:03:38 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Fix sample memory leak
+	  When getting sample from taglist, the memory should be freed after
+	  using it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=756068
+
+2015-10-05 13:10:56 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/cutter/gstcutter.c:
+	  cutter: Fix buffer leak
+	  Buffer is added to the internal cache, and pushed only when accumulated
+	  buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
+	  is not freed. Freeing the cache when the state changes from PAUSED to READY.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754212
+
+2015-08-31 21:10:16 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	  rtpmux: Use default upstream event handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-08-31 21:05:03 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpmux.h:
+	  rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-07-22 09:47:22 +0200  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpmux.h:
+	* tests/check/elements/rtpmux.c:
+	  gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
+	  By not doing this, the muxer is not effectively a rtpmuxer, rather a
+	  funnel, since it should be a single stream that exists the muxer.
+	  If not specified, take the first ssrc seen on a sinkpad, allowing upstream
+	  to decide ssrc in "passthrough" with only one sinkpad.
+	  Also, let downstream ssrc overrule internal configured one
+	  We hence has the following order for determining the ssrc used by
+	  rtpmux:
+	  0. Suggestion from GstRTPCollision event
+	  1. Downstream caps
+	  2. ssrc-Property
+	  3. (First) upstream caps containing ssrc
+	  4. Randomly generated
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-10-02 22:42:20 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Fixup last commit
+
+2015-10-02 22:21:45 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	* gst/udp/gstudpsrc.c:
+	  Update GLib dependency to 2.40.0
+
+2015-06-30 16:56:19 +0200  Miguel París Díaz <mparisdiaz@gmail.com>
+
+	* gst/rtpmanager/rtpstats.c:
+	* gst/rtpmanager/rtpstats.h:
+	  rtpstats: add utility for calculating RTP packet rate
+
+2015-08-10 18:14:39 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: handle empty segments in seeking adjust
+	  If seeking targets an empty segment skip it as there is no media
+	  offset to get from it. Instead look for the next one.
+	  This doesn't make seeking in push-mode work if you seek to an
+	  empty segment but at least won't get you to wrong offsets.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753484
+
+2015-04-17 14:25:43 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: post messages when fragments are being opened and closed
+	  This can be useful for applications that need to track the created fragments
+	  (to log them in a recording database, for example)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750108
+
+2015-04-29 18:23:28 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: allow non-video streams to serve as reference
+	  In the absence of a video stream, the first stream will be used as
+	  reference.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753617
+
+2015-07-22 17:45:12 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: initialize mux_start_time properly
+	  mux_start_time refers to the running_time of the buffer
+	  that goes first in the output file. Normally this time is
+	  0, so this variable is initialized to 0 during the state
+	  change to PAUSED.
+	  However, when dealing with dynamic pipelines and starting
+	  a recording while the pipeline has already run for a while,
+	  the running_time of the first buffer is > 0 and this causes
+	  a problem with detecting the end of the first file(s) when
+	  splitting by duration, because the code will later compare
+	  the threshold_time with (last buffer running_time - mux_start_time)
+	  and will get it wrong until mux_start_time advances enough
+	  to make this difference < threshold_time, creating empty files
+	  in the meantime.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753624
+
+2015-09-16 16:03:02 +0900  Vineeth T M <vineeth.tm@samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Reverse playback does not consider segment.start
+	  During reverse playback, the media should stop playing at segment.start
+	  This does not happen, and avidemux continues to process data even when
+	  current timestamp is less that segment.start.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755094
+
+2015-09-23 12:39:35 +0900  Manasa Athreya <manasa.athreya@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream
+	  If stream has more than one trex box which is not matched to actual
+	  track id, it makes qtdemux crashed.
+	  Author : Manasa Athreya (manasa.athreya@lge.com)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754864
+
+2015-09-04 14:24:45 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/smpte/gstsmpte.c:
+	  smpte: get size, stride info using VideoInfo
+	  Use VideoInfo data to get size stride and
+	  offset, instead of hard coded macros.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754558
+
+2015-09-04 14:18:50 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/smpte/gstsmpte.c:
+	  smpte: free mask
+	  Free the memory allocated to 'mask' to avoid
+	  memory leak.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754555
+
+2015-08-20 11:02:58 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* tests/examples/equalizer/demo.c:
+	* tests/icles/equalizer-test.c:
+	* tests/icles/gdkpixbufoverlay-test.c:
+	* tests/icles/gdkpixbufsink-test.c:
+	* tests/icles/test-oss4.c:
+	* tests/icles/videocrop-test.c:
+	  gstreamer: good: tests: Fix memory leaks when context parse fails.
+	  When g_option_context_parse fails, context and error variables are not getting free'd
+	  which results in memory leaks. Free'ing the same.
+	  And replacing g_error_free with g_clear_error, which checks if the error being passed
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753853
+
+2015-10-02 16:18:15 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: doesn't handle probation and rtp gap in case of sender
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754548
+
+2015-10-02 16:16:32 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/gstrtpsession.h:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals
+	  Allows for applications to get internal source's RTP statistics.
+	  (eg. sender sources for a server/client)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-10-02 14:17:48 +1000  Jan Schmidt <jan@centricular.com>
+
+	* sys/ximage/gstximagesrc.c:
+	  ximagesrc: Gather and coalesce all damaged areas before retrieving.
+	  These days the xserver seems to give us the same damage regions
+	  over and over for entire windows, and we retrieve them multiple
+	  times, which gives time for more damage to appear. Instead, just
+	  quickly gather all damaged areas into a region list and copy
+	  out once.
+
+2015-10-01 16:24:32 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/goom2k1/Makefile.am:
+	* gst/goom2k1/gstgoom.h:
+	  goom2k1: use the new audiovisualizer base class
+	  Rebase to have goom using the GstAudioVisualizer base class in
+	  gst-plugins-base/gst-libs/gst/pbutils
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-01 16:16:08 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/goom/Makefile.am:
+	* gst/goom/gstgoom.h:
+	  goom: use the new audiovisualizer base class
+	  Rebase to have goom using the GstAudioVisualizer base class in
+	  gst-plugins-base/gst-libs/gst/pbutils
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-09-30 17:35:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/interleave/deinterleave.c:
+	* tests/check/elements/deinterleave.c:
+	  deinterleave: implement accept-caps
+	  Avoid using default accept-caps handler that will query downstream
+	  and is more expensive. Just check if the caps is compatible with
+	  the template and check if the channels are the same.
+
+2015-09-30 09:35:39 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/deinterleave.c:
+	  tests: deinterleave: also check for caps query results
+
+2015-09-30 12:30:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/interleave/deinterleave.c:
+	  deinterleave: use the caps query filter
+	  It was being ignored and would lead to wrong results if the
+	  element doing the query would rely on the intersection being made.
+
+2015-09-30 10:00:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/interleave/deinterleave.c:
+	  deinterleave: implement a caps query handler for the sinkpad
+	  It was missing and apparently code relied on having it there
+	  for not allowing a change in the number of channels
+
+2015-09-30 09:05:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/interleave/deinterleave.c:
+	  deinterleave: fix caps leak
+	  Caps from the pad template are being leaked. In any case it is
+	  from a static pad template and will 'leak' in the end, just doing
+	  the cleanup for the good practice.
+
+2015-09-29 11:15:01 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/gdkpixbufoverlay.c:
+	  tests: gdkpixbufoverlay: add minimal unit test
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755773
+
+2015-09-29 11:12:48 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	  gdkpixbufsink: don't leak old pixel buffer when setting a new overlay
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755773
+
+2015-09-28 20:25:22 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: avoid potential string overflow
+	  We don't necessarily have full control over the input tags, so
+	  it's possible that the ISRC tag contains a longer string than
+	  expected, in which case we'd write over the end of the static-size
+	  13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
+	  Make sure to only copy the ISRC if it's not too long, and make
+	  sure the buffer we write to is always NUL-terminated by using
+	  g_strlcpy().
+	  CID 1324931.
+
+2015-09-28 18:03:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Remove leftover assertion from 0.10
+	  We now allocate memory via GstAllocator and as such can handle arbitrary
+	  alignments, not only <= G_MEM_ALIGN.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755708
+
+2015-09-25 10:01:37 +0200  Guillaume Marquebielle <guillaume.marquebielle@parrot.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: fix uninitialized variables in LOAS config reading
+	  On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
+	  "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
+	  'channels' are pointing to uninitialized values can end on setting random values as
+	  rate and channels on src caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755611
+
+2015-09-18 00:58:23 +1000  Jan Schmidt <thaytan@noraisin.net>
+
+	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	  Fix some compiler warnings when building with G_DISABLE_ASSERT
+	  Touches rtpmanager and gdkpixbufsink
+
+2015-08-18 14:30:57 +0100  Chris Bass <floobleflam@gmail.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: support timed-text subtitle tracks.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752818
+
+2015-09-26 00:12:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-parse.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  gst: Don't use deprecated gst_segment_to_position()
+
+2015-09-21 13:47:21 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755125
+
+2015-09-16 19:28:11 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpbin/session: Allow RTCP sync to happen based on capture time or send time
+	  Send time is the previous behaviour and the default, but there are use cases
+	  where you want to synchronize based on the capture time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755125
+
+2015-09-25 23:51:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:15:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.6.0
+
+2015-09-25 22:57:34 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2015-09-25 14:08:09 +0200  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/smpte/gstsmptealpha.c:
+	  smptealpha: Do not set width/height before comparing with old values
+	  Otherwise we end up considering the values did not change and we wrongly
+	  work with the old video format (which will lead to wrong
+	  behaviour/segfaults).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755621
+
+2015-09-23 20:59:00 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Accumulate segments for edit lists before activating the next segment
+	  eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
+	  accumulating segments manually when activating a segment. This is only
+	  needed when handling edit lists, not when activating a segment because of a
+	  seek. Do the accumulation when switching edit list segments instead.
+	  This fixes segment seeks again, while keeping edit lists playback working.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755471
+
+2015-09-23 17:43:51 +0530  Vikram Fugro <vikram.fugro@gmail.com>
+
+	* gst/spectrum/gstspectrum.c:
+	  spectrum: send phase values in the GstMessage for Phase info
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755463
+
+2015-09-22 00:46:01 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: Don't output a warning on MONO multiview mode.
+
+2015-09-19 17:02:18 +0200  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* gst/rtp/gstrtptheoradepay.c:
+	  rtptheoradepay: Fix memory leaks
+	  The same memory leaks were fixed in identical fashion for
+	  vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
+
+2015-09-19 17:04:07 +0200  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtpvorbisdepay.c:
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
+	  * use g_list_free_full(), don't iterate elements maually when freeing
+	  * call gst_rtp_*_pay_clear_packet(), don't duplicate its code
+	  * use gst_buffer_unref() to clarify that it is buffers being released,
+	  instead of refering directly to gst_mini_object_unref()
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
+
+2015-09-19 18:44:22 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755265
+
+=== release 1.5.91 ===
+
+2015-09-18 19:33:13 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.5.91
+
+2015-09-18 19:23:57 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2015-09-18 11:50:31 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/zh_CN.po:
+	  po: Update translations
+
+2015-09-17 10:50:01 +0900  Eunhae Choi <eunhae1.choi@samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Fix taglist leak
+	  gst_tag_list_insert() does not take ownership of the inserted taglist.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=755138
+
+2015-09-16 07:05:36 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: Skip LOAS AAC until a valid config is seen.
+	  It's normal when dropping into the middle of a stream to
+	  not always have the config available immediately, so skip LOAS
+	  until a valid config is seen without either setting invalid
+	  caps or erroring out.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751386
+
+2015-09-13 15:41:38 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: reset just a bit more upon flush_stop
+
+2015-09-13 15:40:09 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: remove dead struct member
+
+2015-09-11 17:09:28 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: fix GError memory leak when hostname resolution fails
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754869
+
+2015-09-10 15:26:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/matroska/ebml-write.c:
+	  matroskamux: drop HEADER flag from output buffers
+	  Drop HEADER flag from output buffers if they are not indeed
+	  headers.
+	  Fixes resending of headers in tcp connection handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754768
+
+2015-09-10 16:00:50 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/ebml-write.c:
+	  matroskamux: fix matroskamux ! matroskademux
+	  Don't carry over DISCONT flags from the input buffers to the
+	  output buffer, or the demuxer might reset its state when it
+	  receives the first data buffer just after parsing the simple
+	  block header, and then expect sane data to follow.
+	  Fixes matroskamux ! demux erroring out.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754768
+	  https://bugzilla.gnome.org/show_bug.cgi?id=657805
+
+2015-09-09 12:51:40 -0700  Martin Kelly <martin@surround.io>
+
+	* gst/rtsp/README:
+	  rtsp: fix small README typo
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754807
+
+2015-09-04 19:45:37 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/audioparsers/gstwavpackparse.c:
+	  wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752106
+
+2015-09-04 19:34:41 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
+	  flac contains the sample offset in the frame header, so after a seek
+	  without index flacparse will know the exact position we landed on and
+	  timestamp buffers accordingly. It only set the pts though, which means
+	  the baseparse-set dts which was set to the seek position prevails, and
+	  since the seek was based on an estimate, there's likely a discrepancy
+	  between where we wanted to land and where we did land, so from here on
+	  that dts/pts difference will be maintained, with dts possibly multiple
+	  seconds ahead of pts, which is just wrong. The easiest way to fix this
+	  is to just set both pts and dts based on the sample offset, but perhaps
+	  parsed audio should just not have dts set at all.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752106
+
+2015-09-06 16:33:02 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	  docs: remove properties and signals that no longer exist
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726443
+
+2013-10-11 15:13:00 +0000  George Chriss <gschriss@gmail.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Make the element count in arrays not include end
+	  One-line removal of tags_written++
+	  This should fix rtmp output to crtmpserver, and hopefully
+	  noone is expecting that the element count includes the end
+	  element, as different bits of documentation say different
+	  things about whether it should or not.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=661624
+
+2015-07-30 00:59:15 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/flv/gstflvmux.c:
+	* gst/flv/gstflvmux.h:
+	  flvmux: Store incoming bitrate tags and send in the metadata
+	  Apparently the Microsoft Azure RTMP server requires that the
+	  videodatarate and audiodatarate metadata be provided, so
+	  set those, even if it's to 0. Use the actual input bitrate
+	  tags if available.
+
+2015-09-04 00:06:29 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Don't parse key data more than needed.
+	  When an auxilliary streams are present in the SDP media,
+	  there's no need to re-parse the SDP attributes multiple
+	  times.
+
+2015-09-03 20:56:55 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
+	  In parse_keymgmt(), don't mutate the input string that's been passed
+	  as const, especially since we might need the original value again if
+	  the same key info applies to multiple streams (RTX, for example).
+	  When a resource is 404, and we have auth info - retry with the auth
+	  info the same as if we had receive unauthorised, in case the resource
+	  isn't even visible until credentials are supplied.
+	  Fix a memory leak handling Mikey data.
+	  When generating a random keystring, don't overrun the 30 byte
+	  buffer by generating 32 bytes into it.
+
+2015-09-04 15:18:05 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Fix build with GLib < 2.44
+	  G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.
+
+2015-09-04 12:01:52 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
+	  This happens on Windows if we use the same socket for sending packets,
+	  and the remote sends ICMP port/host unreachable messages.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754534
+
+2015-09-02 21:12:41 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtptheoradepay.c:
+	* gst/rtp/gstrtpvorbisdepay.c:
+	  rtpvorbis/theoradepay: Fix handling of fragmented packets
+	  This was broken in b1089fb520 by not considering the full packet length of a
+	  fragmented packet but only the length of the first one.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754417
+
+2015-09-01 15:39:22 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/dtmf/gstdtmfsrc.c:
+	* gst/dtmf/gstrtpdtmfsrc.c:
+	  dtmfsrc: Reply to latency query
+
+2015-08-31 16:42:30 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* tests/check/elements/rtpsession.c:
+	  tests: Fix rtpsession test failure
+	  The time of the first RTCP packet is semi-random, so
+	  sometimes it was produced before enough packets from
+	  the second SSRC were received. First drop queued RTCP
+	  packets, then advance the clock enough to ensure
+	  that at least one new RTCP packet is produced.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750731
+
+2015-08-31 13:56:04 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* tests/check/elements/level.c:
+	  level: improve the test for multi-channel mode
+	  Change the test to verify the read-index for multiple messages per buffer.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=754144
+
+2015-08-31 12:46:52 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Align raw video frames to 32 bytes
+	  Outputting unaligned video frames causes videoscale et al to
+	  crash when attempting SIMD-accelerated conversion.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736965
+
+2015-08-26 23:16:46 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/level/gstlevel.c:
+	  level: fix level calculations for mutliple channels
+	  This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.
+
+2015-08-27 10:28:55 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/smpte/gstsmpte.c:
+	  smpte: Fix memory leak
+	  In gst_smpte_collected(), check upfront if input formats are same
+	  or not. This avoids allocation of in1 and in2 buffers and
+	  subsequent memory leak when input formats do not match.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=754153
+
+2015-08-21 11:52:19 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: souphttpsrc: don't try to connect to dead radio server
+
+2015-08-21 16:29:16 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Trivial fix to check correct condition
+	  When checking for describe method, because of missing parentheses, wrong
+	  condition is being checked, which will result in wrong behavior.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753912
+
+2015-08-21 13:19:02 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroska: read: fix tag list memory leak
+	  gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
+	  be unref'ed as soon as the tags are merged to the tocentry
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753904
+
+2015-08-21 12:20:59 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* ext/wavpack/gstwavpackdec.c:
+	  wavpackdec: fix taglist memory leak
+	  When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
+	  by audiodecoder and the caller should free the taglist being passed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753903
+
+2015-08-20 14:45:33 +0200  Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
+
+	* sys/v4l2/gstv4l2transform.c:
+	  v4l2transform: fix pad closing
+	  Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753875
+
+=== release 1.5.90 ===
+
+2015-08-19 13:29:53 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* configure.ac:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-audioparsers.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dtmf.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-imagefreeze.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-isomp4.xml:
+	* docs/plugins/inspect/plugin-jack.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-oss4.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shapewipe.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videofilter.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-vpx.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 1.5.90
+
+2015-08-19 12:47:42 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/en_GB.po:
+	* po/eo.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/gl.po:
+	* po/hr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ro.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
+
+2015-08-19 11:29:55 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/el.po:
+	* po/zh_CN.po:
+	  po: Update translations
+
+2015-08-13 17:29:58 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/multifile/gstmultifilesrc.c:
+	  multifilesrc: fix regression with starting from index set via index property
+	  When we haven't started yet, set the start_index when we set the index property,
+	  so that we start at the right index position after the initial seek. The index
+	  property was never really meant to be for writing, but it used to work, so let's
+	  support it for backwards compatibility.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739472
+
+2015-08-18 10:52:11 +0100  Alex Ashley <bugzilla@ashley-family.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix offset calculation when parsing CENC aux info
+	  Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
+	  DASH common encryption, however commit
+	  bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
+	  shortly before the CENC commit caused the calculation of the CENC
+	  aux info offset to be incorrect.
+	  The base_offset was being added if present, but if the base_offset
+	  is relative to the start of the moof, the offset was being added twice.
+	  The correct approach is to calculate the offset from the start of the
+	  moof and use that offset when parsing the CENC aux info.
+
+2015-08-17 14:28:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: actually return true for accept-caps query handling
+
+2015-08-17 14:07:10 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtp/gstrtpg723pay.c:
+	* gst/rtp/gstrtpgsmpay.c:
+	* gst/rtp/gstrtpklvpay.c:
+	  rtp: copy metadata in the (de)payloaders which is missed before
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753706
+
+2015-08-16 15:21:51 -0400  Dustin Spicuzza <dustin@virtualroadside.com>
+
+	* configure.ac:
+	* sys/directsound/gstdirectsoundsink.c:
+	* sys/directsound/gstdirectsoundsink.h:
+	  directsoundsink: allow specifying audio playback device
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753670
+
+2015-08-16 13:51:47 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: remove single entry if from loop
+	  Iterate from the 2nd channel on and create the 1 channel struct
+	  outside to make loop structure simpler and only slightly faster.
+
+2015-08-16 13:21:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: implement proper accept-caps
+	  Should just compare with what can be immediatelly accepted by
+	  the element. flacenc can't renegotiate so if it has a caps already
+	  it should only accept if it is that caps otherwise just use the
+	  template caps
+
+2015-08-16 13:03:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: improve sink pad template caps
+	  Removes the need for custom caps query handling and makes it more
+	  correct from the beginning on the template. It is a bit uglier
+	  to read because there is 1 entry per channel but makes code easier
+	  to maintain.
+
+2015-08-16 12:41:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/y4m/gsty4mencode.c:
+	  y4mencode: fix gst-launch version in documentation
+
+2015-08-15 22:32:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/speex/gstspeexenc.c:
+	* ext/wavpack/gstwavpackenc.c:
+	* gst/law/alaw-encode.c:
+	* gst/law/mulaw-encode.c:
+	  audioencoders: use template subset check for accept-caps
+	  It is faster than doing a query that propagates downstream and
+	  should be enough
+	  Elements: speexenc, wavpackenc, mulawenc, alawenc
+
+2015-08-15 22:29:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/jpeg/gstjpegenc.c:
+	* ext/libpng/gstpngenc.c:
+	* ext/vpx/gstvp8enc.c:
+	* ext/vpx/gstvp9enc.c:
+	* gst/y4m/gsty4mencode.c:
+	  videoencoders: use template subset check for accept-caps
+	  It is faster than doing a query that propagates downstream and
+	  should be enough
+	  Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
+
+2015-08-16 17:21:24 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/audioparsers/gstmpegaudioparse.c:
+	  mpegaudioparse: use new baseparse API to fix tag handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-03-17 17:50:37 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	  audioparsers: use new base parse API to fix tag handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 14:37:53 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: use new baseparse API and fix tag handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 13:04:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Use signed integer type to be able to check for negative subtraction results
+	  CID 1315829
+
+2015-08-16 11:50:34 +0100  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/rtp/gstrtpvorbisdepay.c:
+	  rtpvorbisdepay: remove dead code
+	  payload_buffer must be NULL in ignore_reserved. Check will always be false.
+	  Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
+	  CID #1316476
+
+2015-08-15 22:45:53 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/law/alaw-encode.c:
+	* gst/law/alaw-encode.h:
+	  alawenc: port to AudioEncoder base class
+
+2015-08-15 09:16:23 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/flac/gstflacdec.c:
+	* ext/speex/gstspeexdec.c:
+	* ext/wavpack/gstwavpackdec.c:
+	* gst/law/alaw-decode.c:
+	* gst/law/mulaw-decode.c:
+	  audiodecoders: use default pad accept-caps handling
+	  Avoids useless check of downstream caps when handling an
+	  accept-caps query
+	  Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
+
+2015-08-15 08:49:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	* ext/libpng/gstpngdec.c:
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp9dec.c:
+	  videodecoders: use default pad accept-caps handling
+	  Avoids useless check of downstream caps when handling an
+	  accept-caps query
+	  Elements: jpegdec, pngdec, vp8dec, vp9dec
+
+2015-08-15 11:31:04 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/law/alaw-decode.c:
+	  alawdec: make error handling a bit nicer
+	  Print the element along with the debug to make it easier to trace
+	  the failures
+
+2015-08-15 11:04:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/law/alaw-decode.c:
+	* gst/law/alaw-decode.h:
+	  alawdec: port to audiodecoder base class
+	  mulawdec was already ported, alawdec was left behind.
+
+2015-08-15 10:34:14 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: only look for more samples in moofs in pull-mode
+	  For playback of some fragmented formats with qtdemux it will
+	  try to look for the next moof after finishing one but it is only
+	  possible for pull-mode. For playback of streaming fragmented formats
+	  such as DASH it should just not try to look for another moof but
+	  instead wait for more data.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752602
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-08-15 12:58:50 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstdcaparse.c:
+	  dcaparse: Don't look for a second syncword
+	  There are streams out there that consistently contain garbage between
+	  every frame so we never ever find a second consecutive syncword.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=738237
+
+2015-08-15 11:12:05 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/vpx/gstvp8enc.c:
+	* ext/vpx/gstvp9enc.c:
+	  vp8enc, vp9enc: reset multipass file index when stopping encoder
+	  Fixes multipass encoding when re-using the same element/pipeline
+	  for subsequent encoding runs.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-15 11:09:42 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvp9enc.h:
+	  vp9enc: provide support for multiple pass cache files
+	  Some files may provide different caps insight of one stream. Since
+	  vp9enc support caps reinit, we should support cache reinit too.
+	  If more then file cache file will be created, the naming will be:
+	  cache cache.1 cache.2 ...
+	  Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-14 11:41:42 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/aacparse.c:
+	  tests: aacparse: use caps query instead of accept-caps
+	  The accept-caps query just does a shallow check at the current
+	  element while at this test we want it to also look at downstream.
+	  So use caps query there.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 11:40:22 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	  audioparsers: enable accept-template flag
+	  Do a quick check with the pad template caps as it is enough. Users
+	  should have figured the appropriate full caps on a previous caps query
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 15:46:53 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: send the User-Agent header
+	  Sometimes it is useful to know this information on the
+	  server side. Other popular implementations (vlc, ffmpeg, ...)
+	  also send this header on every message.
+	  This includes a new "user-agent" property that the user
+	  can set to use a custom User-Agent string. The default
+	  is "GStreamer/<version>"
+	  https://bugzilla.gnome.org/show_bug.cgi?id=750101
+
+2015-08-14 15:42:42 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: wrap gst_rtsp_message_init_request in a local function
+	  This will allow adding common request initialization, like the
+	  user agent string, in just one place.
+
+2015-08-14 09:36:09 +0530  Prashant Gotarne <ps.gotarne@samsung.com>
+
+	* gst/audiofx/audioecho.c:
+	  audioecho: make sure buffer gets reallocated if max_delay changes
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753490
+
+2015-07-09 09:51:26 +0200  Oleksij Rempel <linux@rempel-privat.de>
+
+	* ext/vpx/gstvp8enc.c:
+	* ext/vpx/gstvp8enc.h:
+	  vp8enc: provide support for multiple pass cache files
+	  Some files may provide different caps insight of one stream. Since vp8enc
+	  support caps reinit, we should support cache reinit too.
+	  If more then file cache file will be created, the naming will be:
+	  cache
+	  cache.1
+	  cache.2
+	  ...
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-04-15 22:51:51 +0200  Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+	* gst/rtp/gstrtpmp4gdepay.c:
+	  rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
+	  Use constantDuration to calculate the timestamp of non-first AU in the
+	  RTP packet.
+	  If constantDuration is not present in the MIME parameters, its value
+	  must be calculated based on the timing information from two consecutive
+	  RTP packets with AU-Index equal to 0.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747881
+
+2015-08-14 06:43:13 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: remove unnecessary if, g_free is null safe
+
+2015-08-14 08:33:56 +0100  Alex Ashley <bugzilla@ashley-family.net>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: add property to set HTTP method
+	  To allow souphttpsrc to be use HTTP methods other than GET
+	  (e.g. HEAD), add a "method" property that is a string. If this
+	  property is not set, GET is used.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752413
+
+2015-08-14 11:13:01 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* tests/check/generic/states.c:
+	  check: Rename states unit test
+	  Makes it easier to differentiate from other modules states unit test
+
+2015-08-14 09:21:25 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/goom/gstaudiovisualizer.c:
+	* gst/goom/gstaudiovisualizer.h:
+	* gst/goom2k1/gstaudiovisualizer.c:
+	* gst/goom2k1/gstaudiovisualizer.h:
+	  goom: Rename get_type() function of base class to prevent symbol conflicts
+	  This is a problem when statically linking.
+
+2015-08-13 16:32:55 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
+	  Otherwise we will just output buffers without timestamps after a reset if no
+	  timestamps are provided by upstream, e.g. when using RTSP over TCP.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-08-12 17:16:01 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/matroska/matroska-demux.h:
+	* gst/matroska/matroska-parse.h:
+	  matroska: Remove unused variable
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753556
+
+2015-08-04 20:59:17 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtpL16depay.c:
+	* gst/rtp/gstrtpL24depay.c:
+	* gst/rtp/gstrtpac3depay.c:
+	* gst/rtp/gstrtpac3pay.c:
+	* gst/rtp/gstrtpamrdepay.c:
+	* gst/rtp/gstrtpamrpay.c:
+	* gst/rtp/gstrtpbvdepay.c:
+	* gst/rtp/gstrtpceltdepay.c:
+	* gst/rtp/gstrtpceltpay.c:
+	* gst/rtp/gstrtpdvdepay.c:
+	* gst/rtp/gstrtpdvpay.c:
+	* gst/rtp/gstrtpg722depay.c:
+	* gst/rtp/gstrtpg723pay.c:
+	* gst/rtp/gstrtpg726depay.c:
+	* gst/rtp/gstrtpg729depay.c:
+	* gst/rtp/gstrtpg729pay.c:
+	* gst/rtp/gstrtpgsmdepay.c:
+	* gst/rtp/gstrtpgsmpay.c:
+	* gst/rtp/gstrtpgstdepay.c:
+	* gst/rtp/gstrtpgstpay.c:
+	* gst/rtp/gstrtph261depay.c:
+	* gst/rtp/gstrtph261pay.c:
+	* gst/rtp/gstrtph263depay.c:
+	* gst/rtp/gstrtph263pay.c:
+	* gst/rtp/gstrtph263pdepay.c:
+	* gst/rtp/gstrtph263ppay.c:
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtpilbcdepay.c:
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kpay.c:
+	* gst/rtp/gstrtpjpegdepay.c:
+	* gst/rtp/gstrtpjpegpay.c:
+	* gst/rtp/gstrtpmp1sdepay.c:
+	* gst/rtp/gstrtpmp2tdepay.c:
+	* gst/rtp/gstrtpmp2tpay.c:
+	* gst/rtp/gstrtpmp4adepay.c:
+	* gst/rtp/gstrtpmp4apay.c:
+	* gst/rtp/gstrtpmp4gdepay.c:
+	* gst/rtp/gstrtpmp4gpay.c:
+	* gst/rtp/gstrtpmp4vdepay.c:
+	* gst/rtp/gstrtpmp4vpay.c:
+	* gst/rtp/gstrtpmpadepay.c:
+	* gst/rtp/gstrtpmpapay.c:
+	* gst/rtp/gstrtpmpvdepay.c:
+	* gst/rtp/gstrtpmpvpay.c:
+	* gst/rtp/gstrtppcmadepay.c:
+	* gst/rtp/gstrtppcmudepay.c:
+	* gst/rtp/gstrtpqcelpdepay.c:
+	* gst/rtp/gstrtpqdmdepay.c:
+	* gst/rtp/gstrtpsbcdepay.c:
+	* gst/rtp/gstrtpsbcpay.c:
+	* gst/rtp/gstrtpsirendepay.c:
+	* gst/rtp/gstrtpspeexdepay.c:
+	* gst/rtp/gstrtpspeexpay.c:
+	* gst/rtp/gstrtpsv3vdepay.c:
+	* gst/rtp/gstrtptheoradepay.c:
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtptheorapay.h:
+	* gst/rtp/gstrtputils.c:
+	* gst/rtp/gstrtputils.h:
+	* gst/rtp/gstrtpvorbisdepay.c:
+	* gst/rtp/gstrtpvorbispay.c:
+	* gst/rtp/gstrtpvorbispay.h:
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtp/gstrtpvrawdepay.c:
+	* gst/rtp/gstrtpvrawpay.c:
+	  rtp: Copy metadata in the (de)payloader, but only the relevant ones
+	  The payloader didn't copy anything so far, the depayloader copied every
+	  possible meta. Let's make it consistent and just copy all metas without
+	  tags or with only the video tag.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-10 18:20:15 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix small typo in comment
+
+2015-08-10 16:19:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/goom2k1/gstgoom.c:
+	  goom2k1/doc: Fixup previous commit
+
+2015-08-10 15:55:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* gst/goom2k1/gstgoom.c:
+	* gst/goom2k1/gstgoom.h:
+	  goom2k1/doc: Use GstGoom2k1 namespace
+	  The doc generator isn't happy when we have class name clash. Simply
+	  use it's own namespace.
+
+2015-08-10 17:10:42 +0530  Prashant Gotarne <ps.gotarne@samsung.com>
+
+	* gst/audiofx/audioecho.c:
+	  audioecho: removed unused variable in set_property
+	  unused local variable 'delay' is removed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753450
+
+2015-08-10 12:45:27 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix suboptimal queue iteration code
+
+2015-08-09 17:25:45 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: don't use glib 2.44-only API
+
+2015-07-29 14:14:50 +0100  Alex Ashley <bugzilla@ashley-family.net>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: add support for ISOBMFF Common Encryption
+	  This commit adds support for ISOBMFF Common Encryption (cenc), as
+	  defined in ISO/IEC 23001-7. It uses a GstProtection event to
+	  pass the contents of PSSH boxes to downstream decryptor elements
+	  and attached GstProtectionMeta to each sample.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=705991
+
+2015-08-10 14:13:50 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: checking if depay has sps/pps nals before insertion
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753430
+
+2015-08-08 16:44:49 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: fix outdated comment
+	  The default behaviour was changed in the 0.10 -> 1.x
+	  transition, but the comment was not updated.
+
+2015-08-08 17:42:22 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	  rtptheorapay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:41:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbispay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:34:50 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtptheorapay.h:
+	  rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
+	  We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
+	  and 4:4:4 formats.
+
+2015-08-06 17:46:13 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtp/gstrtpklvdepay.c:
+	* gst/rtp/gstrtpklvpay.c:
+	  rtpklv(de)pay: add "RTP" in the klass string
+	  GstRTSPMedia uses this classification to detect the real payloader
+	  inside a dynpay bin and asserts if it doesn't find it, therefore
+	  it is required
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753325
+
+2015-08-05 11:13:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/rtpaux.c:
+	  tests: rtpaux: use a dynamic pt in the test
+	  1) Tests that using dynamic PT instead of the default ones work
+	  2) If we ever decide to change the codec here we don't need to
+	  worry about change the PT for the default one of the new codec
+	  in the test
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-05 10:53:15 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: print valid type where guint32 is expected
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-06 11:33:37 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtp/gstrtpL16pay.c:
+	* gst/rtp/gstrtpg722pay.c:
+	* gst/rtp/gstrtpg723pay.c:
+	* gst/rtp/gstrtpg729pay.c:
+	* gst/rtp/gstrtpgsmpay.c:
+	* gst/rtp/gstrtph261pay.c:
+	* gst/rtp/gstrtph263pay.c:
+	* gst/rtp/gstrtpjpegpay.c:
+	* gst/rtp/gstrtpmp2tpay.c:
+	* gst/rtp/gstrtpmpapay.c:
+	* gst/rtp/gstrtpmpvpay.c:
+	* gst/rtp/gstrtppcmapay.c:
+	* gst/rtp/gstrtppcmupay.c:
+	  rtppayload: set standard payload type as default
+	  Initialize the PT to the default value of the codec and check if
+	  it is still the default before declaring the pt to be dynamic or
+	  not when setting the caps.
+	  Also use the PT constants from the rtp lib when possible
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747965
+
+2015-07-26 12:07:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: store the moof-offset also for push mode
+	  It will be used in some cases for getting the correct offsets
+	  from trun atoms.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-26 02:09:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_types.h:
+	  qtdemux: handle default-base-is-moof flag
+	  Handle the flag from the tfhd that signals the base offset to
+	  start from the moof atom
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-29 18:54:35 -0600  Glen Diener <grd@loganmill.net>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-read-common.c:
+	* gst/matroska/matroska-read-common.h:
+	  matroskademux: Preserve forward referenced track tags
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752850
+
+2015-08-04 18:07:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/rtpaux.c:
+	  tests: rtpaux: fix test failure
+	  The RTP PT for alaw is 8.
+	  Less than 50 packets are received in the length of this test so it
+	  would never drop a buffer or would drop only the last buffer and
+	  it would fail sometimes when the received wouldn't receive the
+	  retransmission packet in time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-04 20:59:17 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpstreamdepay.c:
+	  rtpstreamdepay: Only allow activation in push mode
+	  We need a proper caps event from upstream with the full RTP caps as we can't
+	  create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
+	  a filesrc or any other element that supports pull mode.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753066
+
+2015-08-04 16:28:17 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  soup: fix typo in translated string
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753240
+
+2015-08-04 12:25:46 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: Put the profile and level into the caps
+
+2015-08-04 12:09:12 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: Only update the srcpad caps if something else than the codec_data changed
+	  h264parse does the same, let's keep the behaviour consistent. As we now
+	  include the codec_data inside the stream too here, this causes less caps
+	  renegotiation.
+
+2015-08-04 11:48:27 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
+	  The spec says:
+	  When a picture parameter set NAL unit with a particular value of
+	  pic_parameter_set_id is received, its content replaces the content of the
+	  previous picture parameter set NAL unit, in decoding order, with the same
+	  value of pic_parameter_set_id (when a previous picture parameter set NAL unit
+	  with the same value of pic_parameter_set_id was present in the bitstream).
+
+2015-08-03 13:45:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: remove extra \n at debug message
+
+2015-08-03 13:42:20 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: prevent deadlock when states change too fast
+	  If the GOP is completed, pads have to start gathering for the
+	  next one but it is possible that the the state might go to
+	  COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
+	  thread has a chance to wake up and proceed, leaving it trapped in
+	  the check_completed_gop loop and deadlocking the other threads
+	  waiting for it to advance.
+	  To solve it, this patch also checks that tha input running time
+	  hasn't changed to prevent this scenario.
+
+2015-08-03 17:55:01 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: Insert SPS/PPS NALs into the stream
+	  h264parse does the same and this fixes decoding of some streams with 32 SPS
+	  (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
+	  the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
+	  As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+	  This looks like a mistake in the part of the spec about the codec_data.
+
+2015-07-30 11:29:27 +0900  Eunhae Choi <eunhae1.choi@samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: handle empty http proxy string
+	  1) If the system http_proxy environment variable is not set
+	  or set to an empty string, we must not set proxy to avoid
+	  http connection error.
+	  2) In case of proxy property setting, if user want to clear
+	  the proxy setting, they should be able to set it to NULL or
+	  an empty string again, so this is fixed too.
+	  3) Check if the proxy string was parsed correctly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752866
+
+2015-07-29 15:46:20 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* ext/dv/gstdvdemux.c:
+	* ext/dv/gstdvdemux.h:
+	  dvdemux: remove unused variable
+	  Remove unused variable 'framecount' from dvdemux
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753008
+
+2015-07-30 15:32:09 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: assertion error due to wrong condition check
+	  In media to caps function, reserved_keys array is being used for variable i,
+	  leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+	  changed it to variable j
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-30 15:21:20 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/rtp/gstrtpmp4vdepay.c:
+	  rtpmp4vdepay: rtpbuffer is being unref'ed twice
+	  process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
+	  the refernce should not be removed here
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753042
+
+2015-07-29 11:26:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Strip keys from the fmtp that we use internally in our caps
+	  Skip keys from the fmtp, which we already use ourselves for the
+	  caps. Some software is adding random things like clock-rate into
+	  the fmtp, and we would otherwise here set a string-typed clock-rate
+	  in the caps... and thus fail to create valid RTP caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 19:28:33 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Support mpegtsmux as a muxer.
+	  As a fallback, look for a pad template sink_%d on
+	  the muxer when requesting pads, to support mpegtsmux
+	  https://bugzilla.gnome.org/show_bug.cgi?id=752999
+
+2015-06-25 01:35:27 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxpartreader.c:
+	* gst/multifile/gstsplitmuxpartreader.h:
+	  splitmuxsrc: Use a separate lock to delay typefind.
+	  Don't hold the main splitmux part lock over
+	  the parent state change function, as it prevents
+	  posting error messages that happen. Since the purpose
+	  is to prevent typefinding from proceeding, use a
+	  separate mutex just for that.
+
+2015-07-29 13:43:50 +0900  Vineeth TM <vineeth.tm@samsung.com>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroska: fix memory leak