ChangeLog

	* 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