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=== release 1.9.1 ===
2016-07-06  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.9.1
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>