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=== release 1.7.2 ===

2016-02-19  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 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

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

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=== release 1.7.1 ===

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2015-12-24 14:16:21 +0100  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/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
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2852 2853 2854 2855 2856 2857 2858 2859 2860 2861 2862 2863 2864 2865 2866 2867 2868 2869 2870 2871 2872 2873 2874 2875 2876 2877 2878 2879 2880 2881 2882 2883 2884 2885 2886 2887 2888 2889 2890 2891 2892 2893 2894 2895 2896 2897 2898 2899 2900 2901 2902 2903 2904 2905 2906 2907 2908 2909 2910 2911 2912 2913 2914 2915 2916 2917 2918 2919 2920 2921 2922 2923 2924 2925 2926 2927 2928 2929 2930 2931 2932 2933 2934 2935 2936 2937 2938 2939 2940 2941 2942 2943 2944 2945 2946 2947 2948 2949 2950 2951 2952 2953 2954 2955 2956 2957 2958 2959 2960 2961 2962 2963 2964 2965 2966 2967 2968 2969 2970 2971 2972 2973 2974 2975 2976 2977 2978 2979 2980 2981 2982 2983 2984 2985 2986 2987 2988 2989 2990 2991 2992 2993 2994 2995 2996 2997 2998 2999 3000 3001 3002 3003 3004 3005 3006 3007

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

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=== release 1.6.0 ===

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2015-09-25 23:15:55 +0200  Sebastian Dröge <sebastian@centricular.com>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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