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=== release 1.5.91 ===

2015-09-18  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.5.91

2015-09-18 11:50:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/zh_CN.po:
	  po: Update translations

2015-09-17 10:50:01 +0900  Eunhae Choi <eunhae1.choi@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix taglist leak
	  gst_tag_list_insert() does not take ownership of the inserted taglist.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755138

2015-09-16 07:05:36 +1000  Jan Schmidt <jan@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Skip LOAS AAC until a valid config is seen.
	  It's normal when dropping into the middle of a stream to
	  not always have the config available immediately, so skip LOAS
	  until a valid config is seen without either setting invalid
	  caps or erroring out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751386

2015-09-13 15:41:38 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: reset just a bit more upon flush_stop

2015-09-13 15:40:09 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: remove dead struct member

2015-09-11 17:09:28 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix GError memory leak when hostname resolution fails
	  https://bugzilla.gnome.org/show_bug.cgi?id=754869

2015-09-10 15:26:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/matroska/ebml-write.c:
	  matroskamux: drop HEADER flag from output buffers
	  Drop HEADER flag from output buffers if they are not indeed
	  headers.
	  Fixes resending of headers in tcp connection handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=754768

2015-09-10 16:00:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/ebml-write.c:
	  matroskamux: fix matroskamux ! matroskademux
	  Don't carry over DISCONT flags from the input buffers to the
	  output buffer, or the demuxer might reset its state when it
	  receives the first data buffer just after parsing the simple
	  block header, and then expect sane data to follow.
	  Fixes matroskamux ! demux erroring out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754768
	  https://bugzilla.gnome.org/show_bug.cgi?id=657805

2015-09-09 12:51:40 -0700  Martin Kelly <martin@surround.io>

	* gst/rtsp/README:
	  rtsp: fix small README typo
	  https://bugzilla.gnome.org/show_bug.cgi?id=754807

2015-09-04 19:45:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
	  https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-04 19:34:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
	  flac contains the sample offset in the frame header, so after a seek
	  without index flacparse will know the exact position we landed on and
	  timestamp buffers accordingly. It only set the pts though, which means
	  the baseparse-set dts which was set to the seek position prevails, and
	  since the seek was based on an estimate, there's likely a discrepancy
	  between where we wanted to land and where we did land, so from here on
	  that dts/pts difference will be maintained, with dts possibly multiple
	  seconds ahead of pts, which is just wrong. The easiest way to fix this
	  is to just set both pts and dts based on the sample offset, but perhaps
	  parsed audio should just not have dts set at all.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-06 16:33:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	  docs: remove properties and signals that no longer exist
	  https://bugzilla.gnome.org/show_bug.cgi?id=726443

2013-10-11 15:13:00 +0000  George Chriss <gschriss@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Make the element count in arrays not include end
	  One-line removal of tags_written++
	  This should fix rtmp output to crtmpserver, and hopefully
	  noone is expecting that the element count includes the end
	  element, as different bits of documentation say different
	  things about whether it should or not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661624

2015-07-30 00:59:15 +1000  Jan Schmidt <jan@centricular.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Store incoming bitrate tags and send in the metadata
	  Apparently the Microsoft Azure RTMP server requires that the
	  videodatarate and audiodatarate metadata be provided, so
	  set those, even if it's to 0. Use the actual input bitrate
	  tags if available.

2015-09-04 00:06:29 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't parse key data more than needed.
	  When an auxilliary streams are present in the SDP media,
	  there's no need to re-parse the SDP attributes multiple
	  times.

2015-09-03 20:56:55 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
	  In parse_keymgmt(), don't mutate the input string that's been passed
	  as const, especially since we might need the original value again if
	  the same key info applies to multiple streams (RTX, for example).
	  When a resource is 404, and we have auth info - retry with the auth
	  info the same as if we had receive unauthorised, in case the resource
	  isn't even visible until credentials are supplied.
	  Fix a memory leak handling Mikey data.
	  When generating a random keystring, don't overrun the 30 byte
	  buffer by generating 32 bytes into it.

2015-09-04 15:18:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix build with GLib < 2.44
	  G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.

2015-09-04 12:01:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
	  This happens on Windows if we use the same socket for sending packets,
	  and the remote sends ICMP port/host unreachable messages.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754534

2015-09-02 21:12:41 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbis/theoradepay: Fix handling of fragmented packets
	  This was broken in b1089fb520 by not considering the full packet length of a
	  fragmented packet but only the length of the first one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754417

2015-09-01 15:39:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmfsrc: Reply to latency query

2015-08-31 16:42:30 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  tests: Fix rtpsession test failure
	  The time of the first RTCP packet is semi-random, so
	  sometimes it was produced before enough packets from
	  the second SSRC were received. First drop queued RTCP
	  packets, then advance the clock enough to ensure
	  that at least one new RTCP packet is produced.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750731

2015-08-31 13:56:04 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  level: improve the test for multi-channel mode
	  Change the test to verify the read-index for multiple messages per buffer.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=754144

2015-08-31 12:46:52 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Align raw video frames to 32 bytes
	  Outputting unaligned video frames causes videoscale et al to
	  crash when attempting SIMD-accelerated conversion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736965

2015-08-26 23:16:46 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: fix level calculations for mutliple channels
	  This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.

2015-08-27 10:28:55 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: Fix memory leak
	  In gst_smpte_collected(), check upfront if input formats are same
	  or not. This avoids allocation of in1 and in2 buffers and
	  subsequent memory leak when input formats do not match.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754153

2015-08-21 11:52:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: don't try to connect to dead radio server

2015-08-21 16:29:16 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Trivial fix to check correct condition
	  When checking for describe method, because of missing parentheses, wrong
	  condition is being checked, which will result in wrong behavior.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753912

2015-08-21 13:19:02 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: read: fix tag list memory leak
	  gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
	  be unref'ed as soon as the tags are merged to the tocentry
	  https://bugzilla.gnome.org/show_bug.cgi?id=753904

2015-08-21 12:20:59 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: fix taglist memory leak
	  When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
	  by audiodecoder and the caller should free the taglist being passed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753903

2015-08-20 14:45:33 +0200  Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fix pad closing
	  Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=753875

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=== release 1.5.90 ===

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2015-08-19 13:29:53 +0300  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/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.5.90

2015-08-19 12:47:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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2015-08-19 11:29:55 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	* po/zh_CN.po:
	  po: Update translations

2015-08-13 17:29:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/multifile/gstmultifilesrc.c:
	  multifilesrc: fix regression with starting from index set via index property
	  When we haven't started yet, set the start_index when we set the index property,
	  so that we start at the right index position after the initial seek. The index
	  property was never really meant to be for writing, but it used to work, so let's
	  support it for backwards compatibility.
	  https://bugzilla.gnome.org/show_bug.cgi?id=739472

2015-08-18 10:52:11 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix offset calculation when parsing CENC aux info
	  Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
	  DASH common encryption, however commit
	  bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
	  shortly before the CENC commit caused the calculation of the CENC
	  aux info offset to be incorrect.
	  The base_offset was being added if present, but if the base_offset
	  is relative to the start of the moof, the offset was being added twice.
	  The correct approach is to calculate the offset from the start of the
	  moof and use that offset when parsing the CENC aux info.

2015-08-17 14:28:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: actually return true for accept-caps query handling

2015-08-17 14:07:10 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpklvpay.c:
	  rtp: copy metadata in the (de)payloaders which is missed before
	  https://bugzilla.gnome.org/show_bug.cgi?id=753706

2015-08-16 15:21:51 -0400  Dustin Spicuzza <dustin@virtualroadside.com>

	* configure.ac:
	* sys/directsound/gstdirectsoundsink.c:
	* sys/directsound/gstdirectsoundsink.h:
	  directsoundsink: allow specifying audio playback device
	  https://bugzilla.gnome.org/show_bug.cgi?id=753670

2015-08-16 13:51:47 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: remove single entry if from loop
	  Iterate from the 2nd channel on and create the 1 channel struct
	  outside to make loop structure simpler and only slightly faster.

2015-08-16 13:21:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: implement proper accept-caps
	  Should just compare with what can be immediatelly accepted by
	  the element. flacenc can't renegotiate so if it has a caps already
	  it should only accept if it is that caps otherwise just use the
	  template caps

2015-08-16 13:03:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacenc.c:
	  flacenc: improve sink pad template caps
	  Removes the need for custom caps query handling and makes it more
	  correct from the beginning on the template. It is a bit uglier
	  to read because there is 1 entry per channel but makes code easier
	  to maintain.

2015-08-16 12:41:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/y4m/gsty4mencode.c:
	  y4mencode: fix gst-launch version in documentation

2015-08-15 22:32:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/speex/gstspeexenc.c:
	* ext/wavpack/gstwavpackenc.c:
	* gst/law/alaw-encode.c:
	* gst/law/mulaw-encode.c:
	  audioencoders: use template subset check for accept-caps
	  It is faster than doing a query that propagates downstream and
	  should be enough
	  Elements: speexenc, wavpackenc, mulawenc, alawenc

2015-08-15 22:29:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngenc.c:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	* gst/y4m/gsty4mencode.c:
	  videoencoders: use template subset check for accept-caps
	  It is faster than doing a query that propagates downstream and
	  should be enough
	  Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc

2015-08-16 17:21:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: use new baseparse API to fix tag handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=679768

2015-03-17 17:50:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: use new base parse API to fix tag handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=679768

2015-08-16 14:37:53 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: use new baseparse API and fix tag handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=679768

2015-08-16 13:04:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use signed integer type to be able to check for negative subtraction results
	  CID 1315829

2015-08-16 11:50:34 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbisdepay: remove dead code
	  payload_buffer must be NULL in ignore_reserved. Check will always be false.
	  Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
	  CID #1316476

2015-08-15 22:45:53 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	  alawenc: port to AudioEncoder base class

2015-08-15 09:16:23 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/flac/gstflacdec.c:
	* ext/speex/gstspeexdec.c:
	* ext/wavpack/gstwavpackdec.c:
	* gst/law/alaw-decode.c:
	* gst/law/mulaw-decode.c:
	  audiodecoders: use default pad accept-caps handling
	  Avoids useless check of downstream caps when handling an
	  accept-caps query
	  Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec

2015-08-15 08:49:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  videodecoders: use default pad accept-caps handling
	  Avoids useless check of downstream caps when handling an
	  accept-caps query
	  Elements: jpegdec, pngdec, vp8dec, vp9dec

2015-08-15 11:31:04 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/law/alaw-decode.c:
	  alawdec: make error handling a bit nicer
	  Print the element along with the debug to make it easier to trace
	  the failures

2015-08-15 11:04:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	  alawdec: port to audiodecoder base class
	  mulawdec was already ported, alawdec was left behind.

2015-08-15 10:34:14 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only look for more samples in moofs in pull-mode
	  For playback of some fragmented formats with qtdemux it will
	  try to look for the next moof after finishing one but it is only
	  possible for pull-mode. For playback of streaming fragmented formats
	  such as DASH it should just not try to look for another moof but
	  instead wait for more data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752602
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2015-08-15 12:58:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: Don't look for a second syncword
	  There are streams out there that consistently contain garbage between
	  every frame so we never ever find a second consecutive syncword.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=738237

2015-08-15 11:12:05 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp9enc.c:
	  vp8enc, vp9enc: reset multipass file index when stopping encoder
	  Fixes multipass encoding when re-using the same element/pipeline
	  for subsequent encoding runs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747728

2015-08-15 11:09:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	  vp9enc: provide support for multiple pass cache files
	  Some files may provide different caps insight of one stream. Since
	  vp9enc support caps reinit, we should support cache reinit too.
	  If more then file cache file will be created, the naming will be:
	  cache cache.1 cache.2 ...
	  Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
	  https://bugzilla.gnome.org/show_bug.cgi?id=747728

2015-08-14 11:41:42 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/aacparse.c:
	  tests: aacparse: use caps query instead of accept-caps
	  The accept-caps query just does a shallow check at the current
	  element while at this test we want it to also look at downstream.
	  So use caps query there.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753623

2015-08-14 11:40:22 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: enable accept-template flag
	  Do a quick check with the pad template caps as it is enough. Users
	  should have figured the appropriate full caps on a previous caps query
	  https://bugzilla.gnome.org/show_bug.cgi?id=753623

2015-08-14 15:46:53 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: send the User-Agent header
	  Sometimes it is useful to know this information on the
	  server side. Other popular implementations (vlc, ffmpeg, ...)
	  also send this header on every message.
	  This includes a new "user-agent" property that the user
	  can set to use a custom User-Agent string. The default
	  is "GStreamer/<version>"
	  https://bugzilla.gnome.org/show_bug.cgi?id=750101

2015-08-14 15:42:42 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: wrap gst_rtsp_message_init_request in a local function
	  This will allow adding common request initialization, like the
	  user agent string, in just one place.

2015-08-14 09:36:09 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* gst/audiofx/audioecho.c:
	  audioecho: make sure buffer gets reallocated if max_delay changes
	  https://bugzilla.gnome.org/show_bug.cgi?id=753490

2015-07-09 09:51:26 +0200  Oleksij Rempel <linux@rempel-privat.de>

	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	  vp8enc: provide support for multiple pass cache files
	  Some files may provide different caps insight of one stream. Since vp8enc
	  support caps reinit, we should support cache reinit too.
	  If more then file cache file will be created, the naming will be:
	  cache
	  cache.1
	  cache.2
	  ...
	  https://bugzilla.gnome.org/show_bug.cgi?id=747728

2015-04-15 22:51:51 +0200  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/rtp/gstrtpmp4gdepay.c:
	  rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
	  Use constantDuration to calculate the timestamp of non-first AU in the
	  RTP packet.
	  If constantDuration is not present in the MIME parameters, its value
	  must be calculated based on the timing information from two consecutive
	  RTP packets with AU-Index equal to 0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747881

2015-08-14 06:43:13 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: remove unnecessary if, g_free is null safe

2015-08-14 08:33:56 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add property to set HTTP method
	  To allow souphttpsrc to be use HTTP methods other than GET
	  (e.g. HEAD), add a "method" property that is a string. If this
	  property is not set, GET is used.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752413

2015-08-14 11:13:01 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/generic/states.c:
	  check: Rename states unit test
	  Makes it easier to differentiate from other modules states unit test

2015-08-14 09:21:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/goom/gstaudiovisualizer.c:
	* gst/goom/gstaudiovisualizer.h:
	* gst/goom2k1/gstaudiovisualizer.c:
	* gst/goom2k1/gstaudiovisualizer.h:
	  goom: Rename get_type() function of base class to prevent symbol conflicts
	  This is a problem when statically linking.

2015-08-13 16:32:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
	  Otherwise we will just output buffers without timestamps after a reset if no
	  timestamps are provided by upstream, e.g. when using RTSP over TCP.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-08-12 17:16:01 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.h:
	  matroska: Remove unused variable
	  https://bugzilla.gnome.org/show_bug.cgi?id=753556

2015-08-04 20:59:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	* gst/rtp/gstrtputils.c:
	* gst/rtp/gstrtputils.h:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	* gst/rtp/gstrtpvp8depay.c:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	  rtp: Copy metadata in the (de)payloader, but only the relevant ones
	  The payloader didn't copy anything so far, the depayloader copied every
	  possible meta. Let's make it consistent and just copy all metas without
	  tags or with only the video tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751774

2015-08-10 18:20:15 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix small typo in comment

2015-08-10 16:19:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/goom2k1/gstgoom.c:
	  goom2k1/doc: Fixup previous commit

2015-08-10 15:55:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/goom2k1/gstgoom.c:
	* gst/goom2k1/gstgoom.h:
	  goom2k1/doc: Use GstGoom2k1 namespace
	  The doc generator isn't happy when we have class name clash. Simply
	  use it's own namespace.

2015-08-10 17:10:42 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* gst/audiofx/audioecho.c:
	  audioecho: removed unused variable in set_property
	  unused local variable 'delay' is removed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753450

2015-08-10 12:45:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix suboptimal queue iteration code

2015-08-09 17:25:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: don't use glib 2.44-only API

2015-07-29 14:14:50 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: add support for ISOBMFF Common Encryption
	  This commit adds support for ISOBMFF Common Encryption (cenc), as
	  defined in ISO/IEC 23001-7. It uses a GstProtection event to
	  pass the contents of PSSH boxes to downstream decryptor elements
	  and attached GstProtectionMeta to each sample.
	  https://bugzilla.gnome.org/show_bug.cgi?id=705991

2015-08-10 14:13:50 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: checking if depay has sps/pps nals before insertion
	  https://bugzilla.gnome.org/show_bug.cgi?id=753430

2015-08-08 16:44:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: fix outdated comment
	  The default behaviour was changed in the 0.10 -> 1.x
	  transition, but the comment was not updated.

2015-08-08 17:42:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtptheorapay: If flushing a packet failed, go out of the loop immediately

2015-08-08 17:41:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtpvorbispay: If flushing a packet failed, go out of the loop immediately

2015-08-08 17:34:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtptheorapay.h:
	  rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
	  We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
	  and 4:4:4 formats.

2015-08-06 17:46:13 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvpay.c:
	  rtpklv(de)pay: add "RTP" in the klass string
	  GstRTSPMedia uses this classification to detect the real payloader
	  inside a dynpay bin and asserts if it doesn't find it, therefore
	  it is required
	  https://bugzilla.gnome.org/show_bug.cgi?id=753325

2015-08-05 11:13:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpaux.c:
	  tests: rtpaux: use a dynamic pt in the test
	  1) Tests that using dynamic PT instead of the default ones work
	  2) If we ever decide to change the codec here we don't need to
	  worry about change the PT for the default one of the new codec
	  in the test
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-08-05 10:53:15 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtpmanager/gstrtprtxsend.c:
	  rtprtxsend: print valid type where guint32 is expected
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-08-06 11:33:37 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtp/gstrtpL16pay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmupay.c:
	  rtppayload: set standard payload type as default
	  Initialize the PT to the default value of the codec and check if
	  it is still the default before declaring the pt to be dynamic or
	  not when setting the caps.
	  Also use the PT constants from the rtp lib when possible
	  https://bugzilla.gnome.org/show_bug.cgi?id=747965

2015-07-26 12:07:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: store the moof-offset also for push mode
	  It will be used in some cases for getting the correct offsets
	  from trun atoms.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2015-07-26 02:09:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.h:
	  qtdemux: handle default-base-is-moof flag
	  Handle the flag from the tfhd that signals the base offset to
	  start from the moof atom
	  https://bugzilla.gnome.org/show_bug.cgi?id=752603

2015-07-29 18:54:35 -0600  Glen Diener <grd@loganmill.net>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroskademux: Preserve forward referenced track tags
	  https://bugzilla.gnome.org/show_bug.cgi?id=752850

2015-08-04 18:07:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpaux.c:
	  tests: rtpaux: fix test failure
	  The RTP PT for alaw is 8.
	  Less than 50 packets are received in the length of this test so it
	  would never drop a buffer or would drop only the last buffer and
	  it would fail sometimes when the received wouldn't receive the
	  retransmission packet in time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746445

2015-08-04 20:59:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpstreamdepay.c:
	  rtpstreamdepay: Only allow activation in push mode
	  We need a proper caps event from upstream with the full RTP caps as we can't
	  create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
	  a filesrc or any other element that supports pull mode.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753066

2015-08-04 16:28:17 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  soup: fix typo in translated string
	  https://bugzilla.gnome.org/show_bug.cgi?id=753240

2015-08-04 12:25:46 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Put the profile and level into the caps

2015-08-04 12:09:12 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Only update the srcpad caps if something else than the codec_data changed
	  h264parse does the same, let's keep the behaviour consistent. As we now
	  include the codec_data inside the stream too here, this causes less caps
	  renegotiation.

2015-08-04 11:48:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
	  The spec says:
	  When a picture parameter set NAL unit with a particular value of
	  pic_parameter_set_id is received, its content replaces the content of the
	  previous picture parameter set NAL unit, in decoding order, with the same
	  value of pic_parameter_set_id (when a previous picture parameter set NAL unit
	  with the same value of pic_parameter_set_id was present in the bitstream).

2015-08-03 13:45:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: remove extra \n at debug message

2015-08-03 13:42:20 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: prevent deadlock when states change too fast
	  If the GOP is completed, pads have to start gathering for the
	  next one but it is possible that the the state might go to
	  COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
	  thread has a chance to wake up and proceed, leaving it trapped in
	  the check_completed_gop loop and deadlocking the other threads
	  waiting for it to advance.
	  To solve it, this patch also checks that tha input running time
	  hasn't changed to prevent this scenario.

2015-08-03 17:55:01 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: Insert SPS/PPS NALs into the stream
	  h264parse does the same and this fixes decoding of some streams with 32 SPS
	  (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
	  the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
	  As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
	  This looks like a mistake in the part of the spec about the codec_data.

2015-07-30 11:29:27 +0900  Eunhae Choi <eunhae1.choi@samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: handle empty http proxy string
	  1) If the system http_proxy environment variable is not set
	  or set to an empty string, we must not set proxy to avoid
	  http connection error.
	  2) In case of proxy property setting, if user want to clear
	  the proxy setting, they should be able to set it to NULL or
	  an empty string again, so this is fixed too.
	  3) Check if the proxy string was parsed correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752866

2015-07-29 15:46:20 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: remove unused variable
	  Remove unused variable 'framecount' from dvdemux
	  https://bugzilla.gnome.org/show_bug.cgi?id=753008

2015-07-30 15:32:09 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: assertion error due to wrong condition check
	  In media to caps function, reserved_keys array is being used for variable i,
	  leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
	  changed it to variable j
	  https://bugzilla.gnome.org/show_bug.cgi?id=753009

2015-07-30 15:21:20 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtp/gstrtpmp4vdepay.c:
	  rtpmp4vdepay: rtpbuffer is being unref'ed twice
	  process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
	  the refernce should not be removed here
	  https://bugzilla.gnome.org/show_bug.cgi?id=753042

2015-07-29 11:26:46 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Strip keys from the fmtp that we use internally in our caps
	  Skip keys from the fmtp, which we already use ourselves for the
	  caps. Some software is adding random things like clock-rate into
	  the fmtp, and we would otherwise here set a string-typed clock-rate
	  in the caps... and thus fail to create valid RTP caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=753009

2015-07-29 19:28:33 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Support mpegtsmux as a muxer.
	  As a fallback, look for a pad template sink_%d on
	  the muxer when requesting pads, to support mpegtsmux
	  https://bugzilla.gnome.org/show_bug.cgi?id=752999

2015-06-25 01:35:27 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	  splitmuxsrc: Use a separate lock to delay typefind.
	  Don't hold the main splitmux part lock over
	  the parent state change function, as it prevents
	  posting error messages that happen. Since the purpose
	  is to prevent typefinding from proceeding, use a
	  separate mutex just for that.

2015-07-29 13:43:50 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska: fix memory leak
	  After adding to tag list, key_val is not being free'd
	  resulting in memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=752992

2015-07-27 13:34:14 +0900  Manasa Athreya <manasa.athreya@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
	  'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
	  be more bits as well, in which case it's just like 'twos'.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752613

2015-07-24 15:10:05 +0200  Dimitrios Katsaros <patcherwork@gmail.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	  v4l2: Allow framerate to be large then 100pfs
	  This limit was arbitrary. We still fixate near 100pfs for compatibility.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752825

2015-07-25 03:25:28 -0400  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/avi/gstavidemux.c:
	  avidemux: Stop without posting error on flushing
	  This could just be a normal pipeline shutdown.

2015-07-23 15:00:08 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
	  https://bugzilla.gnome.org/show_bug.cgi?id=752618

2015-07-16 18:09:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/matroskademux.c:
	  tests: add minmal matroskademux test for subtitle output
	  Some of the subtitle chunks will have embedded
	  NUL-terminators (last three), some don't (first three),
	  some will have markup, some won't, some will be valid
	  UTF-8 (all but last), some won't (last stanza).
	  https://bugzilla.gnome.org/show_bug.cgi?id=752421

2015-07-16 18:49:26 +0300  Dimitrios Christidis <dchristidis@mykolab.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix for subtitle buffers with NUL terminators
	  Commit 45892ec8 created a regression where g_utf8_validate() would fail
	  if the subtitle buffer had a NUL terminator as part of the data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752421

2015-07-21 13:31:05 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: Check available bytes before copy
	  Need to check that the number of bytes we want to copy from the adapter
	  actually is available and handle the error case gracefully. This error
	  may happen if malformed packets are received and we don't have a
	  complete frame.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752663

2015-07-16 09:32:36 +0900  Paul Hyunil <paul.hyunil@lge.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: Support subtitle when track subtype is fourcc_subt
	  https://bugzilla.gnome.org/show_bug.cgi?id=752655

2015-07-20 16:59:40 +0800  Song Bing <b06498@freescale.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Set timestamp when queue buffer.
	  Should set timestamp when queue buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752618

2015-07-16 15:12:17 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* tests/check/elements/rtpmux.c:
	  rtpmux: handle different ssrc's on sinkpads
	  Do this by not putting the ssrc from the src pads in the caps used to
	  probe other sinkpads, and then  intersecting with it later.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752491

2015-07-16 17:19:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/avi/gstavimux.c:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/webm-mux.c:
	  Update mailing list address from sourceforge to freedesktop

2015-07-15 13:44:52 +0300  Dimitrios Christidis <dchristidis@mykolab.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix trailing '*' displayed with some text subtitles
	  The subtitle buffer we push out should not include a NUL terminator
	  as part of the data, we just add such a terminator for safety, but
	  it should not be included in the buffer size.
	  A NUL terminator is not valid UTF-8, so checks will fail if it's
	  included in the size, and the NUL will be replaced by the fallback
	  character specified when converting, i.e. '*'.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752421

2015-07-15 18:23:05 +0200  Wim Taymans <wtaymans@redhat.com>

	* ext/pulse/pulsedeviceprovider.c:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulse: add properties to GstDevice
	  Add the extra properties we get from pulse to the GstDevice we expose
	  with the device monitor

2015-07-15 17:20:20 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audioinvert.c:
	* gst/audiofx/audiowsincband.c:
	  audiofx: Fix typo in example pipelines
	  Fix typo in example pipelines of audiowsincband and audioinvert.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752416

2015-04-15 18:27:04 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: add a "format-location" signal that allows better control over filenames
	  In certain applications, splitting into files named after a base
	  location template and an incremental sequence number is not enough.
	  This signal gives more fine-grained control to the application to
	  decide how to name the files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750106

2015-04-15 20:13:27 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudiosrc: no resampling on OS X
	  Unlike Remote IO, AUHAL doesn't have built-in resampling
	  for sources -- confirmed by Core Audio engineer Doug Wyatt:
	  http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-04-15 18:29:14 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudiosrc: avoid get_channel_layout
	  This only produces a warning and serves no purpose.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-04-07 15:40:14 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: Avoid making a duplicate structure in caps for mono/stereo case
	  For 1ch or 2ch devices, we just need to set the caps to allow both
	  options since CoreAudio will up/downmix appropriately.
	  Also fixes the condition for the 2ch case to be exact, rather than at
	  least 2 channels since the downmix will not take place in the >stereo
	  case.

2015-04-06 16:22:34 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: Don't set the format on an initialized AudioUnit
	  We need to initialize the AudioUnit early to be able to probe the
	  underlying device, but according to the AudioUnitInitialize() and
	  AudioUnitUninitialize() documentation, format changes should be done
	  while the AudioUnit is uninitialized. So we explicitly uninitialize the
	  AudioUnit during a format change and reinitialize it when we're done.

2015-04-06 15:55:59 +0530  Arun Raghavan <arun@centricular.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: Minor spelling fix (unitialize -> uninitialize)

2015-03-21 20:34:25 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudio: Fix lockup in _audio_unit_property_listener
	  _audio_unit_property_listener is called either from a Core Audio thread
	  or as a result of a Core Audio API (e.g. AudioUnitInitialize)
	  from our own thread. In the latter case, osxbuf can be already locked
	  (GStreamer's mutex is not recursive).
	  We introduce the flag cached_caps_valid and use it instead of nullifying
	  cached_caps when we cannot lock on osxbuf.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-03-12 12:15:12 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: Invalidate cached caps on format change
	  Listen for changes in hardware stream format and channel layout, and
	  invalidate cached caps (since they contain the preferred caps).
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-03-09 23:34:06 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxaudioringbuffer.c:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	* sys/osxaudio/gstosxcoreaudiohal.c:
	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: Overhaul of probing caps
	  - Probing caps is unified between source and sink
	  - Hardware stream format is now reported as preferred capabilities
	  (dynamically updated when hardware configuration changes)
	  - Get hardware channel layout from Remote IO just like from HAL
	  - More comprehensive mapping between AudioChannelLabel and
	  GstAudioChannelPosition
	  - Support for unpositioned channel layouts
	  - Announce stereo-mono upmixing/downmixing in caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=743758

2015-03-09 23:15:56 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudio.c:
	  osxaudio: AudioUnitInitialize on open
	  Call AudioUnitInitialize upon open. Otherwise, we cannot get
	  (hardware) stream format nor channel layout from the outer scope.

2015-07-12 14:27:15 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpac3depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpbvdepay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpdvdepay.c:
	* gst/rtp/gstrtpg722depay.c:
	* gst/rtp/gstrtpg723depay.c:
	* gst/rtp/gstrtpg726depay.c:
	* gst/rtp/gstrtpg729depay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph263depay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpj2kdepay.c:
	* gst/rtp/gstrtpjpegdepay.c:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpmp1sdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvdepay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvp8depay.c:
	  rtp: depayloaders: 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.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750235

2015-05-27 19:19:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpvrawdepay.c:
	  rtpvrawdepay: 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 map the base class has already done.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750235

2015-07-10 00:13:32 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix indention

2015-07-09 23:59:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Always estimate DTS from the current clock time
	  Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
	  we would produce wrong DTS. As now the estimated DTS is based on the clock,
	  don't store it in the jitterbuffer items as it would otherwise be used in the
	  skew calculations and would influence the results. We only really need the DTS
	  for timer calculations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-09 09:26:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/.gitignore:
	  gitignore: ignore rtph263 test

2015-07-08 23:47:44 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
	  Replace static constants with macros to make gcc happy
	  CC       elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
	  elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
	  static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
	  ^
	  elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
	  static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
	  ^
	  elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
	  PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;

2015-07-08 23:40:45 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: run indent and fix some comments
	  Fix indent on this file and break some comment lines into two to make
	  it fit 80 chars per line

2015-07-08 15:02:24 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: rework segment event handling for adaptive streaming
	  When a new time segment is received upstream is going to restart
	  with a new atom. Make the neededbytes and todrop variables
	  reflect that to avoid waiting too much or dropping the
	  initial bytes that contain the header.

2015-07-08 12:35:55 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push data from adapter before starting new segment
	  The adapter might have data remaining from the previous segment,
	  push it all before clearing the adapter and starting a new segment.
	  It can accumulate data if it had pushed and got not-linked, returning
	  immediately without processing all the data. Before starting a new
	  segment this data should be handled.

2015-07-08 19:59:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-08 21:08:36 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: fix gap-time calculation and remove "late"
	  The amount of time that is completely expired and not worth waiting for,
	  is the duration of the packets in the gap (gap * duration) - the
	  latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
	  that we make a "multi-lost" packet for.
	  The "late" concept made some sense in 0.10 as it reflected that a buffer
	  coming in had not been waited for at all, but had a timestamp that was
	  outside the jitterbuffer to wait for. With the rewrite of the waiting
	  (timeout) mechanism in 1.0, this no longer makes any sense, and the
	  variable no longer reflects anything meaningful (num > 0 is useless,
	  the duration is what matters)
	  Fixed up the tests that had been slightly modified in 1.0 to allow faulty
	  behavior to sneak in, and port some of them to use GstHarness.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738363

2015-06-30 11:21:31 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
	  This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0.
	  The reverted patch is wrong and introduces a regression because there
	  may still be time to receive some of the packets included in the gap
	  if they are reordered.

2015-07-07 23:53:02 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: flush samples before adding more from moof
	  Avoids accumulating all samples from a fragmented stream that could
	  lead to a 'index-too-big' error once it goes over 50MB of data. It
	  could reach that before 2h of playback so it doesn't take that long.
	  As upstream elements are providing data in time format they should
	  be the ones that have more information about the full media index
	  and should be able to seek if possible.

2015-07-07 23:56:12 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: rename upstream_newsegment to upstream_format_is_time
	  upstream_newsegment isn't really clear on what it means, it is set
	  to TRUE when the upstream element sends a segment in TIME format, so
	  rename it to be more clear about it.
	  It is important to know this because it means that upstream has
	  a notion of time and qtdemux is likely being driven by an upstream
	  element that is reading from a higher level abstraction than a file,
	  such as a DASH, MSS or DLNA element.

2015-07-07 21:31:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix leak by flushing previous sample info from trak
	  In fragmented streaming, multiple moov/moof will be parsed and their
	  previously stored samples array might leak when new values are parsed.
	  The parse_trak and callees won't free the previously stored values
	  before parsing the new ones.
	  In step-by-step, this is what happens:
	  1) initial moov is parsed, traks as well, streams are created. The
	  trak doesn't contain samples because they are in the moof's trun
	  boxes. n_samples is set to 0 while parsing the trak and the samples
	  array is still NULL.
	  2) moofs are parsed, and their trun boxes will increase n_samples and
	  create/extend the samples array
	  3) At some point a new moov might be sent (bitrate switching, for example)
	  and parsing the trak will overwrite n_samples with the values from
	  this trak. If the n_samples is set to 0 qtdemux will assume that
	  the samples array is NULL and will leak it when a new one is
	  created for the subsequent moofs.
	  This patch makes qtdemux properly free previous sample data before
	  creating new ones and adds an assert to catch future occurrences of
	  this issue when the code changes.

2015-07-07 16:46:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix index size check and debug message
	  It is allocating samples_count + n_samples, not only n_samples

2015-07-08 17:02:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Calculate receive time if we don't have any
	  This is required to properly schedule packet loss timers and make
	  sure all our calculations work properly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-08 15:13:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
	  That is, handle DTS==GST_CLOCK_TIME_NONE correctly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749536

2015-07-08 20:31:42 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix event leak
	  when seek fails in avidemux, event is not being freed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752117

2015-07-08 12:02:22 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtph263depay.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtph263.c:
	  rtph263depay: Make sure payload is large enough
	  Plus new unit test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752112

2015-07-08 08:59:49 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/rtp/gstrtpklvdepay.c:
	  rtpklvdepay: fix printf format compiler warning
	  v_len is of type guint64, but while print the value(16 + len_size + v_len)
	  G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT
	  https://bugzilla.gnome.org/show_bug.cgi?id=752100

2015-07-07 20:25:47 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-rtp.xml:
	  docs: add new RTP elements to docs

2015-07-07 20:07:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: add basic unit test for KLV payloading
	  Also make it so that the mtu is always set if specified, not
	  only in case of the rather weird bufferlist test code path.
	  This allows us to easily make the payloader fragment a payload
	  across multiple output packets by setting a small MTU on it.

2015-07-07 19:58:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvdepay.h:
	  rtpklvdepay: improve start detection and handle fragmented KLV units

2015-07-05 20:25:10 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpklvdepay.c:
	* gst/rtp/gstrtpklvdepay.h:
	  rtp: add SMPTE 336M KLV metadata depayloader
	  http://tools.ietf.org/html/rfc6597

2014-08-09 10:08:42 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpklvpay.c:
	* gst/rtp/gstrtpklvpay.h:
	  rtp: add SMPTE 336M KLV metadata payloader
	  http://tools.ietf.org/html/rfc6597

2015-07-07 16:59:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/properties.h:
	* gst/matroska/matroska-mux.c:
	* gst/rtpmanager/rtpsource.c:
	  docs: fix "Symbol name not found at the start of the comment block"
	  Add symbols or change comment into a regular comment.

2015-07-07 16:58:53 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audioparsers/gstamrparse.h:
	  docs: remove outdated doc strings

2015-07-03 23:10:40 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  docs: add missing plugins and ensure master doc is sorted

2015-07-07 15:54:41 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* gst/imagefreeze/gstimagefreeze.c:
	  Revert "imagefreeze: Remove impossible error condition"
	  This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94.
	  pad only handle EOS events but not EOS flow, and will push the buffer again
	  resulting in an assertion error. So we should not handle the buffer
	  and return EOS flow.

2015-07-07 15:50:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpg729depay.c:
	  rtpg729depay: unmap rtp buffer in error path

2015-07-07 15:48:40 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: fix buffer leak
	  The handle_buffer vfunc takes ownership of the input buffer.
	  Fixes elements/rtp-payloading under valgrind.

2015-07-02 08:52:43 +0200  Tobias Mueller <muelli@cryptobitch.de>

	* gst/goom/goom_core.c:
	  goom: Initialised variables to remove compiler warnings
	  goom_core.c: In function 'goom_update':
	  goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
	  ^
	  goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
	  goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
	  ^
	  https://bugzilla.gnome.org/show_bug.cgi?id=752053

2015-07-07 09:18:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: fix indentation

2015-07-06 19:11:00 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: Fix uninitialized variable compiler error
	  endpos variable does not correctly understand in the
	  4.6.3 GCC version. So compile error appears when we do
	  compile rtph261pay using jhbuild.
	  This patch is fixed the compile error in 4.6.3 GCC version.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751985

2014-11-12 12:08:58 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Handle seek flags properly
	  Allows for non-keyframe seeks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=738570

2015-02-24 10:50:52 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: avoid looping reading the 'moof' atom forever
	  It gets stuck if it only finds a moof and no mfra/mfro or moov
	  atoms. Skip the moof to continue the parsing to have it either
	  play or error out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745089

2015-06-26 13:24:17 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/flac/gstflacdec.c:
	  flacdec: improve error handling
	  for files which have corrupted header, libflac is not able to
	  process the metadata properly. We just try to ignore the error
	  and continue with the processing, since metadata parsing is not
	  making much of a difference to libflac
	  https://bugzilla.gnome.org/show_bug.cgi?id=751334

2015-07-06 20:16:38 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* sys/ximage/ximageutil.c:
	  ximagesrc: add meta transform function
	  ximage metadata can't be transformed or copied, but provide an empty
	  transformation function instead of NULL to allow unconditional calling
	  of metas' transform functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751778

2014-06-16 16:14:28 +0200  Stian Selnes <stian.selnes@gmail.com>

	* gst/rtp/gstrtph263pdepay.c:
	  rtph263pdepay: init debug category
	  https://bugzilla.gnome.org/show_bug.cgi?id=752012

2014-06-20 10:59:14 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpv8depay: ignore reserved bit in payload descriptor
	  Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
	  R: Bit reserved for future use.  MUST be set to zero and MUST be
	  ignored by the receiver.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751929

2015-07-04 20:56:42 +0200  Stian Selnes <stian@pexip.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: rtph261depay: Add documentation
	  https://bugzilla.gnome.org/show_bug.cgi?id=751982

2015-07-03 21:58:14 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From f74b2df to 9aed1d7

2015-07-03 14:29:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: Fix compiler warning
	  gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
	  gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
	  GObjectClass *gobject_class;

2015-07-03 14:03:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph261depay.c:
	  rtph261depay: Let the base class push the buffer so it can deal with the flow return

2015-07-03 14:11:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph261pay.c:
	  rtph261pay: Remove unused adapter

2015-07-03 13:17:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpspeexpay.c:
	  speexpay: Directly attach payload to the output buffer instead of copying it

2015-07-03 13:07:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpsbcpay.c:
	  sbcpay: Attach payload directly to the output instead of copying

2014-12-01 14:18:40 +0100  Stian Selnes <stian@pexip.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtph261depay.c:
	* gst/rtp/gstrtph261depay.h:
	* gst/rtp/gstrtph261pay.c:
	* gst/rtp/gstrtph261pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtp: add H.261 RTP payloader and depayloader
	  Implementation according to RFC 4587.
	  Payloader create fragments on MB boundaries in order to match MTU size
	  the best it can. Some decoders/depayloaders in the wild are very strict
	  about receiving a continuous bit-stream (e.g. no no-op bits between
	  frames), so the payloader will shift the compressed bit-stream of a
	  frame to align with the last significant bit of the previous frame.
	  Depayloader does not try to be fancy in case of packet loss. It simply
	  drops all packets for a frame if there is a loss, keeping it simple.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751886

2015-07-03 12:18:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmpvdepay.c:
	  rtpmpvdepay: Don't forget to unmap the input buffer

2015-07-03 12:14:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmpvpay.c:
	  rtpmpvpay: Create buffer lists instead of pushing each buffer individually

2015-07-03 12:03:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmpapay.c:
	  rtpmpapay: Use buffer lists instead of pushing each fragment individually

2015-07-03 10:51:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpmp4apay.c:
	  rtpmp4apay: Create buffer lists and don't copy payload memory

2015-06-29 16:14:18 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT
	  When there are a lot of small gaps, we can consider that there is
	  a big gap (too losses) to reset the buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751636

2015-06-29 15:53:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: If possible, always update the current time before looping over all timers
	  If we have a clock, update "now" now with the very latest running time we have.
	  If timers are unscheduled below we otherwise wouldn't update now (it's only updated
	  when timers expire), and also for the very first loop iteration now would otherwise
	  always be 0.
	  Also the time is used for the timeout functions, e.g. to calculate any times
	  for the next timeouts and we would otherwise pass too old times there.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751636

2015-07-02 14:34:57 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: fix memory leak
	  tmp needs to be freed before going out of scope in 'done'.
	  CID #1308954

2015-07-02 12:23:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it

2015-07-02 09:48:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263pdepay.c:
	  rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us

2015-07-02 09:17:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pay.h:
	  rtph263pay: Stop using an adapter and directly use the buffer
	  We always pushed one buffer into the adapter, then handled exactly that one
	  buffer and flushed it from the adapter. Now also don't memcpy() the actual
	  payload but just attach the input buffer's data to the output buffer.
	  This code still needs some serious refactoring/rewriting.

2015-07-01 21:57:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgsmpay.c:
	  rtpgsmpay: Remove non-existing includes for now
	  git add -p mistake.

2015-07-01 19:29:07 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Use the return value of gst_buffer_append()

2015-07-01 19:19:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpgsmpay.c:
	  rtpgsmpay: Attach payload to the output buffer instead of copying it

2015-07-01 17:58:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpg729pay.c:
	  rtpg729pay: Attach payload directly to output buffers instead of copying

2015-07-01 17:43:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpg723pay.c:
	  rtpg723pay: Attach payload buffer to the output instead of copying

2015-07-01 17:30:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpdvdepay.c:
	  rtpdvdepay: Map the output buffer once instead of once every 80 bytes

2015-07-01 21:46:46 +0900  Jimmy Ohn <yongjin.ohn@lge.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix return type of index_entry_offset_search()
	  It's a compare function and may return a negative value,
	  so should for correctness and consistency return a signed
	  integer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751780

2015-07-01 14:12:57 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor handle_next_buffer
	  The goal of this patch is making handle_next_buffer function
	  more readable avoiding unnecesary gotos and adding other
	  cosmetic changes.

2015-07-01 15:40:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpac3pay.c:
	  rtpac3pay: Attach the payload to the output buffer instead of copying it
	  Might also want to produce buffer lists here if needed.

2015-07-01 15:38:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpsirendepay.c:
	  rtp: Fix indention

2015-07-01 12:37:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-VP8-OPUS.sh:
	* tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh:
	  rtp: Add examples with VTS/ATS for VP8/OPUS
	  Let's have an example with modern codecs.

2015-06-30 18:11:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()

2015-06-30 14:06:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpvp8depay.c:
	  vp8depay: Don't lock/map every non-keyframe buffer twice
	  Just copy the complete header instead of first looking at the first byte
	  and then at the remaining 10 bytes.

2015-06-29 16:05:44 +0100  Luis de Bethencourt <luis@debethencourt.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: document fallthrough cases
	  Pacify coverity and document fallthrough cases in switch statements.
	  CID #1308948, #1308947, #1308946

2015-06-29 10:36:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout"
	  This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182.
	  If we have multiple immediate timers, we want to first handle the one with the
	  lowest sequence number... which would be broken now.
	  Instead of this we should just use a GSequence for the timers, and have them
	  sorted first by timestamp, and for equal timestamps by sequence number. Then
	  we would always only have to take the very first timer from the list and never
	  have to look at any others.

2015-06-29 10:14:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout
	  If we have lots of such immediate timeouts, we would otherwise have quadratic
	  runtime in the number of timeouts.

2015-06-19 18:01:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: sticky events are sent automatically from the pad
	  No need to send them explicitly from the element
	  https://bugzilla.gnome.org/show_bug.cgi?id=751240

2015-06-19 18:00:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: make sure to push sticky events before adding pad
	  It allows the caps to be set on the pad before being added for
	  dynamic autoplugging to work.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751240

2015-06-26 00:05:29 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
	  Enable to use new ntp-time-source property of rtpbin
	  https://bugzilla.gnome.org/show_bug.cgi?id=751496

2015-06-25 23:19:58 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin/session: fix description
	  https://bugzilla.gnome.org/show_bug.cgi?id=751496

2015-06-25 10:57:25 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/imagefreeze/gstimagefreeze.c:
	* gst/matroska/matroska-demux.c:
	* tests/examples/shapewipe/shapewipe-example.c:
	  docs: decodebin2 -> decodebin

2015-06-25 10:47:06 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: update example pipeline
	  Update reference to decodebin2 to decodebin

2015-06-25 10:45:35 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: remove dead assignments
	  Values in fields_required and same_buffer are overwritten before used. Removing
	  assignment

2015-06-25 10:06:07 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/Makefile.am:
	* ext/mikmod/Makefile.am:
	* ext/mikmod/README:
	* ext/mikmod/drv_gst.c:
	* ext/mikmod/gstmikmod.c:
	* ext/mikmod/gstmikmod.h:
	* ext/mikmod/mikmod_reader.c:
	* ext/mikmod/mikmod_types.c:
	* ext/mikmod/mikmod_types.h:
	* m4/Makefile.am:
	* m4/libmikmod.m4:
	* win32/MANIFEST:
	* win32/vs8/libgstmikmod.vcproj:
	  mikmod: remove ancient unported plugin
	  This hasn't been touched in 11 years, and
	  clearly no one's been missing it.

2015-06-23 20:15:13 +0900  Gilbok Lee <gilbok.lee@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: does not detect orientation
	  Most files don't contain the values for transposing the coordinates
	  back to the positive quadrant so qtdemux was ignoring the rotation
	  tag. To be able to properly handle those files qtdemux will also ignore
	  the transposing values to only detect the rotation using the values
	  abde from the transformation matrix:
	  [a b c]
	  [d e f]
	  [g h i]
	  https://bugzilla.gnome.org/show_bug.cgi?id=738681

2015-06-25 00:04:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

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=== release 1.5.2 ===

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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/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.5.2
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2015-06-24 22:56:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-06-24 11:15:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/nl.po:
	  po: Update translations

2015-06-23 18:42:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/qtmux.c:
	  qtmux: Correctly test each segments
	  In presence of gaps, qtdemux will emit multiple segments. The
	  second segment start should match the CTTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 17:54:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Correctly calculate the elst media start
	  The media start has nothing to do with the shift we have applied
	  but with the value of the first PTS. This is defined as:
	  Dt(0) = 0
	  Ct(0) = Dt(0) + CTTS(0)
	  So the media start is always the first CTTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 11:49:32 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: accumulate previous edts entries into segment.base
	  Allows playing edts editted files with proper synchronization of
	  streams. This patch fixes the regression introduced by
	  bf95f93c0189aa04f18e264b86b6527e431c5d53 that was added to fix
	  segment seeks handling.
	  Having the accumulated_base separated from the main segment.base
	  allows handling both segment seeks and edts editted files.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 00:56:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: improve some debug messages
	  Those messages are about the stream, use the pad as the
	  debug object to make it clear from the logs
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-22 22:22:09 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: store last_dts of the first buffer
	  Buffers need not to start at running-time 0 so the last_dts needs
	  to be the value of the first buffer's dts as it is used to compute
	  the duration of the buffers. If it was left at 0 the first buffer
	  would have a larger duration when it shouldn't
	  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 17:11:57 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: fix possible memory leak
	  when buffer is stored to seektable, and stop gets called due to
	  corrupt flac file, then the seektable is not being released
	  https://bugzilla.gnome.org/show_bug.cgi?id=751364

2015-06-23 16:28:40 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  Revert "splitmuxsink: Mask async-start/done while switching files."
	  This reverts commit d61e5393f110ed482815d77807245d78b52eff46.
	  Causes failures muxing larger GOP sizes for some reason. Reverting
	  while I figure it out

2015-06-18 23:22:06 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Fix startup and shutdown races.
	  Fix 2 startup races when things happen too quickly, and 1
	  at shutdown by holding a ref to the pads in use until the
	  loop functions exit.
	  Handle errors activating file parts and publish them on
	  the bus.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-18 09:26:13 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Mask async-start/done while switching files.
	  Sometimes, extra async-start/done from the internal sink
	  while the element is still starting up can cause splitmuxsink
	  to stall in PAUSED state when it has been set to PLAYING
	  by the app. Drop the child's async-start/done messages while
	  switching, so they don't cause state changes at the
	  splitmuxsink level.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-15 16:12:10 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Use gst_video_multiview_guess_half_aspect()
	  Use the gst_video_multiview_guess_half_aspect() utility function
	  to set the half-aspect flag (or not) on stereoscopic frame-packed
	  videos.

2015-06-15 16:10:37 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Move multiview caps calculations, add half-aspect heuristics
	  Move the multiview caps calculations to the configure_stream()
	  function, so the rest of the video info is available, and
	  use the gst_video_multiview_guess_half_aspect() function to
	  determine if the half-aspect flag should be set on frame-packed
	  video.

2015-06-18 16:06:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add cslg support
	  The cslg atom provide information about the DTS shift. This is
	  needed in recent version of ctts atom where the offset can be
	  negative. When cslg is missing, we parse the CTTS table as proposed
	  in the spec to calculate these values.
	  In this implementation, we only need to know the shift. As GStreamer
	  cannot transport negative timestamps, we shift the timestamps forward
	  using that value and adapt the segment to compensate. This patch also
	  removes bogus offset of ctts_soffset, this offset shall be included
	  in the edit list.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751103

2015-06-19 18:37:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/qtmux.c:
	  qtmux: Test gaps at start of stream
	  https://bugzilla.gnome.org/show_bug.cgi?id=751242

2015-06-19 18:40:43 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use PTS to figure-out presence of gaps
	  We need to look at the presentation timestamp in order to conclude if
	  there is a gap at the start of a stream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751242

2015-06-19 16:45:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Set edit list to compensate DTS shift
	  We shift DTS forward to avoid negative timestamps which cannot be
	  represented with version 0 of the CTTS table. To stick with that
	  version (backward compatibility), the spec recommend using an
	  edit list entry to move back the presentation time to where it
	  should be.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751242

2015-06-22 14:35:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Insert AVC end of sequence
	  This FLV specific mark is needed to prevent Flow Player (most likely
	  all Flash base player) from going into buffering state when near EOS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751320

2015-06-22 13:05:29 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	  matroska: remove useless check
	  No need to check for context availability while freeing. We are inside
	  inside a code block with a condition that dereferences context.
	  if (context->type == 0 ...
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 19:35:57 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/matroska/lzo.c:
	  lzo: fix memory leak
	  the opened file is not being closed during test, which will result
	  in memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 19:30:58 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* ext/mikmod/mikmod_reader.c:
	  mikmod_reader: Possible null pointer dereference:
	  gst_reader variable is being used before actually checking if it
	  allocated properly
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 19:45:14 +0900  Sangkyu Park <sk1122.park@samsung.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Minor clean-up
	  1. Fix the code which is wrong coding style.
	  2. Fix a typing error of comment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751316

2015-06-22 11:28:13 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Do not try to push NULL buffers
	  If update_receiver_stats() fails, we can't really do anything with this buffer
	  anymore and have to drop it. This happens if there's a big seqnum
	  discontinuity for example.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2015-06-22 13:10:02 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: trivial cleanup
	  trivial patch to add proper ( while checking for if(G_UNLIKELY())
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 13:16:08 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstdcaparse.c:
	  dcaparse: initialize size variable
	  size can be used in cleanup without being initialized. Hence
	  setting it to 0 when declaring
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-22 13:13:29 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: initialze bpf variable
	  bpf variable might be used in cleanup without being intialized.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751306

2015-06-19 14:50:59 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtprtxqueue.c:
	  rtprtxqueue: reverse pending list before pushing buffers
	  With this we send the RTX buffers in the same order
	  that they were requested.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751297

2015-06-21 19:22:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/flv/gstflvmux.c:
	  flvmux: Fix DTS validity check
	  This check was up-side-down, causing a bad timestamp at start
	  and then all timestamp being delayed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751298

2015-06-17 15:19:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  cslg: Add Composition Shift Least Greatest Atom
	  This simply add fourcc and dump function for the cslg Atom.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751103

2015-06-17 15:18:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/qtdemux_dump.c:
	  ctts_dump: Fix signess issues
	  It didn't bug, but use correct signess in traces. The number of
	  entries is unsigned while the offset can be signed according to
	  recent spec.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751103

2015-06-16 17:48:08 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* common:
	  Automatic update of common submodule
	  From 6015d26 to f74b2df

2015-06-16 11:43:39 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: gst_rtp_buffer_ext_timestamp() modifies its first argument, keep a copy around

2015-06-16 10:30:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Compare ext RTP times, not plain RTP time and ext RTP time when calculating elapsed time
	  Otherwise all RTP times after a wraparound would be considered as going
	  backwards, they will always be smaller than the ext RTP time.

2015-06-15 19:25:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: The default rtp-profile should be AVP, not AVPF

2015-06-15 14:32:21 +0900  Sangkyu Park <sk1122.park@samsung.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Minor cleanup
	  1. Add Null check in 'free_item' function.
	  2. Fix a typing error of comment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750965

2015-06-12 17:44:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	  flmux: Make sure best_time is initialized

2015-06-12 23:29:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	  rtpbin/session: Add new ntp-time-source property and deprecate use-pipeline-clock property
	  The new property allows to select the time source that should be used for the
	  NTP time in RTCP packets. By default it will continue to calculate the NTP
	  timestamp (1900 epoch) based on the realtime clock. Alternatively it can use
	  the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's
	  clock time. The latter is especially useful for synchronizing multiple
	  receivers if all of them share the same clock.
	  If use-pipeline-clock is set to TRUE, it will override the ntp-time-source
	  setting and continue to use the running time plus 70 years. This is only kept
	  for backwards compatibility.

2015-04-07 16:03:42 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/qtmux.c:
	  tests: qtmux: test for muxing with DTS outside the segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-06-11 17:26:49 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Adjust segment according to ctts offset
	  In presence of a CTTS, the segment start/stop must be offset so
	  the segment start/stop include the PTS. This is needed since the
	  PTS cannot be negative in this format. This fixes issues where the
	  running time of the first buffer isn't at the start.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-04-03 20:34:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: Handle DTS with negative running time
	  As QT works with duration, simply bring back first DTS to 0 and shift
	  forward the PTS of the same amount.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-06-10 18:15:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Add negative runtime DTS support
	  This is done by using new feature of the CollectPad clip function
	  which sets the DTS as a gint64 in the collected data. It also simplify
	  the code a bit.
	  https://bugzilla.gnome.org/show_bug.cgi?id=740575

2015-06-12 23:06:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Rename some variables and debug output to make more sense
	  Local and remote were mixed up in a few places, and the time we store here is
	  not UNIX time (1970 epoch), but NTP time (1900 epoch) in nanoseconds.

2015-06-12 19:21:10 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* sys/osxaudio/gstosxcoreaudioremoteio.c:
	  osxaudio: fix latency property query on RemoteIO
	  AudioUnitGetProperty would fail with kParamErr (-50) every time,
	  simply because size wasn't initialized.
	  Now it returns zero latency, but at least it doesn't fail.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750868

2015-04-07 14:06:16 +0530  Arun Raghavan <git@arunraghavan.net>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Fix mapping of latency parameters to buffer attributes

2015-06-12 01:56:37 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroska-demux: Actually set detected 3D info into output caps.
	  Use the information read from the StereoMode info
	  to configure multiview-mode and multiview-flags in the
	  video caps.

2015-06-11 13:36:54 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Take released-but-not-yet-output bytes into account
	  When deciding whether it's time to switch to a new file, take into
	  account data that's been released for pushing, but hasn't yet
	  been pushed - because downstream is slow or the threads haven't been
	  scheduled.
	  Fixes a race in the unit test and probably in practice - sometimes
	  failing to switch when it should for an extra GOP or two.
	  Also fix a problem in splitmuxsrc where playback sometimes
	  stalls at startup if types are found too quickly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-11 01:04:51 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  atoms: remove custom gst_buffer_new function in favor of core version
	  Remove a custom specialized version of gst_buffer_new_wrapped by
	  using gst_buffer_new_wrapped_full inside a macro to simplify
	  parameters and give it a more meaningful name.
	  It is only used to create temporary buffers to have its data copied.

2015-06-11 00:14:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  atoms: simplify free form data atoms creation
	  Avoid creating an intermediary buffer or memory area just
	  to copy into an atom's data area.

2015-06-10 22:27:27 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: add AC-3 muxing support
	  Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.
	  One extra feature that was added was the ability to add extension
	  atoms after set_caps as the AC-3 extension atom needs some data
	  that has to be extracted from the stream itself and is not
	  present on caps.

2015-06-10 22:36:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	  qtmux: remove unused type MP4S

2015-06-10 22:29:01 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove duplicate attribute value set
	  It is also set a few lines below

2015-06-11 00:22:54 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroska: Implement basic stereoscopic video support
	  Implement support for the packed video formats WebM
	  uses, not all the values that Matroska might use.
	  In practice, it's really hard to find any samples in the
	  wild of any.
	  Supported in both the muxer and demuxer.

2015-06-10 01:26:15 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add basic support for MPEG-A stereoscopic video
	  The MPEG-A format provides an extension to the ISO base media
	  file format to store stereoscopic content encoded with different
	  codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
	  atom declares the presence and storage method for the video.
	  Stereo video information for MPEG-A can also be supplied through
	  the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
	  is not implemented in this patch.
	  Also missing is support for stereo video encoded as separate video tracks
	  for now.
	  Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=611157

2015-06-02 16:15:35 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add tls-database property
	  https://bugzilla.gnome.org/show_bug.cgi?id=750298

2015-06-10 14:33:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstasteriskh263.c:
	* gst/rtp/gstrtpac3pay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpceltdepay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpdvpay.c:
	* gst/rtp/gstrtpg723pay.c:
	* gst/rtp/gstrtpg729pay.c:
	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpj2kpay.c:
	* gst/rtp/gstrtpjpegpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtpmp4apay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmparobustdepay.c:
	* gst/rtp/gstrtpmpvpay.c:
	* gst/rtp/gstrtpqcelpdepay.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpsbcpay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvrawdepay.c:
	* gst/rtp/gstrtpvrawpay.c:
	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
	  The mix between all these in the RTP code is confusing, let's try to be
	  consistent.

2015-06-10 14:49:50 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: clarify negative lost packets in stats
	  Also:
	  - Move notes on units before field documentation.
	  - Unify documentation style.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750653

2015-06-10 06:38:39 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix getter of "ssl-use-system-ca-file"
	  https://bugzilla.gnome.org/show_bug.cgi?id=750298

2015-06-10 09:49:47 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix reverse playback
	  When performing seek, segment->start is being updated with desired_offset,
	  but in case of reverse playback segment->start should be 0 and
	  segment->stop should be updated with desired offset.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750675

2015-01-21 18:09:03 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2allocator.c:
	* sys/v4l2/gstv4l2allocator.h:
	* sys/v4l2/gstv4l2bufferpool.c:
	  gstv4l2bufferpool: handle -EPIPE from DQBUF to signal EOS
	  The V4L2 decoder signals EOS by returning -EPIPE from DQBUF after the
	  last buffer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=743338

2015-06-06 21:09:19 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add a GTlsInteraction property
	  It can be used for TLS client authentication.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750471

2015-01-09 11:36:11 +0100  Enrico Jorns <ejo@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2: Allow scaling in the v4l2*convert element
	  This is inspired of videoscale and videoconvert elements.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742917

2015-06-09 19:02:55 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: document units of stats and arguments
	  Also, minor spelling and style corrections.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750653

2015-06-09 14:42:27 +0200  Stefan Sauer <ensonic@users.sf.net>

	* Makefile.am:
	  cruft: add the obsolete tmpl dir to cruft-dirs

2015-06-09 11:30:22 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Automatic update of common submodule
	  From d9a3353 to 6015d26

2015-06-09 07:04:07 +0200  Edward Hervey <bilboed@bilboed.com>

	* common:
	  Fix common version
	  Was accidently downgraded by 87a4884acd8655a6591d735a1d944ecb5ea3de16

2015-06-08 19:11:41 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Also set colorimetry on output devices
	  This completes the code that set the colorimetry on output
	  device.

2015-06-08 19:10:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* common:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add missing SMTP240M matrix
	  This is missing in the doc, but was in the header.

2015-06-08 23:00:16 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom/goom_core.c:
	  goom: possible uninitialized variables warning
	  Build fails with the latest snapshot of gcc-4.9 because param1 and param2 might
	  possibly be used uninitialized. They are set depending on the cases of a switch
	  statement and the compiler sees this as not a complete guarantee.
	  Set them to 0 if the switch statement falls down to the default case.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750566#c6

2015-06-08 17:24:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fully implement colorimetry support
	  This fixes wrong mapping for sRGB as in GStreamer sRGB correctly
	  apply to RGB formats, while in V4L2 it's an alias for sYCC. Also
	  add support for the new quantization (range), ycbcr_encoding (matrix)
	  and xfer_func (transfer) enumeration.

2015-06-08 17:01:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/ext/types-compat.h:
	* sys/v4l2/ext/v4l2-common.h:
	* sys/v4l2/ext/v4l2-controls.h:
	* sys/v4l2/ext/videodev2.h:
	  v4l2: Update kernel headers to latest from media tree
	  This is the latest from media tree. This should enable more development
	  of the v4l2 elements. This includes new flags requires to fix draining
	  path in decoder, colorimetry and much more.

2015-06-08 23:07:55 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From d37af32 to d9a3353

2015-06-08 19:42:30 +0100  Chris Clayton <chris2553@googlemail.com>

	* gst/rtp/gstrtpvp8pay.c:
	  rtpvp8depay: potential access beyond end of array
	  Compiling (with gcc-4.9-20150603) produces an error because of an access beyond
	  the end of an array. This patch fixes the error by initializing the loop
	  control/array index variable (i) to 1 and returning i - 1 when a match is found.
	  Also, because the values stored in the array increase in value as the index
	  increases, the >= test unnecessary, so it is removed.

2015-04-30 02:52:58 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Don't accumulate more than 2 GOPs
	  Don't allow large amounts of data to queue up - we only need
	  the GOP we're writing, and the GOP we're accumulating.

2015-04-16 10:44:49 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: fsync after sending updates in robust mode
	  Use the new GstBuffer SYNC_AFTER flag to trigger an fsync
	  after updating the moov or mdat atom, and after updating the free
	  atom to make it visible.

2015-04-03 00:57:20 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: Only set moov header into streamheader at EOS
	  Only update the moov header into the caps if it's the finalised
	  moov at EOS time. Avoids posting a bogus moov at startup and
	  repeated updates in robust-recording mode

2015-04-03 01:44:15 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/qtmux.c:
	  tests: Update mp4 mux test for mdat placeholder change
	  The mp4 muxer now writes a place-holder mdat as a free
	  atom followed by a 0-byte mdat that covers the rest of the
	  file, making it possible to rewrite it as 64-bit, or leave
	  it as-is if nothing else is written afterward

2015-04-01 11:15:38 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/atomsrecovery.c:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  isomp4: Implement robust muxing using ping-pong strategy
	  Implement a robust recording mode, where the output
	  file is always in a playable state, seeking and rewriting
	  the moov header at a configurable interval. Rewriting
	  moov is done using reserved space at the start of
	  the file, and a ping-pong strategy where the moov
	  is replaced atomically so it's never invalid.
	  Track when tags have actually changed, and don't write them into
	  the moov unless they've changed. Clear any existing tags when
	  re-writing them, so we can do progressive moov updating in robust
	  recording mode.
	  Write placeholder mdat as a free atom plus a 32-bit mdat
	  with '0' size, which means "rest of the file" in the spec.
	  Re-write it later to a full 64-bit extended size atom if needed.

2015-04-01 00:58:52 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  isomp4: Update edit list when re-writing moov
	  Correctly update any edit lists each time the moov is recalculated,
	  updating existing table entries if they already exist instead of just
	  adding new ones.

2015-04-08 01:41:18 +1000  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: Remove an extra bracket in a comment.

2015-03-19 20:29:44 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Protect total_duration state variable with the object lock.
	  Prevent deadlocks from downstream querying duration from the streaming thread.

2015-06-07 23:06:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From 21ba2e5 to d37af32

2015-06-07 19:24:20 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom/gstaudiovisualizer.c:
	  goom: clean dereferences of private structure
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-06-07 19:20:04 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/goom2k1/gstaudiovisualizer.c:
	  goom2k1: clean dereferences of private structure
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-06-07 17:32:01 +0200  Stefan Sauer <ensonic@users.sf.net>

	* common:
	  Automatic update of common submodule
	  From c408583 to 21ba2e5

2015-06-07 17:01:37 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  docs: remove variables that we define in the snippet from common
	  This is syncing our Makefile.am with upstream gtkdoc.

2015-06-07 17:16:19 +0200  Stefan Sauer <ensonic@users.sf.net>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From d676993 to c408583

2015-06-07 16:44:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

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=== release 1.5.1 ===

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2015-06-07 10:46:34 +0200  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/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/deinterlace/tvtime-dist.c:
	* gst/videomixer/videomixerorc-dist.c:
	* win32/common/config.h:
	  Release 1.5.1

2015-06-07 10:38:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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