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=== release 1.7.1 ===

2015-12-24  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.7.1

2015-12-24 12:22:32 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/cs.po:
	* po/de.po:
	* po/el.po:
	* po/hu.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2015-12-21 09:57:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: drop flushes from our own offset seek
	  Prevents downstream from receiving flushes for a seek only in
	  upstream. Those seeks are only to start reading from the right
	  offset when skipping or returning to qt atoms.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758928

2015-11-11 16:53:19 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Always set the channel mask for PCM streams
	  Just use the gst_audio_channel_get_fallback_mask function for now as
	  the specification is too complicated and nobody implements it.

2015-12-21 11:37:26 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix sleep for buffer-time lower than 200000
	  https://bugzilla.gnome.org/show_bug.cgi?id=748680

2015-12-21 12:31:19 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Use -Bsymbolic-functions if available
	  While this is more useful for libraries, some of our plugins with multiple
	  files and some internal API can also benefit from this.

2015-12-18 15:34:52 +0000  William Manley <will@williammanley.net>

	* gst/debugutils/progressreport.c:
	* gst/debugutils/progressreport.h:
	  progressreport: add support for using format=buffers with do-query=false
	  This is useful for investigating and debugging pipelines which are
	  producing buffers at a slower/faster rate than you would expect.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759635

2015-12-18 15:49:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Update formats table
	  This change add all the new RGB based format. Those format removes the
	  ambiguity with the ALPHA channel. Some other missing multiplanar format
	  has been added with some additional cleanup.

2015-12-18 05:17:15 +1100  Jan Schmidt <jan@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't write invalid edit list start time.
	  Avoid writing a negative number as a large positive
	  integer in an edit list when the first_ts is smaller
	  than the first_dts - which can happen when the first
	  packet received has a PTS but no DTS.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759615

2015-12-04 23:16:45 +1100  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Only update running time when it increases.
	  Don't increment running time from every buffer. The correct
	  logic to only increment when running time advances is a
	  little further down, so delete this left-over line.

2015-11-18 11:01:20 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Implement prores support
	  https://bugzilla.gnome.org/show_bug.cgi?id=758258

2015-11-18 16:20:38 +1100  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroska-demux: Play ProRes video streams
	  Generate video/x-prores caps for ProRes video streams.
	  Every frame needs an 8 byte header prepended, as described in
	  http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout
	  so do that in a post-processing callback.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758258

2015-12-18 10:18:09 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* ext/dv/gstdvdec.h:
	  dvdec: Remove unused fields
	  Remove unused fields frame_len and space
	  https://bugzilla.gnome.org/show_bug.cgi?id=759614

2015-12-17 16:03:04 +0100  Vincent Dehors <vincent.dehors@openwide.fr>

	* gst/rtp/gstrtpj2kdepay.c:
	  rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=758943

2015-12-16 11:43:58 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/raw1394/gstdv1394src.c:
	* ext/raw1394/gsthdv1394src.c:
	  dv1394: log error if failed to set socket status flag
	  Log an error message if failed to set write or read socket as
	  non-blocking.
	  CID 1139608
	  CID 1139609

2015-12-15 17:10:00 +0000  Dave Craig <davecraig@unbalancedaudio.com>

	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstac3parse.c:
	* gst/audioparsers/gstamrparse.c:
	* gst/audioparsers/gstdcaparse.c:
	* gst/audioparsers/gstflacparse.c:
	* gst/audioparsers/gstmpegaudioparse.c:
	* gst/audioparsers/gstsbcparse.c:
	* gst/audioparsers/gstwavpackparse.c:
	  audioparsers: Check for NULL return value of gst_pad_get_current_caps()
	  https://bugzilla.gnome.org/show_bug.cgi?id=759503

2015-12-16 09:35:53 +0100  Sebastian Dröge <sebastian@centricular.com>

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

2015-12-15 14:27:22 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/Makefile.am:
	  vpx: Add missing headers in Makefile.am
	  This fixes distcheck.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755510

2015-09-24 12:57:00 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp8enc.c:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/gstvpxenc.c:
	* ext/vpx/gstvpxenc.h:
	  vpx: created common baseclass GstVPXEnc
	  GstVP8Enc and GstVP9Enc has almost 80% code in common.
	  created common baseclass GstVPXEnc for GstVP8Enc and GstVP9Enc
	  https://bugzilla.gnome.org/show_bug.cgi?id=755510

2015-12-15 12:57:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxdec.h:
	  vpxdec: Remove unneeded add video_meta
	  This also remove copies for VP8, which was not correctly in place
	  in previous related patch.

2015-12-15 09:49:24 +0530  Prashant Gotarne <ps.gotarne@samsung.com>

	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvpxdec.c:
	* ext/vpx/gstvpxdec.h:
	  vpx: created common base class GstVPXdec for vpx decoders
	  Base class for the vp8dec and vp9dec.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755510

2015-06-10 09:17:08 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* configure.ac:
	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: Add GTlsInteraction property
	  https://bugzilla.gnome.org/show_bug.cgi?id=750709

2015-12-14 09:05:06 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Retry connection if tunneling needs authentication
	  Leverage response from gst_rtsp_connection_connect_with_response to
	  determine if the connection should be retried using authentication.  If
	  so, add the appropriate authentication headers based upon the response
	  and retry the connection.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749596

2015-12-14 14:19:05 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: check port-range format
	  The string could exist but with a wrong format, in that case we still want
	  to reset the values of client_port_range.min and max like we do if there is
	  no string.
	  CID 1139593

2015-12-14 14:55:12 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Check device property and fail if device can't be found
	  Don't use default if a specific device is set but it can't be found.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759452

2015-12-14 14:15:00 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix handling of the mute property
	  - set mute value at startup
	  - correct set and get mute functions
	  https://bugzilla.gnome.org/show_bug.cgi?id=755106

2015-12-11 11:23:13 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Check the return value of GetStatus() too to decide if there was an error
	  If GetStatus() fails, the status itself won't be very meaningful but we also
	  have to look at its return value. This fixes blocking pipelines when removing
	  sound devices or during other errors, where we wouldn't notice the error and
	  then wait forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734098

2015-12-10 17:41:46 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  isomp4: remove unused parameters in build_*_extension
	  AtomTRAK parameter is not used by build_mov_alac_extension(),
	  build_jp2h_extension(), or build_mov_alac_extension()  and can be
	  removed.

2015-12-10 15:11:07 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/gstqtmux.c:
	  isomp4: replace variable only used once
	  Replace has_shift variable with value since it is only use once.

2015-12-09 12:24:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix packet dropping after a big discont
	  We would queue 5 consective packets before considering a reset and a proper
	  discont here. Instead of expecting the next output packet to have the current
	  seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
	  going to drop all queued up packets.

2015-12-09 11:49:02 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/interleave/interleave.h:
	  interleave: Remove unsed field
	  Remove unused field collect_event in interleave.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759226

2015-12-07 16:33:14 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Stop pushing data as soon as possible in push-mode
	  When working in push-mode, we attempt to push out everything currently
	  buffered in the adapter.
	  This has two pitfalls:
	  * We could stop earlier (the moment we get a non-ok or non-not-linked)
	  * We return the last combined flow return, which might be completely
	  different from the previous combined flow return

2015-12-07 09:08:09 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From b319909 to 86e4663

2015-12-07 14:41:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: Add a warning if an empty RTCP packet is tried to be sent
	  https://bugzilla.gnome.org/show_bug.cgi?id=759119

2015-11-30 19:20:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9dec.h:
	  vpxdec: Use GstMemory to avoid copies
	  With the VPX decoders it's not simple to use downstream buffer pool,
	  because we don't know the image size and alignment when buffers get
	  allocated. We can though use GstAllocator (for downstream, or the system
	  allocator) to avoid a copy before pushing if downstream supports
	  GstVideoMeta. This would still cause a copy for sink that requires
	  specialized memory and does not have a GstAllocator for that, though
	  it will greatly improve performance for sink like glimagesink and
	  cluttersink. To avoid allocating for every buffer, we also use a
	  internal buffer pool.
	  https://bugzilla.gnome.org/show_bug.cgi?id=745372

2015-11-30 08:42:35 +0100  Edward Hervey <edward@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Avoid over-skipping when checking LOAS config
	  There might be multiple LOAS config in a row in a full frame. The first
	  one might be a multi-layer config (which we can't properly parse yet)...
	  but then followed by a valid (single-layer) one.
	  The code was previously skipping whole frames (instead of just the LOAS
	  config we failed to read) resulting in multiple frames (seen up to 6s in
	  some situation) being dropped before finally getting the configuration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758826

2015-11-25 17:08:56 +0100  Edward Hervey <edward@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Properly set SPARSE stream flags for subpicture/subtitle
	  And while we're at it, also detect 'DXSA' as being a variant fourcc
	  of 'DXSB' for XSUB

2015-11-30 21:23:52 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: grammar fix

2015-11-30 21:01:17 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* tests/check/elements/souphttpsrc.c:
	  tests: souphttpsrc: switch shoutcast stream provider
	  Fixes failing ICY test. Previous provider has
	  streaming disabled outside UK.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758114

2015-11-18 16:10:11 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: don't crash if we never got audio caps before stopping
	  auds.blockalign is set once the first caps arrive. If
	  gst_avi_mux_stop_file() is called before this happens then auds.blockalign
	  is zero and gst_avi_mux_audsink_set_fields() cause a crash:
	  [...]
	  avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign;
	  [...]
	  https://bugzilla.gnome.org/show_bug.cgi?id=758912

2015-12-01 18:20:23 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: don't block when resurecting a buffer
	  When we are resurecting a buffer, don't block. instead let us copy a
	  buffer.

2015-12-01 00:30:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: remove extra variable to improve readability
	  Makes it easier to see that the event is being replaced/unrefed

2015-12-01 00:22:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: respect seqnum in seek events
	  Propagate the original seek seqnum to events originated from
	  seeking to make sure they have the same value

2015-12-01 00:03:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: flush upstream when seeking in pull mode
	  Makes sure upstream will unblock and return the thread so that
	  seeking can continue
	  https://bugzilla.gnome.org/show_bug.cgi?id=758861

2015-11-27 09:27:29 +0100  Anton Bondarenko <antonbo@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: add "send SPS/PPS with every key frame" mode
	  It's not enough to have timeout or event based SPS/PPS information sent
	  in RTP packets. There are some scenarios when key frames may appear
	  more frequently than once a second, in which case the minimum timeout
	  for "config-interval" of 1 second for sending SPS/PPS is not sufficient.
	  It might also be desirable in general to make sure the SPS/PPS is
	  available with every keyframe (packet loss aside), so receivers can
	  actually pick up decoding immediately from the first keyframe if
	  SPS/PPS is not signaled out of band.
	  This patch adds the possibility to send SPS/PPS with every key frame. This
	  mode can be enabled by setting "config-interval" property to -1. In this
	  case the payloader will add SPS and PPS before every key (IDR) frame.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757892

2015-11-27 09:03:51 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtph264pay: change config-interval property type from uint to int
	  This way we can use -1 as special value, which is nicer than MAXUINT.
	  This is backwards compatible even with the GValue API, as shown by
	  a unit test.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757892

2015-11-26 21:46:11 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for Opus
	  Add support for demuxing Opus encapsulated in MP4 files, based on the
	  following spec: https://www.opus-codec.org/docs/opus_in_isobmff.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=742643

2015-11-25 22:48:32 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use macro for codec_name
	  Use _codec() macro instead of duplicating code.

2015-03-25 16:32:55 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: videodec: choose format from caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=733827

2015-03-27 15:02:33 +0100  Philipp Zabel <p.zabel@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: add gst_v4l2_object_probe_caps
	  Add a variant of gst_v4l2_object_get_caps that bypasses the probed_caps cache.
	  https://bugzilla.gnome.org/show_bug.cgi?id=733827

2015-11-19 17:20:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	  v4l2-probe: Skip devices without supported formats

2015-11-13 12:35:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* configure.ac:
	* sys/v4l2/gstv4l2.c:
	  v4l2: Track /dev/video* to triggered required probe
	  If something in /dev/video* get added, removed or replaced, we need to
	  probe the devices again in order to ensure the dynamic devices are up to
	  date.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758085

2015-11-25 14:51:40 +1100  Alessandro Decina <alessandro.d@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: rtpsession: don't send empty RTCP packets
	  generate_rtcp can produce empty packets when reduced size RTCP is turned on.
	  Skip them since it doesn't make sense to push them and they cause errors with
	  elements that expect RTCP packets to contain data (like srtpenc).

2015-11-24 10:57:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: restore the segment on case of soft reset
	  When seeking back to restore the mdat position a flush is pushed
	  through and it resets downstream segment information. Make sure
	  that after the flush (that does a soft reset) a segment will
	  be pushed again
	  Fixes regressions spotted at
	  https://ci.gstreamer.net/job/GStreamer-master-validate/2100/

2015-11-20 12:44:22 +0000  Graham Leggett <minfrin@sharp.fm>

	* gst/multifile/gstmultifilesink.c:
	  multifilesink: fix spelling of variable
	  https://bugzilla.gnome.org/show_bug.cgi?id=758390

2015-11-20 11:05:51 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: unite duplicate FourCC
	  Unite in fourcc.h the FourCCs that are used twice or more in qtdemux

2015-11-19 15:33:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	  v4l2: Fix capture/output-io-mode properties
	  There was some miss-match in the implementation. This makes it
	  concistent, though functionally it worked, except the video decoder
	  output-io-mode getter.

2015-11-19 19:48:06 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  atoms: remove unused argument of build_mov_wave_extension()
	  AtomTrak * trak argument of build_move_wave_extension() isn't used.
	  Removing it.

2015-11-19 19:28:20 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	  qtdemux: remove duplicate FourCC
	  Use the available FourCCs in fourcc.h instead of duplicating them.

2015-11-19 18:36:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	  isomp4: centralize all FourCC
	  10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c
	  already exist in fourcc.h. Don't duplicate these and use them directly.
	  Plus moving 6 to fourcc.h, to centralize them all.

2015-11-19 17:32:12 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/matroska/webm-mux.c:
	  matroska/webmmux: fix outdated example launch lines
	  Update gst-launch-0.10 lines to gst-launch-1.0

2015-11-16 13:26:50 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  isomp4: add support for Opus in mp4mpux
	  Add support for muxing MP4 files containing Opus. Based on the spec
	  detailed here:
	  https://www.opus-codec.org/docs/opus_in_isobmff.html
	  https://bugzilla.gnome.org/show_bug.cgi?id=742643

2015-11-18 19:10:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Replace tabs with spaces

2015-11-18 19:07:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers
	  This fixes seeking if the first entries in the samples table are negative. The
	  binary search would always fail on this as the array would not be sorted if
	  interpreting the negative numbers as huge positive numbers. This caused us to
	  always output buffers from the beginning after a seek instead of close to the
	  seek position.
	  Also add a case to the comparison function for equality.

2015-11-18 16:01:48 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: remove duplicate check
	  We want 1 or 2 streamheaders, the check  if (bufarr->len != 1 &&
	  bufarr->len != 2) is enough. Not need to check if bufarr->len is <= 0 or
	  > 255.

2015-11-18 14:48:36 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Fix error leak and handle error
	  g_thread_try_new allows for possiblity of failures. In case it fails,
	  error is not handled and leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758260

2015-11-15 17:16:29 -0800  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: Properly handle backward compat for event deserialization
	  Actual code is checking for a NULL terminator and a ';' terminator,
	  for backward compat, in a chained way that cause all events being rejected.
	  The proper condition is to reject the events when terminator isn't
	  in ['\0', ';'] set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758151

2015-11-15 17:11:02 -0800  Josep Torra <n770galaxy@gmail.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: Test for handling of custom events in rtpgst
	  Add a simple test that checks proper serialization/deserialization
	  of custom events with rtpgstpay and rtpgstdepay.

2015-11-16 16:23:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  vpxdec: Use threads on multi-core systems
	  This adds an automatic mode to the threads property of vpxdec in order to
	  use as many threads as there is CPU on the platform. This brings back
	  GStreamer VPX decoding performance closer to what is achieved by other
	  players, including Chromium.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758195

2015-11-16 10:58:32 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only send initial gaps for non-fragmented streams
	  It would be unusual to have the header segment with an 'edts' atom
	  indicating gaps at the beginning when handling fragmented streams.
	  The header usually doesn't contain any timestamping information, this
	  should come from the playlist/manifest and the segments with media
	  in those scenarios.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758171

2015-11-17 09:41:34 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  Revert "Revert "qtdemux: respect qt segments in push-mode for empty starts""
	  This reverts commit d842ff288a9d01214a046becbfd9cbff3a4acea0.
	  This was reverted by accident

2015-11-17 12:39:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Add "loop" property for enabling/disabling multicast loopback
	  On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it
	  is a setting for the receiver socket. As such we will need it on udpsrc too to
	  allow filtering out our own multicast packets.

2015-11-16 13:52:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  Revert "qtdemux: respect qt segments in push-mode for empty starts"
	  This reverts commit 142d8e2d23e5602e7382977af1043d621625f8c8.

2015-11-16 16:56:04 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix string memory leak
	  The string got using g_strdup_printf will be allocated memory
	  and should be freed after use.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758161

2015-11-14 21:51:11 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2/object: remove unnecessary NULL check before g_free()

2015-11-14 21:45:29 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/oss/gstosssrc.c:
	  osssrc: remove unnecessary NULL check before g_free()

2015-11-14 21:43:24 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/sunaudio/gstsunaudiosrc.c:
	  sunaudiosrc: remove unnecessary NULL checks before g_free()

2015-11-14 21:36:30 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: remove unnecessary NULL checks before g_free()

2015-11-14 21:31:08 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: remove unnecessary NULL checks before g_free()

2015-11-14 21:26:21 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/matroska/matroska-read-common.c:
	  matroska/read-common: remove unnecessary NULL checks before g_free()

2015-11-14 20:43:10 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/isomp4/atoms.c:
	  isomp4/atoms: remove unnecessary NULL checks before g_free()

2015-11-14 20:35:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtptheorapay.c:
	  rtp/theorapay: remove unnecessary NULL checks before g_free()

2015-11-14 20:33:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtpvorbispay.c:
	  rtp/vorbispay: remove unnecessary NULL checks before g_free()

2015-11-14 20:31:34 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtpjpegpay.c:
	  rtp/jpegpay: remove unnecessary NULL checks before g_free()

2015-11-14 20:27:04 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: remove unnecessary NULL checks before g_free()

2015-11-14 20:22:09 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: remove unnecessary NULL checks before g_free()

2015-11-14 20:14:25 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/flx/gstflxdec.c:
	  flxdec: remove unnecessary NULL check before g_free()

2015-11-14 20:09:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstop.c:
	  effectv/optv: remove unnecessary NULL checks before g_free()

2015-11-14 20:05:03 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstshagadelic.c:
	  effectv/shagadelictv: remove unnecessary NULL checks before g_free()

2015-11-14 20:01:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstripple.c:
	  effectv/ripple: remove unnecessary NULL checks before g_free()

2015-11-14 19:56:57 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gstradioac.c:
	  effectv/radioac: remove unnecessary NULL checks before g_free()

2015-11-14 19:52:12 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/effectv/gststreak.c:
	  effectv/streak: remove unnecessary NULL check before g_free()

2015-11-14 17:04:55 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/shout2/gstshout2.c:
	  shout2: remove unnecessary NULL checks before g_free()

2015-11-14 16:57:13 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/vpx/gstvp9enc.c:
	  vp9enc: remove unnecessary NULL check before g_free()

2015-11-14 16:54:42 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/vpx/gstvp8enc.c:
	  vp8enc: remove unnecessary NULL check before g_free()

2015-11-14 16:20:33 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: remove unnecessary NULL checks before g_free()

2015-11-13 13:34:02 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: add support of NV16, NV61 and NV24 formats
	  Mapped respectively to V4L2_PIX_FMT_NV16/V4L2_PIX_FMT_NV16M,
	  V4L2_PIX_FMT_NV61,V4L2_PIX_FMT_NV61M and V4L2_PIX_FMT_NV24 v4l2 formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758058

2015-11-11 14:10:53 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxpartreader: Fix GCond leak
	  inactive_cond is not being cleared resulting in memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757924

2015-08-06 12:44:20 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: fix output state memory leak
	  When jpeg_finish_decompress is called, output state reference is being created.
	  But if there is any failures in finishing decompress, it jumps to setjmp,
	  and at that point state was not referenced. Resulting in leak of output state.
	  Hence adding another setjmp after output state is referenced.
	  Similarly adding another setjmp to unmap the frame in case error happens before
	  finish_decompress
	  https://bugzilla.gnome.org/show_bug.cgi?id=753087

2015-08-10 11:23:45 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: respect qt segments in push-mode for empty starts
	  In push-mode it is hard to support qt segments overall but it is
	  possible to support when the file isn't heavily edited but just contain
	  a segment to indicate a gap at the beginning. This also allows properly
	  timestamping data that has negative DTS in push-mode.
	  It is relevant to support those for 2 scenarios:
	  1) fragmented streaming
	  2) HTTP playback of 'regular' mp4
	  https://bugzilla.gnome.org/show_bug.cgi?id=753484

2015-11-05 18:39:33 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse: Don't leak caps and structures in the device provider

2015-11-04 19:01:20 +0530  Arun Raghavan <arun@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: Document properties that are expressed in bits per second
	  This changed in 928cd110bcea5d143cab3ea747991851d52ecbad and
	  73c0c2920f9aca96982a4de0c20b3417aa148b81 but was not documented.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747863

2015-11-04 18:51:32 +0530  Arun Raghavan <arun@centricular.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: Trivial gst-indent fixes

2015-08-12 13:35:40 +0200  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: support for cenc auxiliary info parsing outside of moof box
	  When the cenc aux info index is out of moof boundaries, keep track of
	  it and parse the beginning of the mdat box, before the first sample.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755614

2015-11-03 20:33:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Use codecutils helpers for creating Opus caps
	  Also fix up codec data with values from the container.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 14:51:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: There is no multistream field for Opus anymore
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 12:42:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	  matroska/webmmux: Support Opus in webmmux and VP9 in matroskamux
	  https://bugzilla.gnome.org/show_bug.cgi?id=729950

2015-11-03 12:40:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Parse and handle CodecDelay, SeekPreroll and DiscardPadding
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2015-11-03 12:18:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: Write CodecDelay, DiscardPadding and SeekPreroll for Opus
	  And also adjust timestamps and durations according to the codec delay, both
	  should include it for whatever reason.
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2015-11-03 11:49:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Opus headers are not in-band
	  https://bugzilla.gnome.org/show_bug.cgi?id=727305

2015-11-03 22:01:07 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/v4l2/gstv4l2.c:
	  v4l2: Set O_CLOEXEC on the device fd
	  This is needed to make sure that child processes don't inherit the video
	  device fd which can cause problems with some drivers.

2015-11-03 14:46:30 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpmanager: switch G_GINT64_FORMAT for GST_STIME_ARGS
	  No need to use G_GINT64_FORMAT for potentially negative values of
	  GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
	  Plus it creates more readable values in the logs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-03 14:26:29 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to manually handle negative values of diff, GST_STIME_ARGS does
	  exactly this.

2015-11-02 16:53:15 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to manually handle negative values of diff, GST_STIME_ARGS does
	  exactly this.

2015-11-02 16:43:46 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: use GST_STIME_ARGS for GstClockTimeDiff
	  No need to manually handle negative values of diff, GST_STIME_ARGS is
	  available for this.

2015-10-30 10:05:37 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audiochebband.c:
	  audiochebband: Fix typo in example pipeline
	  Fix typo in example pipeline.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757340

2015-10-28 23:47:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2: fix double-unref in the v4l2 device provider

2015-10-27 10:48:00 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-ids.c:
	  matroskamux: don't drop JPEG frames that only have PTS but no DTS set
	  For the MS/VfW codec ids, we want to write DTS timestamps instead
	  of PTS because that's what everyone else seems to do (and it's also
	  how it is in AVI). So for those input formats we use the buffer DTS
	  instead of the PTS. However, if there's no DTS set but only the PTS
	  then just take the PTS instead of dropping the input buffer. This
	  is useful especially for I-frame only codecs like JPEG and huffyuv,
	  but should also be fine as fallback in general.
	  Fixes regression with input JPEG frames that only have PTS set on them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756967

2015-10-24 23:57:38 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/splitmux.c:
	  tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
	  https://bugzilla.gnome.org/show_bug.cgi?id=753622

2015-10-24 23:57:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: do not destroy the multiqueue & muxer when going to NULL
	  Instead, delay it until all request pads have been released. This is
	  because the release_pad() vfunc requires the multiqueue and muxer to
	  be there in order to release their request pads as well. If those
	  elements are destroyed earlier, release_pad() does not work, no
	  pads are released and some resources are leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753622

2015-10-20 15:28:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Read buffer timestamp *after* actually setting it
	  https://bugzilla.gnome.org/show_bug.cgi?id=756809

2015-10-24 17:14:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: Fix handling of rate < 0
	  We have to reverse all samples in a buffer before processing them to properly
	  have continuous data from one buffer to another. As a result we will have a
	  negative applied rate and a rate of 1.0.
	  Also make sure that input buffers are correctly clipped to the segment,
	  otherwise our calculations are going to go wrong.
	  Also copy over the segment event's sequence number to the output segment while
	  we're at it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757033

2015-10-19 18:04:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: break as soon as non-interlaced if found
	  It looks for a non-interlaced entry on the filter caps, break
	  as soon as one is found to avoid wasting cpu

2015-10-19 17:50:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: implement accept-caps
	  Implement accept-caps handler to avoid doing a full caps query
	  downstream to handle it.
	  This commit implements accept-caps as a simplification of the _getcaps
	  function, so it exposes the same limitations that getcaps would.
	  For example, not accepting renegotiation to caps with capsfeatures when
	  it was last configured to a caps that it has to deinterlace.

2015-10-19 17:06:28 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/deinterlace.c:
	  tests: deinterlace: fix small typo in comment

2015-10-26 00:41:28 +1100  Jan Schmidt <jan@centricular.com>

	* tests/files/Makefile.am:
	  check: Dist splitvideo0[012].ogg test files.

2015-10-23 20:16:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: Add support for F64

2015-10-22 17:40:38 -0700  Mischa Spiegelmock <mspiegelmock@gmail.com>

	* docs/plugins/inspect/plugin-rtp.xml:
	* gst/multipart/multipartdemux.c:
	* gst/rtp/README:
	* gst/rtp/gstrtpvp8pay.c:
	* gst/rtpmanager/gstrtprtxreceive.c:
	* gst/udp/gstudpsrc.c:
	  docs: Minor fixes in various places
	  https://bugzilla.gnome.org/show_bug.cgi?id=756996

2015-10-21 17:43:31 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/goom/plugin_info.c:
	  goom: remove compiler trick
	  After commit 2cb6cfed22166b262ae50cb58f3ff11dd8ba91f9 there is no need to
	  trick the compiler anymore about the usage of variable cpuFlavour.

2015-10-21 14:35:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* common:
	  Automatic update of common submodule
	  From b99800a to b319909

2015-10-21 17:41:38 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audiofxbaseiirfilter.h:
	  audiofx: remove unused variable
	  Remove unsued variable have_coeffs in audiofxbaseiirfilter
	  https://bugzilla.gnome.org/show_bug.cgi?id=756905

2015-10-20 17:29:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Use new GST_ENABLE_EXTRA_CHECKS #define
	  https://bugzilla.gnome.org/show_bug.cgi?id=756870

2015-10-21 14:25:55 +0300  Sebastian Dröge <sebastian@centricular.com>

	* README:
	* common:
	  Automatic update of common submodule
	  From 9aed1d7 to b99800a

2015-10-21 11:53:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: relax creation time parsing
	  Parse wrong timestamps like we used to write as well,
	  e.g. 10:9:42, and the hour might be without a leading
	  zero in any case.

2015-10-21 11:45:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix indentation

2015-10-21 11:44:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: extract both creation date and time
	  Before we only extracted the date part.

2015-10-21 11:16:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: fix writing of creation time
	  Don't write time as e.g. 11:9:42

2015-10-13 12:42:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: update fragment offset
	  It was always being set to 0, making the resulting stream broken
	  for the receiver
	  https://bugzilla.gnome.org/show_bug.cgi?id=756422

2015-10-19 15:36:37 +0300  Ryan Hendrickson <ryan.hendrickson@alum.mit.edu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't unconditionally use strnlen()
	  It's not available on older OSX and we can as well use memchr() here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756154

2015-10-19 17:38:32 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/auparse/gstauparse.c:
	  auparse: Fix event memory leak
	  Free the event after being handled to prevent memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756799

2015-10-19 09:14:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: unify raw audio caps into a single caps structure

2015-10-14 15:42:50 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for FFV1 coded streams in mov
	  https://bugzilla.gnome.org/show_bug.cgi?id=752495

2015-10-14 15:53:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: EOS immediately if we have an empty seek segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=748316

2015-10-14 10:43:19 +0300  Stavros Vagionitis <stavrosv@digisoft.tv>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: Make non-inclusive segment boundaries inclusive
	  The problem is that the filesrc and souphttpsrc are behaving
	  differently regarding the calculation of the segment boundaries. The
	  filesrc is using a non-inclusive boundaries, while the souphttpsrc
	  uses inclusive. Currently the hlsdemux calculates the boundaries as
	  inclusive, so for this reason there is no problem with the souphttpsrc,
	  but there is an issue in the filesrc.
	  The GstSegment is non-inclusive, so the proposed solution is to use
	  non-inclusive boundaries in the hlsdemux in order to be consistent.
	  Make the change in the hlsdemux, will break the souphttpsrc, which
	  will expect inclusive boundaries, but the hlsdemux will offer
	  non-inclusive. This change makes sure that the non-inclusive
	  boundaries are converted to inclusive.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748316

2015-10-11 22:07:54 +0000  Graham Leggett <minfrin@sharp.fm>

	* ext/soup/gstsouphttpclientsink.c:
	* ext/soup/gstsouphttpclientsink.h:
	  souphttpclientsink: Add the retry and retry-delay properties
	  These allow a failed request to be retried after the given number of seconds
	  instead of failing the pipeline. Take account of the Retry-After header if
	  present. Add retries parameter that controls the number of times an HTTP
	  request will be retried before failing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756318

2015-10-14 12:03:15 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix caps leak
	  If the QtDemuxStream are re-used they may already have caps which used
	  to be leaked.
	  Reproduced using the
	  validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
	  scenario.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756561

2015-10-14 09:29:50 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix taglist memory leak
	  Free the stream and its sub items instead of just the stream
	  https://bugzilla.gnome.org/show_bug.cgi?id=756544

2015-10-11 12:06:26 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Allow negotiating to S8 as a raw format but stop making it best choice
	  Negotiation to audio/x-raw,format=S8 was not possible because S8 does
	  not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
	  https://bugzilla.gnome.org/show_bug.cgi?id=756387

2015-10-11 09:18:40 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Add prores support
	  https://bugzilla.gnome.org/show_bug.cgi?id=756388

2015-10-12 18:56:32 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: add GST_PLUGINS_BASE_LIBS for flvdemux check
	  So it pulls in the right libgsttag-1.0.

2015-10-11 22:27:47 +0100  Julien Isorce <j.isorce@samsung.com>

	* gst/goom/Makefile.am:
	* gst/goom/gstaudiovisualizer.c:
	* gst/goom/gstaudiovisualizer.h:
	* gst/goom/gstgoom.h:
	* gst/goom2k1/Makefile.am:
	* gst/goom2k1/gstaudiovisualizer.c:
	* gst/goom2k1/gstaudiovisualizer.h:
	* gst/goom2k1/gstgoom.h:
	  goom/goom2k1: remove obsolete left over files
	  They now use the new GstAudioVisualizer base class
	  from gst-plugins-base/gst-libs/gst/pbutils
	  Also fixed undefined reference to gst_audio_visualizer_get_type
	  Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-10-12 10:48:23 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Fix buffer memory leak during failures
	  mapped buffer is not being unmapped during failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=756231

2015-10-12 11:18:51 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Check if soup message is created
	  If soup message is not created then the same should not be passed
	  on, which is resulting in segfault. Hence throwing a warning message
	  and returning
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-12 11:15:15 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Check if location being set is valid
	  Adding a check in set_property to find if the location uri is valid
	  and printing warning if not valid.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-12 11:09:30 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Fix memory leaks during failures
	  freeing streamheader_buffers and sent_buffers during failure cases.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-12 11:03:17 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/soup/gstsouphttpclientsink.c:
	  souphttpclientsink: Replace redundant free_buffer_list function
	  Removing free_buffer_list and replacing it with already available function
	  g_list_free_full
	  https://bugzilla.gnome.org/show_bug.cgi?id=755326

2015-10-11 16:40:01 +0200  Edward Hervey <bilboed@bilboed.com>

	* tests/check/Makefile.am:
	  check: Don't forget base CFLAGS for flvdemux check
	  elements/flvdemux.c:25:25: fatal error: gst/tag/tag.h: No such file or directory

2015-10-11 11:37:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/ebml-write.c:
	* gst/matroska/ebml-write.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/matroska-mux.h:
	  matroskamux: Create a TIME segment when creating streamable output
	  Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which
	  does the same for flvmux.

2015-09-23 13:50:52 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/Makefile.am:
	* gst/flv/gstflvdemux.c:
	* tests/check/Makefile.am:
	* tests/check/elements/flvdemux.c:
	  flvdemux: output speex vorbiscomment as a GstTagList
	  This is what speexdec expects.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755478

2015-09-22 22:59:16 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
	  Or else flvdemux don't understand it
	  https://bugzilla.gnome.org/show_bug.cgi?id=754435

2015-09-02 10:44:59 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: use time segment and copy timestamps when streamable
	  Add a basic test using speex data to verify timestamping.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754435

2015-09-23 13:14:03 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: speex is also always 16KHz
	  This is just a cosmetic change for the logs, since the right caps
	  for Speex is being set elsewhere.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755479

2015-07-14 15:19:44 +0200  Stian Selnes <stian@pexip.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  rtpmanager: Add 'source-stats' to stats and notify
	  Add statitics from each rtp source to the rtp session property.
	  'source-stats' is a GValueArray where each element is a GstStructure of
	  stats for one rtp source.
	  The availability of new stats is signaled via g_object_notify.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752669

2015-06-05 17:20:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Implement sending of reduced size RTCP packets
	  https://bugzilla.gnome.org/show_bug.cgi?id=750456

2015-10-08 15:01:13 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/audiofx/audiodynamic.h:
	  audiofx: Remove unused variable
	  Remove unused variable 'degree' in audiodynamic
	  https://bugzilla.gnome.org/show_bug.cgi?id=756234

2015-10-08 14:44:07 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix memory leak for corrupted file
	  Free brands before overriding them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756226

2015-10-08 11:44:04 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbufdec: Fix pixbuf_loader leak during failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=756219

2015-10-07 23:23:45 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	  rtpbin: Add missing break

2015-10-07 13:03:02 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2015-10-07 13:02:12 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpmanager: add "max-dropout-time" and "max-misorder-time" props
	  https://bugzilla.gnome.org/show_bug.cgi?id=751311

2015-10-07 17:14:57 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix date memory leak
	  When getting date from taglist, the memory should be freed after
	  using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756171

2015-10-05 11:03:38 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix sample memory leak
	  When getting sample from taglist, the memory should be freed after
	  using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756068

2015-10-05 13:10:56 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/cutter/gstcutter.c:
	  cutter: Fix buffer leak
	  Buffer is added to the internal cache, and pushed only when accumulated
	  buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
	  is not freed. Freeing the cache when the state changes from PAUSED to READY.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754212

2015-08-31 21:10:16 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Use default upstream event handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=752694

2015-08-31 21:05:03 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	  rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set
	  https://bugzilla.gnome.org/show_bug.cgi?id=752694

2015-07-22 09:47:22 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/rtpmanager/gstrtpmux.c:
	* gst/rtpmanager/gstrtpmux.h:
	* tests/check/elements/rtpmux.c:
	  gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
	  By not doing this, the muxer is not effectively a rtpmuxer, rather a
	  funnel, since it should be a single stream that exists the muxer.
	  If not specified, take the first ssrc seen on a sinkpad, allowing upstream
	  to decide ssrc in "passthrough" with only one sinkpad.
	  Also, let downstream ssrc overrule internal configured one
	  We hence has the following order for determining the ssrc used by
	  rtpmux:
	  0. Suggestion from GstRTPCollision event
	  1. Downstream caps
	  2. ssrc-Property
	  3. (First) upstream caps containing ssrc
	  4. Randomly generated
	  https://bugzilla.gnome.org/show_bug.cgi?id=752694

2015-10-02 22:42:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fixup last commit

2015-10-02 22:21:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	* gst/udp/gstudpsrc.c:
	  Update GLib dependency to 2.40.0

2015-06-30 16:56:19 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpstats.c:
	* gst/rtpmanager/rtpstats.h:
	  rtpstats: add utility for calculating RTP packet rate

2015-08-10 18:14:39 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handle empty segments in seeking adjust
	  If seeking targets an empty segment skip it as there is no media
	  offset to get from it. Instead look for the next one.
	  This doesn't make seeking in push-mode work if you seek to an
	  empty segment but at least won't get you to wrong offsets.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753484

2015-04-17 14:25:43 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: post messages when fragments are being opened and closed
	  This can be useful for applications that need to track the created fragments
	  (to log them in a recording database, for example)
	  https://bugzilla.gnome.org/show_bug.cgi?id=750108

2015-04-29 18:23:28 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: allow non-video streams to serve as reference
	  In the absence of a video stream, the first stream will be used as
	  reference.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753617

2015-07-22 17:45:12 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: initialize mux_start_time properly
	  mux_start_time refers to the running_time of the buffer
	  that goes first in the output file. Normally this time is
	  0, so this variable is initialized to 0 during the state
	  change to PAUSED.
	  However, when dealing with dynamic pipelines and starting
	  a recording while the pipeline has already run for a while,
	  the running_time of the first buffer is > 0 and this causes
	  a problem with detecting the end of the first file(s) when
	  splitting by duration, because the code will later compare
	  the threshold_time with (last buffer running_time - mux_start_time)
	  and will get it wrong until mux_start_time advances enough
	  to make this difference < threshold_time, creating empty files
	  in the meantime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753624

2015-09-16 16:03:02 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Reverse playback does not consider segment.start
	  During reverse playback, the media should stop playing at segment.start
	  This does not happen, and avidemux continues to process data even when
	  current timestamp is less that segment.start.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755094

2015-09-23 12:39:35 +0900  Manasa Athreya <manasa.athreya@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream
	  If stream has more than one trex box which is not matched to actual
	  track id, it makes qtdemux crashed.
	  Author : Manasa Athreya (manasa.athreya@lge.com)
	  https://bugzilla.gnome.org/show_bug.cgi?id=754864

2015-09-04 14:24:45 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: get size, stride info using VideoInfo
	  Use VideoInfo data to get size stride and
	  offset, instead of hard coded macros.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754558

2015-09-04 14:18:50 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst/smpte/gstsmpte.c:
	  smpte: free mask
	  Free the memory allocated to 'mask' to avoid
	  memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754555

2015-08-20 11:02:58 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/examples/equalizer/demo.c:
	* tests/icles/equalizer-test.c:
	* tests/icles/gdkpixbufoverlay-test.c:
	* tests/icles/gdkpixbufsink-test.c:
	* tests/icles/test-oss4.c:
	* tests/icles/videocrop-test.c:
	  gstreamer: good: tests: Fix memory leaks when context parse fails.
	  When g_option_context_parse fails, context and error variables are not getting free'd
	  which results in memory leaks. Free'ing the same.
	  And replacing g_error_free with g_clear_error, which checks if the error being passed
	  https://bugzilla.gnome.org/show_bug.cgi?id=753853

2015-10-02 16:18:15 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: doesn't handle probation and rtp gap in case of sender
	  https://bugzilla.gnome.org/show_bug.cgi?id=754548

2015-10-02 16:16:32 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* docs/plugins/gst-plugins-good-plugins.signals:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpsession.h:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals
	  Allows for applications to get internal source's RTP statistics.
	  (eg. sender sources for a server/client)
	  https://bugzilla.gnome.org/show_bug.cgi?id=746747

2015-10-02 14:17:48 +1000  Jan Schmidt <jan@centricular.com>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: Gather and coalesce all damaged areas before retrieving.
	  These days the xserver seems to give us the same damage regions
	  over and over for entire windows, and we retrieve them multiple
	  times, which gives time for more damage to appear. Instead, just
	  quickly gather all damaged areas into a region list and copy
	  out once.

2015-10-01 16:24:32 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/goom2k1/Makefile.am:
	* gst/goom2k1/gstgoom.h:
	  goom2k1: use the new audiovisualizer base class
	  Rebase to have goom using the GstAudioVisualizer base class in
	  gst-plugins-base/gst-libs/gst/pbutils
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-10-01 16:16:08 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/goom/Makefile.am:
	* gst/goom/gstgoom.h:
	  goom: use the new audiovisualizer base class
	  Rebase to have goom using the GstAudioVisualizer base class in
	  gst-plugins-base/gst-libs/gst/pbutils
	  https://bugzilla.gnome.org/show_bug.cgi?id=742875

2015-09-30 17:35:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	* tests/check/elements/deinterleave.c:
	  deinterleave: implement accept-caps
	  Avoid using default accept-caps handler that will query downstream
	  and is more expensive. Just check if the caps is compatible with
	  the template and check if the channels are the same.

2015-09-30 09:35:39 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/elements/deinterleave.c:
	  tests: deinterleave: also check for caps query results

2015-09-30 12:30:59 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: use the caps query filter
	  It was being ignored and would lead to wrong results if the
	  element doing the query would rely on the intersection being made.

2015-09-30 10:00:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: implement a caps query handler for the sinkpad
	  It was missing and apparently code relied on having it there
	  for not allowing a change in the number of channels

2015-09-30 09:05:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/interleave/deinterleave.c:
	  deinterleave: fix caps leak
	  Caps from the pad template are being leaked. In any case it is
	  from a static pad template and will 'leak' in the end, just doing
	  the cleanup for the good practice.

2015-09-29 11:15:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	* tests/check/elements/.gitignore:
	* tests/check/elements/gdkpixbufoverlay.c:
	  tests: gdkpixbufoverlay: add minimal unit test
	  https://bugzilla.gnome.org/show_bug.cgi?id=755773

2015-09-29 11:12:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufsink: don't leak old pixel buffer when setting a new overlay
	  https://bugzilla.gnome.org/show_bug.cgi?id=755773

2015-09-28 20:25:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: avoid potential string overflow
	  We don't necessarily have full control over the input tags, so
	  it's possible that the ISRC tag contains a longer string than
	  expected, in which case we'd write over the end of the static-size
	  13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
	  Make sure to only copy the ISRC if it's not too long, and make
	  sure the buffer we write to is always NUL-terminated by using
	  g_strlcpy().
	  CID 1324931.

2015-09-28 18:03:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove leftover assertion from 0.10
	  We now allocate memory via GstAllocator and as such can handle arbitrary
	  alignments, not only <= G_MEM_ALIGN.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755708

2015-09-25 10:01:37 +0200  Guillaume Marquebielle <guillaume.marquebielle@parrot.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix uninitialized variables in LOAS config reading
	  On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
	  "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
	  'channels' are pointing to uninitialized values can end on setting random values as
	  rate and channels on src caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755611

2015-09-18 00:58:23 +1000  Jan Schmidt <thaytan@noraisin.net>

	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpsession.c:
	  Fix some compiler warnings when building with G_DISABLE_ASSERT
	  Touches rtpmanager and gdkpixbufsink

2015-08-18 14:30:57 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: support timed-text subtitle tracks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752818

2015-09-26 00:12:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-parse.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  gst: Don't use deprecated gst_segment_to_position()

2015-09-21 13:47:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
	  https://bugzilla.gnome.org/show_bug.cgi?id=755125

2015-09-16 19:28:11 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	  rtpbin/session: Allow RTCP sync to happen based on capture time or send time
	  Send time is the previous behaviour and the default, but there are use cases
	  where you want to synchronize based on the capture time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755125

2015-09-25 23:51:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

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=== release 1.6.0 ===

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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.6.0

2015-09-25 22:57:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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2015-09-25 14:08:09 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Do not set width/height before comparing with old values
	  Otherwise we end up considering the values did not change and we wrongly
	  work with the old video format (which will lead to wrong
	  behaviour/segfaults).
	  https://bugzilla.gnome.org/show_bug.cgi?id=755621

2015-09-23 20:59:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Accumulate segments for edit lists before activating the next segment
	  eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
	  accumulating segments manually when activating a segment. This is only
	  needed when handling edit lists, not when activating a segment because of a
	  seek. Do the accumulation when switching edit list segments instead.
	  This fixes segment seeks again, while keeping edit lists playback working.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755471

2015-09-23 17:43:51 +0530  Vikram Fugro <vikram.fugro@gmail.com>

	* gst/spectrum/gstspectrum.c:
	  spectrum: send phase values in the GstMessage for Phase info
	  https://bugzilla.gnome.org/show_bug.cgi?id=755463

2015-09-22 00:46:01 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Don't output a warning on MONO multiview mode.

2015-09-19 17:02:18 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: Fix memory leaks
	  The same memory leaks were fixed in identical fashion for
	  vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277

2015-09-19 17:04:07 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
	  * use g_list_free_full(), don't iterate elements maually when freeing
	  * call gst_rtp_*_pay_clear_packet(), don't duplicate its code
	  * use gst_buffer_unref() to clarify that it is buffers being released,
	  instead of refering directly to gst_mini_object_unref()
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277

2015-09-19 18:44:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=755265

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=== release 1.5.91 ===

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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.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.5.91

2015-09-18 19:23:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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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
Sebastian Dröge's avatar
Sebastian Dröge committed
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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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2015-06-24 23:30:41 +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: