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=== release 1.5.1 ===

2015-06-07  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.5.1

2015-06-07 09:35:03 +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/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: Update translations

2015-06-05 16:44:08 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasepayload.c:
	  rtpbasepayload: Always prefer downstream's ssrc suggestion if any
	  Otherwise ssrc changes via rtpsession's (deprecated!) internal-ssrc property
	  are not possible anymore. rtpsession was now patched to only suggest an ssrc
	  if it makes sense to do so.
	  In 2.0 we should get rid of all the properties that are also negotiated via
	  caps, the code and behaviour is too confusing otherwise.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749581

2015-06-05 10:16:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	* win32/common/libgstrtp.def:
	  rtcpbuffer: Improve documentation of new functions a bit
	  Also actually add them to the documentation.

2015-06-03 11:20:35 +0200  Jose Antonio Santos Cadenas <santoscadenas@gmail.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	* tests/check/libs/rtp.c:
	  rtcpbuffer: Update package validation to support reduced size rtcp packets
	  According to this section of the rfc.
	  https://tools.ietf.org/html/rfc5506#section-3.4.2
	  The validation should be updated to accept more types of RTCP
	  packages, with this mask change feedback packages will be also
	  accepted.
	  Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868

2015-06-04 19:03:51 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/audioresample/gstaudioresample.c:
	  audioresample: copy metadata that only has the "audio" tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750406

2015-06-04 19:00:45 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst-libs/gst/audio/gstaudiofilter.c:
	  audiofilter: copy metadata that only has the "audio" tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750406

2015-06-04 17:59:17 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: copy metadata that only has the "audio" tag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750406

2015-05-20 18:16:07 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Serialize the top level DiscovererInfo
	  Which contains fields such as duration, uri and tags.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749673

2015-06-04 16:31:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: Add AAC channel configurations 11, 12 and 14 and levels 6 and 7

2015-06-02 16:14:39 +0200  Edward Hervey <edward@centricular.com>

	* tests/check/generic/clock-selection.c:
	* tests/check/libs/allocators.c:
	* tests/check/libs/audio.c:
	* tests/check/libs/fft.c:
	* tests/check/libs/navigation.c:
	* tests/check/libs/rtp.c:
	* tests/check/libs/rtsp.c:
	* tests/check/libs/rtspconnection.c:
	* tests/check/libs/tag.c:
	* tests/check/libs/xmpwriter.c:
	* tests/check/pipelines/basetime.c:
	* tests/check/pipelines/capsfilter-renegotiation.c:
	* tests/check/pipelines/gio.c:
	* tests/check/pipelines/simple-launch-lines.c:
	* tests/check/pipelines/theoraenc.c:
	* tests/check/pipelines/vorbisdec.c:
	* tests/check/pipelines/vorbisenc.c:
	  check: Use GST_CHECK_MAIN () macro everywhere
	  Makes source code smaller, and ensures we go through common initialization
	  path (like the one that sets up XML unit test output ...)

2015-06-02 12:47:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: add description for video/x-cavs caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=727731

2015-06-02 12:28:19 +0200  Edward Hervey <bilboed@bilboed.com>

	* win32/common/libgstpbutils.def:
	  win32: Update def file for new encoding API

2015-05-29 14:15:31 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	  rtpbuffer: optimise payload mapping for buffers with one memory
	  Micro-optimisation: if the buffer consist of just one memory, we
	  know we have already mapped that memory to read the headers, so
	  no need to map it another time to get to the payload data, we
	  can just set up the payload data details right there and then
	  and avoid another map call in gst_rtp_buffer_get_payload().
	  Adds up when receiving RTP-payloaded raw video which can easily
	  be thousands of packets per frame.

2015-05-21 13:59:55 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	* gst-libs/gst/rtp/gstrtpbasedepayload.h:
	  rtpbasedepayload: provide chain_list function on sink pad
	  Implement a chain_list function, which avoids lots of locking
	  compared to the default fallback implementation in GstPad.
	  We may also want to do some more sophisticated timestamp
	  tracking here at some point, but for now leave it up to the
	  jitterbuffer and/or subclasses (in case buffers in the
	  buffer list have no timestamp set on them, there may only
	  be a timestamp for the whole list on the first buffer).
	  This provides the exact same behaviour as the default
	  fallback implementation.

2015-05-07 10:26:47 +0200  Thibault Saunier <tsaunier@gnome.org>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/encoding-profile.c:
	* gst-libs/gst/pbutils/encoding-profile.h:
	* gst/encoding/gstencodebin.c:
	  encodebin: Add a way to enable/disabled a GstEncodingProfile
	  Summary:
	  So that the user can easily use the same encoding profile to render
	  with/without audio/video stream.
	  API:
	  gst_encoding_profile_is_disabled
	  gst_encoding_pofile_set_enabled
	  https://bugzilla.gnome.org/show_bug.cgi?id=749056

2015-05-30 15:34:51 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* tools/gst-play.c:
	  tools: gst-play: remove unnecessary variable
	  The second assignment of sret is never used. We can remove the first assignment
	  and use the value directly instead.

2015-05-30 08:12:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/tag/id3v2frames.c:
	  id3v2frames: Fix compiler warnings
	  id3v2frames.c:951:20: error: unused variable 'utf16enc' [-Werror,-Wunused-const-variable]
	  static const gchar utf16enc[] = "UTF-16";
	  ^
	  id3v2frames.c:952:20: error: unused variable 'utf16leenc' [-Werror,-Wunused-const-variable]
	  static const gchar utf16leenc[] = "UTF-16LE";
	  ^
	  id3v2frames.c:953:20: error: unused variable 'utf16beenc' [-Werror,-Wunused-const-variable]
	  static const gchar utf16beenc[] = "UTF-16BE";
	  ^

2015-05-30 01:03:46 +1000  Jan Schmidt <jan@centricular.com>

	* docs/design/part-stereo-multiview-video.markdown:
	  part-stereo-multiview-video: Add a section of open design questions

2015-05-30 00:58:38 +1000  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/video/video-format.h:
	  video-format: Fix minor docs typo

2015-03-16 19:37:26 +1100  Jan Schmidt <jan@centricular.com>

	* gst/videotestsrc/gstvideotestsrc.h:
	  videotestsrc: Document the solid-color pattern

2015-03-16 19:28:35 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstplay-enum.h:
	  playback: Document GST_PLAY_FLAG_SOFT_COLORBALANCE

2014-10-09 01:13:29 +1100  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/video/gstvideometa.c:
	* gst-libs/gst/video/gstvideometa.h:
	* win32/common/libgstvideo.def:
	  video: Make gst_buffer_get_video_meta() a real function, Return lowest id
	  Instead of returning the first video meta found on a buffer, return the
	  one with the lowest id (which is usually the same thing, except on
	  multi-view buffers)

2015-05-29 15:30:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: don't crash on unknown info types when deserializing
	  Handle unknown info types when deserializing instead of
	  dereferencing NULL pointers.
	  Coverity CID 1302394

2015-05-29 13:15:59 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst-libs/gst/sdp/gstsdpmessage.c:
	  sdp: prevent the sdp message parser from reading past the end of the buffer
	  Otherwise, a malformed SDP message could crash the application,
	  or even maliciously gather data from the memory located after
	  this buffer...
	  https://bugzilla.gnome.org/show_bug.cgi?id=750096

2015-05-28 19:49:31 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/videorate.c:
	  tests: add test for videorate caps renegotiation after a framerate has been calculated and added to caps
	  The original 0/1 framerate must still be allowed to be configured
	  on the upstream side of videorate, otherwise future caps renegotiation
	  is going to fail.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750032

2015-05-28 12:51:35 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/videorate/gstvideorate.c:
	  videorate: update the caps framerate only in the GST_PAD_SINK transform_caps direction
	  When a stream has a variable framerate, videorate calculates it and
	  forces it on the output caps. However, the code in _transform_caps()
	  currently also does that if the transform is going in the opposite
	  direction (GST_PAD_SRC), so during a renegotiation it tries to force
	  upstream to use the calculated framerate and it fails.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750032

2015-05-26 08:06:50 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstplaysink.c:
	  playsink: use queue to avoid lock in audiotee audio branches
	  This part of pipeline is:
	  tee name=t ! visualizationbin ! streamsynchronizer name=s
	  t. ! s.
	  streamsynchronizer might block and it could starve the visualization
	  branch of the pipeline when it is enabled.
	  The visualization bin has queues internally but the other branch
	  that links the audiotee directly to the synchronizer is vulnerable
	  to block. Adding a queue between "t. ! s." fixes deadlocks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749676

2015-05-26 13:11:00 +0300  Claudiu Florin Lazar <lazar.claudiu.florin@gmail.com>

	* ext/pango/gstbasetextoverlay.c:
	  basetextoverlay: make deltax and deltay properties controllable
	  This will be more useful once we have absolute direct
	  control bindings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749824

2015-05-05 18:01:46 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: fix chain leak
	  Don't leak the building_chain when destroying.
	  Fix leaks with the validate.http.playback.reverse_playback.vorbis_theora_1_ogg
	  scenario.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748964

2015-05-25 22:37:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/tag/id3v2frames.c:
	  tag: id3v2: fix parsing of UTF-16 text on systems with crippled iconv
	  Use g_utf16_to_utf8() instead of the more generic g_convert(), so
	  that we can extract text in UTF-16 format even on embedded systems
	  with crippled iconv support.
	  This code path is exercised by the id3demux test_unsync_v23
	  check in gst-plugins-good.
	  https://bugzilla.gnome.org/show_bug.cgi?id=741144

2015-05-25 22:37:06 +0100  Tim-Philipp Müller <tim@centricular.com>

	* .gitignore:
	  Add new generated rtp enum files to .gitignore

2015-05-24 18:58:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play.c:
	  tools: gst-play: keep configured playback rate and trick mode when seeking
	  Instead of resetting rate to 1.0

2015-05-24 18:47:25 +0100  Tim-Philipp Müller <tim@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/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: update for new translatable strings

2015-05-24 18:46:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play.c:
	  tools: gst-play: mark more strings for translation

2015-05-23 01:50:11 +0900  danny song <danny.song.ga@gmail.com>

	* tools/gst-play.c:
	  tools: gst-play: add keyboard shortcut help
	  https://bugzilla.gnome.org/show_bug.cgi?id=749740

2015-05-23 12:02:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/Makefile.am:
	  tests: add back videoscale unit test
	  Has been removed in 835422b2 as part of porting
	  things over to the new videoscale API.

2015-05-21 12:10:40 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play-1.0.1:
	* tools/gst-play.c:
	  tools: gst-play: enable interative mode by default
	  And change --interactive option to --no-interactive.

2015-05-21 13:07:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/Makefile.am:
	  rtp: Clean G-I files on make clean too

2015-05-20 16:23:46 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/Makefile.am:
	  rtp: Add builddir to the include path for gobject-introspection
	  And also add missing headers/sources
	  https://bugzilla.gnome.org/show_bug.cgi?id=749632

2015-05-20 15:40:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* win32/common/libgstrtp.def:
	* win32/common/libgstrtsp.def:
	  win32: Update exports

2015-05-20 13:36:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtp/gstrtpdefs.h:
	* gst-libs/gst/rtp/rtp.h:
	  rtp: Add GstRTPProfile enum

2015-05-20 13:35:13 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/gstrtsptransport.h:
	  rtsp: Add FIXME 2.0 comment about GstRTSPTransport being an enum instead of flags

2015-05-20 13:33:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/rtsp/gstrtsptransport.c:
	* gst-libs/gst/rtsp/gstrtsptransport.h:
	  rtsp: Use glib-mkenums to generate GstRTSPProfile and GstRTSPLowerTrans GTypes

2015-05-20 10:22:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/ogg/gstoggdemux.c:
	  Revert "oggdemux: Prevent seeks when _SCHEDULING_FLAG_SEQUENTIAL is set"
	  This reverts commit 76647f2710d718e27f207b005956b7dba72c2d19.
	  Avoiding pull mode activation is a feature regression, and
	  demuxers should always use pull mode where that is possible,
	  e.g. if there's an upstream queue2 with a ring buffer or
	  a download buffer.
	  This patch made reverse playback no longer possible over http.
	  If the goal is to minimise seeks, then that can still be done
	  by making the demuxer behave differently in pull mode if
	  the SEQUENTIAL flag is set. If there are bugs, like the demuxer
	  needlessly scanning the entire file on start-up in pull mode,
	  then those should be fixed instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=746010

2015-05-19 19:48:54 +0100  Tim-Philipp Müller <tim@centricular.com>

	* win32/common/libgstpbutils.def:
	  win32: update .def file for new API

2014-10-24 17:49:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: don't use soon-to-be-deprecated g_cancellable_reset()
	  From the API documentation: "Note that it is generally not
	  a good idea to reuse an existing cancellable for more
	  operations after it has been cancelled once, as this
	  function might tempt you to do. The recommended practice
	  is to drop the reference to a cancellable after cancelling
	  it, and let it die with the outstanding async operations.
	  You should create a fresh cancellable for further async
	  operations."
	  https://bugzilla.gnome.org/show_bug.cgi?id=739132

2014-10-24 17:49:23 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/gio/gstgiobasesink.c:
	* gst/gio/gstgiobasesrc.c:
	  gio: don't use soon-to-be-deprecated g_cancellable_reset()
	  From the API documentation: "Note that it is generally not
	  a good idea to reuse an existing cancellable for more
	  operations after it has been cancelled once, as this
	  function might tempt you to do. The recommended practice
	  is to drop the reference to a cancellable after cancelling
	  it, and let it die with the outstanding async operations.
	  You should create a fresh cancellable for further async
	  operations."
	  https://bugzilla.gnome.org/show_bug.cgi?id=739132

2014-10-24 17:48:54 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/tcp/gstmultioutputsink.c:
	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gsttcpclientsink.c:
	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpserversrc.c:
	  tcp: don't use soon-to-be-deprecated g_cancellable_reset()
	  From the API documentation: "Note that it is generally not
	  a good idea to reuse an existing cancellable for more
	  operations after it has been cancelled once, as this
	  function might tempt you to do. The recommended practice
	  is to drop the reference to a cancellable after cancelling
	  it, and let it die with the outstanding async operations.
	  You should create a fresh cancellable for further async
	  operations."
	  https://bugzilla.gnome.org/show_bug.cgi?id=739132

2015-05-19 18:53:09 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst-libs/gst/pbutils/gstdiscoverer.h:
	  gstdiscoverer: Add since annotation.
	  Forgot to add the since annotation to the
	  GstDiscovererSerializeFlags in the previous commit.

2015-05-03 03:18:28 +0200  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/pbutils/gstdiscoverer.h:
	* tests/check/libs/discoverer.c:
	* win32/common/libgstpbutils.def:
	  discoverer: Add serialization methods.
	  [API] gst_discoverer_info_to_variant
	  [API] gst_discoverer_info_from_variant
	  [API] GstDiscovererSerializeFlags
	  + Serializes as a GVariant
	  + Adds a test
	  + Does not serialize potential GstToc (s)
	  https://bugzilla.gnome.org/show_bug.cgi?id=748814

2015-05-19 16:32:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasepayload.c:
	  rtpbasepayload: Try harder to reuse previously configured caps values and give more preference to anything set as properties
	  This affects the pt, ssrc, seqnum-offset and timestamp-offset properties. If
	  they were set from a property, or we configured caps before, we try to use
	  that value for them. Even if the first structure of the downstream caps
	  specifies a different value, we check if the value is supported by other
	  structures.
	  Only if all this fails, we use the values given by downstream in the first
	  structure, i.e. if no properties were set and these are the first caps we
	  negotiate or downstream does not support our values.
	  By doing this we ensure that we don't spuriously change ssrcs or other fields
	  in the middle of the stream (and also consider property values more). Ssrc
	  changes would currently happen after sending an RTX packet (thus creating a
	  new internal source inside the rtpsession), and then renegotiating the
	  payloader (which then gets the RTX ssrc from rtpsession).
	  https://bugzilla.gnome.org/show_bug.cgi?id=749581

2015-05-18 21:09:25 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/video-scaler.c:
	  docs: a random set of trivial fixes for the library docs
	  Warnings down to 35, unused symbols doen to 112.

2015-05-18 20:56:28 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/allocators/gstfdmemory.c:
	* gst-libs/gst/allocators/gstfdmemory.h:
	  docs: add fdmemory to docs

2015-05-18 20:45:45 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/allocators/gstfdmemory.h:
	* gst-libs/gst/video/colorbalance.h:
	* gst-libs/gst/video/video-scaler.c:
	  docs: a random set of trivial fixes for the library docs
	  All those where super straight forward from the warnings gtkdoc prints. It kind
	  of makes sense to apply them before the list of warnings is >100 and people
	  complain that gtkdoc is noisy.

2015-05-18 20:31:30 +0200  Stefan Sauer <ensonic@users.sf.net>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/sdp/gstmikey.h:
	  mikey: fix a bunch of doc warnings
	  Rename header/source mismatch of parameters. Update the exposed API in
	  sections.txt.

2015-05-18 20:01:49 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/playback/gstplaybin2.c:
	  Revert "doc: Workaround gtkdoc issue"
	  This reverts commit df7ef3c35d34352257a28307c07d4673f239452e.
	  This is fixed by the gtk-doc 1.23 release.

2015-05-18 11:23:16 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/app/gstappsrc.c:
	* tests/check/elements/appsrc.c:
	  appsrc: optimise caps changing when previously-set caps have not taken effect yet
	  Only negotiate/change caps once when setting caps twice and
	  the first-set caps have not been used yet.
	  Based on patch by Eunhae Choi.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747517

2015-05-18 16:16:10 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* sys/xvimage/xvimagesink.c:
	  xvimagesink: fix pool leak
	  During set caps when config fails, the referenced newpool
	  is not unref ed.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749530

2015-05-18 15:45:01 +0900  eunhae choi <eunhae1.choi@samsung.com>

	* gst/playback/gstplaybin2.c:
	  playbin: check the flags before set again
	  check the previous flags of playsink to avoid the reconfigure of playsink repeatedly
	  https://bugzilla.gnome.org/show_bug.cgi?id=749528

2015-05-16 23:33:55 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* gst/playback/gstplaybin2.c:
	  doc: Workaround gtkdoc issue
	  With gtkdoc 1.22, the XML generator fails when a itemizedlist is
	  followed by a refsect2. Workaround the issue by wrapping the refsect2
	  into para.

2015-05-15 14:49:47 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/playback/gstplaybin2.c:
	* gst/playback/gstsubtitleoverlay.c:
	  playback: use the new gst_object api
	  Use gst_object_has_as_anchestor instead of the now deprecated _has_ancestor.

2015-05-10 11:42:21 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/ogg/gstoggmux.c:
	  docs: fix up example pipeline

2015-05-09 22:33:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	* ext/ogg/gstoggdemux.c:
	* ext/pango/gstbasetextoverlay.c:
	* ext/pango/gstclockoverlay.c:
	* ext/pango/gsttextoverlay.c:
	* ext/pango/gsttextrender.c:
	* ext/pango/gsttimeoverlay.c:
	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoraenc.c:
	* ext/theora/gsttheoraparse.c:
	* ext/vorbis/gstvorbisdec.c:
	* ext/vorbis/gstvorbisenc.c:
	* ext/vorbis/gstvorbisparse.c:
	* ext/vorbis/gstvorbistag.c:
	* gst/adder/gstadder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audiorate/gstaudiorate.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/gio/gstgiosink.c:
	* gst/gio/gstgiosrc.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gstsubtitleoverlay.c:
	* gst/tcp/gsttcpclientsink.c:
	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpserversink.c:
	* gst/tcp/gsttcpserversrc.c:
	* gst/videoconvert/gstvideoconvert.c:
	* gst/videorate/gstvideorate.c:
	* gst/videoscale/gstvideoscale.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/volume/gstvolume.c:
	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	  docs: update element example pipelines
	  - gst-launch -> gst-launch-1.0
	  - use autoaudiosink and audiovideosink more often
	  - review pipeline examples and descriptions

2015-05-10 10:51:09 +1000  Jan Schmidt <jan@centricular.com>

	* win32/common/libgstvideo.def:
	  video: Update win32 exports for new libgstvideo API

2015-05-08 15:21:16 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/videoconvert/gstvideoconvert.c:
	* gst/videoconvert/gstvideoconvert.h:
	  videoconvert: Expose some properties from the videoconverter API
	  Expose chroma resampler, alpha mode, alpha value, chroma mode, matrix mode,
	  gamma mode and primaries mode from the videoconverter API.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749105

2015-05-08 14:57:03 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst-libs/gst/video/video-converter.c:
	* gst-libs/gst/video/video-converter.h:
	* gst-libs/gst/video/video-resampler.h:
	* gst/videoscale/gstvideoscale.c:
	  video-converter: Change some implicit string enums to real enums
	  GST_VIDEO_CONVERTER_OPT_ALPHA_MODE, GST_VIDEO_CONVERTER_OPT_CHROMA_MODE,
	  GST_VIDEO_CONVERTER_OPT_MATRIX_MODE, GST_VIDEO_CONVERTER_OPT_GAMMA_MODE and
	  GST_VIDEO_CONVERTER_OPT_PRIMARIES_MODE were G_TYPE_STRING with only a few valid
	  options. Changed those to real enums.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749104

2015-05-08 15:06:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: Also negotiate with downstream if needed before handling a GAP event

2015-05-08 15:02:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Also negotiate with downstream if needed before handling a GAP event

2015-05-06 12:40:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Try to be smarter when clipping buffers without duration/framerate to the segment
	  2 second frame duration is rather unlikely... but if we don't clip
	  away buffers that far before the segment we can cause the pipeline to
	  lockup. This can happen if audio is properly clipped, and thus the
	  audio sink does not preroll yet but the video sink prerolls because
	  we already outputted a buffer here... and then queues run full.
	  In the worst case we will clip one buffer too many here now if no
	  framerate is given, no buffer duration is given and the actual
	  framerate is less than 0.5fps.
	  Fixes seeking on HLS/DASH streams, when seeking into the middle of
	  fragments and having no framerate/buffer duration.

2015-05-04 17:59:30 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/xvimage/xvimagesink.c:
	  xvimagesink: fix navigation event leak when early returning
	  Create the event *after* the early return check so it's not leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748903

2015-05-04 18:00:18 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* sys/xvimage/xvimagesink.c:
	  xvimagesink: fix navigation event leak when not handled
	  gst_navigation_message_new_event() is *not* consuming the event so we should
	  always drop our extra reference.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748903

2015-05-04 17:58:38 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst-libs/gst/video/navigation.c:
	  navigation: fix structure leak if subclass doesn't implement send_event()
	  The send_event() implementation is supposed to consume @structure.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748903

2015-05-05 15:35:46 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Don't override segment.base from upstream with 0
	  Upstream might want to use it to properly map timestamps to running/stream
	  times, if we just override it with 0 synchronization will be just wrong.
	  For this we remove some old 0.10 code related to segment accumulation, and
	  remove some more code that is useless now, and accumulate the group start time
	  (aka segment.base offset) manually now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=635701

2015-05-05 13:14:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepayload: Add some debug output

2015-03-19 10:50:22 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* docs/design/part-mediatype-video-raw.txt:
	* gst-libs/gst/video/video-converter.c:
	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-format.h:
	* gst-libs/gst/video/video-info.c:
	* gst-libs/gst/video/video-scaler.c:
	  video: add NV61 format support
	  https://bugzilla.gnome.org/show_bug.cgi?id=746466

2015-05-04 20:33:23 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: add new video API to docs

2015-05-04 02:18:22 +1000  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/video/video-info.c:
	* gst-libs/gst/video/video-info.h:
	  video: check colorimetry and chroma_site equality in gst_video_info_is_equal()
	  Add VideoInfo accessors for colorimetry and chroma_site and use them
	  when checking the equality of two GstVideoInfo

2015-05-04 02:10:17 +1000  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/video/video-color.c:
	* gst-libs/gst/video/video-color.h:
	* win32/common/libgstvideo.def:
	  video-color: Add gst_video_colorimetry_is_equal()
	  Add a function for comparing the equality of 2 colorimetry
	  structures.

2015-04-10 16:05:45 +0900  Young Han Lee <y.lee@lge.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: remove unused code
	  These lines have done nothing for about 10 years.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748820

2015-04-10 15:24:28 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  pbutils: Use more strict profile checking for hevc
	  Use the profile_idc value to set the profile string in caps.
	  Don't use compatibility flags for this purpose.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747613

2015-04-30 14:55:14 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst-libs/gst/video/video-converter.c:
	  video-converter: Remove unused macro
	  Remove unused macro GET_TMP_LINE
	  https://bugzilla.gnome.org/show_bug.cgi?id=748687

2015-04-29 15:44:59 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play.c:
	  gst-play: add some more key navigation mappings
	  And don't feed multi-character key descriptors to the
	  event handler, it won't be what it expects.

2015-04-29 15:30:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/video/navigation.c:
	* gst-libs/gst/video/navigation.h:
	* win32/common/libgstvideo.def:
	  navigation: sprinkle some since markers and add new API to .def file
	  https://bugzilla.gnome.org/show_bug.cgi?id=747245

2015-04-02 16:16:58 +0200  Edward Hervey <edward@centricular.com>

	* tools/gst-play.c:
	  tools: Add mouse/keyboard handling from messages
	  Allows the user to control playback with the window in focus
	  https://bugzilla.gnome.org/show_bug.cgi?id=747245

2015-04-02 16:10:32 +0200  Edward Hervey <edward@centricular.com>

	* sys/xvimage/xvimagesink.c:
	  xvimagesink: Post unhandled navigation events on the bus
	  https://bugzilla.gnome.org/show_bug.cgi?id=747245

2015-04-02 16:09:13 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/video/navigation.c:
	* gst-libs/gst/video/navigation.h:
	  video: Add a new "event" navigation message type
	  This will be useful for elements that wish to post unhandled navigation
	  events on the bus to give the application a chance to do something with
	  it
	  https://bugzilla.gnome.org/show_bug.cgi?id=747245

2015-04-28 12:01:02 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-info.c:
	* gst-libs/gst/video/video-info.h:
	* win32/common/libgstvideo.def:
	  video-info: expose InterlaceMode conversion to/from string
	  Expose the methods used to convert a GstVideoInterlaceMode to and
	  from a string.

2015-04-27 11:26:10 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/audioconvert/gstaudioconvert.c:
	* gst/audiorate/gstaudiorate.c:
	* gst/encoding/gstsmartencoder.c:
	  Rename property enums from ARG_ to PROP_
	  Property enum items should be named PROP_ for consistency and readability.

2015-04-27 11:06:58 +0200  Matthieu Bouron <matthieu.bouron@collabora.com>

	* gst/videoconvert/gstvideoconvert.c:
	  videoconvert: Keep colorimetry and chroma-site fields if passthrough
	  https://bugzilla.gnome.org/show_bug.cgi?id=748141

2015-04-27 10:08:17 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.h:
	  audio: Change the remaining "samples" in the ::delay() vfunc docs to "frames"
	  https://bugzilla.gnome.org/show_bug.cgi?id=748289

2015-04-26 20:13:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/pipelines/tcp.c:
	  tests: tcp: remove SOCK_CLOEXEC which causes build problems on OS/X
	  It's not needed here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747692

2015-04-26 21:08:14 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioringbuffer.h:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.h:
	  audio: The delay vfunc returns the number of frames, not samples
	  https://bugzilla.gnome.org/show_bug.cgi?id=748289

2015-04-26 17:49:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* Android.mk:
	* android/NOTICE:
	* android/alsa.mk:
	* android/app.mk:
	* android/app_plugin.mk:
	* android/audio.mk:
	* android/audioconvert.mk:
	* android/audioresample.mk:
	* android/audiotestsrc.mk:
	* android/decodebin.mk:
	* android/decodebin2.mk:
	* android/gdp.mk:
	* android/pbutils.mk:
	* android/playbin.mk:
	* android/queue2.mk:
	* android/riff.mk:
	* android/rtp.mk:
	* android/rtsp.mk:
	* android/sdp.mk:
	* android/tag.mk:
	* android/tcp.mk:
	* android/typefindfunctions.mk:
	* android/video.mk:
	* android/videoconvert.mk:
	* android/videoscale.mk:
	* android/videotestsrc.mk:
	* ext/ogg/Makefile.am:
	* ext/vorbis/Makefile.am:
	* gst-libs/gst/allocators/Makefile.am:
	* gst-libs/gst/app/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/fft/Makefile.am:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/riff/Makefile.am:
	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtsp/Makefile.am:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/tag/Makefile.am:
	* gst-libs/gst/video/Makefile.am:
	* gst/adder/Makefile.am:
	* gst/app/Makefile.am:
	* gst/audioconvert/Makefile.am:
	* gst/audiorate/Makefile.am:
	* gst/audioresample/Makefile.am:
	* gst/audiotestsrc/Makefile.am:
	* gst/encoding/Makefile.am:
	* gst/playback/Makefile.am:
	* gst/tcp/Makefile.am:
	* gst/typefind/Makefile.am:
	* gst/videoconvert/Makefile.am:
	* gst/videorate/Makefile.am:
	* gst/videoscale/Makefile.am:
	* gst/videotestsrc/Makefile.am:
	* gst/volume/Makefile.am:
	* tools/Makefile.am:
	  Remove obsolete Android build cruft
	  This is not needed any longer.

2015-04-26 14:37:56 +0100  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/gst/typefindfunctions.c:
	  tests: typefindfunctions: add test for UTF-16 MSS manifest typefinding

2015-04-26 14:44:33 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefinding: don't read more data than needed in MSS typefinder

2015-04-26 14:27:30 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefinding: detect MSS manifests without using g_convert()
	  Embedded systems often have limited charset conversion
	  functionality, so don't rely on g_convert() (i.e. iconv)
	  for UTF-16 to UTF-8 conversions, we can easily enough do
	  that ourselves by converting to native endianness and
	  then using GLib's helper functions.

2015-04-25 18:45:50 +0200  Stefan Sauer <ensonic@users.sf.net>

	* ext/libvisual/gstaudiovisualizer.c:
	* ext/libvisual/gstaudiovisualizer.h:
	  audiovisualizer: fix the license from GPL to LGPL
	  This was a copy'n'paste buf in the initial commit done by myself.

2015-04-24 14:59:21 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst-libs/gst/tag/gstxmptag.c:
	  xmptag: fix invalid reads in GST_DEBUG statement
	  Don't try to print a string that is not NUL-terminated. This
	  log line does not really seem useful so let's just drop it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=748413

2015-04-24 17:10:59 +0100  Luis de Bethencourt <luis.bg@samsung.com>

	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/encoding/gstencodebin.c:
	* gst/playback/gstdecodebin2.c:
	* gst/playback/gstplaybin2.c:
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