Release 1.5.11.5.1

42 files changed, 8127 insertions, 247 deletions

diff --git a/ChangeLog b/ChangeLog
index 95c21b1ab..e20813746 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,7295 @@
+=== 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:
+	* gst/playback/gstplaysink.c:
+	* gst/playback/gsturidecodebin.c:
+	* gst/tcp/gstmultifdsink.c:
+	* gst/tcp/gstmultihandlesink.c:
+	* gst/tcp/gstmultioutputsink.c:
+	* gst/videotestsrc/gstvideotestsrc.c:
+	  remove unused enum items PROP_LAST
+	  This were probably added to the enums due to cargo cult programming and are
+	  unused. Removing them.
+
+2015-04-03 00:44:12 +0900  Wonchul Lee <chul0812@gmail.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudiodecoder.h:
+	  audiodecoder: Add sink and src query virtual method
+	  API: GstAudioDecoderClass::src_query()
+	  API: GstAudioDecoderClass::sink_query()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747293
+
+2015-04-23 15:57:37 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	  tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+	  Make sure the test environment is set up.
+	  https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 15:42:41 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* configure.ac:
+	  configure: bump automake requirement to 1.14 and autoconf to 2.69
+	  This is only required for builds from git, people can still
+	  build tarballs if they only have older autotools.
+	  https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 15:14:07 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* .gitignore:
+	* tests/check/libs/.gitignore:
+	* tests/check/pipelines/.gitignore:
+	  Update .gitignore
+
+2015-04-23 09:50:12 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: n_lines member should be a guint not a boolean
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748348
+
+2015-04-21 15:27:57 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: fix event leaks
+	  gst_event_replace() takes its own reference on the event so we should drop
+	  ours after creating and storing an event using it.
+	  This fix leaks which can be reproduced using the
+	  validate.http.media_check.vorbis_theora_1_ogg scenario.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748247
+
+2015-04-22 10:34:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* INSTALL:
+	  Remove INSTALL file
+	  autotools automatically generate this, and when using different versions
+	  for autogen.sh there will always be changes to a file tracked by git.
+
+2015-04-22 10:33:58 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* LICENSE_readme:
+	  Remove LICENSE_readme
+	  It's completely outdated and just confusing, better if people are
+	  forced to look at the actual code in question than trusting this file.
+
+2015-04-21 13:31:44 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix YUY2 scaling some more
+	  Take into account the different steps between Y and UV when calculating
+	  the line size for vertical resampling or else we might not resample
+	  enough pixels and leave bad lines.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=747790
+
+2015-04-21 13:16:29 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: scale enough pixels in YUY2 (and friends) mode
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=747790
+
+2015-04-17 16:21:05 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* tests/check/libs/rtpbasedepayload.c:
+	  tests: rtpbasedepayload: fix crash in test when passing varargs
+	  Need to pass 64 bits where 64 bits are expected.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748027
+
+2015-04-17 11:18:22 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Remove unused variables
+	  Remove unused variables n_taps, max_taps in setup_scale()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=748021
+
+2015-04-16 10:03:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/gstvideoutils.h:
+	  video: add missing part of documentation text
+
+2015-03-31 13:26:21 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* gst-libs/gst/pbutils/gstdiscoverer.c:
+	  discoverer: fix GstToc leak when parsing toc messages
+	  gst_message_parse_toc() returns a reffed GstToc which is owned by the
+	  GstDiscovererInfo. But we have to make sure we unref its previous value before
+	  setting the new one.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747103
+
+2015-04-17 11:45:34 +0200  Edward Hervey <edward@centricular.com>
+
+	* win32/common/libgstallocators.def:
+	  win32: Update defs for new API
+
+2015-04-17 09:31:40 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/allocators/gstdmabuf.c:
+	* gst-libs/gst/allocators/gstfdmemory.c:
+	* gst-libs/gst/allocators/gstfdmemory.h:
+	  allocators: make GstFdAllocator non-abstract
+	  Make the GstFdAllocator non-abstract because it is perfectly possible
+	  to make memory from a generic fd. Mark the memory as simply "fd".
+
+2015-04-15 11:24:17 +0200  Bernhard Miller <bernhard.miller@streamunlimited.com>
+
+	* gst/audioconvert/gstchannelmix.c:
+	  audioconvert: fix mixed usage of gint and gint32 in int matrix
+	  This is a fixup for b2db18cda2e4e7951655cb2a34108a8523b6eca9
+	  audioconvert: avoid float calculations when mixing integer-formatted channels
+	  The int matrix was using gint and gint32 synonymously, which can theoretically
+	  cause problems if gint and gint32 are actually different types.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747005
+
+2015-04-14 12:47:07 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* common:
+	* gst/gio/gstgio.c:
+	  gio: fix gvfs plugin dependencies
+	  Try harder to look for gvfs backend changes in the right
+	  place, to make sure the plugin gets reloaded when backends
+	  are removed or installed. We watch the gvfs mounts directory
+	  because the files there contain absolute paths to the
+	  backend executables, and those may not be in the usual gio
+	  path.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747841
+
+2015-04-14 15:08:09 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/scrubby.c:
+	  examples: disconnect scale callback in scrubby
+	  When the position slider's button is released, disconnect the "value_changed"
+	  callback to avoid triggering false seek callbacks.
+
+2015-04-13 17:35:36 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/scrubby.c:
+	  examples: keep scrubby command consistent
+	  scrubby has two options, wav and playbin. Wav takes a file location so make
+	  the playbin option take a file location as well instead of an uri. This also
+	  means the usage help string will be correct for the playbin option.
+
+2015-04-13 17:28:45 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/scrubby.c:
+	  examples: no need to set intermediate states
+
+2015-04-13 16:09:26 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/scrubby.c:
+	  examples: wavparse doesn't need dynamic linking
+	  In scrubby, there is no need to link wavparse with the sink dynamically.
+	  The pad is available when the element is generated.
+	  Change video and audio sinks to the automatically detected sinks.
+
+2015-04-11 19:51:54 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: Break instead of return if default negotiation on GAP events fails
+	  Otherwise we're going to leak the event.
+
+2015-04-11 00:03:29 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/app/Makefile.am:
+	* gst/videorate/Makefile.am:
+	  app, videorate: fix CFLAGS and LIBADD order
+	  Make sure local headers are included before installed -base.
+
+2015-04-10 14:30:36 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/playrec/playrec.c:
+	  examples: remove reference to 0.10 in playrec
+
+2015-04-10 13:41:39 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/overlay/gtk-videooverlay.c:
+	  examples: remove deprecated function in gtk-videooverlay
+	  gtk_widget_set_double_buffered () has been deprecated since GTK 3.14.
+	  Also, widgets are realized automatically and gtk_wiget_realize () is only
+	  meant to be used in widget implementations.
+
+2015-04-09 17:03:11 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: fix buffer leak in chain function
+	  If we don't consume the buffer by passing its reference to
+	  overlay->text_buffer then we need to unref it.
+	  Fix a leak with validate.file.playback.fast_forward.test5_mkv
+	  when running inside Valgrind.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747602
+
+2015-04-08 18:32:29 +0300  Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+	* gst-libs/gst/app/gstappsrc.c:
+	  appsrc: docs grammar fixes
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747516
+
+2015-04-09 16:49:44 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/gio/giosrc-mounting.c:
+	  examples: add example description to giosrc-mounting
+	  Also, use GST_MESSAGE_TYPE instead of accessing the GstMessage structure
+
+2015-04-09 13:00:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: fix ring buffer leak on open failure
+
+2015-04-09 12:59:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstaudiobasesrc.c:
+	  audiobasesrc: fix ring buffer leak on open failure
+
+2015-04-09 11:23:25 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/encoding/encoding.c:
+	  examples: reuse variables in encoding example
+
+2015-04-08 20:49:24 -0700  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Don't post error messages while holding the stream lock
+
+2015-04-08 20:48:39 -0700  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Don't get and parse the current srcpad caps
+	  We only get here if we don't have any srcpad caps, and we're going
+	  to override the GstAudioInfo a few lines below anyway without ever
+	  using it if for whatever reason we get caps here.
+
+2015-04-08 20:45:58 -0700  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: Try to invent default caps instead of setting none at all when getting a GAP event before CAPS
+	  Otherwise we would forward the GAP event without ever providing any caps,
+	  which then would make decodebin expose a srcpad without any caps set. That's
+	  confusing for applications and can lead to all kinds of interesting bugs.
+	  Instead do the same as already is done in GstAudioDecoder, and try to invent
+	  caps based on the sinkpad caps and the caps allowed by downstream and the
+	  srcpad template caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747190
+
+2015-04-08 20:44:15 -0700  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Also log the pointer value of sticky events in debug output
+	  Makes it easier to follow them in the debug logs.
+
+2015-04-08 17:12:22 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/dynamic/addstream.c:
+	  examples: remove unused return value in addstream
+	  Removing unused return value of pause_play_stream ().
+	  Fixing code style to satisfy the git hook.
+
+2015-04-08 15:31:39 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/dynamic/sprinkle.c:
+	  examples: avoid sprinkle running endlessly
+	  Quit sprinkle when there are no more frequencies to remove.
+	  Also rename for readability the check for linking elements.
+
+2015-04-08 16:15:43 +0200  Edward Hervey <edward@centricular.com>
+
+	* common:
+	* tests/check/Makefile.am:
+	  tests: Use AM_TESTS_ENVIRONMENT
+	  Needed by the new automake test runner
+
+2015-04-07 16:43:59 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/rtp/gstrtcpbuffer.h:
+	  rtp: rtcpbuffer: fix typo in enum
+	  and in docs. Spotted by Rob Swain.
+
+2015-04-07 15:32:35 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/app/appsink-src2.c:
+	  tests: remove unused filename string from appsink-src2
+
+2015-04-07 15:30:30 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/app/appsink-src.c:
+	  tests: check file exists before running appsink-src
+
+2015-04-07 15:16:41 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/app/appsink-src.c:
+	* tests/examples/app/appsink-src2.c:
+	* tests/examples/app/appsrc_ex.c:
+	  tests: add missing license headers for example apps
+
+2015-04-06 19:20:00 -0700  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  {audio,video}decoder: Forward SEGMENT_DONE events immediately and drain decoders
+	  Otherwise we're going to wait with draining until the next data comes, which
+	  is a bit suboptimal and might take a long time... or maybe never happens.
+
+2015-04-05 13:53:38 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/appsrc.c:
+	  tests: appsrc: clean up block_deadlock test and make it work in valgrind
+	  Remove all the bus watch and main loop code from the block_deadlock
+	  test, it's not needed: neither pipeline will ever post an EOS or ERROR
+	  message on the bus, and we're the only ones posting an error, from a
+	  timeout. Might just as well just sleep for a bit and then do whatever
+	  we want to do.
+	  Don't gratuitiously set tcase timeout, just use whatever is the
+	  default (or set via the environment).
+	  Make individual pipeline runs shorter.
+	  Check for valgrind and only do a handful iterations when running
+	  in valgrind, not 100 (each iteration takes about 4s on a core i7).
+	  Make videotestsrc output smaller buffers than the default resolution,
+	  we don't care about the buffer contents here anyway.
+	  Fixes test timeouts when run in valgrind.
+
+2015-04-05 12:30:39 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/multisocketsink.c:
+	  tests: multisocketsink: fix flaky unit test
+	  On slower systems, or under high system load (e.g. check-valgrind),
+	  the sending_buffers_with_9_gstmemories test would sometimes fail,
+	  because the read call only returns 32 bytes instead of the full
+	  36 bytes expected. This is because multisocketsink might end up
+	  doing a partial write of 32 bytes first, and then write the
+	  missing 4 bytes later, but since we don't wait for all of data
+	  to be written, there's a short window where our read call in the
+	  unit test might then only receive the 32 bytes written so far,
+	  which makes it deeply unhappy.
+	  Instead, make sure we loop to read all bytes.
+
+2015-04-04 21:38:40 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/tcp/gstmultisocketsink.c:
+	  tcpserversink: don't error out if clients send us something, just ignore it
+	  We don't expect clients to send us any data, but if they do, just
+	  ignore it. Web browsers might send us an HTTP request for example,
+	  but some will still be happy if we just send them data without
+	  a proper HTTP response.
+	  There was a bug in the reading code path. We only have a small
+	  read buffer and would provoke an EWOULDBLOCK trying to read
+	  because we don't bail out of the loop early enough.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=743834
+
+2015-04-04 01:23:48 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/pipelines/basetime.c:
+	  tests: basetime: fix timeouts when running under valgrind
+	  This test sets a rather short timeout, increase this when
+	  we run under valgrind. Also add a short sleep to the
+	  fakesrc ! fakesink pipeline to avoid thrashing the CPU,
+	  which would often not stop the main loop when it should.
+	  Also fix wrong (0.10) return value from pad probe callback.
+
+2015-04-04 00:46:46 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/videorate/gstvideorate.c:
+	  videorate: downgrade left-over ERROR debug message
+
+2015-04-04 00:42:52 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/videorate/gstvideorate.c:
+	* tests/check/elements/videorate.c:
+	  videorate: fix a couple of memory leaks
+	  tests: videorate: fix leak in unit test
+
+2015-04-03 18:18:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	  doc: Add gst_video_encoder_get_allocator() to doc
+
+2015-04-03 21:00:53 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/tag/gstexiftag.c:
+	  tag: exiftag: don't try to convert utf-8 to latin1 if string is ASCII already
+	  Bypass g_convert/iconv if there's nothing to convert. That way,
+	  conversion won't fail on systems where iconv doesn't support
+	  converting utf-8 to latin1 and there's nothing to convert.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723252
+
+2015-04-03 18:57:43 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From bc76a8b to c8fb372
+
+2015-03-12 16:01:48 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	* ext/ogg/gstoggdemux.h:
+	  oggdemux: fix wrong duration on partial streams with a skeleton index
+	  When a stream has a skeleton index, the stream time is taken from that
+	  index. However, when part of the stream is captured, the index is
+	  invalid as its offsets are now wrong. To avoid this, we ignore the index
+	  when the last offset points beyond the end of the stream (when its
+	  byte length is known).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744070
+
+2015-03-18 16:32:53 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  textoverlay: fix disappearing text with high deltax
+	  When deltax is large enough to cause the text to push past the
+	  width of the frame, it would disappear due to a bug in setting
+	  the layout width.
+	  While there, fix a log printing an incorrect width to set.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739689
+
+2014-12-17 12:17:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: fix deadlock when not pulling a buffer from collectpads
+	  oggmux keeps a cached buffer per pad, and pulls buffers from
+	  collectpads to this cached buffer for all pads before processing
+	  the best pad. In some cases, the move from collectpads buffer
+	  to cached buffer is delayed till next call. However, when there
+	  is only one pad, this can't be delayed till next call as there
+	  will be a deadlock since collectpads has no other pad to push to.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740565
+
+2015-03-25 15:36:38 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin2: fix deadlock on chain shutdown
+	  When shutting down the chain, we can get a deadlock when removing
+	  a pad, if that chain was being busy streaming but blocked (eg, while
+	  waiting for a queue to have free space).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746480
+
+2015-04-03 13:20:58 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/scrubby.c:
+	  examples: add license header to scrubby
+
+2015-03-19 10:48:15 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  audio,video: use gst_segment_is_equal instead of memcmp
+	  memcmp will blindly compare the reserved fields, as well as any
+	  padding the compiler may choose to sprinkle in GstSegment.
+	  Fixes valgrind complaints in unit tests, as well as some found via
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738216
+
+2014-04-04 12:32:14 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* sys/xvimage/xvimageallocator.c:
+	  xvimagsink: fix failure to allocate large shared memory blocks
+	  A previous patch increased allocations by 15 bytes in order to ensure
+	  16 byte alignment for g_malloc blocks. However, shared memory is
+	  already block aligned, and this extra 15 bytes caused allocation
+	  to fail when we were already allocating to the shared memory limit,
+	  which is a lot smaller than typical available RAM.
+	  Fix this by removing the alignment slack when allocating shared
+	  memory.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=706066
+
+2014-04-04 12:40:14 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* sys/ximage/ximagepool.c:
+	  ximage: do not allocate extra alignment slack for shared memory
+	  A previous patch increased allocations by 15 bytes in order to ensure
+	  16 byte alignment for g_malloc blocks. However, shared memory is
+	  already block aligned, and this extra 15 bytes is not needed. Since
+	  shared memory limits are low compared to RAM, we remove this waste.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727236
+
+2015-04-03 13:56:28 +0900  Chihyoung Kim <chihyoung2.kim@lge.com>
+
+	* configure.ac:
+	  tests: require Gtk+ 3.10 for examples
+	  Fixes build of playback and seek tests when an
+	  older Gtk+ version is present on the system.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747283
+
+2014-12-09 13:18:42 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/videorate/gstvideorate.c:
+	* gst/videorate/gstvideorate.h:
+	* tests/check/elements/videorate.c:
+	  videorate: Detect framerate if not forced to variable downstream
+	  In case upstream does not provide videorate with framerate information,
+	  it will detect the current framerate from the buffer it received,
+	  but if downstream forces the use of variable framerate (most probably
+	  through the use of a caps filter with framerate = 0 / 1), videorate will
+	  respect that.
+	  And add some unit tests
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734424
+
+2014-12-09 11:31:30 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/videorate/gstvideorate.c:
+	  videorate: Do not loop forever pushing first buffer when variable framerate
+	  In the case the framerate is variable (represented by framerate=0/1),
+	  we currently end up loop pushing the first buffer and then recompute
+	  diff1 and diff2 without updating the videorate->next_ts at all
+	  leading to infinitely looping pushing that first buffer.
+	  In the case of variable framerate, we should just compute the next_ts
+	  as previous_pts + previous_duration.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734424
+
+2015-04-02 14:32:15 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/playback/playback-test.c:
+	  playback-test: update deprecated API
+
+2015-04-02 11:33:12 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/icles/test-colorkey.c:
+	* tests/icles/test-videooverlay.c:
+	  tests: fix deprecated API in colorkey and videooverlay
+
+2015-04-02 11:14:08 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/scrubby.c:
+	  examples: fix deprecated API in scrubby
+
+2015-03-19 14:34:07 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: don't use GST_ERROR() for debug messages
+	  Fix https://bugzilla.gnome.org/show_bug.cgi?id=746457
+
+2015-04-01 15:58:28 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/audio/volume.c:
+	  tests: use elapsed label of volume example
+
+2015-03-30 11:24:46 +0200  Bernhard Miller <bernhard.miller@streamunlimited.com>
+
+	* gst/audioconvert/audioconvert.h:
+	* gst/audioconvert/gstchannelmix.c:
+	  audioconvert: avoid float calculations when mixing integer-formatted channels
+	  The patch calculates a second channel mixing matrix from the current one. The
+	  matrix contains the original values * (2^10) as integers. This matrix is used
+	  when integer-formatted channels are mixed.
+	  On a ARM Cortex-A8, single core, 800MHz this improves performance in a
+	  testcase from 29s to 9s for downmixing 6 channels to stereo.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747005
+
+2015-04-01 15:02:13 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/audio/volume.c:
+	  tests: fix deprecated API in audio volume example
+
+2015-04-01 14:37:23 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/jsseek.c:
+	  jsseek: update deprecated GTK API
+
+2015-04-01 13:50:51 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/seek/jsseek.c:
+	  jsseek: switch deprecated GtkTable for GtkGrid
+
+2015-04-01 11:01:57 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* tests/examples/audio/audiomix.c:
+	  tests: update deprecated GTK API in audiomix
+
+2015-03-31 11:21:25 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* 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:
+	  introspection: Don't use g-ir-scanner cache at compile time
+	  It pollutes user directories and we don't need to cache it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=747095
+
+2014-04-10 12:03:05 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst-libs/gst/tag/id3v2frames.c:
+	  id3v2: ignore RVA2 tags with more than 64 peak bits
+	  The spec for this does not say nor imply how this should be
+	  interpreted.  The previous code would try to shift by 64 bits,
+	  which is undefined.
+	  Coverity 1195119
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727955
+
+2015-03-30 10:50:45 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: avoid possible deference of null pointer
+	  For safety, check the pointer playbin->curr_group is valid before
+	  reading parameters of the structure.
+	  CID #1291624
+
+2015-03-28 16:59:23 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: resurrect some flow return handling
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-27 20:16:28 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst-libs/gst/app/gstappsrc.c:
+	  appsrc: handle a sample not having caps or a buffer more gracefully
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746908
+
+2015-03-27 16:22:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
+	* tests/check/libs/rtpbasedepayload.c:
+	  basedepay: Handle initial gaps and no clock-base
+	  When generating segment, we can't assume the first buffer is actually
+	  the first expected one. If it's not, we need to adjust the segment to
+	  start a bit before.
+	  Additionally, we if don't know when the stream is suppose to have
+	  started (no clock-base in caps), it means we need to keep everything in
+	  running time and only rely on jitterbuffer to synchronize.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-03-26 23:53:44 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: improve debug message by printing the object
+	  Print the pad object that EOS'd too early
+
+2015-03-27 13:39:43 +0800  Song Bing <b06498@freescale.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: Keep sticky events around when doing a soft reset
+	  The current code will first discard all frames, and then tries to copy
+	  all sticky events from the (now discarded) frames. Let's change the order.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746865
+
+2015-03-26 18:03:12 -0700  David Schleef <ds@schleef.org>
+
+	* gst-libs/gst/riff/riff-ids.h:
+	  riff: Add FLLR tag
+
+2015-03-25 18:40:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
+	* tests/check/libs/rtpbasedepayload.c:
+	  basedepayload: Fix generated segment
+	  This fixes playback position in RTSP.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-03-25 08:20:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: ignore new pads if it is shutting down
+	  If a new pad is added after playbin has been put to READY/NULL it
+	  should ignore new pads as it is shutting down.
+	  This can happen when the pipeline fails to preroll (is still in READY)
+	  and the user gives up on waiting or an error that doesn't reach
+	  the demuxer occurs (on some event handling) and it will continue to
+	  work and exposing pads while playbin has been put to NULL.
+	  Without this check an input-selector is created and set to PAUSED
+	  state, preventing playbin from properly shutting down in case it
+	  has data blocked inside it.
+
+2015-03-24 15:47:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* ext/theora/gsttheoradec.c:
+	  Revert "theoradec: Disable usage of crop meta"
+	  This reverts commit da52868f468bd75ddb595a3eb52aaa38ecbbac41.
+
+2015-03-24 15:18:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* gst/videorate/gstvideorate.c:
+	  videorate: Don't leak the pools
+	  gst_query_set_nth_alloction_pool() is transfer none on the pool, so we must
+	  unref the pool when done.
+
+2015-03-01 11:44:22 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* ext/theora/gsttheoradec.c:
+	  theoradec: Disable usage of crop meta
+	  This is a temporary workaround that simply disables usage of crop
+	  meta for now.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741030
+
+2015-03-24 17:28:51 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+	* gst/audioconvert/gstaudioquantize.c:
+	  audioconvert: Eliminate unsigned quantizers
+	  audio_convert_convert unpacks to default format (signed) before calling
+	  quantize, and the unsigned variants were equivalent to signed anyway,
+	  so we just get rid of them.
+
+2015-03-24 03:01:22 +0200  Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+	* gst/audioconvert/gstaudioquantize.c:
+	* gst/audioconvert/gstfastrandom.h:
+	  audioconvert: Avoid int division in quantization
+	  Since range size is always 2^n, we can simply use modulo (implemented
+	  with a bitmask).
+	  The previous implementation used 64-bit integer division, which is
+	  done in software on ARMv7. Although the divisor was constant, the
+	  division could not be transformed into "multiplication by magic number"
+	  since the dividend was 64-bit.
+	  The now-unused and not-so-fast gst_fast_random_(u)int32_range functions
+	  were removed.
+	  Also, implementing bug fixes:
+	  1) ADD_DITHER_TPDF_HF_I no longer discards bias.
+	  2) We change TPDF's noise range to be the same as RPDF's. Previously,
+	  RPDF's noise ranged:
+	  { bias - dither, bias + dither }
+	  while TPDF's noise ranged:
+	  { bias/2 - dither/2, bias/2 + dither/2 - 1 } +
+	  { bias/2 - dither/2, bias/2 + dither/2 - 1 } =
+	  { bias - dither, bias + dither - 2 }
+	  Now, both range:
+	  { bias - dither, bias + dither - 1 }
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746661
+
+2015-02-16 09:25:03 +1000  Duncan Palmer <dpalmer@digisoft.tv>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin2: Set multiqueue sizes before use-buffering.
+	  This fixes a race where the use-buffering property on a multiqueue was
+	  set before the queue depth was changed from it's high preroll limits to
+	  lower playback limits. This resulted in buffering messages being emitted
+	  by the multiqueue in the short window between use-buffering being
+	  set and the queue depth being reset.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744308
+
+2015-03-24 10:46:44 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/allocators/gstfdmemory.c:
+	  Revert "fdmemory: freed pointer will always be 0"
+	  This reverts commit 7fbcefb753f944a79eae6957ea2789c960eb9eea.
+
+2015-03-24 10:19:05 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/allocators/gstfdmemory.c:
+	  fdmemory: freed pointer will always be 0
+
+2015-03-20 17:45:03 +0900  Wonchul Lee <chul0812@gmail.com>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: Fix compiler warning
+	  gstoggdemux.c:1233:11: error: format specifies type 'long' but the argument has type 'ogg_int64_t' (aka 'long long') [-Werror,-Wformat]
+	  granule);
+	  ^~~~~~~
+	  https://bugzilla.gnome.org/show_bug.cgi?id=746512
+
+2015-03-19 13:31:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* win32/common/libgstallocators.def:
+	  defs: update
+
+2015-03-19 12:42:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-convert: fix clamping for 16 bits alpha mult
+
+2015-03-18 20:38:20 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-frame.c:
+	  video-frame: fix height/width assertions
+	  As commit 274984e8 states:
+	  When doing CROP META it is expected that the width and/or height
+	  in the GstVideoMeta is bigger or equal to the caps negotiated size.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741030
+
+2015-03-18 15:12:03 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/allocators/Makefile.am:
+	* gst-libs/gst/allocators/gstdmabuf.c:
+	* gst-libs/gst/allocators/gstfdmemory.c:
+	* gst-libs/gst/allocators/gstfdmemory.h:
+	  fdmemory: make a base class for allocating fd-backed memory
+	  Make a base class that can help with allocating fd-backed memory.
+	  Make dmabuf extend from the base class.
+	  We can now make methods to check if memory has an fd and get the fd for
+	  all the different types of fd-backed memory.
+
+2015-03-16 20:41:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/multisocketsink.c:
+	  multisocketsink: Allocate enough memory on the stack in the test
+	  Otherwise we just overwrite other things on the stack and cause crashes.
+
+2015-03-16 11:53:24 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: fix playback regression on streams with clipped data at start
+	  The code that was calculating the start granule from packet durations
+	  was interpreting a negative value as an error, but this is actually a
+	  valid case, to indicate clipping of data at start.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=743900
+
+2015-03-15 17:27:33 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/allocators/gstdmabuf.c:
+	* gst-libs/gst/allocators/gstfdmemory.c:
+	* gst-libs/gst/allocators/gstfdmemory.h:
+	  fdmemory: add flags to control behaviour
+	  Add some flags to the GstFdMemory to control how memory is mapped and
+	  unmapped.
+
+2015-03-15 16:41:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/libs/allocators.c:
+	  allocators: add allocators test
+
+2015-03-15 15:16:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/allocators/Makefile.am:
+	* gst-libs/gst/allocators/gstdmabuf.c:
+	* gst-libs/gst/allocators/gstfdmemory.c:
+	* gst-libs/gst/allocators/gstfdmemory.h:
+	  fdmemory: add fd backed GstMemory to separate file
+	  Make a separate file for the code to handle the fd backed memory.
+	  This would make it possible later to add other allocators also using
+	  fd backed memory.
+
+2015-03-14 18:08:15 +0000  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gststreamsynchronizer.c:
+	  streamsynchronizer: fix deadlock condition
+	  The variables could have changed when the lock was released
+	  to push a gap event. Streamsynchronizer needs to check them
+	  again before going to sleep.
+	  Bonus: fix a comment typo
+
+2015-03-13 18:07:12 +0000  Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+	* gst/playback/gstplaysink.c:
+	  playsink: remove redundant else statements
+
+2015-03-13 18:23:46 +0000  Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: don't escape percent sign in documentation code sample
+
+2014-11-03 12:47:18 +0000  William Manley <will@williammanley.net>
+
+	* configure.ac:
+	* tests/check/Makefile.am:
+	* tests/check/pipelines/tcp.c:
+	  Add test_that_multisocketsink_and_socketsrc_preserve_meta
+	  This test is in a seperate commit to the previous two because it depends
+	  on and tests the functionality in both.
+
+2015-03-13 16:19:28 +0000  William Manley <will@williammanley.net>
+
+	* gst/tcp/gstsocketsrc.c:
+	  socketsrc: Add support for GstNetControlMessageMeta
+	  multisocketsink now understands the new GstNetControlMessageMeta to allow
+	  sending control messages (ancillary data) with data when writing to Unix
+	  domain sockets.
+	  Thanks to glib's `GSocketControlMessage` abstraction the code introduced
+	  in this commit is entirely portable and doesn't introduce and additional
+	  dependencies or conditionally compiled code, even if it is unlikely to be
+	  of much use on non-UNIX systems.
+
+2014-10-30 17:53:15 +0000  William Manley <will@williammanley.net>
+
+	* configure.ac:
+	* gst/tcp/gstmultisocketsink.c:
+	  multisocketsink: Add support for GstNetControlMessageMeta
+	  multisocketsink now understands the new GstNetControlMessageMeta to allow
+	  sending control messages (ancillary data) with data when writing to Unix
+	  domain sockets.
+	  A later commit will introduce a new socketsrc element which will similarly
+	  understand `GstNetControlMessageMeta`.  This, when used with a
+	  `GSocketControlMessage` of type `GUnixFDMessage` will allow GStreamer to
+	  send and receive file-descriptions in ancillary data, the first step to
+	  using memfds to implement zero-copy video IPC.
+	  Thanks to glib's `GSocketControlMessage` abstraction the code introduced
+	  in this commit is entirely portable and doesn't introduce and additional
+	  dependencies or conditionally compiled code, even if it is unlikely to be
+	  of much use on non-UNIX systems.
+
+2015-03-13 13:56:13 +0000  William Manley <will@williammanley.net>
+
+	* gst/tcp/gstsocketsrc.c:
+	* gst/tcp/gstsocketsrc.h:
+	* tests/check/pipelines/tcp.c:
+	  socketsrc: Add `connection-closed-by-peer` signal
+	  This provides notification that the socket in use was closed by the peer
+	  and gives an opportunity to replace it with a new one which is not
+	  closed, allowing reading from many sockets in order.
+	  I use this in pulsevideo to implement reconnection logic to handle the
+	  pulsevideo service dieing, such that is can be restarted without
+	  disrupting downstream.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2015-03-13 13:43:59 +0000  William Manley <will@williammanley.net>
+
+	* gst/tcp/gstsocketsrc.c:
+	  socketsrc: Tidy up usage of `g_object_unref`/`g_clear_object` and locking
+	  This is clearer, and should make future changes safer.  No functional
+	  change intended.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2015-03-13 13:30:48 +0000  William Manley <will@williammanley.net>
+
+	* gst/tcp/gstsocketsrc.c:
+	  socketsrc: Refactor to simplify
+	  * Don't bother polling, just do a blocking read, the `GCancellable` will
+	  take care of unlocking.  This should also be faster on MS Windows where
+	  the GIO documentation for `g_socket_get_available_bytes` states: "Note
+	  that on Windows, this function is rather inefficient in the UDP case".
+	  * Implement `GstPushSrc.fill` rather than `GstPushSrc.create`.  This means
+	  that we will be using the downstream allocator which may be more
+	  efficient.  It also means that socketsrc is likely to respect its
+	  "blocksize" property (assuming that there is enough data available).
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2014-11-03 02:47:14 +0000  William Manley <will@williammanley.net>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-base-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-base-plugins-sections.txt:
+	* docs/plugins/inspect/plugin-tcp.xml:
+	* gst/tcp/Makefile.am:
+	* gst/tcp/gstsocketsrc.c:
+	* gst/tcp/gstsocketsrc.h:
+	* gst/tcp/gsttcpplugin.c:
+	* tests/check/pipelines/tcp.c:
+	* win32/vs7/libgsttcp.vcproj:
+	* win32/vs8/libgsttcp.vcproj:
+	  tcp: Add element socketsrc
+	  `socketsrc` can be considered a source counterpart to `multisocketsink`.
+	  It can be considered a generalization of `tcpclientsrc` and
+	  `tcpserversrc`:  it contains all the logic required to communicate over
+	  the socket but none of the logic for creating the sockets/establishing
+	  the connection in the first place, allowing the user to accomplish this
+	  externally in whatever manner they wish making it applicable to other
+	  types of sockets besides TCP.
+	  This commit essentially copies the implementation directly from
+	  tcpserversrc.  Later patches will tidy the implementation up and
+	  re-implement `tcpclientsrc` and `tcpserversrc` in terms of `socketsrc`.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=739546
+
+2015-03-13 23:24:23 +0530  Arun Raghavan <git@arunraghavan.net>
+
+	* gst-libs/gst/audio/gstaudioringbuffer.c:
+	  audioringbuffer: Log with the ringbuffer object where possible
+
+2015-03-13 12:49:31 +0000  William Manley <will@williammanley.net>
+
+	* gst/tcp/gstmultisocketsink.c:
+	* tests/check/elements/multisocketsink.c:
+	  multisocketsink: Map `GstMemory`s individually when sending
+	  If a buffer is made up of non-contiguous `GstMemory`s `gst_buffer_map`
+	  has to copy all the data into a new `GstMemory` which is contiguous.  By
+	  mapping all the `GstMemory`s individually and then using scatter-gather
+	  IO we avoid this situation.
+	  This is a preparatory step for adding support to multisocketsink for
+	  sending file descriptors, where a GstBuffer may be made up of several
+	  `GstMemory`s, some of which are backed by a memfd or file, but I think this
+	  patch is valid and useful on its own.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=746150
+
+2015-03-13 10:30:43 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/video/video-frame.c:
+	  video-frame: Relax width/height assertion
+	  When doing CROP META it is exepcted that the width and/or height in the
+	  GstVideoMeta is bigger or equal to the caps negotiated size.
+
+2015-03-12 16:32:31 +0000  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/video/gstvideopool.c:
+	  videopool: Choose the biggest buffer size
+	  We should respect what has been negotiated.
+
+2015-03-12 10:06:15 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: recover from EOS when searching for chain in push mode
+	  If we get EOS when we're trying to build a chain, we disable seeking
+	  and continue instead of posting an error. This can happen for corner
+	  cases such as a stream with a video that stops before the end, for
+	  instance.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745980
+
+2015-03-11 16:46:38 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: fix seeking in files with a "missing" stream
+	  When looking for pages when seeking, we stop looking for non sparse
+	  streams if we don't find one within a given threshold. This fixes
+	  seeking filling up queues and blocking in corner cases such as an
+	  audio file with a pathological 1 frame video stream (yes, I saw one).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745980
+
+2015-03-13 01:06:57 +1100  Jan Schmidt <jan@centricular.com>
+
+	* docs/libs/gst-plugins-base-libs-docs.sgml:
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/gstvideometa.c:
+	* gst-libs/gst/video/video-chroma.c:
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-dither.c:
+	* gst-libs/gst/video/video-resampler.c:
+	* gst-libs/gst/video/video-resampler.h:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst/videoscale/gstvideoscale.h:
+	  docs: Add new video functions and objects. Cleanup a little.
+	  Add GstVideoChroma, GstVideoDither, GstVideoScaler and friends to the docs.
+	  Remove and clean up a few obsolete/deleted refs and typos
+
+2015-03-12 12:17:11 +0000  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Disconnect signals and invalidate group if it fails to activate
+	  Otherwise playbin might move to the group directly after EOS of the next
+	  group, and then error out again.
+
+2015-02-01 03:39:07 +1100  Jan Schmidt <jan@centricular.com>
+
+	* ext/theora/gsttheoradec.c:
+	* ext/theora/gsttheoradec.h:
+	  theoradec: Fix decoding in the presence of GstVideoCropMeta
+	  Store the video info of the internal frame decode width/height
+	  separate to the exposed (cropped) frame info, so that it can be
+	  used for mapping the downstream allocated video frame buffer correctly
+	  when using GstVideoCropMeta.
+	  Fixes playback of files with sizes that aren't a multiple of 16-pixels
+	  width or height.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741030
+
+2015-03-03 15:18:04 +0800  Song Bing <b06498@freescale.com>
+
+	* tests/check/pipelines/streamsynchronizer.c:
+	  streamsynchronizer: Should wait state change complete before start another state change
+	  Should wait state change complete before start another state change.
+	  Can't ensure can received async-done message when state change from PLAYING to PAUSED.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-02-27 16:40:23 +0800  Song Bing <b06498@freescale.com>
+
+	* gst/playback/gststreamsynchronizer.c:
+	  streamsynchronizer: Remove unnecessary ERROR message.
+	  Remove unnecessary ERROR message.
+	  Push GAP will fail as flushing. Needn't ERROR message.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-03-05 17:42:53 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	* ext/ogg/gstoggdemux.h:
+	  oggdemux: do not send seek events from the streaming thread
+	  This will usually deadlock, despite this patch being in master for
+	  quite some time and working fine. Nevertheless, we deem it to be
+	  not working, disregarding facts.
+	  As such, we fix it by keeping track of seek events, and sending
+	  them upstream from a separate thread. Buffers are then discarded
+	  till we get a new segment with the expected seqnum.
+
+2015-02-23 13:07:41 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	* ext/ogg/gstoggdemux.h:
+	  oggdemux: set correct seqnum on segment events after a seek in push mode
+	  There is already a seqnum field for this, which was used to overwrite
+	  the seqnum that was set by the push specific code.
+
+2015-02-23 11:30:36 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: try harder to query duration from upstream
+	  READY->PAUSED can be too early as souphttpsrc can get the HTTP
+	  headers after this. Try again in the chain function.
+	  Also use seeking query to disable seeking if upstream reports
+	  being unseekable.
+
+2014-10-31 10:55:14 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: add non flushing time seeking in push mode
+	  Some resetting code has to be done in the NEW_SEGMENT
+	  event handler, instead of the missing FLUSH_STOP one.
+	  Segment base was also wrongly accounted for. This was hidden
+	  by the fact that flushing resets the base.
+	  A discontinuity is now also signalled on seeking. We have to
+	  also ensure that the discontinuity "sticks" till a buffer
+	  with a valid timestamp goes out, or the audio decoder base
+	  class will ignore the discontinuity for purposes of keeping
+	  track of the current time.
+	  This allows using non flushing segment seeks for looping
+	  HTML audio in particular, and more generally non flushing seeks.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=729198
+
+2015-02-04 17:13:44 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: fix wrong first granule
+	  The code was using the first nonnegative granulepos to seed the
+	  granule tracking, which appeared to work since headers have zero
+	  granulepos. However, this does not work for files with a hole at
+	  start, which are common in live streaming.
+	  The correct behavior is to look for the first granule, and subtract
+	  the duration of all the packets finishing on this page.
+	  The function which does this relies on the fact that the ogg_stream
+	  structure can be duplicated by shallow copy, in order to pull the
+	  packets from the first page(s) on the copy without affecting the
+	  original stream state.
+
+2015-03-11 09:48:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix border handling of YUY2 and friends
+	  Don't draw the border in groups of 4 pixels for YUY2 but instead in
+	  groups of 2 with alternating U and V. This avoids a crash on odd width
+	  borders.
+
+2015-03-11 09:47:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: force yuv conversion for border
+	  Make sure we always do yuv conversion for the border.
+
+2015-03-10 17:29:51 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: fix A422 subsampling description
+
+2015-03-10 15:12:30 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add table based matrix8 implementation
+	  Based on patch from Mozzhuhin Andrey <nopscmn at gmail.com>
+	  Add a table based matrix8 multiplication implementation. The algorithm
+	  does not do any clipping so we need to make sure we never call this on
+	  input that might need to be clipped. In general, this algorithm is
+	  2 times faster than the orc optimized one and would be chosen for all
+	  RGB -> YUV conversions and some YUV->YUV and RGB->RGB conversions.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732186
+
+2015-03-10 11:55:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videotestsrc/gstvideotestsrc.c:
+	* gst/videotestsrc/gstvideotestsrc.h:
+	* gst/videotestsrc/videotestsrc.c:
+	* gst/videotestsrc/videotestsrc.h:
+	  videotestsrc: add all colors mode
+
+2015-03-10 10:19:22 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* 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:
+	  video: Add support for 10 bit planar AYUV formats
+
+2015-03-10 09:27:08 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/vorbis/gstvorbisparse.c:
+	* gst-libs/gst/rtsp/gstrtsprange.c:
+	* gst/playback/gstsubtitleoverlay.c:
+	* gst/volume/gstvolume.c:
+	* sys/xvimage/xvimagepool.c:
+	* tests/check/libs/rtpbasedepayload.c:
+	* tests/check/libs/video.c:
+	  Fix double semicolons
+
+2015-03-09 21:35:59 -0400  Olivier Crete <olivier.crete@collabora.com>
+
+	* gst/videorate/gstvideorate.c:
+	  videorate: Accept any capsfeatures
+
+2015-03-09 16:28:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-info.c:
+	  video-info: validate parsed colorimetry
+	  Validate the parsed colorimetry and reset to defaults when we get RGB
+	  with a matrix or YUV without a matrix.
+
+2015-03-09 16:01:19 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: detect identity matrix
+	  Do nothing if we have an identity matrix conversion.
+
+2015-03-09 15:58:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-info.c:
+	  video-info: use default colorimetry on error
+	  When we fail to parse the colorimetry property, fall back to the default
+	  colorimetry for the format and dimension instead of leaving things
+	  undefined.
+
+2015-03-09 11:25:41 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: unused value
+	  Value set in ret is immediately overwritten in the next line outside of the if
+	  block. Run reset but don't store return.
+	  CID #1226470
+
+2015-03-09 12:13:44 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: only convert to/from rgb when needed
+	  Only use the YUV->RGB matrix when we have YUV as input and only use the
+	  matrix when we need to make YUV output.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745780
+
+2015-03-09 11:12:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/rtp/gstrtpbuffer.c:
+	  rtpbuffer: Link to an explanation why the seqnum comparison function does the right thing even for wraparounds
+
+2015-02-22 21:13:35 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: only return EOS upon clipping if applicable
+	  See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-02-22 21:11:50 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: only return EOS upon clipping if applicable
+	  See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-03-07 16:49:07 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: Update orc generated C files
+
+2015-03-06 12:54:56 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add transfer full annotation for config
+
+2015-03-06 09:30:51 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: correct right-border location for YUY2, YVYU, UYVY
+	  Remove 'r_border /= 2' in convert_fill_border(). It doesn't
+	  take the right border to correct location.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745719
+
+2015-03-05 12:31:06 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/volume/gstvolume.c:
+	  volume: Explicitly cast integers to doubles and then back to integers after multiplication
+	  gcc 4.9.1 on ARM seems to have a bug that causes it to cast the float to an
+	  integer first, resulting in a 0 scale factor for volume < 1.0.
+	  As a side effect this change here will also improve accuracy of the result a
+	  bit because we go via doubles instead of floats.
+	  https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65325
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745667
+
+2015-03-05 09:52:18 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: avoid scaler when size is unchanged
+
+2015-03-04 16:45:35 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add horizontal 2tap u16 orc function
+	  Add slightly faster u16 horizontal resampler orc function.
+
+2015-03-04 12:28:47 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/libs/video.c:
+	  check: add another generic converter test
+	  Run conversion and scaling with borders.
+
+2015-03-04 12:21:33 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* tests/check/libs/video.c:
+	  video-converter: don't reuse the input line when adding borders
+	  When we need to add borders, we need a writable input line, so
+	  don't reuse the source memory directly.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745207
+
+2015-03-03 16:36:20 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  textoverlay: Re-render if video size changed
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745554
+
+2015-03-03 22:56:37 +0530  Arun Raghavan <arun@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	* gst-libs/gst/audio/gstaudiosink.c:
+	  audiobasesink: Reset audio clock if necessary
+	  When the ringbuffer is deactivated and then acquired, if the audio clock
+	  provided by the sink gets reset to zero, we need to add an offset to the
+	  clock to make sure that subsequent samples are written out at the right
+	  times. While we need to leave this to derived classes to take care of
+	  when they provide their own clock (since that clock may or may not be
+	  reset to zero), we can do this ourselves if we know the provided clock
+	  is our own (which does reset to zero on a re-acquire).
+
+2015-03-02 16:42:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: avoid making scalers for outsize == 0
+
+2015-03-02 16:33:09 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-converter: v-resample enough pixels
+	  When we are using the fast linear resampler, use the ->inc to calculate
+	  the first and last pixel we need so that we can do vertical resampling
+	  on the right amount of pixels.
+
+2015-03-02 15:07:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: fix unpack functions for RGB/RGB15 on BE
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745337
+
+2015-03-02 13:27:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-format: more fixes for big endian
+
+2015-03-02 12:26:23 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-format: add big-endian versions of RGB/BGR 15/16 pack/unpack
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745337
+
+2015-02-28 13:31:41 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play.c:
+	  gst-play: fix compiler warning
+	  ‘return’ with no value, in function returning non-void
+
+2015-02-28 12:26:21 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play-1.0.1:
+	* tools/gst-play.c:
+	  gst-play: add keyboard shortcut to cycle through trick modes
+	  Make "t" activate trick modes and cycle through the various
+	  modes.
+
+2015-02-28 11:37:27 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play.c:
+	  gst-play: fix indentation
+	  Prevent gst-indent from messing up indentation, it
+	  really doesn't like the G_GNUC_PRINTF thing here.
+
+2015-02-27 20:22:59 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/audiodecoder.c:
+	* tests/check/libs/audioencoder.c:
+	* tests/check/libs/videodecoder.c:
+	* tests/check/libs/videoencoder.c:
+	  tests: fix crashes in {audio,video}{decoder,encoder} tests on 32-bit
+	  Don't feed 64-bit integer variable into vararg function that expects
+	  an unsigned integer to go with GST_TAG_TRACK_NUMBER. This would
+	  cause crashes on 32-bit platforms, and if not that then test
+	  failures if the comparisons fail later (at least on big endian
+	  platforms).
+
+2015-02-27 15:07:36 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	  pbutils: description: Make static strings static
+	  Otherwise, they're not guaranteed to still be valid when leaving the scope.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-27 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/pbutils.c:
+	  tests: pbutils: more checking of returned description strings
+	  https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-27 00:36:43 +0530  Arun Raghavan <arun@accosted.net>
+
+	* gst/adder/gstadder.c:
+	  adder: Drop custom latency querying logic
+	  The default latency query handler now implements the same logic already.
+
+2015-02-26 14:47:28 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: remove check for below zero for unsigned int
+	  CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
+	  number since it in an unsigned integer. Removing that check and only checking
+	  if it is bigger than max and setting it appropriately.
+	  CID #1271606
+
+2015-02-26 12:06:23 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  playback: Fix broken GList modification
+	  When we modify a GList (via g_list_delete_link), always reassign the
+	  new head to the original GList. Otherwise we end up with
+	  filtered_errors being corrupt (the head might have been the element
+	  removed)
+
+2015-02-26 11:06:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play-1.0.1:
+	  gst-play: add new keyboard shortcuts to man page
+
+2015-02-26 10:57:56 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play.c:
+	  gst-play: more fine-grained playback rate control
+	  Use smaller steps for lower rates to allow more
+	  fine-grained control. Handle jump across 0 properly
+	  from both sides (just flip direction where we would
+	  have gone down to 0 instead). Don't artificially
+	  limit rates to +/- 10x. Print new rate.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745174
+
+2015-02-26 10:20:20 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play.c:
+	  gst-play: stash current playback rate in app structure
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745174
+
+2015-02-25 18:52:11 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* tools/gst-play.c:
+	  gst-play: support changing the playback rate in interactive mode
+	  It is fun to have this feature, also it is useful for testing decoders.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745174
+
+2015-02-25 17:00:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: we can use the scaler without scalers to copy
+
+2015-02-25 16:50:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: only make a scaler when we are scaling
+	  Only make a scaler when we are actually doing any scaling. Without
+	  scalers, the scale function will simply do a copy.
+
+2015-02-25 16:49:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add support for copy
+	  When no scalers are given, simply do a copy of the requested area.
+
+2015-02-25 16:15:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: activate scaler fastpath depending on method
+	  Only activate the scaler fastpath for x2 up and downscale when the
+	  scaler method is respectively nearest and linear because that is what
+	  those fastpaths really implement.
+
+2015-02-25 15:33:26 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add scaler optimization
+	  If we are vertically downscaling, it is better to first downscale and
+	  then do the horizontal scaling in most cases.
+
+2015-02-25 15:32:57 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: remove unused case
+
+2015-02-25 11:38:17 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: don't overwrite border alpha
+	  Let border alpha and image alpha be independent.
+
+2015-02-24 17:33:57 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: use 1.0 as default alpha
+
+2015-02-24 17:26:31 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: add alpha handling
+	  Add support for alpha. Make it possible to copy, set and multiply the
+	  alpha value of a frame during conversion.
+	  Set the border alpha to 0xff by default.
+	  Go over some of the fastpaths and add alpha handling.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=745006
+
+2015-02-24 17:20:53 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix chroma subsampling
+	  Also adjust the output line number with the offset.
+
+2015-02-24 10:01:18 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: disable fastpath when scaling and gamma
+	  Disable the fastpath when scaling and doing gamma remap.
+
+2015-02-24 09:54:18 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: don't do gamma on alpha channel
+	  The alpha channel is not supposed to be gamma encoded.
+
+2015-02-24 16:06:08 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: fix deadlock when resetting buffering
+	  This function is static, and only ever called with the expose lock
+	  taken. It thus has no reason to take this lock itself.
+	  This was introduced by one of my locking fixes from 741355.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-24 12:38:10 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: minor docs fix
+
+2014-05-27 13:54:06 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: forward template and ring buffer settings to existing decodebins
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744844
+
+2015-02-23 17:24:52 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: move null check
+	  Check if dbin->decode_chain is NULL before running drain_and_switch_chains()
+	  because if it is, we shouldn't run that function or it will segfault.
+	  CID #1271074
+
+2015-02-23 01:32:14 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Don't send pending events before decode
+	  Make sure to update the output segment to track the segment
+	  we're decoding in, but don't actually push it downstream until
+	  after buffers are decoded.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744806
+
+2015-02-08 05:19:25 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideodecoder.h:
+	  videodecoder: Add drain() vfunc
+	  drain() is a new vfunc which does what finish() does, while
+	  explicitly requiring the decoder be able to continue processing
+	  data afterward.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2015-02-22 16:57:57 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  Revert "videodecoder: drain current segment upon new one to ensure correct flow return"
+	  This reverts commit cc1b4eaf9ebe4568f9c2c64338cef1b2edbdca3f.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2015-02-22 16:57:50 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  Revert "audiodecoder: drain current segment upon new one to ensure correct flow return"
+	  This reverts commit 696b8cdc40f033ff0a45ebe620279130152fb2f8.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2015-02-21 17:42:08 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: drain current segment upon new one to ensure correct flow return
+	  See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-02-21 17:41:50 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: drain current segment upon new one to ensure correct flow return
+	  See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
+
+2015-02-20 12:34:11 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Only consider non-parser factories for generating the post-parser capsfilter caps
+	  Otherwise if there are multiple parsers we would most likely break negotiation
+	  of the stream-format/alignment wanted by the decoders as parsers generally
+	  support all possible stream-formats and alignments.
+
+2015-02-19 15:51:19 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  audio: video: fix a few GI annotations
+	  transfer-full -> transfer full
+	  @Since -> Since
+
+2015-02-05 12:07:50 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: fix deadlock between downward state change and pad addition
+	  If caps on a newly added pad are NULL, analyze_new_pad will try to
+	  acquire the chain lock to add a probe to the pad so the chain can
+	  be built later. This comes from the streaming thread, in response
+	  to headers or other buffers causing this pad to be added, so the
+	  stream lock is taken.
+	  Meanwhile, another thread might be destroying the chain from a
+	  downward state change. This will cause the chain to be freed with
+	  the chain lock taken, and some elements are set to NULL here, which
+	  can include the parser. This causes pad deactivation, which tries
+	  to take the element's pad's stream lock, deadlocking.
+	  Fix this by keeping track of which elements need setting to NULL,
+	  and only do this after the chain lock is released. Only the chain
+	  manipulation needs to be locked, not the elements' state changes.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-04 11:46:09 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: guard against the decode chain going while a pad is added
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-03 17:06:43 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: possible fix for deadlock when spamming "next song"
+	  There was a deadlock between a thread changing decodebin/demuxer
+	  state from PAUSED to READY, and another thread pushing data
+	  when starting.
+	  From the stack trace at
+	  https://bug741355.bugzilla-attachments.gnome.org/attachment.cgi?id=292471,
+	  I deduce the following is happening, though I did not reproduce the
+	  problem so I'm not sure this patch fixes it.
+	  The streaming thread (thread 2 in that stack trace) takes the demuxer's
+	  sink pad's stream lock in gst_ogg_demux_perform_seek_pull and will
+	  activate a new chain. This ends up causing the expose lock being taken
+	  in _pad_added_cb in decodebin.
+	  Meanwhile, a state changed is triggered on thread 1, which takes the
+	  expose lock in decodebin in gst_decode_bin_change_state, then frees
+	  the previous chain, which ends up calling gst_pad_stop_task on the
+	  demuxer's task, which in turn takes the demuxer's sink pad's stream
+	  lock, deadlocking as both threads are now waiting for each other.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741355
+
+2015-02-18 20:58:15 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/tag/gsttagdemux.c:
+	  tagdemux: ensure tags have been fetched before pulling data
+	  Otherwise upstream can get confused about offsets as there will
+	  be a jump once the tags have been parsed due to the stripped area.
+	  If upstream pulls from 0 to 100, and then tagdemux does the
+	  tag reading and finds out that the first 200 bytes are the tag, the
+	  next pull from upstream will have an offset of 200 bytes. So
+	  upstream will get the following data:
+	  0 - 100, 300 - (EOS), as it will continue requesting from where
+	  it has last stopped, but tagdemux will add an offset to skip the
+	  tags.
+	  This patch makes sure that the tags have been parsed and skipped
+	  since the first pull range call.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744580
+
+2015-02-19 01:30:05 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Reset the default query return value when the iterator has to resync
+
+2015-02-19 01:21:47 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Let the latency query fail if one of the source queries fails
+
+2015-02-18 11:34:15 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	  pbutils: description: fix MPEG-2 video profiles in description
+	  We would accidentally use the profile nick as profile name
+	  in the description for MPEG video that's not version 4.
+
+2015-01-29 18:49:45 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Pass object, not GValue to debug print
+
+2015-02-16 23:54:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  audiovisualizer: don't use private GMutex implementation details
+	  Don't use private GMutex implementation details to check
+	  whether it has been freed already or not. Just turn dispose
+	  function into finalize function which will only be called
+	  once, that way we can just clear the mutex unconditionally.
+
+2015-02-15 13:51:36 +0800  Song Bing <b06498@freescale.com>
+
+	* gst/playback/gststreamsynchronizer.c:
+	  streamsynchronizer: Use the same waiting function for EOS and stream switches
+	  Also improve the waiting condition for stream switches, which was assuming
+	  before that the condition variable will only stop waiting once when it is
+	  signaled. But the documentation says that there might be spurious wakeups.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-01-26 11:14:13 +0800  Song Bing <b06498@freescale.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/pipelines/streamsynchronizer.c:
+	  streamsynchronizer: Unit test for streamsynchronizer's EOS handling
+	  Test that a pipeline can change from PLAYING to PAUSED and back in
+	  the following scenarios:
+	  1. One track reach EOS after pushed some buffers while another track
+	  still pushes buffers
+	  2. One track reach EOS without buffers while another track still pushes
+	  buffers
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-01-12 17:40:25 +0800  Song Bing <b06498@freescale.com>
+
+	* gst/playback/gststreamsynchronizer.c:
+	  streamsynchronizer: Send GAP events from the pads' streaming threads
+	  Change the GAP events that are currently sent from the chain function of
+	  the current pad to all other EOS pads. They should instead be sent from
+	  their own streaming threads.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-01-12 16:08:33 +0800  Song Bing <b06498@freescale.com>
+
+	* gst/playback/gststreamsynchronizer.c:
+	* gst/playback/gststreamsynchronizer.h:
+	  streamsynchronizer: Send GAP event to finish preroll when change state from PLAYING to PAUSED
+	  Wait in the event function when EOS is received until all pads are EOS
+	  and then forward the EOS event from each pads own event function.
+	  Also send a new GAP event for EOS pads from the event function whenever
+	  going from PLAYING->PAUSED by shortly waking up the GCond. This is needed
+	  to allow sinks to pre-roll again, as they did not receive EOS yet because
+	  we blocked that, but also will never get data again.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736655
+
+2015-02-16 09:48:03 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/pbutils/codec-utils.c:
+	  Revert "codec-utils: Handle the two rext profiles for h265"
+	  This reverts commit 19b93566801a56e7b043a670b7edcf8f2da06619.
+	  These two "profiles" are actually a complete set of profiles, which we will
+	  need to handle separately. Unfortunately it seems like we need information
+	  from the SPS to detect the exact profile.
+
+2015-02-15 20:08:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	  pbutils: description: move some code into utility function
+
+2015-02-15 20:05:13 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	* tests/check/libs/pbutils.c:
+	  pbutils: descriptions: add H.265 profile to description if available
+	  https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-15 19:03:38 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	* tests/check/libs/pbutils.c:
+	  pbutils: descriptions: add MPEG-4 video profile to description if available
+	  https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-15 18:37:38 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	* tests/check/libs/pbutils.c:
+	  pbutils: descriptions: add Dirac/VC-2 profile to description if available
+	  https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-15 18:14:18 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	* tests/check/libs/pbutils.c:
+	  pbutils: descriptions: add H.264 profile to description if available
+	  https://bugzilla.gnome.org/show_bug.cgi?id=673976
+
+2015-02-13 22:56:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/install-plugins.c:
+	  install-plugins: fix indentation and add Since marker
+	  Forgot to squash this into the actual patch before pushing.
+
+2015-02-13 22:49:04 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* win32/common/libgstpbutils.def:
+	  install-plugins: add new API to exports .def and to docs
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744465
+
+2015-02-03 10:47:11 +0100  Kalev Lember <kalevlember@gmail.com>
+
+	* gst-libs/gst/pbutils/install-plugins.c:
+	* gst-libs/gst/pbutils/install-plugins.h:
+	  install-plugins: Add API to suppress confirmation before searching
+	  The new gst_install_plugins_context_set_confirm_search() API can be used
+	  to pass a hint to modify the behaviour of the external installer
+	  process.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744465
+
+2015-02-02 16:16:46 +0100  Kalev Lember <kalevlember@gmail.com>
+
+	* gst-libs/gst/pbutils/install-plugins.c:
+	* gst-libs/gst/pbutils/install-plugins.h:
+	  install-plugins: Add API for passing desktop ID and startup ID
+	  The new gst_install_plugins_context_set_desktop_id() and
+	  gst_install_plugins_context_set_startup_notification_id() API can be
+	  used to pass extra details to the external installer process.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744465
+
+2015-02-12 12:08:16 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video-orc: update with new methods
+
+2015-02-12 11:38:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-format: add orc function for RGB15/16 unpack
+
+2015-02-10 21:57:02 -0800  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: improve debug log
+	  Log the human readable pad_link_return desc as well.
+
+2015-02-11 15:57:54 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/pbutils/codec-utils.c:
+	  codec-utils: Handle the two rext profiles for h265
+	  These values are for now taken from x265 and need to be checked against
+	  the spec. Especially we need to check if information from other fields
+	  need to be taken into consideration too, e.g. the bit depth and chroma
+	  index from the SPS.
+	  This however makes 4:4:4 output of x265enc actually work.
+
+2015-02-11 13:43:11 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/app/gstappsrc.c:
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	* gst/adder/gstadder.c:
+	* gst/playback/gsturidecodebin.c:
+	  Improve and fix LATENCY query handling
+	  This now follows the design docs everywhere, especially the maximum latency
+	  handling.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=744106
+
+2015-02-11 13:32:25 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	* win32/common/libgstvideo.def:
+	  video-scaler: add 2d scaler
+	  Make a convenience function that combines 2 scalers to perform a 2d
+	  scale. This removes quite a bit of overhead in method calls when doing a
+	  typical scale and it also can reuse a piece of unused memory in the
+	  vertical scaler.
+	  Use the 2d scaler in video-converter and remove the other scalers and
+	  temp memory.
+
+2015-02-10 16:43:03 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Fix YUY2 formats and friends
+	  Only merge scalers for selected formats.
+	  Use nearest neighbour scaling for chroma when doing nearest neighbour
+	  for the luma.
+	  Also fastpath GRAY16_OE in nearest neighbour.
+	  configure parameters correctly for packed fastpath.
+
+2015-02-10 16:40:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: Small performance tweaks
+	  Small performance tweaks for RGB and friends.
+	  Add, but ifdef out, alternative nearest neighbour scaling, it is slower
+	  than the current table based version.
+	  Use memcpy instead of orc_memcpy because it is measurably faster.
+	  Fix YUY2 and friends vertical scaling.
+
+2015-02-10 16:44:38 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: Guard against (impossible) bits!=16 && bits!=8 case to fix compiler warning with clang
+	  video-scaler.c:1331:14: error: variable 'func' is used uninitialized whenever 'if' condition is false
+	  [-Werror,-Wsometimes-uninitialized]
+	  } else if (bits == 16) {
+	  ^~~~~~~~~~
+	  video-scaler.c:1348:3: note: uninitialized use occurs here
+	  func (scale, src_lines, dest, dest_offset, width, n_elems);
+	  ^~~~
+	  video-scaler.c:1331:10: note: remove the 'if' if its condition is always true
+	  } else if (bits == 16) {
+	  ^~~~~~~~~~~~~~~~
+	  video-scaler.c:1260:27: note: initialize the variable 'func' to silence this warning
+	  GstVideoScalerVFunc func;
+	  ^
+	  = NULL
+
+2015-02-10 16:38:05 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Use correct enum type to fix compiler warnings with clang
+	  video-converter.c:3406:12: error: implicit conversion from enumeration type 'GstVideoFormat' to different
+	  enumeration type 'GstFormat' [-Werror,-Wenum-conversion]
+	  format = convert->fformat[plane];
+	  ~ ^~~~~~~~~~~~~~~~~~~~~~~
+	  video-converter.c:3413:44: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+	  type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+	  gst_video_scaler_horizontal (h_scaler, format,
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~            ^~~~~~
+	  video-converter.c:3471:12: error: implicit conversion from enumeration type 'GstVideoFormat' to different
+	  enumeration type 'GstFormat' [-Werror,-Wenum-conversion]
+	  format = convert->fformat[plane];
+	  ~ ^~~~~~~~~~~~~~~~~~~~~~~
+	  video-converter.c:3487:42: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+	  type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+	  gst_video_scaler_vertical (v_scaler, format, lines, d + out_x, i,
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~            ^~~~~~
+	  video-converter.c:3551:12: error: implicit conversion from enumeration type 'GstVideoFormat' to different
+	  enumeration type 'GstFormat' [-Werror,-Wenum-conversion]
+	  format = convert->fformat[plane];
+	  ~ ^~~~~~~~~~~~~~~~~~~~~~~
+	  video-converter.c:3569:46: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+	  type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+	  gst_video_scaler_horizontal (h_scaler, format,
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~            ^~~~~~
+	  video-converter.c:3577:42: error: implicit conversion from enumeration type 'GstFormat' to different enumeration
+	  type 'GstVideoFormat' [-Werror,-Wenum-conversion]
+	  gst_video_scaler_vertical (v_scaler, format, lines, d + out_x, i,
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~            ^~~~~~
+
+2015-02-10 15:25:04 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-converter: bits variable always set
+	  In function gst_video_scaler_vertical() the bits variable is always
+	  set to either 8 or 16 in every possible format. No need to initialize it.
+	  If the format isn't valid it goes to no_func, so there is no need to
+	  handle the case of bits not being 8 or 16.
+	  CID #1268401
+
+2015-02-10 11:15:22 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: only enable backlog for interlaced video
+	  Skip lines we don't need.
+
+2015-02-10 09:30:44 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add fastpath for NV formats
+
+2015-02-10 09:20:12 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: fix pstride of NV16 and NV24 formats
+
+2015-02-09 18:01:30 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/rtsp/gstrtspmessage.c:
+	* tests/check/libs/rtsp.c:
+	  rtspmessage: map headers we know that are added by string to their enum
+	  That way we can look them up by their field enum later as well.
+
+2015-02-09 17:49:12 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/rtsp.c:
+	  tests: rtsp: add some unit tests for new GstRTSPMessage API
+
+2015-02-09 16:24:19 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/rtsp/gstrtspmessage.c:
+	* gst-libs/gst/rtsp/gstrtspmessage.h:
+	* win32/common/libgstrtsp.def:
+	  rtspmessage: add API to add and get custom headers
+	  Add API to add and get custom headers that are not
+	  covered by our header fields enum. This is backwards
+	  compatible in that it will also work for our defined
+	  fields, so if we ever add a new header field to the
+	  enum, get_header_by_name() for the same header string
+	  will still work.
+	  API: gst_rtsp_message_add_header_by_name()
+	  API: gst_rtsp_message_take_header_by_name()
+	  API: gst_rtsp_message_remove_header_by_name()
+	  API: gst_rtsp_message_get_header_by_name()
+
+2015-02-09 17:51:00 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: Add more fastpaths
+	  Add fastpaths for all planar conversion and scaling.
+	  Improve gray and alpha handling.
+	  Add option to specify the chroma resampler method and set to linear as
+	  default.
+
+2015-02-09 13:20:43 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add generic planar scaler/converter
+	  Add code to convert and scale between any planar format and use it in
+	  the fastpaths of some planare converters.
+
+2015-02-09 10:20:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Fix compiler warnings by using the correct enum type
+	  video-converter.c:3645:24: error: implicit conversion from enumeration type
+	  'GstFormat' to different enumeration type 'GstVideoFormat'
+	  [-Werror,-Wenum-conversion]
+	  convert->fformat = fformat;
+	  ~ ^~~~~~~
+	  video-converter.c:3667:24: error: implicit conversion from enumeration type
+	  'GstFormat' to different enumeration type 'GstVideoFormat'
+	  [-Werror,-Wenum-conversion]
+	  convert->fformat = fformat;
+	  ~ ^~~~~~~
+	  video-converter.c:3963:50: error: implicit conversion from enumeration type
+	  'const GstVideoFormat' to different enumeration type 'GstFormat'
+	  [-Werror,-Wenum-conversion]
+	  if (!setup_scale (convert, transforms[i].fformat))
+	  ~~~~~~~~~~~           ~~~~~~~~~~~~~~^~~~~~~
+
+2015-02-07 03:56:05 +1100  Jan Schmidt <jan@centricular.com>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: Don't pass GstCollectData as a GstObject to GST_DEBUG
+
+2015-02-06 13:39:04 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-converter: add more scaler fastpaths
+
+2015-02-06 13:25:51 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: fix loading of param
+	  param loading ignores the x4, loading only part of the param.
+
+2015-02-06 12:35:01 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add border and crop to more fastpaths
+
+2015-02-06 12:28:54 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix border for YUY2 and friends
+	  Convert as many pixels as the max subsampling so that we convert a
+	  complete group of pixels.
+
+2015-02-06 15:39:14 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: support AYUV border
+	  Convert the border color from ARGB to AYUV, using
+	  colorimetry matrix when output format is YUV.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741640
+
+2015-02-06 10:57:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix swapped border width
+	  And also do nothing when there is no border.
+
+2015-02-06 10:56:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: actually draw the border in some fastpaths
+	  Don't forget to draw the border after doing the fastpath conversion.
+
+2015-02-06 10:53:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: clamp width and heigth
+	  Clamp the width and height based on the in and out offsets.
+
+2015-02-06 10:50:09 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: add unaligned fallbacks
+	  Add fallback C implementations for when we can't call the ORC function
+	  because of bad alignment.
+
+2015-01-28 05:20:19 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Where possible, skip decode for GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO
+	  If we have timestamps on input buffers and are in trickmode no-audio
+	  mode, then don't pass anything to the subclass for decode and simply
+	  send gap events downstream
+	  Only for forward playback for now - reverse requires accumulating
+	  GAP events and pushing out in reverse order.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-02-05 17:44:59 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: Re-work GAP buffer and trick-mode handling
+	  In trickmode no-audio mode, or when receiving a GAP buffer,
+	  discard the contents and render as a GAP event instead.
+	  Make sure when rendering a gap event that the ring buffer will
+	  restart on PAUSED->PLAYING by setting the eos_rendering flag.
+	  This mostly reverts commit 8557ee and replaces it. The problem
+	  with the previous approach is that it hangs in wait_preroll()
+	  on a PLAYING-PAUSED transition because it doesn't commit state
+	  properly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-02-03 20:38:44 +1100  Jan Schmidt <jan@centricular.com>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: Add a little timestamping debug output
+
+2015-02-03 01:19:05 +1100  Jan Schmidt <jan@centricular.com>
+
+	* ext/theora/gsttheoradec.c:
+	  theora: If no header packets in stream, look for them in the caps
+	  Makes theora work in cases where the header packets are only in the caps
+	  (because theoradec was connected to oggdemux late and missed the
+	  beginning of the stream)
+
+2015-02-02 22:23:51 +1100  Jan Schmidt <jan@centricular.com>
+
+	* ext/theora/gsttheoradec.c:
+	  theora: Remove FIXME and return GST_CUSTOM_FLOW_DROP for header packet handling
+	  This FIXME is easily fixed :)
+
+2015-01-31 05:12:10 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Remove pointless else{} around some code
+
+2015-01-31 05:09:46 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Fix reverse playback when there's only one gather set.
+	  The decoder can fail to drain on EOS if there was only one gather
+	  set, because it will never have sent the segment event downstream
+	  and set the output segment, and fail to detect that the rate < 0.0
+	  Make sure to send pending events before sending all the gather data
+	  for decode.
+
+2014-10-09 03:31:58 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/video/video-frame.h:
+	  video: Fix simple typo in GstVideoFrameMapFlags docs
+
+2015-02-05 17:49:55 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add crop and border to some fastpaths
+
+2015-02-05 17:18:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: add support for borders in scale fastpath
+	  Add support for borders and cropping in the scaler fastpaths.
+
+2015-02-05 15:03:24 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: disable fastpath for crop and border
+	  Add crop and border properties to the fastpath table and only select
+	  fastpath functions when it can handle the cropping or borders.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=744028
+
+2015-02-04 18:01:51 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-converter: add fastpath for some gray formats
+
+2015-02-04 17:44:31 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-converter: add fastpath for some more RGB formats
+	  Add fastpath for RGB and BGR.
+	  Add fastpath for nearest resampling for RGB15 and RGB16 formats.
+
+2015-02-04 16:37:22 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: skip lines we don't need
+	  Make sure to skip unused lines instead of doing a useless horizontal
+	  resampling.
+
+2015-02-04 12:08:21 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	  videoscale: fix memory leak
+	  In gst_video_scale_fixate_caps () it can goto done without freeing the memory
+	  of the tmp GstStructure. This makes it go out of scope and leak.
+	  CID #1265766
+
+2015-02-04 11:25:54 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	  video-resampler: make sure params.envelope is initialized
+	  In gst_video_resampler_init () if method is GST_VIDEO_RESAMPLER_METHOD_NEAREST
+	  then params.envelope is not initialized but still used later in line 382.
+	  Make sure this variable is initiliazed to avoid undefined behaviour.
+	  CID #1256568
+
+2015-02-03 12:23:06 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  video{enc,dec}oder: Don't reset latency all the time and handle max=GST_CLOCK_TIME_NONE correctly
+	  max=NONE means that *this* element has no maximum latency. If upstream had a
+	  maximum latency we must not override it with NONE.
+
+2015-02-03 12:15:25 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	  audio{enc,dec}oder: Always directly post latency messages on the bus when the subclass sets the latency
+	  Instead of doing it only in setcaps for the encoder, and never at all for the
+	  decoder.
+
+2015-02-03 12:12:18 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	  audio{enc,dec}oder: Handle max_latency == GST_CLOCK_TIME_NONE
+	  And initialize the latencies with 0 and NONE.
+
+2015-01-28 05:26:06 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: Don't render a GAP silence buffer
+	  Don't render out silence samples to a buffer, just
+	  start the clock running, since any buffer with the
+	  GAP flag will be discarded in render() now anyway.
+
+2015-01-28 22:42:17 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: Make sure the ringbuffer is started before waiting
+	  Don't call the basesink wait_event implementation until we're sure
+	  the ringbuffer is running, because it might wait on a non-running
+	  clock.
+
+2015-01-27 02:04:22 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: drop GAP buffers, or all buffers in trickmode no-audio mode
+	  Make the base audio sink throw away buffers marked GAP, or all
+	  incoming buffers when performing a trick play with
+	  GST_SEGMENT_TRICKMODE_NO_AUDIO flag set, and make sure to start
+	  the ringbuffer when that happens so the clock starts running.
+	  Preserve the timing calculations when rendering, so state is all
+	  updated the same, but just don't render samples.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-01-29 17:58:27 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: do not throw a flow error on flushing
+	  If the streaming task attempts to read a chain while the pipeline
+	  is stopping (which can happen if the pipeline stops shortly after
+	  start or a new URI being setup in gapless playback case), it will
+	  see a flushing return from upstream, and should then also return
+	  flushing to the caller, rather than emit a flow error.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722442
+
+2015-01-28 17:44:57 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Fix compiler warnings
+	  video-converter.c:3073:48: error: implicit conversion from enumeration type 'GstFormat' to different enumeration type 'GstVideoFormat'
+	  [-Werror,-Wenum-conversion]
+	  gst_video_scaler_horizontal (h_scaler, format,
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~~~            ^~~~~~
+	  video-converter.c:3081:44: error: implicit conversion from enumeration type 'GstFormat' to different enumeration type 'GstVideoFormat'
+	  [-Werror,-Wenum-conversion]
+	  gst_video_scaler_vertical (v_scaler, format, lines, d, i, out_w);
+	  ~~~~~~~~~~~~~~~~~~~~~~~~~            ^~~~~~
+	  video-converter.c:3137:24: error: implicit conversion from enumeration type 'const GstVideoFormat' to different enumeration type 'GstFormat'
+	  [-Werror,-Wenum-conversion]
+	  convert->fformat = GST_VIDEO_INFO_FORMAT (in_info);
+	  ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+	  ../../../gst-libs/gst/video/video-info.h:125:43: note: expanded from macro 'GST_VIDEO_INFO_FORMAT'
+	  ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+	  ../../../gst-libs/gst/video/video-format.h:361:59: note: expanded from macro 'GST_VIDEO_FORMAT_INFO_FORMAT'
+	  ~~~~~~~~^~~~~~
+	  video-converter.c:3157:24: error: implicit conversion from enumeration type 'GstVideoFormat' to different enumeration type 'GstFormat'
+	  [-Werror,-Wenum-conversion]
+	  convert->fformat = GST_VIDEO_FORMAT_GRAY8;
+
+2015-01-28 17:43:59 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: Update orc files
+
+2015-01-28 17:37:35 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* win32/common/libgstvideo.def:
+	  defs: update
+
+2015-01-28 17:32:12 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	  video-converter: add fast-path scaler for some packed YUV formats
+	  Add fast path scaling for YUY2 and other packed YUV formats. Add a new
+	  method to merge the scalers of the Y and UV components into one scaler.
+	  Add faster horizontal 2tap scaler.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=741987
+
+2015-01-28 17:30:53 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	  videoscale: don't do dithering
+
+2015-01-28 17:30:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: the default is BAYER dithering
+
+2015-01-28 17:29:45 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: don't do dither when set to NONE
+
+2015-01-28 11:38:16 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix taps calculation for pstride == 1
+	  Take pstride into consideration when calculating the scaler taps.
+
+2015-01-28 04:51:25 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: Make sure the ringbuffer really starts when we need it to
+	  Some audio sink sub-classes (pulsesink) don't start their clock
+	  when the ringbuffer starts, but always have to on EOS. When we
+	  explicitly need to start the ringbuffer, make sure sub-classes will
+	  do it by (ab)using the existing eos_rendering flag.
+
+2014-12-11 01:54:07 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/examples/playback/playback-test.c:
+	  playback-test: Support new skip seek flags
+	  Support the new SEEK_TRICKMODE_KEY_UNITS and SEEK_TRICKMODE_NO_AUDIO
+	  flags added to core
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735666
+
+2015-01-27 13:39:14 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/audio/gstaudiopack-dist.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst/adder/gstadderorc-dist.c:
+	* gst/audioconvert/gstaudioconvertorc-dist.c:
+	* gst/videotestsrc/gstvideotestsrcorc-dist.c:
+	* gst/volume/gstvolumeorc-dist.c:
+	  orc: update orc files
+
+2015-01-27 10:28:35 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add fastpath for planar scaling
+	  Add fastpaths for scaling of planar subsampled formats.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=741987
+
+2015-01-27 10:04:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add support for monochroma formats
+	  Add support for scaling of images with pstride == 1. This can be used
+	  to scale individual planes later.
+	  Rework some of the scaling code to take the pstride as a parameter.
+
+2015-01-27 09:51:47 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	  videoscale: disable chroma and matrix operations
+	  Ignore chroma subsampling and color matrix transformations like the
+	  old videoscale used to do. This is to make the performance like it was
+	  before.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=741987
+
+2015-01-26 12:52:40 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: fix GBR unpack
+
+2015-01-27 01:31:50 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudiodecoder.h:
+	  audiodecoder: Fix typo in documentation
+	  Fix a couple of harmless warnings in the gtk-doc parsing
+
+2015-01-23 12:46:41 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst-libs/gst/video/video-dither.c:
+	  video: Fix leaked dither object in error cases
+	  Coverity CID : 1256564
+
+2015-01-21 15:22:15 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  visual: fix caps leak
+	  Fix leak of caps event and of caps objects when setting caps on sink and src
+	  pads. Sync audiovisualizer class implementation to the one in gst-plugins-bad.
+	  This commit matches c5ef1bee7318f057aa1f542d5a1474b75e85131a in that module.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-01-21 14:46:15 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  visual: post QoS messages when dropping frames due to QoS
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-01-21 09:49:47 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/cdparanoia/gstcdparanoiasrc.h:
+	* gst-libs/gst/video/video-format.c:
+	* gst/audioconvert/audioconvert.c:
+	* gst/audioconvert/gstaudioquantize.c:
+	* gst/audioresample/gstaudioresample.c:
+	* gst/audioresample/resample.c:
+	  Constify some static arrays everywhere
+
+2015-01-21 09:42:21 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/alsa/gstalsa.c:
+	  alsa: Constify channel position table
+
+2015-01-21 09:41:23 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/alsa/gstalsa.c:
+	  alsa: Fix indention
+
+2015-01-21 08:33:57 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* ext/alsa/gstalsa.c:
+	  alsa: Allow to use 8 bit samples with ALSA
+	  8 bit samples have no (0) as endianness, not the native endianness.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739446
+
+2015-01-21 09:39:30 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/audio-format.c:
+	  audio-format: Constify the audio format table
+
+2015-01-21 09:37:30 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiosrc.c:
+	  audiosrc: Fill in the correct silence
+	  For unsigned raw formats this is not all zeroes, and for non-raw formats
+	  we just continue to assume all zeroes for now.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739446
+
+2015-01-21 08:47:26 +0100  Thomas Roos <thomas.roos@industronic.de>
+
+	* gst-libs/gst/audio/gstaudiosink.c:
+	  audiosink: Fill in the correct silence
+	  For unsigned raw formats this is not all zeroes, and for non-raw formats
+	  we just continue to assume all zeroes for now.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739446
+
+2015-01-20 19:14:21 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/app/gstappsink.c:
+	  appsink: Only emit EOS signal after all buffers are consumed
+	  Otherwise the application will possibly shut down the pipeline already
+	  because EOS is received, while there are still some buffers pending.
+
+2015-01-20 15:08:24 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  dcodebin2: fix lock/unlock mismatch on multiqueue overrun
+
+2015-01-13 16:07:06 +0100  Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+	* gst/audioresample/resample.c:
+	  audioresample: Try to prevent endless looping
+	  Speex may decide not to consume any samples because it can't write any. I've
+	  seen a hang during draining caused by the resample loop never terminating.
+	  In that case, resampling happened as normal until olen was 0 but ilen was
+	  still 1. _process_native then reduced ichunk to 0, so ilen never decreased
+	  below 1 and the loop never terminated.
+	  Instead of reverting 684cf44 ({audioresample: don't skip input samples),
+	  break only if all output samples have been produced and speex refuses
+	  to consume any more input samples.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=732908
+
+2015-01-19 11:17:18 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videorate/Makefile.am:
+	  videorate: Add $(GST_PLUGINS_BASE_CFLAGS) to be able to find gst/video/video.h
+
+2015-01-18 14:58:36 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* gst/videorate/Makefile.am:
+	* gst/videorate/gstvideorate.c:
+	  videorate: Implement allocation query
+	  The videorate element keeps 1 buffer internally. This buffer need
+	  to be requested during allocation query otherwise the pipeline may
+	  stall.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738302
+
+2015-01-18 14:17:07 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* gst/videorate/Makefile.am:
+	* gst/videorate/gstvideorate.c:
+	  Revert "videorate: Implement allocation query"
+	  This reverts commit 3c04db4a307048db70ee1d08c1d62e26ad9569d8.
+
+2015-01-18 11:02:00 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* gst/videorate/Makefile.am:
+	* gst/videorate/gstvideorate.c:
+	  videorate: Implement allocation query
+	  VideRate keeps 1 buffer in order to duplicate base on closest buffer
+	  relative to targeted time. This extra buffer need to be request
+	  otherwise the pipeline may stall when fixed size buffer pool is used.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738302
+
+2015-01-17 14:51:48 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Fix compilation
+
+2015-01-12 14:38:09 +0100  Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: do call set_queue_size in no_more_pads_cb
+	  Consider pipeline: gst-launch-1.0 playbin uri=http://example.com/a.ogg
+	  Consider 128kbit audio stream.
+	  As soon as uridecodebin detects the bitrate, it configures its input
+	  queue2 max-size to 32000 bytes.
+	  The 2MB buffer in multiqueue is nearly 2 orders of magnitude bigger.
+	  This non-deterministically drives queue2 buffer anywhere from
+	  100% to 0% until multiqueue is filled.
+	  This patch sets multiqueue size to 5 buffers early in no_more_pads_cb.
+	  Partly reverts commit db771185ed750627a6a1824c42b651d739e1b4a4.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740689
+
+2015-01-16 15:21:14 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: free old groups when switching groups
+	  Old groups are freed with one switch's delay when switching groups.
+	  They're freed in a scratch thread to avoid delaying the switch.
+
+2014-12-12 17:02:35 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: fix clipped duration determination for non 0 based segments
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740422
+
+2015-01-15 10:51:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudioutilsprivate.c:
+	  audio: Keep caps features when building the downstream filter
+	  Based on 5fd4e3e0b6cc4f30d7b1489a105db946b43f1a9f for video
+	  by Alessandro Decina.
+
+2015-01-15 13:54:14 +1100  Alessandro Decina <alessandro.d@gmail.com>
+
+	* gst-libs/gst/video/gstvideoutilsprivate.c:
+	  videoutils: keep caps features in account when building the downstream filter
+	  See 00c2ce6 and https://bugzilla.gnome.org/show_bug.cgi?id=741263 for reference.
+
+2015-01-14 10:35:34 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/examples/playback/playback-test.c:
+	  examples: playback: add labels with supported seek range
+	  Add the supported seeking range in the advanced seek area.
+	  Also implement seeking querying the pipeline to retrieve those
+	  values and show to the user. It is done in a smaller frequency
+	  compared to the position/duration querying.
+
+2015-01-13 19:25:52 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: disable pad link checks as it has already been done
+	  Decodebin has already added the element to the bin and should only
+	  select caps compatible pads. It should disable the pad link checks
+	  to avoid doing those again.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742885
+
+2015-01-13 16:58:34 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  visual: cleanup
+	  Shameful fix to a silly mistake in the previous commit. Above email address for
+	  any mockery
+
+2015-01-13 16:36:09 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  visual: handle the return of the setup function
+	  Make the baseclass future proof by handling the gboolean return of the setup
+	  function. So if/when a child class uses this the base class is ready.
+
+2015-01-13 16:09:49 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  Revert "visual: remove unnecessary variable"
+	  This reverts commit a91d521a3602f33083405467db9454d422b9da1b.
+	  Being a base class it is better to check the value instead of ignoring it since
+	  a child class could be created that returns valuable information.
+
+2015-01-13 15:07:56 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  visual: remove unnecessary variable
+	  klass->setup (scope) will always return TRUE since all children of this class
+	  do so, no need to store the return. Besides, the value is overwritten a few
+	  lines down before it is ever used. Save the unnecessary memory and instructions.
+	  CID #1226467
+
+2015-01-12 15:27:18 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/libvisual/gstaudiovisualizer.c:
+	  visual: use unused value
+	  ret is assigned but not used and in the next cycle of the loop it is overwritten
+	  with default_prepare_output_buffer (). If there is a flow error the function
+	  should return instead.
+	  CID #1226475
+
+2015-01-12 15:56:06 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From f2c6b95 to bc76a8b
+
+2015-01-08 21:20:14 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/audio/gstaudioringbuffer.c:
+	  audioringbuffer: start ringbuffer if needed upon commit
+	  ... to provide for a running clock.
+
+2015-01-02 14:34:41 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: fix comment typo
+
+2015-01-09 15:38:09 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/video-dither.c:
+	  video-dither: remove check for below zero for unsigned value
+	  CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
+	  number since it is an unsigned integer. Removing that check and only checking if
+	  it is bigger than max and setting it appropriately.
+	  CID 1256559
+
+2015-01-09 15:28:06 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	  video-resampler: remove check for below zero for unsigned value
+	  CLAMP checks both if n_taps is '< 0' and '> max_taps'. n_taps will never be a
+	  negative number because it is an unsigned integer. Removing that check and only
+	  making sure it isn't set bigger than max.
+	  CID 1256558
+
+2015-01-08 10:45:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-color.c:
+	* gst-libs/gst/video/video-color.h:
+	* gst-libs/gst/video/video-info.c:
+	  video: Add support for BT2020 colorspace (UHD)
+
+2015-01-07 15:54:58 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: remove useless debug
+
+2015-01-07 15:52:57 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: add options to control chroma resampling
+	  Add an option to disable chroma resampling.
+	  Improve the matrix option values so that you can choose to use the input
+	  or output matrix or disable conversion.
+
+2015-01-02 15:27:23 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: remove unused enum
+
+2014-12-31 19:40:20 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: fix silly GQueue iteration code
+
+2014-12-26 20:48:55 +0000  Sam Thursfield <sam@afuera.me.uk>
+
+	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
+	  Fix documentation that incorrectly says a return value should be freed
+	  The gst_discoverer_info_get_missing_elements_installer_details()
+	  documentation and annotation says that the return value should be freed
+	  with g_strfreev(), but actually it's owned by the GstDiscovereInfo
+	  object and should definitely not get freed by the caller as well.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=742006
+
+2014-12-27 14:44:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesrc.c:
+	  audiobasesrc: Explicitly document that buffer-time and latency-time may be ignored
+
+2014-12-26 18:55:08 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: only clip by duration if end of buffer is ahead of segment
+	  It might happen that the timestamp is before the segment and the
+	  check would succeed. In this case reducing the duration makes no
+	  sense and would lead to broken results.
+
+2014-12-22 22:04:41 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videotestsrc/gstvideotestsrc.c:
+	  videotestsrc: Report our latency properly in live mode
+	  While we have no latency at all in theory, any other live source has the
+	  duration of one buffer as minimum latency. Do the same in videotestsrc.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741879
+
+2014-12-22 22:00:26 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiotestsrc/gstaudiotestsrc.c:
+	  audiotestsrc: Report our latency properly in live mode
+	  While we have no latency at all in theory, any other live source has the
+	  duration of one buffer as minimum latency. Do the same in audiotestsrc.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741879
+
+2014-12-22 09:25:04 -0500  Song Bing <b06498@freescale.com>
+
+	* gst-libs/gst/video/gstvideopool.c:
+	* sys/ximage/ximagepool.c:
+	* sys/xvimage/xvimagepool.c:
+	  videopool: update video alignment after applying
+	  Video buffer pool will update video alignment to respect stride alignment
+	  requirement. But haven't updated it to video alignment in configure.
+	  Which will cause user get wrong video alignment.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741501
+
+2014-11-28 14:36:23 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: get the internal time before the clock reset
+	  Otherwise calls to get the clock time might change its internal state
+	  and the internal/external time for calibration get unbalanced leading to
+	  a clock jump
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740834
+
+2014-12-22 11:45:53 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* MAINTAINERS:
+	  MAINTAINERS: Update my mail address
+
+2014-12-22 11:38:20 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  video{en,de}coder: Call reset() before the start() vfunc
+	  This makes sure that the element is in the same state before start() is called
+	  the very first time and every future call after the element was used already.
+	  Also it ensure that we always have a clean state before start(), cleaned the
+	  same way in every case.
+
+2014-12-22 11:36:58 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	  audioencoder: Call reset() before the start() vfunc to guarantee a clean state
+	  The same was done already in the decoder, and we cleaned some state just above
+	  manually that would also be taken care of by reset().
+	  This makes sure that the element is in the same state before start() is called
+	  the very first time and every future call after the element was used already.
+
+2014-12-22 11:33:14 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  video{en,de}coder: Reset the codec after calling the stop() vfunc
+	  The stop() vfunc might mess with some of our fields we have just
+	  reset, which could cause memory leaks or invalid state taken over
+	  to later.
+	  Also the stop() vfunc, or anything called until it from another thread,
+	  might want to be able to use the fields that were just resetted and
+	  become confused because of that.
+	  In the decoder we already had a workaround for things like this happening,
+	  this workaround is not needed anymore.
+
+2014-12-22 10:45:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	* gst-libs/gst/audio/gstaudiobasesrc.c:
+	  audiobase{sink,src}: Don't hold the object lock while calling create_ringbuffer() vfunc
+	  The implementation of that vfunc might want to use the object lock for
+	  something too. It's generally not a good idea to keep the object lock while
+	  calling any function implemented elsewhere.
+	  Also the ringbuffer can only be NULL at this point, remove a useless if block.
+	  And in the sink actually hold the object lock while setting the ringbuffer on
+	  the instance. Code accessing this is expected to use the object lock, so do it
+	  here ourselves too.
+
+2014-12-18 13:24:22 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/riff/riff-media.c:
+	  riff-media: Error out early if we observe an invalid audio format
+
+2014-12-18 13:22:17 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/riff/riff-media.c:
+	  riff: Also handle invalid block aligns for raw audio
+	  Fixes audio playback of
+	  http://demo.archermind.com/Test%20Sample/Video/MPEG%204/Divx3/Low-Motion/576-320.avi
+	  Audio and video together is still broken because of other issues.
+
+2014-12-18 10:57:13 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst-libs/gst/audio/Makefile.am:
+	  audio: Fix private header include/dist
+	  We want to dist it, but we don't want to install it.
+	  Fixes make dist/distcheck
+
+2014-12-18 10:53:20 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From ef1ffdc to f2c6b95
+
+2014-12-17 19:14:38 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  video: audio: fix GI annotations for proxy caps function
+	  Add the annotations to parameters that can be null and also for stating
+	  the ownership of the returned caps
+
+2014-12-17 15:21:48 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/libs/audiodecoder.c:
+	  tests: audiodecoder: tests for caps query implementation
+	  Copied from videodecoder tests and updated to audio features
+
+2014-12-17 15:21:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* gst-libs/gst/audio/gstaudiodecoder.h:
+	* win32/common/libgstaudio.def:
+	  audiodecoder: expose getcaps virtual function
+	  Allows subclasses to do custom caps query replies.
+	  Also exposes the standard caps query handler so subclasses can just
+	  extend on top of it instead of reimplementing the caps query proxying.
+
+2014-12-16 18:36:57 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: implement caps and accept-caps queries
+	  Allows decoders to proxy downstream restrictions on caps.
+	  Also implements accept-caps query to prevent regressions caused by the
+	  new fields on the return of a caps query that would cause the accept-caps
+	  to fail as it uses subset caps comparisons
+
+2014-12-16 11:13:40 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/Makefile.am:
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	* gst-libs/gst/audio/gstaudioutilsprivate.c:
+	* gst-libs/gst/audio/gstaudioutilsprivate.h:
+	  audioencoder: refactor getcaps proxy function to be reusable
+	  Makes the audioencoder's getcaps function that proxies downstream
+	  restriction available to other elements in the audio module to use it
+
+2014-12-17 14:18:03 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideodecoder.h:
+	* tests/check/libs/videodecoder.c:
+	* win32/common/libgstvideo.def:
+	  videodecoder: expose getcaps virtual function
+	  Allows subclasses to do custom caps query replies.
+	  Also exposes the standard caps query handler so subclasses can just
+	  extend on top of it instead of reimplementing the caps query proxying.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-15 18:46:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: accept-caps should only require fields from the template
+	  With the new caps query results the caps returned might have extra fields
+	  that are not required by the decoder (framerate for image decoders) and it
+	  causes a regression making, for example, jpegdec reject caps that don't
+	  have framerates.
+	  The accept-caps implementation will do 2 checks:
+	  1) Do subset check with the template caps, making sure all the required
+	  fields that are present on the template are present on the received caps.
+	  2) Do a intersection check with the result of a caps query, making sure
+	  that downstream can accept the fields in the received caps.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-09 16:08:12 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/gstvideoutilsprivate.c:
+	  videoutils: proxy filter when doing a caps query downstream
+	  Allows downstream to use the filter and possibly reduce caps complexity
+	  to speed up negotiation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-09 16:05:27 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/gstvideoutilsprivate.c:
+	  videoutils: return empty if the element has no possible allowed caps
+	  Instead of returning the template caps and having a failure happen
+	  later because there are no possible caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-08 16:33:33 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideoencoder.c:
+	* gst-libs/gst/video/gstvideoutilsprivate.c:
+	* gst-libs/gst/video/gstvideoutilsprivate.h:
+	* tests/check/libs/videodecoder.c:
+	  videodecoder: implement caps query
+	  Refactor the encoder's caps query proxying function to a common place
+	  and use it in the videodecoder to proxy downstream restrictions.
+	  The new function is private to the gstvideo lib.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741263
+
+2014-12-17 12:01:19 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* configure.ac:
+	  configure: require release version of orc now that there is one
+
+2014-12-16 12:57:55 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* sys/ximage/ximagesink.c:
+	* sys/xvimage/xvimagesink.c:
+	  ximagesink: clear src and dest rectangles
+	  Now that the center function also takes into account the x and y
+	  coordinates of the dest rectangle, better clear all the fields before
+	  using them.
+
+2014-12-16 12:10:53 +0100  Song Bing <b06498@freescale.com>
+
+	* gst-libs/gst/video/gstvideopool.c:
+	* sys/ximage/ximagepool.c:
+	* sys/xvimage/xvimagepool.c:
+	  videopool: update buffer size after video alignment
+	  Update the new buffer size after alignment in the pool configuration
+	  before calling the parent set_config. This ensures that the parent knows
+	  about the buffer size that we will allocate and makes the size check
+	  work in the release_buffer method.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741420
+
+2014-12-15 20:57:14 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.h:
+	* gst-libs/gst/audio/gstaudiobasesrc.h:
+	  audiobasesrc/sink: Add _CAST macros
+
+2014-12-15 14:10:17 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst-libs/gst/video/gstvideosink.c:
+	* tests/check/libs/video.c:
+	  video: Fix non-default usage of gst_video_sink_center_rect
+	  Make sure we take into account non-0 x/y destination rectangles
+
+2014-12-15 12:12:44 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/examples/playback/playback-test.c:
+	  examples: improve playback-test help text a little
+	  And allow pipeline type to be specified as string.
+
+2014-12-15 10:35:35 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/pango/gstbasetextoverlay.h:
+	  pango: Add license/copyright header to header file
+
+2014-12-15 09:45:43 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  Revert "decodebin: Only emit the drain signal for the main decode chain, not any subchains"
+	  This reverts commit a391dfe17f1a325f60e1d51a6d40c1a68eb196de.
+	  It breaks gapless playback: https://bugzilla.gnome.org/show_bug.cgi?id=740045
+
+2014-12-09 03:18:37 +0100  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/audiorate/gstaudiorate.c:
+	  audiorate: Fill gap events
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741281
+
+2014-12-10 16:10:58 +0530  Sanjay NM <sanjay.nm@samsung.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audio: Add error handling to gst_audio_decoder_drain()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740686
+
+2014-12-13 16:14:49 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudioclock.c:
+	  audioclock: Fix redundant definitions compiler warning
+	  gstaudioclock.c:51:31: error: redundant redeclaration of 'gst_audio_clock_init' [-Werror=redundant-decls]
+	  G_DEFINE_TYPE (GstAudioClock, gst_audio_clock, GST_TYPE_SYSTEM_CLOCK);
+	  gstaudioclock.c:51:31: error: redundant redeclaration of 'gst_audio_clock_class_init' [-Werror=redundant-decls]
+	  G_DEFINE_TYPE (GstAudioClock, gst_audio_clock, GST_TYPE_SYSTEM_CLOCK);
+
+2014-12-13 16:04:40 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudioclock.c:
+	  audioclock: No need to get the parent class in class_init, G_DEFINE_TYPE does that for us
+
+2014-12-13 16:01:44 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudioclock.c:
+	  audioclock: Use G_DEFINE_TYPE instead of a custom get_type() function
+
+2014-12-12 08:32:15 -0800  Zaheer Abbas Merali <zaheermerali@gmail.com>
+
+	* gst-libs/gst/rtp/gstrtcpbuffer.c:
+	  rtcpbuffer: fix spelling of word in comment
+
+2014-12-12 14:59:49 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/rtpbasedepayload.c:
+	  tests: rtpbasepayload: fix indentation
+
+2014-12-12 14:59:03 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/audiodecoder.c:
+	  tests: audiodecoder: fix indentation
+
+2014-12-12 14:56:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/audiodecoder.c:
+	  tests: audiodecoder: fix broken refcounting in unit test
+	  The set_format vfunc does not pass ownership of the caps
+	  to the decoder, so we mustn't unref the caps there.
+	  gst_event_new_caps() does not take ownership of the caps
+	  passed, so we must unref the caps afterwards.
+	  Fixes leaks when running test in valgrind in 1.4 branch.
+
+2014-12-12 10:02:43 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	  video: Update disted orc source files
+
+2014-12-12 10:01:36 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  Revert "video-converter: Fix compiler warning because of missing prototype of non-static function"
+	  This reverts commit 406f32a9468c837a4d71f988de10dc2198a8edc9.
+	  The problem was apparently that my video-orc.h was not updated and did not
+	  include the prototype for that function. Only a "make clean" caused it to
+	  be regenerated.
+
+2014-12-12 09:51:05 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Fix compiler warning because of missing prototype of non-static function
+	  video-converter.c:838:1: error: no previous prototype for function
+	  '_custom_video_orc_matrix8' [-Werror,-Wmissing-prototypes]
+
+2014-12-09 22:47:31 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: do not use fixed caps on source pad
+	  decoders can change the caps on their source pads, so they don't
+	  use fixed caps. Having fixed caps can cause renegotiation issues.
+
+2014-12-09 22:46:42 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: do not use fixed caps on source pad
+	  decoders can change the caps on their source pads, so they don't
+	  use fixed caps. Having fixed caps can cause renegotiation issues.
+
+2014-12-11 13:45:38 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Do not mix up stream type when getting stream combiner element
+	  We were always returning the video stream combiner whatever stream type
+	  combiner was wanted.
+
+2014-12-10 13:23:23 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin2: always unref the combiner sinkpad when removing the srcpad
+	  Create a function to do the pad cleanup of the GstSourceCombine struct
+	  and use it to not forget to also cleanup the sink pad and fix a memory
+	  leak.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741198
+
+2014-12-10 16:42:12 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: make RGB pack/unpack faster
+	  Avoid all the merging and splitting and use a pair of shifts and or
+
+2014-12-11 01:53:15 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.h:
+	  videodecoder: Add GST_VIDEO_DECODER_CAST macro
+	  It's used in some macros already, so let's make it exist.
+
+2014-11-25 13:31:48 +0100  Göran Jönsson <goranjn@axis.com>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	  rtspconnection: No remove child if destroyed.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740730
+
+2014-12-08 18:53:35 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/icles/test-reverseplay.c:
+	  reverse-play: fix seek to end when starting reverse
+	  Start reverse playback by actually seeking to the end of
+	  the file.
+
+2014-12-06 21:02:37 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: set bits and format after conversion
+	  Update the current format, bits and pstride.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741187
+
+2014-12-05 22:09:45 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: free dither_lines
+	  Avoid a memory leak
+
+2014-12-05 18:16:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	  Bump ORC requirement to 4.22.1
+	  We now depend on git commit f1cfa5, "orcc: allow setting custom
+	  backup function"
+
+2014-12-05 14:51:28 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: use custom backup function
+	  Use the new orc feature to set a custom backup function.
+
+2014-12-05 12:18:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: improve matrix8 function
+	  Avoid using a constant.
+	  Avoid doing saturated adds, results are not supposed to overflow here.
+	  Rework the C backup function a little in preparation for custom backup
+	  functions in ORC.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=741015
+
+2014-11-28 15:06:27 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	* tests/check/libs/audiodecoder.c:
+	  audiodecoder: Push pending events before sending EOS.
+	  Segments are added to the pending events, and pushing a segment
+	  is mandatory before sending EOS.
+	  + Adds a test.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740853
+
+2014-11-27 05:53:20 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: Fix seeking before the first frame.
+	  The previous code was setting keytarget to target
+	  to make sure the keyframe found for each pad was
+	  indeed before the target.
+	  Then if target == keytarget, it assumed a keyframe had been
+	  found, which was not the case if target was before the first frame
+	  in the file.
+	  This patch checks that a keyframe was indeed found, and if not
+	  seeks to 0, without bisecting again.
+	  Assuming default gst qa assets in $HOME/gst-validate
+	  seek_before_first_frame.scenario:
+	  description, seek=true, handles-states=true
+	  pause, playback-time=0.0
+	  seek, playback-time=0.0, start=0.0, flags=accurate+flush
+	  seek, playback-time=0.0, start=0.01, flags=accurate+flush
+	  seek, playback-time=0.0, start=0.1, flags=accurate+flush
+	  GST_DEBUG=*theoradec*:2 gst-validate-1.0 playbin \
+	  uri=file://$HOME/gst-validate/gst-qa-assets/medias/ogg/vorbis_theora.0.ogg \
+	  --set-scenario seek_before_first_frame.scenario
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741097
+
+2014-10-08 08:54:57 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Only check sinks which are in >= GST_STATE_READY
+	  Otherwise we endup with bogus caps intersection (from the pad template
+	  caps and not from what the actual hardware/device supports)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738131
+
+2014-12-03 10:15:18 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix chroma resampling check
+	  Decide if we need chroma resampling by checking if we have a progressive
+	  or interlaced chroma resampler.
+
+2014-12-03 10:14:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: only do dithering when needed
+	  Only do dithering when one of the quantizers is > 1.
+
+2014-12-02 15:58:00 -0500  Chad <crh184@psu.edu>
+
+	* gst/audiorate/gstaudiorate.c:
+	  audiorate: Use gst_util_uint64_scale_int_round()
+	  Using gst_util_uint64_scale_int() causes slight drift
+	  which accumulates over time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=741045
+
+2014-12-02 13:39:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* win32/common/libgstvideo.def:
+	  defs: update defs file
+
+2014-12-02 11:51:19 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoconvert/gstvideoconvert.c:
+	* gst/videoconvert/gstvideoconvert.h:
+	  videoconvert: add dither-bits option
+	  Fix the dither option.
+	  Add a new option to set the quantizer
+
+2014-12-02 11:48:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add where orc functions could go
+	  Add the disabled orc functions in #if 0 lines for when we can enable
+	  them.
+
+2014-12-02 11:40:59 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	* gst-libs/gst/video/video-dither.c:
+	  video-converter: add dithering
+	  Use the new dither object to perform dithering.
+	  Add option to select dithering method.
+	  Add option to quantize to a specific value
+
+2014-12-02 11:39:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add palette when needed
+
+2014-12-02 11:32:28 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/video-dither.c:
+	* gst-libs/gst/video/video-dither.h:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video.h:
+	  video-dither: add video dither helper object
+	  Add a new object that implements various dithering methods.
+
+2014-12-01 22:28:52 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tools/gst-play.c:
+	  gst-play: do not set system's volume to 100% by default
+	  Only change the volume if requested
+
+2014-12-01 09:50:24 +0100  Thomas Klausner <wiz@danbala.tuwien.ac.at>
+
+	* ext/alsa/gstalsasink.c:
+	* ext/alsa/gstalsasrc.c:
+	  alsa: Use EPIPE instead of ESTRPIPE if the latter does not exist
+	  NetBSD does not have ESTRPIPE.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740952
+
+2014-11-28 14:28:06 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/alsa/gstalsasrc.c:
+	* ext/ogg/gstoggmux.c:
+	* ext/vorbis/gstvorbisdec.c:
+	* gst-libs/gst/audio/gstaudioringbuffer.c:
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	* gst-libs/gst/tag/gsttagdemux.c:
+	* gst-libs/gst/tag/id3v2frames.c:
+	* gst-libs/gst/video/navigation.c:
+	* gst-libs/gst/video/video-converter.c:
+	* gst/adder/gstadder.c:
+	* gst/encoding/gstencodebin.c:
+	* gst/playback/gstdecodebin2.c:
+	* gst/playback/gstplaysink.c:
+	* gst/playback/gstsubtitleoverlay.c:
+	* gst/playback/gsturidecodebin.c:
+	* gst/subparse/gstsubparse.c:
+	* gst/tcp/gstmultihandlesink.c:
+	* gst/tcp/gstmultioutputsink.c:
+	* tests/examples/playback/playback-test.c:
+	* tests/examples/seek/jsseek.c:
+	* tools/gst-discoverer.c:
+	  Don't compare booleans for equality to TRUE and FALSE
+	  TRUE is 1, but every other non-zero value is also considered true. Comparing
+	  for equality with TRUE would only consider 1 but not the others.
+
+2014-11-16 15:54:56 +0100  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:
+	* win32/common/libgstpbutils.def:
+	  encodebin: Add a way to disable caps renegotiation for output stream format
+	  In some cases, the user might want the stream outputted by encodebin to
+	  be in the exact same format during all the stream. We should let the
+	  user specify when this is the case. This commit add some API in the
+	  GstEncodingProfile to determine whether the format can be renegotiated
+	  after the encoding started or not.
+	  API:
+	  gst_encoding_profile_set_allow_dynamic_output
+	  gst_encoding_profile_get_allow_dynamic_output
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740214
+
+2014-11-28 13:31:39 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	  tests: remove libs/video and videoconvert test from valgrind blacklist
+	  Seem to work fine.
+
+2014-11-28 13:29:37 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	  tests: don't run orc/* tests under valgrind
+	  They just seem to blow up for some reason that needs investigating.
+
+2014-11-28 13:11:33 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/tag/gsttagmux.c:
+	  tagmux: fix criticals when there are no tags at all
+
+2014-11-21 01:47:35 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/icles/test-reverseplay.c:
+	  test-reverseplay: Use uridecodebin for input
+	  Work with any installed URI handler
+	  Add some more debug output
+
+2014-11-28 10:27:28 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-frame.c:
+	  video-frame: Mapping a frame with inconsistent values between GstVideoMeta and GstVideoInfo is a bug
+	  It will cause the frame to be initialized with inconsistent values that then
+	  later can cause crashes or any other kind of interesting and hard to debug
+	  bugs.
+
+2014-11-27 17:10:31 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From 7bb2bce to ef1ffdc
+
+2014-11-27 15:28:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  video-blend: make use of x offset when unpacking overlay image pixels
+	  Now that it's implemented we can use it, which is a minor
+	  optimisation when the image to overlay gets cropped on the
+	  left.
+
+2014-11-27 15:04:12 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: sprinkle some 'restrict' keywords in pack/unpack functions
+	  In cases where we just call orc directly this is somewhat
+	  superfluous, but let's do it anyway for consistency. In
+	  other cases the compiler can hopefully use this to optimise
+	  memory access a little.
+
+2014-11-27 13:01:03 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: handle x offset in unpack
+	  Add support for x offset in almost all unpack methods.
+	  Fix naming of source and dest pixels.
+	  Add const to source pixels.
+
+2014-11-27 10:51:58 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: improve unpack i420
+	  unpack_i420 does not need extra code to handle odd widths, the orc code
+	  already handles it fine.
+
+2014-11-27 09:45:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	  videoscale: use old property name
+	  Unbreak ABI by changing to the old property name again.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740798
+
+2014-11-25 13:39:07 +0100  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Analyze source pad before setting to PAUSED for 'simple demuxers'
+	  Before we were setting them to PAUSED and (much) later connecting to
+	  their source pad caps notify signal.
+	  There was a race where that demuxer was pushing a caps and later a buffer
+	  on its source pad when we were not even connected to its source pad caps notify
+	  signal leading to decodebin missing the information and not keeping on
+	  building the pipeline on CAPS event thus the demuxer was posting an ERROR
+	  (not linked) message on the bus. This need to be done for 'simple
+	  demuxers' because those have one ALWAYS source pad, not like usual demuxers
+	  that have several dynamic source pads.
+	  A "simple demuxer" is a demuxer that has one and only one ALWAYS source
+	  pad.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740693
+
+2014-11-25 16:46:50 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin2: Take STREAM_LOCK before sending sticky events.
+	  There was a race where:
+	  1) we would put the element to PAUSED
+	  2) It would get data sent to it from upstream
+	  3) It would thus send caps
+	  3) caps_notify_cb would continue autoplugging
+	  4) caps would flow downstream, the last pad would get exposed
+	  5) we were still not done sending the sticky events
+	  Taking the stream lock on the new element's sinkpad and only
+	  releasing it when sticky events have all been sent prevents
+	  the caps from reaching the source pad of the element before
+	  we're all set.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740694
+
+2014-08-06 19:31:25 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefindfunctions: detect mp4 common file format variant
+	  Used e.g. by UltraViolet.
+
+2014-11-25 22:01:08 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/alsa/gstalsasrc.c:
+	  alsasrc: debug message fixes
+	  In the same vein as 74e9640a.
+
+2014-11-25 17:42:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scale: combine adds when max_taps equals combine size
+	  When the amount of pixels/lines matches the amount we can combine,
+	  combine the adds and multiplies and do the scale as a separate
+	  operation.
+
+2014-11-25 17:25:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: combine scaling operations
+	  Combine add and scale of multiple lines/pixels to reduce the amount of
+	  read and writes to temporary memory.
+
+2014-11-25 14:45:23 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/pango/gsttimeoverlay.c:
+	* ext/pango/gsttimeoverlay.h:
+	  timeoverlay: add "time-line" property
+	  So we can also show running time or stream time, not just the
+	  buffer time stamps.
+
+2014-11-25 11:54:51 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	* gst/videoscale/gstvideoscale.h:
+	  videoscale: add property to do scaling after gamma-decode
+
+2014-11-25 11:28:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	* gst/videoscale/gstvideoscale.h:
+	  videoscale: add more scaling filters
+	  Adjust the filter parameters so that they use the same number of taps
+	  and method as the old ones.
+	  Add some new filters
+
+2014-11-25 10:36:13 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	  video-resampler: remove print
+
+2014-11-25 10:32:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	  video-resampler: improve variable taps
+	  Improve quality of variable taps on all methods by reusing the lanczos
+	  parameters where possible.
+
+2014-11-25 09:11:31 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	  video-resampler: Fix lanczos parameters for variable taps
+	  when using variable taps and when we are limiting the number of taps,
+	  recalculate the lanczos parameters to match the clamped value.
+	  Set the max number of taps to 128
+
+2014-11-25 11:38:34 +0300  Andrei Sarakeev <sarakusha@gmail.com>
+
+	* gst/playback/gstplaysink.c:
+	  playsink: Reset mute property of the sink to playsink's value when setting up the audio chain
+	  Otherwise the following can happen:
+	  1. set mute=true
+	  2. play media1 (Ok)
+	  3. play media without audio (audiochain removed)
+	  4. play media2 (audiochain created, mute=*false*)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740675
+
+2014-11-25 11:38:34 +0300  Andrei Sarakeev <sarakusha@gmail.com>
+
+	* gst-libs/gst/pbutils/gstdiscoverer.h:
+	  discoverer: fix typo in header file
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740675
+
+2014-11-25 09:08:18 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	  pbutils: add description for audio/x-audible
+
+2014-11-25 01:02:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefind: improve 'audible' audio typefinder a little
+	  Don't return NEARLY_CERTAIN just based on 4 bytes.
+	  Also change media type to audio/x-audible.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=715050
+
+2013-11-23 11:36:43 +1000  Jonathan Matthew <jonathan@d14n.org>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefindfunctions: add audio/audible typefinder
+	  https://bugzilla.gnome.org/show_bug.cgi?id=715050
+
+2014-06-16 11:46:18 +0200  Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+	* ext/alsa/gstalsasink.c:
+	* ext/alsa/gstalsasrc.c:
+	  alsa: Change the log messages in xrun_recovery() from DEBUG to WARNING
+	  xrun_recovery() runs when there is an error
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740615
+
+2014-11-24 12:47:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: keep track of required temp lines
+	  Make a small object to hold a pool of allocated temp lines.
+	  Keep track of how many temp lines each conversion stage needs and use
+	  this to allocate just enough temp lines from the temp lines object. from
+	  the temp lines object.
+
+2014-11-24 12:45:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: use err line in fastpath
+	  Use the error line for temporary storage in the fastpath so that we
+	  don't have to allocate any other temp lines.
+
+2014-11-22 21:51:33 +0100  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: don't complain about PTS != DTS on keyframes
+	  It is valid for streams with b-frames
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740556
+
+2014-11-21 16:06:54 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: handle mixed interlaced
+	  When dealing with mixed interlaced, setup a scaler and chroma-resampler
+	  for both interlaced and progressive frames and switch between them
+	  depending on the interlace mode of the input frame.
+
+2014-11-21 16:04:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Cleanup options parsing
+	  Cleanup option parsing
+	  Add some debug
+
+2014-11-21 15:59:47 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: there is no need to apply x offset to temp lines
+
+2014-11-21 15:58:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: ensure both fields have the same number of taps
+
+2014-11-21 11:15:04 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: rework the options a little
+	  Rework the options a little to make it nicer to set defaults.
+
+2014-11-21 11:12:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	* gst-libs/gst/video/video-resampler.h:
+	  video-resampler: add option to limits taps
+	  Add an option to limit the number of taps to use in automatic mode. The
+	  problem is that for lanczos, we might use more taps than what we can
+	  handle with the current precision.
+	  Rework the other options a little to make it nicer to set defaults.
+
+2014-11-20 18:20:00 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: update orc files
+
+2014-11-20 15:53:23 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* win32/common/libgstvideo.def:
+	  win32: Update defs file
+
+2014-11-19 21:18:04 +0900  Hyunjun Ko <zzoonis@gmail.com>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.h:
+	  rtspconnection: fix warning on param name mismatch
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740013
+
+2014-11-18 00:04:59 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/icles/.gitignore:
+	* tests/icles/Makefile.am:
+	* tests/icles/test-reverseplay.c:
+	  tests: Add reverse playback verification test
+	  Plays a requested URI forward to EOS, then backward and
+	  checks that the same timestamp range(s) are covered.
+
+2014-11-12 15:23:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videorate/gstvideorate.c:
+	  videorate: Operate in a zero-latency mode if drop-only is set to TRUE
+	  There's no reason why we would have to wait for the next buffer to decide
+	  whether to output the current one or not. We just have to check if the
+	  current one is earlier than our expected next time, which is the previous
+	  frame timestamp plus the expected frame duration.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=740018
+
+2014-11-19 14:38:03 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Use correct enum, GstVideoFormat instead of GstFormat
+
+2014-11-19 13:25:13 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix size check
+	  Add some debug, fix size check that decides what scaling to do first and
+	  when to do conversion.
+
+2014-11-19 12:53:03 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: avoid primaries conversion when asked
+	  Don't do conversion between primaries when the option is disabled.
+	  Only do some matrix code when needed.
+
+2014-11-19 12:41:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-info.c:
+	  video-info: add a note about subsampled formats
+	  Add a note about gst_video_info_set_format() and interlaced formats.
+
+2014-11-19 12:05:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-info.c:
+	  video-info: handle interlaced size correctly
+	  Refactor GstVideoInfo init, make function to set default colorimetry.
+	  Call fill_planes after we configure the GstVideoInfo with parameters
+	  from the caps.
+	  The size of the chroma planes for interlaced vertically subsampled
+	  formats needs to be rounded up to 2, we have 2 fields with each
+	  the same anount of chroma lines.
+
+2014-11-19 12:04:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-color.c:
+	  video-color: return FALSE on unparsable colorimetry
+
+2014-11-19 09:40:05 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: handle unpack interlaced subsampled formats
+	  For interlaced vertically subsampled formats the check for even lines
+	  needs to take into account the two fields.
+
+2014-11-19 09:39:32 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix interlaced shift
+
+2014-11-19 09:30:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: keep a small backlog of lines
+	  Allow lines to jump backwards slightly, usefull for interlaced content.
+
+2014-11-19 09:28:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	  video-chroma: Fix interlaced chroma resampling
+	  Use the interlaced flag to select the right resampler.
+
+2014-11-18 16:36:08 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-resampler.c:
+	* gst-libs/gst/video/video-scaler.c:
+	  video: add some more debuging
+
+2014-11-18 16:35:13 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix interlacing some more
+	  Use the right phase.
+	  Take the right lines from interlaced content.
+
+2014-11-18 12:53:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: fix dither method
+
+2014-11-18 12:52:27 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: fix some leaks
+	  And remove some unused fields.
+
+2014-11-18 12:20:26 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: add support for gamma and primaries
+	  Keep only 1 structure with all matrix information.
+	  Add structure to hold gamma information.
+	  Add more options to control gamma, primaries and color matrix handling.
+	  Add functions to compute transformations to and from XYZ and use this
+	  to convert between primaries.
+	  Merge gamma into the convert to and from RGB stage.
+	  Fix border val.
+	  Simplify the fastpath table, remove unused fields, add some more checks.
+
+2014-11-18 11:09:40 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-color.c:
+	* gst-libs/gst/video/video-color.h:
+	  video-color: add method to get primaries info
+
+2014-11-18 11:08:10 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-color.c:
+	* gst-libs/gst/video/video-info.c:
+	  video-color: fix default 601 primaries
+
+2014-11-18 11:06:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix interlaced taps setup
+
+2014-11-14 09:15:22 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-color.c:
+	* gst-libs/gst/video/video-color.h:
+	* gst-libs/gst/video/video-info.c:
+	  video-color: make sRGB colorimetry the default for RGB
+
+2014-11-13 12:03:26 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: split YUV to and from RGB conversions
+	  Prepare for doing full gamma corrected conversion and scaling by first
+	  splitting the conversions from and to RGB into separate steps.
+	  split scaling in downscaling and upscaling steps to be performed before
+	  and after conversion respectively.
+
+2014-11-13 12:02:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: don't convert too much
+	  because we do conversion after downscaling we only need to convert the
+	  smallest width.
+
+2014-11-13 12:00:05 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: add orc splat functions to draw border
+
+2014-11-05 21:52:44 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  Revert "basetextoverlay: Fix segfault when overlay outside the frame"
+	  This is not correct. overlay->silent is a property and we
+	  should not just flip the property forever because one text
+	  we render is outside of the frame. The next one might not
+	  be, the positioning properties can be changed after all.
+	  The lower layers should handle clipping, and now do.
+	  This reverts commit 1cc311156cc3908d1d9888fbcda67305fc647337.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738984
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-05 21:46:47 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  Revert "basetextoverlay: segfault when xpos >= video size"
+	  This is not right, even if it might avoid a crash. We don't
+	  want to just set xpos/ypos to 0 in those cases. Clipping
+	  should be done properly, see bug #739281 for that.
+	  This reverts commit 900d0267d511e9553eec44d948d7e33ead7dc903.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738984
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-16 23:26:45 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  video-blend: minor optimisation
+	  Only need to run matrix on those pixels which
+	  will actually be used.
+
+2014-11-16 19:28:54 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/icles/Makefile.am:
+	* tests/icles/test-overlay-blending.c:
+	  tests: make overlay blending test slightly less boring
+
+2014-11-16 16:34:31 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  video-blend: fix clipping of overlay images on the left
+	  Fix clipping of images that are partially left of the video
+	  surface, they would get clipped on the right side instead of
+	  the left side, because the video unpack functions currently
+	  ignore the x offset parameter. Work around that until that
+	  is implemented.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-16 16:31:45 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  video-blend: fix allocation of temp src line for wide sources
+	  Fix allocation of temporary source line buffers for source
+	  images that are wider than the video overlay surface.
+
+2014-11-16 01:34:09 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/icles/.gitignore:
+	* tests/icles/Makefile.am:
+	* tests/icles/test-overlay-blending.c:
+	  tests: add visual overlay composition blending test
+	  Shows visual result of blending a logo on top of
+	  a video surface, esp. when the logo is partially
+	  outside of the video surface and needs to be
+	  clipped.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-16 01:32:55 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/video.c:
+	  tests: fix leak in video unit test
+
+2014-11-10 16:36:35 +0530  Vineeth T M <vineeth.tm@samsung.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  video-blend: fix blending of rectangles partially or fully outside of the video
+	  In case of overlay being completely or partially outside
+	  the video frame, the offset calculations are not right,
+	  which resulted in the overlay not being displayed as
+	  expected, or crashes due to invalid memory access.
+	  When the overlay rectangle is completely outside,
+	  we need not render the overlay at all.
+	  For partial display of overlay rectangles, src_yoff
+	  was not being calculated, hence it was always clipping
+	  the bottom half of the overlay, By calculating the
+	  src_yoff, now the overlay is clipped properly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-10 12:12:42 +0530  Vineeth T M <vineeth.tm@samsung.com>
+
+	* tests/check/libs/video.c:
+	  tests: video: add video blend test
+	  Add test to check rendering of overlays of different sizes
+	  that are completely or partially outside the video surface.
+	  Once the overlay is blended to the video, verify if the
+	  position of the blended overlay is as expected, by comparing
+	  the pixels of the blended video with the expected values.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739281
+
+2014-11-15 23:15:06 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/plugins/gst-plugins-base-plugins.args:
+	* docs/plugins/gst-plugins-base-plugins.hierarchy:
+	* docs/plugins/gst-plugins-base-plugins.signals:
+	* docs/plugins/inspect/plugin-adder.xml:
+	* docs/plugins/inspect/plugin-alsa.xml:
+	* docs/plugins/inspect/plugin-app.xml:
+	* docs/plugins/inspect/plugin-audioconvert.xml:
+	* docs/plugins/inspect/plugin-audiorate.xml:
+	* docs/plugins/inspect/plugin-audioresample.xml:
+	* docs/plugins/inspect/plugin-audiotestsrc.xml:
+	* docs/plugins/inspect/plugin-cdparanoia.xml:
+	* docs/plugins/inspect/plugin-encoding.xml:
+	* docs/plugins/inspect/plugin-gio.xml:
+	* docs/plugins/inspect/plugin-libvisual.xml:
+	* docs/plugins/inspect/plugin-ogg.xml:
+	* docs/plugins/inspect/plugin-pango.xml:
+	* docs/plugins/inspect/plugin-playback.xml:
+	* docs/plugins/inspect/plugin-subparse.xml:
+	* docs/plugins/inspect/plugin-tcp.xml:
+	* docs/plugins/inspect/plugin-theora.xml:
+	* docs/plugins/inspect/plugin-typefindfunctions.xml:
+	* docs/plugins/inspect/plugin-videoconvert.xml:
+	* docs/plugins/inspect/plugin-videorate.xml:
+	* docs/plugins/inspect/plugin-videoscale.xml:
+	* docs/plugins/inspect/plugin-videotestsrc.xml:
+	* docs/plugins/inspect/plugin-volume.xml:
+	* docs/plugins/inspect/plugin-vorbis.xml:
+	* docs/plugins/inspect/plugin-ximagesink.xml:
+	* docs/plugins/inspect/plugin-xvimagesink.xml:
+	  docs: update to git
+
+2014-11-15 23:13:42 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/gio/gstgiostreamsink.c:
+	* gst/gio/gstgiostreamsrc.c:
+	* gst/playback/gstplaybin2.c:
+	  docs: fix some gtk-doc warnings
+	  Deprecated entities found in documentation for xyz:Long_description
+	  .
+
+2014-11-12 09:57:38 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: take offset into account when unpacking
+	  When we can directly take the input line from the source frame when
+	  unpacking, also take into account the x offset.
+
+2014-11-12 09:57:12 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add some notes
+
+2014-11-11 16:19:03 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* win32/common/libgstvideo.def:
+	  defs: update defs and docs
+
+2014-11-11 16:11:15 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-color.c:
+	* gst-libs/gst/video/video-color.h:
+	* tests/check/libs/video.c:
+	  video-color: add gamma encode/decode functions
+	  Add functions to encode and decode gamma.
+	  Add unit test to check that encode and decode are eachothers inverse
+	  and that the limits are respected.
+
+2014-11-10 14:53:13 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/libs/video.c:
+	  test: add scaling test
+	  Sort pack and unpack performance measurements
+
+2014-11-10 12:01:48 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: update disted file
+	  and disable one failing function
+
+2014-10-24 17:08:43 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videoscale/Makefile.am:
+	* gst/videoscale/gstvideoscale.c:
+	* gst/videoscale/gstvideoscale.h:
+	* gst/videoscale/gstvideoscaleorc-dist.c:
+	* gst/videoscale/gstvideoscaleorc-dist.h:
+	* gst/videoscale/gstvideoscaleorc.orc:
+	* gst/videoscale/vs_4tap.c:
+	* gst/videoscale/vs_4tap.h:
+	* gst/videoscale/vs_fill_borders.c:
+	* gst/videoscale/vs_fill_borders.h:
+	* gst/videoscale/vs_image.c:
+	* gst/videoscale/vs_image.h:
+	* gst/videoscale/vs_lanczos.c:
+	* gst/videoscale/vs_scanline.c:
+	* gst/videoscale/vs_scanline.h:
+	* tests/check/Makefile.am:
+	  videoscale: port to new API
+
+2014-11-10 11:40:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: use faster saturating conversions
+	  saturating conversions are generally faster.
+
+2014-11-07 15:45:04 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-chroma: add ORC version of UP_H2_CS
+	  It is however slower than the C version and thus disabled.
+
+2014-11-09 14:44:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/descriptions.c:
+	  pbutils: add description for Apple Core Audio Format
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739840
+
+2014-11-09 12:53:32 +0100  Peter G. Baum <peter@dr-baum.net>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefind: recognize Apple Core Audio Format
+	  (CAF) Specification 1.0
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739840
+
+2014-11-09 10:47:14 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/pipelines/capsfilter-renegotiation.c:
+	  capsfilter-renegotiation: Use assertions from libcheck for more information on failures
+
+2014-11-07 12:06:10 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	* tests/check/libs/video.c:
+	  video-chroma: ORCify 2x vertical upsampling
+	  Make an ORC version of the 2x vertical upsampling code.
+	  Improve unit tests, test chroma up and down sampling.
+	  memset buffer in conversion to make valgrind happy.
+
+2014-11-06 14:14:22 +0000  William Manley <will@williammanley.net>
+
+	* gst/tcp/gstmultihandlesink.c:
+	* gst/tcp/gsttcpserversink.c:
+	  tcpserversink: Don't leak a `GSocket` and a `GInetSocketAddress`
+	  when accepting a connection.
+	  Discovered by `make check-valgrind` with the new `socketintegrationtest`.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739544
+
+2014-11-03 01:08:27 +0000  William Manley <will@williammanley.net>
+
+	* tests/check/Makefile.am:
+	* tests/check/pipelines/.gitignore:
+	* tests/check/pipelines/tcp.c:
+	  tests: Add TCP pipelines test
+	  There don't seem to be any unit tests for the socket handling elements.  As
+	  I am about to attempt some refactorings I've added some basic tests which
+	  exercise some of the happy-paths in tcpclientsrc, tcpserversrc,
+	  tcpserversink and tcpclientsink.  They should let me know if I've caused
+	  serious breakage.
+	  They are far from exhaustive but are sufficient for me to have caught a few
+	  memory-leaks in the existing code.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739544
+
+2014-11-06 18:18:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/libs/video.c:
+	  tests: add video conversion test
+	  Go through all conversions and make a list of performance.
+
+2014-11-06 18:13:12 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-info.c:
+	  video-info: use h-cosited chroma for HD video by default
+
+2014-11-06 18:09:04 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: clamp lines
+
+2014-11-06 16:29:16 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video-orc: update disted files
+
+2014-11-06 16:18:25 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: ORCify 8<->16 conversion
+
+2014-11-06 15:30:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: unpack into the destination when needed
+	  Make sure we write into the destination line when we can propose the
+	  dest allocator.
+
+2014-11-06 15:29:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add more debug
+
+2014-11-06 15:01:27 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: Update disted orc files
+
+2014-11-06 13:08:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	* gst-libs/gst/video/video-orc.orc:
+	* tests/check/libs/video.c:
+	  video-chroma: optimize chroma subsampling a little
+	  Combine multiplies in 4x filters.
+	  Rename conversion functions to make them nicer in orc.
+	  Add ORC versions for various downsampling algorithms
+	  Add unit test chroma resampler
+
+2014-11-06 10:43:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/libs/video.c:
+	  tests: make pack/unpack test
+	  Make a more complete pack/unpack test, check if the image after
+	  pack/unpack has the same color and precision, and has correctly
+	  duplicated subsampled pixels.
+
+2014-11-06 10:42:09 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/libs/video.c:
+	  tests: get the correct number of video formats
+	  Make a method to get the number of formats (including the last one).
+
+2014-11-06 09:44:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.h:
+	  video-format: update some docs and add a FIXME(2.0)
+
+2014-11-06 09:38:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: add range extension to BGR_10XE format
+
+2014-11-06 09:34:59 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-format: fix pack of 4:2:0 formats
+	  When packing 4:2:0 formats, we need to take the chroma from the even
+	  lines, for the odd lines we only take luminance.
+
+2014-11-06 09:32:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: fix range extension of UYVP
+	  We need to shift the top 6 bits to the lower 6 bits
+
+2014-11-06 09:28:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	  video-chroma: do h subsampling after v subsampling
+	  We only need to do the horizontal subsampling on 1 line if we do it
+	  after vertical subsampling and we also avoid doing vertical subsampling
+	  on unused pixels.
+
+2014-11-06 09:39:08 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/Makefile.am:
+	  tests: dist header file needed for ABI checks on powerpc32
+	  Fixes 'make check' on debian powerpc32 buildbot:
+	  libs/libsabi.c:95:26: fatal error: struct_ppc32.h: No such file or directory
+
+2014-11-05 04:34:44 +0900  Danny Song <danny.song.ga@gmail.com>
+
+	* tests/check/elements/adder.c:
+	  test : fix leaks in adder unit test
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739640
+
+2014-11-05 11:54:31 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: keep separate lines with border
+	  Make separate with a border around them so that we can avoid a memcpy.
+
+2014-11-05 11:52:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: avoid memcpy when not needed
+
+2014-11-05 11:51:44 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: pass output line correctly
+
+2014-11-04 09:30:45 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: rework the converter to allow more optimizations
+	  Rework the converter, keep track of the conversion steps by chaining the
+	  cache objects together. We can then walk the chain and decide the
+	  optimal allocation pattern.
+	  Remove the free function, we're not going to need this anytime soon.
+	  Keep track of what output line we're constructing so that we can let the
+	  allocator return a line directly into the target image when possible.
+	  Directly read from the source pixels when possible.
+
+2014-11-04 11:03:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix temp line allocation
+	  We need to allocate the templine with the amount of pixels we are going
+	  to handle, which we only know for the vertical resampler when we are
+	  asked to resample.
+
+2014-11-04 11:02:49 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix taps in interlaced mode
+
+2014-11-04 11:01:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix phases in interlaced mode
+
+2014-11-04 09:29:58 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: fix v_2tap_u16
+
+2014-11-03 16:18:41 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add extra pixels for the border
+	  We need extra pixels for the border.
+
+2014-11-03 15:36:26 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add support for 16bits formats
+	  Add scaler functions for 16 bits formats.
+	  Rename the scaler functions so that 16bits versions don't look too
+	  weird.
+	  Remove old unused h_2tap functions
+	  Fix v_ntap functions, it was using 1 tap too little.
+
+2014-11-03 15:33:24 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: Add support for 16 bits formats
+	  Rework the way we track the current state of the video through the
+	  different conversion phases and use this to make sure we use the right
+	  format and pstride where needed.
+
+2014-10-22 13:37:40 +0100  William Manley <will@williammanley.net>
+
+	* gst-libs/gst/allocators/gstdmabuf.c:
+	  docs: gst_dmabuf_allocator_alloc: Improve documentation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739545
+
+2014-11-03 10:07:56 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	  video-orc: comment out unused function
+	  A faster version of 4tap horizontal scaling causes segfaults in ORC
+	  presumably because it uses too many registers so disable it to avoid
+	  crashing in the ORC tests.
+
+2014-11-02 21:45:30 +0100  Andreas Frisch <fraxinas@opendreambox.org>
+
+	* gst/playback/gstsubtitleoverlay.c:
+	  subtitleoverlay: return available factory CAPS instead of ANY on CAPS query
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739536
+
+2014-11-03 08:12:44 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: Fix compiler warning
+	  video-scaler.c:151:58: error: implicit conversion from enumeration type
+	  'GstVideoScalerFlags' to different enumeration type
+	  'GstVideoResamplerFlags' [-Werror,-Wenum-conversion]
+	  gst_video_resampler_init (&scale->resampler, method, flags, out_size,
+	  ~~~~~~~~~~~~~~~~~~~~~~~~                             ^~~~~
+
+2014-11-01 20:08:01 +0000  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/rtp/gstrtpbuffer.c:
+	  rtp: Do not use deprecated gtk-doc 'Rename to' tag
+	  GObject introspection GTK-Doc tag "Rename to" has been deprecated, changing to
+	  rename-to annotation.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739514
+
+2014-11-01 14:58:13 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	  video: fix some g-i / gtk-doc warnings
+
+2014-11-01 14:47:26 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: update disted orc backup functions
+	  Fixes build without orc.
+
+2014-11-01 14:28:55 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/video-blend.c:
+	  video: add video blend helper functions to docs
+	  I don't think those were ever meant to be made public,
+	  but they are, so we might as well document them.
+
+2014-11-01 13:14:32 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: ORCify vertical ntap function
+
+2014-11-01 12:58:01 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: handle 4tap interlaced
+
+2014-10-31 16:53:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video-orc: update dist files
+
+2014-10-31 16:49:43 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add ORC optimized ntap horizontal scalers
+
+2014-10-29 16:28:28 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* tests/icles/playback/test.c:
+	* tests/icles/playback/test2.c:
+	* tests/icles/playback/test4.c:
+	  tests/playback: quit from main loop
+	  Listen for eos and error signal to quit main loop.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739346
+
+2014-10-29 16:26:07 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* tests/icles/playback/test2.c:
+	* tests/icles/playback/test4.c:
+	  tests/playback: correct state change checking
+	  Correct the test apps check if result of state change is not failure as the
+	  state change can happen async
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739346
+
+2014-10-31 22:52:43 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: Update disted orc files for new functions.
+	  Fixes the build when building without ORC
+
+2014-10-31 11:07:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: align offsets to subsampling
+	  Only apply an offset that is a multiple of the subsampling. To handle
+	  arbitrary offsets in the future, we need to be able to chroma-resample
+	  part of the borders.
+
+2014-10-31 10:38:15 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: clamp output lines
+
+2014-10-31 10:34:46 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-format.c:
+	  video-format: add alignment checks
+	  Some of the ORC functions need specific alignment
+
+2014-10-31 10:33:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix offset check
+
+2014-10-30 18:41:01 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: also chroma up/downsample when scaling
+
+2014-10-30 18:40:43 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: clamp input lines correctly
+
+2014-10-30 23:53:39 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: fix build without orc
+	  https://bugzilla.gnome.org/show_bug.cgi?id=739433
+
+2014-10-30 17:30:33 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add border color
+
+2014-10-30 16:57:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: add support for src/dest regions
+	  Add support for cropping the source and placing the converted image
+	  into a rectangle in the destination frame.
+	  Add an option to add a border and border color.
+
+2014-06-05 14:50:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/vorbis/gstvorbisenc.c:
+	  vorbisenc: push an updated segment stop time when we know it
+	  When encoding, libvorbis will tell us how many samples are encoded
+	  in the buffer it returns. This number may be less than the maximum
+	  of samples in the block, if this is the last packet. In we have no
+	  segment end time, we set it to the end time of that last sample to
+	  tell downstream that the buffer contains less samples.
+
+2014-06-05 14:54:31 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggmux.c:
+	  oggmux: set correct granpos on last page when samples are clipped
+	  Samples may be clipped at the end, and this is conveyed by a
+	  granulepos that's smaller than it would otherwise be. Use the
+	  segment stop time to detect this, and calculate the right
+	  granulepos.
+
+2014-06-05 11:26:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	* ext/ogg/gstoggdemux.h:
+	  oggdemux: fix last buffer timestamp when samples are clipped
+	  The end of a stream can be clipped by setting the granulepos of
+	  the last page to a lower value that it otherwise would be.
+
+2014-10-30 14:48:45 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/libs/video.c:
+	  tests: fix test
+
+2014-10-03 12:42:46 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* tools/gst-discoverer.c:
+	  gst-discoverer: error out on failure to copy
+	  This should not really fail, but let's check return value
+	  anyway as it guards against future changes.
+	  Coverity 1135731
+
+2014-10-03 12:28:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst-libs/gst/rtp/gstrtpbuffer.c:
+	  rtpbuffer: add a const where appropriate
+
+2014-10-03 12:08:05 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefind: remove unneeded test
+	  We've already bailed out if we have less than 5 bytes.
+	  Coverity 1226441
+
+2014-10-30 11:33:17 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* win32/common/libgstvideo.def:
+	  Update libgstvideo.def for resampler -> video_resample renaming
+
+2014-10-30 11:46:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add more ORC functions
+	  Add the old ORC functions for nearest and linear. Label them as Low
+	  quality because they are not as accurate but ORC lacks opcodes to
+	  express this for now.
+
+2014-10-30 11:43:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/resampler.c:
+	* gst-libs/gst/video/resampler.h:
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-resampler.c:
+	* gst-libs/gst/video/video-resampler.h:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	  video-scaler: rename resampler to video-resampler
+	  Prefix the resampler with video-. It we would like to reuse the
+	  resampler for audio later, we can copy/move it and deprecate this
+	  one.
+
+2014-10-29 17:38:33 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	  video-scaler: remove color range argument
+	  We just need to clip to the format limits, if there is extra headroom in
+	  the range we can use that without problems.
+
+2014-10-29 17:14:51 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* win32/common/libgstvideo.def:
+	  defs: update defs
+
+2014-10-29 16:20:56 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add ORC optimized versions
+	  Add ORC optimized versions of 2 and 4tap vertical scaling. Provide
+	  a high quality 12 bits and a low quality 6 bits version.
+
+2014-10-29 16:13:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-scaler.c:
+	  video-scaler: add precision to make_s16_taps
+
+2014-10-29 13:19:00 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: copy config fields
+	  When setting a new config, copy all the fields into our own config and
+	  not only the ones we know about.
+
+2014-10-29 13:17:39 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/resampler.c:
+	* gst-libs/gst/video/resampler.h:
+	* gst-libs/gst/video/video-scaler.c:
+	  resampler: make offset/phase/n_taps uint32
+	  Make various resizer fields uint32 so that we can use them in ORC
+	  functions later.
+
+2014-10-27 11:59:14 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: don't convert too much
+	  Always convert the smallest width.
+
+2014-10-27 10:13:47 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/resampler.c:
+	* gst-libs/gst/video/video-scaler.c:
+	* tests/check/libs/video.c:
+	  resampler: make shift easier to use
+
+2014-10-26 05:58:56 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/resampler.c:
+	* gst-libs/gst/video/resampler.h:
+	* gst-libs/gst/video/video-converter.c:
+	  resampler: add parameters to cubic filter
+	  Improve cubic filter and add parameters. Switch to mitchell filter
+	  by default.
+
+2014-10-24 16:51:37 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	* tests/check/libs/video.c:
+	  video-scaler: add extra options
+
+2014-10-24 16:42:11 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	  video-converter: define some options
+
+2014-10-24 16:23:53 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/resampler.c:
+	* gst-libs/gst/video/resampler.h:
+	  resampler: add some options
+
+2014-10-24 15:42:31 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/resampler.c:
+	  resampler: limit max number of taps
+	  Don't use more taps than the input size.
+
+2014-10-24 15:28:22 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: add scaling support
+	  Add scaling support for the video-converter object
+
+2014-10-24 15:25:33 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/video-scaler.c:
+	* gst-libs/gst/video/video-scaler.h:
+	* gst-libs/gst/video/video.h:
+	* tests/check/libs/video.c:
+	  video-scaler: add video scaler helper object
+	  Add a video scaler object build on top of the resampler. It has
+	  implementation to deal with interlaced video as well as horizontal and
+	  vertical scaling functions.
+
+2014-10-24 13:01:12 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/resampler.c:
+	* gst-libs/gst/video/resampler.h:
+	  video: add generic resampler
+	  Add an object that can generate a set of resample filter coefficients.
+
+2014-10-24 12:11:43 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  video-converter: rework the generic converter function
+	  Use a LineCache object to track and process lines between unpack,
+	  upsample, convert, downsample and pack stages. This simplifies the
+	  main core processing function a lot and allows for future additions
+	  easily.
+	  Add support for interlaced formats in chroma up and downsampling.
+
+2014-10-24 11:45:13 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	* gst/videoconvert/gstvideoconvert.c:
+	  video-convert: swap src and dest
+	  It is more natural and consistent with other uses.
+
+2014-10-24 11:35:31 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	  video-chroma: fix typo
+
+2014-10-27 17:56:51 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From 84d06cd to 7bb2bce
+
+2014-10-23 14:41:13 +0530  Vineeth T M <vineeth.tm@samsung.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  video-blend: segfault when xpos >= video size
+	  When the xpos is given as greater than or equal to the video size,
+	  we get a segfault, due to improper condition.
+	  Hence adding proper conditions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738984
+
+2014-10-23 14:38:07 +0530  Vineeth T M <vineeth.tm@samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: segfault when xpos >= video size
+	  When the xpos is given as greater than or equal to the video size,
+	  we get a segfault, due to improper condition.
+	  Hence adding proper conditions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738984
+
+2014-10-26 21:31:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/examples/app/.gitignore:
+	  examples: add new appsink example to .gitignore
+
+2014-10-26 11:04:47 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  Revert "decodebin: fix the autoplugging of parser elements"
+	  This reverts commit 2b0d3927410ae24e6b0fce100bd4ebbbe805a66f.
+	  This breaks cases where an actual second parser is required after the parser,
+	  e.g. to do timestamp corrections.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-26 11:04:38 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  Revert "decodebin: Fix locking"
+	  This reverts commit aa94d5dc9aa6ef381da6b60a67f218117c662958.
+
+2014-10-24 13:09:42 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/playbin-complex.c:
+	  tests: fix playbin-complex test on big endian
+
+2014-10-24 13:04:07 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/libs/struct_ppc32.h:
+	  tests: fix expected GstRTSPTimeRange structure size for ABI test for ppc32
+	  Also see https://bugzilla.gnome.org/show_bug.cgi?id=695276
+
+2014-10-24 12:26:40 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/adder.c:
+	  tests: fix adder check on big-endian
+
+2014-10-24 10:17:47 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* android/rtsp.mk:
+	* gst-libs/gst/rtsp/.gitignore:
+	* gst-libs/gst/rtsp/Makefile.am:
+	* gst-libs/gst/rtsp/gstrtsp-marshal.list:
+	* gst-libs/gst/rtsp/gstrtspextension.c:
+	  rtsp: use generic marshaller
+
+2014-10-23 11:22:35 +0200  Thibault Saunier <tsaunier@gnome.org>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: Make GstBaseTextOverlay::font-desc readable
+
+2014-10-21 13:01:16 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From a8c8939 to 84d06cd
+
+2014-10-21 13:30:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Fix locking
+	  The chain mutex needs to be locked when looking at chain->elements. Move code
+	  around a bit to require only one lock() and unlock().
+
+2014-10-21 12:58:41 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: fix the autoplugging of parser elements
+	  If there are two parser elements available for the same media format,
+	  then decodebin is autoplugging an extra capsfilter and parser irrespective
+	  of caps and rank. So restrict the decodebin from autoplugging multiple parser
+	  elements back to back in adjacent positions with in a single DecodeChain
+	  for the same media format.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-21 12:57:59 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* README:
+	* common:
+	  Automatic update of common submodule
+	  From 6e75498 to a8c8939
+
+2014-10-21 14:43:30 +0530  Vineeth T M <vineeth.tm@samsung.com>
+
+	* gst/videotestsrc/gstvideotestsrc.c:
+	* gst/videotestsrc/gstvideotestsrc.h:
+	  videotestsrc: assertion error
+	  timestamp_offset is being declared as an int64 variable,
+	  for which the min
+	  value of G_MININT64 is -9223372036854775808
+	  Changing the minimum and maximum limit for the offset variable.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738568
+
+2014-10-13 00:03:55 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: optimize the code a bit by avoiding unnecessary string comparisons
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-13 00:03:20 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Fix typo in comment
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738416
+
+2014-10-01 15:04:09 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	  rtspconnection: call watch notify before freeing any watch resources
+	  This gives control to the notify function allowing it to finish other
+	  watch related functionality.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737752
+
+2014-10-20 15:31:29 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/app/gstappsink.c:
+	  appsink: Fix gst_app_sink_pull() docs to transfer full for the return value
+	  Also we get a GstSample, not a GstBuffer here.
+
+2014-10-17 12:10:44 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefind: use gslice for typefine data
+	  Also use our free function in the failure case.
+
+2014-10-13 15:58:56 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/encoding/gstencodebin.c:
+	  encodebin: fix some leaks in error code path
+	  Fixes test_encodebin_sink_pads_nopreset_static
+	  running under valgrind.
+
+2014-10-13 05:08:41 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* Makefile.am:
+	* common:
+	  tests: parallelise 'make valgrind'
+	  Use $(MAKE) instead of 'make' inside the Makefile,
+	  otherwise the make will run as if -j1 had been
+	  specified and complain about the job server not
+	  being available, and with $(MAKE) in inherits the
+	  parent make's settings it seems.
+	  Upgrade common submodule for parallel check-valgrind.
+
+2014-10-03 12:57:52 +0200  Peter G. Baum <peter@dr-baum.net>
+
+	* gst-libs/gst/riff/riff-media.c:
+	  riff-media: allow more channel_masks
+	  Allow partial valid channel masks.
+	  Set channel mask to 0 for non-valid channel masks.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733405
+
+2014-10-03 12:54:17 +0200  Peter G. Baum <peter@dr-baum.net>
+
+	* gst-libs/gst/audio/audio-channels.c:
+	  audio-channels: allow partially valid channel_mask
+	  Since WAVEFORMATEXTENSIBLE allows to have more channels than
+	  bits in the channel mask we should allow this, too, to avoid
+	  loss of information.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733405
+
+2014-10-13 22:24:31 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: should post DECODE errors and not ENCODE
+	  Fix error code for audio decoder
+
+2014-10-10 18:49:29 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst-libs/gst/video/video-blend.c:
+	  videoblend: Avoid assigning a negative value to a guint
+	  There are some few but certain conditions where it is possible for the
+	  dest_width to be smaller than x. So we check this before assigning a negative
+	  value to src_width, which is a unsigned and would be promoted to a number that
+	  can segfault videoblend.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738242
+
+2014-10-10 10:05:19 +0530  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: Fix segfault when overlay outside the frame
+	  When the textoverlay is set outside the video frame by deltax or deltay the
+	  calculation segfaults, but it is also  unnecessary since it doesn't need to be
+	  displayed. So we should clip the text.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738242
+
+2014-10-10 17:32:41 -0400  Olivier Crête <olivier.crete@ocrete.ca>
+
+	* gst-libs/gst/pbutils/missing-plugins.c:
+	  pbutils: Rename clock-base/seqnum-base to timestamp-offset/seqnum-offset
+	  To match how they were renamed elsewhere.
+
+2014-10-10 12:14:17 +0300  Heinrich Fink <hfink@toolsonair.com>
+
+	* gst/playback/gstplaysink.c:
+	  playsink: Use correct property enum value for video-filter property installation
+
+2014-10-08 16:50:52 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	  videoscale: remove FIXME about NV21 support
+	  NV21 is already supported so removing FIXME about adding support for it.
+
+2014-10-08 11:26:24 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videotestsrc/gstvideotestsrc.c:
+	* gst/videotestsrc/gstvideotestsrc.h:
+	* gst/videotestsrc/videotestsrc.c:
+	* gst/videotestsrc/videotestsrc.h:
+	  videotestsrc: add gradient pattern
+	  Makes a gradient between background and foreground color.
+
+2014-10-06 15:17:42 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-chroma.c:
+	  video-chroma: improve 4x downsampling coefficients
+
+2014-10-06 22:13:00 +0200  Peter G. Baum <peter@dr-baum.net>
+
+	* gst/audioresample/gstaudioresample.h:
+	  audioresample: remove unused variables
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738026
+
+2014-10-07 05:50:56 +0900  Danny Song <danny.song.ga@gmail.com>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefindfunctions: Remove leftover #define from 0.10
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738018
+
+2014-10-07 12:10:42 +0400  Andrei Sarakeev <sarakusha@gmail.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Only emit the drain signal for the main decode chain, not any subchains
+	  https://bugzilla.gnome.org/show_bug.cgi?id=738064
+
+2014-10-06 10:15:13 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Free factories array when delaying autoplugging due to non-final caps
+
+2014-10-06 10:11:05 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	  videoconverter: Free the converter config in free()
+
+2014-10-02 21:20:48 +0200  Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: unref decode pad after usage
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737757
+
+2014-10-04 23:09:19 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: Stop storing if we received EOS
+	  This was never reset when going from PAUSED->READY and resulted
+	  in encoders being not reusable after EOS. They just rejected any
+	  buffer because they received EOS in their previous life.
+	  The flag wasn't used anywhere except for rejecting buffers after
+	  EOS, and this is now handled by GstPad directly.
+
+2014-10-02 00:14:03 +0200  Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+	* ext/vorbis/gstvorbisdeclib.c:
+	  vorbisdec: don't reorder streams with channels count greater than eight
+	  vorbis_reorder_map is defined for eight channels max. If we have more
+	  than eight channels, it's the application which shall define the order.
+	  Since we set audio position to none, we just interleave all the channels
+	  without any particular reordering.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737742
+
+2014-03-04 16:51:11 +0200  Andres Gomez <agomez@igalia.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Removed setting "iradio-mode" property in the source element
+	  The "iradio-mode" property used to have a default FALSE value in HTTP
+	  source elements but now it should default to TRUE or just do not exist
+	  as a property so it is not really needed to set it any more in
+	  uridecodebin.
+	  Apart from that this code could've never worked as uridecodebin looks for a
+	  string-typed iradio-mode property, but it's a boolean in all sources.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725383
+
+2014-10-02 02:46:58 +1000  Jan Schmidt <jan@centricular.com>
+
+	* docs/design/part-stereo-multiview-video.markdown:
+	  design: Add a proposal for handling stereoscopic 3D and multiview
+
+2014-10-01 11:16:30 +0200  Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: release frame in finish_frame when no output state is configured
+	  Otherwise, frame is leaked.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737706
+
+2014-09-25 17:32:32 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	* gst-libs/gst/video/video-orc.orc:
+	  video-converter: add orc optimized matrix8 function
+	  Add an ORC implementation of the matrix8 function.
+	  Regenerate video-orc-dist.[ch]
+
+2014-09-29 19:45:22 +0530  Arun Raghavan <arun@accosted.net>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audio: Fix up a comment in GstAudioBaseSink
+	  Rewrote the comment to not be PulseAudio-specific.
+
+2014-09-27 20:05:38 +0200  Rico Tzschichholz <ricotz@ubuntu.com>
+
+	* gst-libs/gst/video/Makefile.am:
+	  video: Make sure to link against libm
+
+2014-09-27 15:58:51 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/xvimage/xvimagepool.c:
+	* sys/xvimage/xvimagepool.h:
+	  xvimagesink: get rid of unnecessary private struct for pool
+
+2014-09-27 15:53:43 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/ximage/ximagepool.c:
+	* sys/ximage/ximagepool.h:
+	  ximagesink: get rid of unnecessary private struct for pool
+	  This is not exposed as API after all.
+
+2014-09-24 20:38:31 +0530  Arun Raghavan <arun@accosted.net>
+
+	* gst-libs/gst/audio/gstaudioiec61937.c:
+	  audio: Trivial comment for unhandled MPEG-2 payloading case
+	  The spec mentions a version of the MPEG-2 frame with a base frame and
+	  extension frame. I don't have IEC 13818-3 to figure out what that is,
+	  and don't see any references in search results, so it's a FIXME for now.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736797
+
+2014-09-24 20:11:49 +0530  Arun Raghavan <arun@accosted.net>
+
+	* gst-libs/gst/audio/gstaudioiec61937.c:
+	  audio: Fixes for MPEG-2 LSF IEC61937 payloading
+	  The low sample frequency case for MPEG-2 is <=12kHz (the 32kHz number
+	  applies to MPEG-1).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736797
+
+2014-09-17 17:40:04 +0530  Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+	* gst-libs/gst/audio/gstaudioiec61937.c:
+	  audio: correct condition for MPEG case.
+	  Signed-off-by: Anuj Jaiswal <anuj.jaiswal@samsung.com>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736797
+
+2014-09-26 18:14:11 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-orc.orc:
+	  video: improve YUV -> RGB conversion
+	  Reorganize orc instructions to free up some registers.
+	  We can reuse the ORC code to implement the generic AYUV->ARGB matrix.
+
+2014-09-26 16:35:51 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/videotestsrc/gstvideotestsrcorc.orc:
+	  videotestsrc: storel is better then copyl
+	  It is better to use storel to splat the variable into the destination.
+	  ORC doesn't know when a variable is last written to so it can't yet optimize
+	  away the copy operation.
+
+2014-09-26 15:00:12 +0100  Luis de Bethencourt <luis.bg@samsung.com>
+
+	* gst/videoscale/vs_lanczos.c:
+	  videoscale: avoid recalculating values
+	  Avoid recalculating values used multiple times as base of index. Plus some style
+	  fixes.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737400
+
+2014-09-26 09:14:51 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	* gst/videoscale/vs_image.h:
+	* gst/videoscale/vs_lanczos.c:
+	  videoscale: support lanczos method for NV formats
+	  Support lanczos scaling method for NV12 and NV21 formats.
+	  Scale the 'Y' plane and scale 'NV' plane.
+	  Implementation for submethods - int16, int32, float and double
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737400
+
+2014-09-25 15:19:21 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/video/video-orc-dist.c:
+	* gst-libs/gst/video/video-orc-dist.h:
+	  video: update disted orc backup files
+
+2014-09-24 16:19:30 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/video-converter.c:
+	* gst-libs/gst/video/video-converter.h:
+	* gst-libs/gst/video/video-convertor.c:
+	* gst-libs/gst/video/video-convertor.h:
+	* gst-libs/gst/video/video.h:
+	* gst/videoconvert/gstvideoconvert.c:
+	* gst/videoconvert/gstvideoconvert.h:
+	* win32/common/libgstvideo.def:
+	  video: convertor -> converter
+
+2014-09-24 15:49:42 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/Makefile.am:
+	* gst-libs/gst/video/video-convertor.c:
+	* gst-libs/gst/video/video-convertor.h:
+	* gst-libs/gst/video/video-orc.orc:
+	* gst-libs/gst/video/video.h:
+	* gst/videoconvert/Makefile.am:
+	* gst/videoconvert/gstcms.c:
+	* gst/videoconvert/gstcms.h:
+	* gst/videoconvert/gstvideoconvert.c:
+	* gst/videoconvert/gstvideoconvert.h:
+	* gst/videoconvert/gstvideoconvertorc-dist.c:
+	* gst/videoconvert/gstvideoconvertorc-dist.h:
+	* gst/videoconvert/gstvideoconvertorc.orc:
+	* gst/videoconvert/videoconvert.c:
+	* gst/videoconvert/videoconvert.h:
+	* tests/check/Makefile.am:
+	* win32/common/libgstvideo.def:
+	  video: move videoconvert code to video library
+	  Move the conversion code used in videoconvert to the video library
+	  and expose a simple but generic API to do arbitrary conversion. It can
+	  currently do colorspace conversion but the plan is to add videoscale to
+	  it as well.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=732415
+
+2014-09-24 11:04:15 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/video-color.c:
+	* gst-libs/gst/video/video-color.h:
+	* gst/videoconvert/videoconvert.c:
+	* win32/common/libgstvideo.def:
+	  video-color: add gst_video_color_matrix_get_Kr_Kb()
+	  Move the function to get the color matrix coefficients from
+	  videoconvert to the video library.
+
+2014-09-23 14:14:36 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst-libs/gst/audio/gstaudiosink.c:
+	  audiosink: compensate for segment restart with clock's time_offset
+	  When playing chained data the audio ringbuffer is released and
+	  then acquired again. This makes it reset the segbase/segdone
+	  variables, but the next sample will be scheduled to play in
+	  the next position (right after the sample from the previous media)
+	  and, as the segdone is at 0, the audiosink will wait the duration
+	  of this previous media before it can write and play the new data.
+	  What happens is this:
+	  pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0
+	  it will have to wait the length of 698 samples before being able to write.
+	  In a regular sample playback it looks like:
+	  pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0
+	  In this case it will write to the next available position and it
+	  doesn't need to wait or fill with silence.
+	  This solution is borrowed from pulsesink that resets the clock to
+	  start again from 0, which makes it reset the time_offset to the time
+	  of the last played sample. This is used to correct the place of
+	  writing in the ringbuffer to the new start (0 again)
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737055
+
+2014-09-21 13:16:43 +0200  Ognyan Tonchev <otonchev@gmail.com>
+
+	* gst-libs/gst/video/gstvideopool.c:
+	  videopool: add missing annotation for gst_video_buffer_pool_new()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=737072
+
+2014-09-23 23:12:19 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videoscale/vs_4tap.c:
+	  videoscale Use stride instead of width in more places
+
+2014-09-19 12:31:49 +0530  Sanjay NM <sanjay.nm@samsung.com>
+
+	* gst/videoscale/vs_4tap.c:
+	  videoscale: Use width instead of stride in buffer offset calculation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736944
+
+2014-09-23 11:56:33 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	  audioencoder: reshuffle code in error handling
+	  Move the assert to the error handling block at the end of the function so the
+	  the logging is still triggered. Reword the logging slightly and add another
+	  comment to hint what went wrong.
+	  Fixes #737138
+
+2014-09-22 20:15:13 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: log the timestamps if we are unhappy about them
+	  When complaining about the DTS!=PTS on keyframes log the actualy timestamps.
+
+2014-09-22 10:42:47 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/Makefile.am:
+	  tests: add orc test for videoconvert
+
+2014-09-22 10:40:01 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* tools/gst-play.c:
+	  gst-play: Fix format string compiler warning
+	  gst-play.c:92:28: error: format string is not a string literal
+	  [-Werror,-Wformat-nonliteral]
+	  len = g_vasprintf (&str, format, args);
+	  ^~~~~~
+
+2014-09-19 14:58:20 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* tests/examples/overlay/gtk-videooverlay.c:
+	  example/overlay: Specify minimum gdk version
+	  Avoids deprecation warnings (such as for gtk_widget_set_double_buffered()
+	  which became deprecated from 3.14)
+
+2014-09-19 18:29:54 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tools/gst-play.c:
+	  gst-play: add --quiet option to suppress output
+
+2014-09-05 13:49:46 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: Do not fail the negotiation if query fails
+	  The allocation query failure doesn't mean that the negotiation
+	  has failed as the element can allocate buffers itself.
+	  Instead, only fail if the pads are flushing and the allocation
+	  query failed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735844
+
+2014-09-18 15:45:43 +0530  Sanjay NM <sanjay.nm@samsung.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	* gst/videoscale/vs_4tap.c:
+	* gst/videoscale/vs_4tap.h:
+	  videoscale: Added NV support for 4Tap resize
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736845
+
+2014-09-18 12:29:37 +0400  Andrei Sarakeev <sarakusha@gmail.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Don't leak input-selector sinkpads
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736861
+
+2014-09-18 12:39:48 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Simplify code a bit
+
+2014-09-17 14:34:25 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/encoding/gststreamsplitter.c:
+	  streamsplitter: do not leak events when flushing them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 14:18:49 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	  audioencoder: do not leak events when flushing them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 14:11:21 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: do not leak events when flushing them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 14:08:17 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: do not leak events when flushing them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736796
+
+2014-09-17 12:17:27 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* tests/check/libs/audiodecoder.c:
+	  audiodecoder: extend flush_events test to check for event leaks
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736788
+
+2014-09-17 12:17:53 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Don't leak events
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736788
+
+2014-09-16 13:32:52 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/audio/gstaudiocdsrc.c:
+	  audiocdsrc: do not leak uid after parsing TOC select event
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736739
+
+2014-09-17 10:51:59 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefind: correct the condition for irap flag
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736779
+
+2014-09-16 21:42:46 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaysink.c:
+	  playsink: Add audio/videoconvert in front of the audio/video-filters
+	  audioresample and videoscale is something the application will have to do if
+	  required, but we can at least help here by adding the
+	  audioconvert/videoconvert elements.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735748
+
+2014-09-16 01:07:18 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-frame.c:
+	  video-frame: Don't ref buffers twice when mapping
+
+2014-09-16 00:41:55 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/app/gstappsink.h:
+	* gst-libs/gst/app/gstappsrc.h:
+	  app: Add FIXME comment for making the instance/class structs private
+
+2014-09-15 21:51:15 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/app/gstappsrc.h:
+	  appsrc: fix recent ABI breakage caused by GstAppSrc structure size increase
+	  Also fixes 'make check'.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=728379
+
+2014-09-15 16:23:57 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: do not leak pool and allocator in error case
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736679
+
+2014-09-12 14:41:01 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideofilter.c:
+	  videofilter: Use new GST_VIDEO_FRAME_MAP_FLAG_NO_REF
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736118
+
+2014-09-12 14:39:16 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/video-frame.c:
+	* gst-libs/gst/video/video-frame.h:
+	  video-frame: Add GST_VIDEO_FRAME_MAP_FLAG_NO_REF
+	  This makes sure that the buffer is not reffed another time when
+	  storing it in the GstVideoFrame, keeping it writable if it was
+	  writable.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736118
+
+2014-09-12 14:27:44 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideofilter.c:
+	  videofilter: Unref buffers before calling the transform_frame functions
+	  GstVideoFrame has another reference, so the buffer looks unwriteable,
+	  meaning that we can't attach any metas or anything to it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736118
+
+2014-09-05 09:54:10 -0700  Garg <aksg86@gmail.com>
+
+	* gst-libs/gst/audio/gstaudiobasesink.c:
+	  audiobasesink: Fix deadlock caused by holding object lock while calling clock functions
+	  Issue:
+	  During a PAUSED->PLAYING transition when we are rendering an audio buffer in AudioBaseSink
+	  we make adjustments to the sink's provided clock i.e. fix clock calibration using the external
+	  pipeline clock, within "gst_audio_base_sink_sync_latency function inside gstaudiobasesink.c".
+	  For the calibration adjustment we need to get the sink clock time using "gst_audio_clock_get_time".
+	  But before calling "gst_audio_clock_get_time" we acquire the Object Lock on the Sink. If sink is
+	  a pulsesink, "gst_audio_clock_get_time" internally calls "gst_pulsesink_get_time" which needs to
+	  acquire Pulse Audio Main Loop Lock before querying Pulse Audio for its stream time using
+	  "pa_stream_get_time". Please see "gst_pulsesink_get_time in pulsesink.c".
+	  So the situation here is we have acquired the Object lock on Sink and need PA Main Loop Lock.
+	  Now Pulse Audio Main Thread itself might be in the process of posting a stream status
+	  message after Paused to Playing transition which in turn acquires the PA Main loop lock and
+	  needs the Object Lock on Pulse Sink. This causes a deadlock with the earlier render thread.
+	  Fix:
+	  Do not acquire the object Lock on Sink before querying the time on PulseSink clock. This is
+	  similar to the way we have used get_time at other places in the code. Acquire it after the
+	  get_time call. This way PA Main loop will be able to post its stream status message by
+	  acquiring the Sink Object lock and will eventually release its Main Loop lock needed for
+	  gst_pulsesink_get_time to continue.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736071
+
+2014-09-04 11:56:50 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* tests/examples/app/Makefile.am:
+	* tests/examples/app/appsink-src2.c:
+	  appsrc: Add example that shows gst_app_src_push_sample() usage
+
+2014-09-05 11:14:51 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/app/gstappsrc.c:
+	* gst-libs/gst/app/gstappsrc.h:
+	* win32/common/libgstapp.def:
+	  appsrc: Add push_sample() convenience function for easy appsink -> appsrc use
+	  https://bugzilla.gnome.org/show_bug.cgi?id=728379
+
+2014-09-11 22:19:05 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/xvimage/xvcontext.c:
+	* sys/xvimage/xvcontext.h:
+	  xvimagesink: only try to set XV_ITURBT_709 port attribute if it exists
+	  Don't try to set port attribute that's not advertised by the
+	  adaptor. Fixes videotestsrc ! xvimagesink aborting with
+	  X Error of failed request:  BadMatch (invalid parameter attributes)
+	  Major opcode of failed request:  151 (XVideo)
+	  Minor opcode of failed request:  13 ()
+	  on intel HD4600 graphics with kernel 3.16, xserver 1.15,
+	  intel driver 2.21.15.
+
+2014-09-11 16:58:35 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: protect buffering message handling
+	  Use the object lock to avoid concurrent processing which leads
+	  to small disasters (assertions or crashes)
+
+2014-09-09 11:37:26 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	  rtspconnection: ignore timeout in session request header
+	  The timeout parameter is only allowed in a session response header
+	  but some clients, like Honeywell VMS applications, send it as part
+	  of the session request header. Ignore everything from the semicolon
+	  to the end of the line when parsing session id.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736267
+
+2014-03-28 13:02:54 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: filter out buffering messages when switching uri
+	  When switching URI from about-to-finish, playbin starts decoding the new
+	  URI and the queue2 inside uridecodebin starts emitting buffering messages
+	  immediately. However, the queue(s) inside playsink still have buffers to
+	  play and the pipeline doesn't need to pause for buffering, so we should
+	  not send those buffering messages up to the application, otherwise there
+	  is an audible glitch caused by pausing the pipeline for a very short time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727255
+
+2014-07-08 12:37:41 -0400  Kipp Cannon <kipp.cannon@ligo.org>
+
+	* gst/audioresample/resample.c:
+	  audioresample: don't skip input samples
+	  when downsampling, the output buffer can be filled before all the input
+	  samples are consumed.  this is correct:  when downsampling, several input
+	  samples are needed for each output sample, so when only a small number of
+	  input samples are available the number of output samples produced can be 0.
+	  the resampler, however, was discarding those extra input samples instead of
+	  clocking them into its filter history for the next iteration.  this patch
+	  fixes this by removing the check that the output buffer is full.  the code
+	  now always loops until all input samples are consumed, and relies on the
+	  calling code to have provided a suitably sized location for the output.
+	  note that there are already other checks in place in the calling code to
+	  ensure that this is the case.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=732908
+
+2013-01-31 13:49:00 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: get framerate from previously parsed video info
+
+2013-01-31 13:47:35 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: do not ask for a bufferpool when checking for composition meta
+
+2014-09-04 15:06:31 +0200  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: schedule reconfigure on source pad when negotiation fails
+	  The source pad might be flushing while negotiating, resulting in
+	  set_caps or the ALLOCATION query failing. In this case set the
+	  reconfigure flag on the source pad so that negotiation is retried on the
+	  next buffer.
+
+2013-01-31 15:38:18 +0100  Arnaud Vrac <avrac@freebox.fr>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: just forward the seek event to sink pads like other events
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735844
+
+2014-09-04 12:13:45 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: remove unneeded cairo transparence setting
+	  he code here:
+	  http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/pango/gstbasetextoverlay.c#n1554
+	  should make transparent the box that contains the text, I think this code is
+	  not correct, it should be:
+	  if (overlay->want_shading) {
+	  double alpha = overlay->shading_value / 255.0;
+	  cairo_paint_with_alpha (cr, alpha);
+	  }
+	  however I think this code could be removed, we already do a shaded background,
+	  why shade the box behind the text with cairo too? only one shading is needed so
+	  we must shade with cairo or with methods like these:
+	  http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/pango/gstbasetextoverlay.c#n1642
+	  not both
+	  https://bugzilla.gnome.org/show_bug.cgi?id=736028
+
+2014-09-02 13:10:34 +0200  Nicola Murino <nicola.murino@gmail.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: Make shading_value a property
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735879
+
+2014-09-03 15:23:26 +0530  Vineeth T M <vineeth.tm@samsung.com>
+
+	* gst/videorate/gstvideorate.c:
+	  videorate: GstStructure refcount critical message
+	  s3 is not being initialized when run in a loop
+	  and the same was being freed, which resulted in the crash
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735952
+
+2014-09-02 15:37:38 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Also include the raw caps in the error message, not just the human readable description
+
+2014-09-02 12:59:18 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Include codec description for missing plugins in the error message
+	  If we had plugins and an error occurred we only include the error message
+	  caused by this, otherwise we will include the codec description as generated
+	  from the caps.
+	  This allows to detect which exact codec was missing instead of getting a
+	  generic "no suitable decoders found" error message.
+
+2014-09-01 15:23:27 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/elements/textoverlay.c:
+	  tests: textoverlay: add test to reproduce fakesink scenario
+	  Adds a new test to textoverlay to make sure it can properly handle
+	  elements that have ANY caps but fail to add the overlay meta in
+	  the allocation query.
+	  This test verifies that textoverlay won't use the caps features even
+	  knowing that the overlay meta is accepted when querying the downstream
+	  caps because it also needs downstream to confirm by putting the meta
+	  in the allocation query.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735800
+
+2014-09-01 12:38:02 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: properly fallback to non-overlay caps
+	  When downstream claims to accept the overlay meta but fails to
+	  provide it in the allocation query, properly fallback to setting
+	  a new caps without the overlay meta as that is not going to be used.
+	  Only do this if the original caps doesn't have the overlay already,
+	  otherwise there isn't much that can be done.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735800
+
+2014-09-01 15:06:51 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: don't set segment.base in pad_submit_packet()
+	  Setting segment.base in the segment sent from gst_ogg_demux_handle_page() is
+	  enough to ensure that chained oggs are played corretly (see bgo#706569).
+	  Tweaking the base in gst_ogg_pad_submit_packet() as well result in delays when
+	  playing a file with start != -1.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735808
+
+2014-09-01 12:28:24 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  textoverlay: Don't hold any mutexes while calling negotiate
+	  It's not done in any other code calling negotiate and will cause deadlocks
+	  as it is sending events and queries in the pipeline.
+	  Specifically this pipeline was deadlocking:
+	  gst-launch-1.0 videotestsrc ! textoverlay ! textoverlay ! fakesink
+
+2014-08-29 14:00:06 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: accumulate base time
+	  Base time should be accumulated so non flushing seeks have the expected base.
+	  Not accumulating result in segments appearing as "too late" and so are not
+	  played by the sink.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735509
+
+2014-08-29 19:15:56 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  textoverlay: remove code that can't be reached
+	  If this code could ever be reached, it would leak
+	  memory (CID 1231978), but gst_caps_get_features()
+	  never returns NULL, so that can't happen.
+
+2014-08-29 18:18:10 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/encoding/gstencodebin.c:
+	  encoding: remove assignment that's no longer needed
+	  CID 1231980
+
+2014-07-23 21:25:24 +0200  Peter G. Baum <peter@dr-baum.net>
+
+	* gst-libs/gst/riff/riff-ids.h:
+	* gst-libs/gst/riff/riff-read.c:
+	  riff: Recognize RF64 as RIFF file
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735631
+
+2014-08-27 13:45:57 +0200  Göran Jönsson <goranjn@axis.com>
+
+	* gst-libs/gst/rtsp/gstrtspconnection.c:
+	  rtspconnection: Protect readsrc, writesrc and controllsrc with a mutex
+	  Fixes a crash when controlsrc, readsrc or writesrc are modified from
+	  gst_rtsp_source_dispatch_read/write and gst_rtsp_watch_reset at the
+	  same time.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=735569
+
+2014-08-28 17:13:05 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaysinkconvertbin.c:
+	  playsinkconvertbin: setcaps() always returns TRUE and the return value is unused
+	  Change it to a void return value. The caps are forwarded afterwards via
+	  gst_pad_event_default() and not inside this function.
+	  CID 1226477
+
+2014-08-28 17:06:22 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: Fix broken boolean expression
+	  We can seek with end_type==NONE and end_type==SET && end_position=-1. The
+	  check for end_type!=NONE made the second condition impossible.
+	  CID 1226440
+
+2014-08-28 17:00:26 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Fix broken boolean expression
+	  We can seek with end_type==NONE and end_type==SET && end_position=-1. The
+	  check for end_type!=NONE made the second condition impossible.
+	  CID 1226439
+
+2014-08-25 20:59:40 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	* gst/playback/gsturidecodebin.c:
+	  decodebin: Include information from the error messages of tried but failed elements in the missing plugin errors
+
+2014-08-25 16:22:46 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Initialize local variables for every retry
+
+2014-08-25 15:15:06 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Remove error case that resulted in two error messages
+	  We already send one in gst_decode_bin_expose() for this case. Only
+	  if we're unable to typefind the caps another error message is needed.
+
+2014-08-24 22:36:59 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/typefind/gsttypefindfunctions.c:
+	  typefinding: tighten checks for 'freeform mp3' a little
+	  Freeform mp3s typically have bitrates higher than the
+	  otherwise max allowed rate. Prevents misdetection of
+	  some truetype font files as mp3.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=732923
+
+2014-08-25 13:14:36 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Don't ignore ::start/stop return values
+
+2014-08-18 13:04:31 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-plugins-base.spec.in:
+	  spec: add gst-device-monitor-1.0 to RPM .spec file
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734944
+
+2014-08-14 16:57:01 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/playback/gstplaysinkconvertbin.c:
+	  playsinkconvertbin: only intersect with the filter at the end
+	  Otherwise we might change some capsfeatures from ANY to the specific
+	  value from the filter and do not filter those out in case the
+	  sink doesn't support them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734822
+
+2014-08-15 13:31:53 +0200  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst-libs/gst/pbutils/gstdiscoverer.c:
+	  discoverer: Set 'processing = FALSE' when done discovering SYNC
+	  This avoids a race where we would get new tag but we are already
+	  prerolled and analyzing results.
+	  It is the way it is supposed to be handled as stated in comment:
+	  "If preroll is complete, drop these tags - the collected information is
+	  possibly already being processed and adding more tags would be racy"
+
+2014-08-14 17:21:44 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* win32/common/libgstvideo.def:
+	  gstvideo: add missing entry to win32 .def
+	  gst_video_guess_framerate
+
+2014-08-14 23:53:16 +1000  Jan Schmidt <jan@centricular.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/video.c:
+	* gst-libs/gst/video/video.h:
+	  video: Add gst_video_guess_framerate() function
+	  Takes a nominal frame duration and returns a standard
+	  FPS if it matches closely enough (< 0.1%), or else
+	  calculates a framerate that'll do.
+
+2014-08-15 01:04:45 +1000  Jan Schmidt <jan@centricular.com>
+
+	* docs/libs/gst-plugins-base-libs-sections.txt:
+	* gst-libs/gst/video/gstvideometa.h:
+	* gst-libs/gst/video/gstvideoutils.h:
+	* gst-libs/gst/video/video-format.c:
+	* gst-libs/gst/video/video-frame.h:
+	* gst-libs/gst/video/video-overlay-composition.c:
+	  video: Various simple docs fixes
+
+2014-08-08 20:01:20 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	* gst-libs/gst/video/gstvideodecoder.h:
+	  videodecoder: Reset last_timestamp_out on new segment
+	  Reset last_timestamp_out when applying the output segment
+	  change, to avoid decoder confusion over new timestamp timelines when
+	  a seamless segment change happens.
+	  Move some locks/unlocks to later when they're actually needed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734617
+
+2014-07-14 12:29:50 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: handle group switching for deadend group
+	  Gracefully handle switching groups that all pads are deadend.
+	  This can happen when quickly switching programs on mpegts as the
+	  output is unaligned it can happen that not enough data was accumulated at
+	  parsers to generate any buffers, causing the stream to receive EOS before
+	  any data can be decoded.
+	  To handle this scenario, the _expose function now also gets if there is
+	  any next group to be exposed along with the list of endpads. If there are
+	  no endpads and there is another group to expose it will switch to this next
+	  group and then retry exposing the streams.
+	  Also, the requirement to only switch from the chain that has the endpad had
+	  to be modified to care for when the drainpad is NULL
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733169
+
+2014-07-11 18:51:44 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: consider all deadend pads as drained
+	  Otherwise when switching out a group with a deadend pad it will block
+	  as it would be waiting for EOS on a deadend that already got one
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733169
+
+2014-08-12 13:41:04 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: fix caps negotiation filter
+
+2014-08-13 14:28:05 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaysinkconvertbin.c:
+	  playsinkconvertbin: Make sure to intersect raw caps with our converter caps
+	  Otherwise we end up allowing video/x-raw with arbitrary caps features that are
+	  not handled by our converters.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734683
+
+2014-08-12 23:18:57 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Don't drain and flush on SEGMENT events.
+	  As was done for the base video decoder in commit 695675, don't
+	  flush out the decoder on a new SEGMENT event. Segment events
+	  may be a new segment, but are also often segment updates for
+	  the current segment where the old data should be kept. For new
+	  segments, a STREAM_START event will already trigger a drain, but
+	  make sure to flush any remaining partial data then as well.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734666
+
+2014-08-11 10:15:14 +0530  Sanjay NM <sanjay.nm@samsung.com>
+
+	* gst/videoscale/gstvideoscale.c:
+	  videoscale: Add NV21 support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734650
+
+2014-08-11 18:21:26 +0200  Matthieu Crapet <mcrapet@gmail.com>
+
+	* tests/icles/playback/decodetest.c:
+	* tests/icles/playback/test.c:
+	* tests/icles/playback/test5.c:
+	  tests: fix decodebin signal used in icles/playback/ decodetest, test and test5
+	  Since release 1.1.4, "new-decoded-pad" no longer exists.
+
+2014-08-08 12:46:47 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	* tests/check/elements/textoverlay.c:
+	  basetextoverlay: rework caps negotiation
+	  Make textoverlay negotiate caps more correctly.
+	  1) Check what caps we received in the video-sink
+	  2) If it already has the overlay meta -> use it directly
+	  3) If it doesn't, textoverlay try adding the overlay meta and using it,
+	  if downstream doesn't support it, just use what is received in the
+	  video-sink
+	  4) Check if the allocation query also supports the meta to enable
+	  really using it
+	  Before it wasn't really doing renegotiation of any kind, just
+	  re-checking if it should use the overlay meta or not
+	  Also had to update the caps in the test as memory:SystemMemory seems
+	  to be required when you use a caps feature otherwise intersection/subset
+	  checks will fail.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733916
+
+2014-08-07 17:35:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* ext/pango/gstbasetextoverlay.c:
+	  basetextoverlay: always intersect with the filter caps
+	  Avoids returning values that upstream can't produce
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733916
+
+2014-07-30 16:59:15 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* gst/encoding/gstencodebin.c:
+	* tests/check/elements/encodebin.c:
+	  encodebin: delay missing encoder error as passthrough is still possible
+	  Set up a fakesink with a pad probe to replace the missing encoder to detect
+	  if encoding was really required and only error out in this case. Otherwise
+	  just let passthrough branch work.
+	  This delays the error posting from the set_state function to when buffers
+	  are really flowing. Unit test updated accordingly
+	  https://bugzilla.gnome.org/show_bug.cgi?id=650652
+
+2014-08-11 10:57:43 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Remove buffering special casing for adaptive streaming demuxers
+	  They output smaller buffers now and we should be able to handle the buffering
+	  limits like in every other situation now.
+
+2014-08-07 10:44:03 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: Don't set decoding timestamps on raw video
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733720
+
+2014-08-07 18:10:41 +0300  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: In reverse playback, flush the output queue after decoding each keyframe chain
+	  This fixes the reverse playback scenario when upstream is not fully
+	  parsing the stream and does not send every keyframe chain separately
+	  with the DISCONT flag on the keyframe.
+	  To explain this, let's suppose we have this stream:
+	  0 1 2 3 4 5 6 7 8
+	  K     K     K
+	  In most circumstances, the upstream parser will chain in the
+	  decoder the buffers in the following order:
+	  6 7 8 3 4 5 0 1 2
+	  D     D     D
+	  In this case, GstVideoDecoder will flush the parse queue every time
+	  it receives discont (D) and we will eventually get in the output queue:
+	  (flush here) 8 7 6  (flush here) 5 4 3 (flush here) 2 1 0
+	  In case the upstream parser doesn't do this work, though,
+	  GstVideoDecoder will receive the whole stream at once and will flush
+	  the parse queue afterwards:
+	  0 1 2 3 4 5 6 7 8
+	  D
+	  During the flush, it will look backwards for keyframes and will
+	  decode in this order:
+	  6 7 8 3 4 5 0 1 2
+	  This is the same order that it would receive from upstream if
+	  upstream was parsing and looking for the keyframes, only that now
+	  there is no flushing of the output queue in between keyframes,
+	  which will result in the output queue looking like this:
+	  2 1 0 6 5 3 8 7 6
+	  This will confuse downstream obviously and will play incorrectly.
+	  This patch forces the decoder to flush the output queue every time
+	  it picks a new keyframe to decode, so it will end up decoding 6 7 8
+	  and then flushing before picking 3 for decoding, so the output will
+	  get 8 7 6 before 6 5 3 and the video will play back correctly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734441
+
+2014-08-10 17:30:18 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* configure.ac:
+	  configure: use pkg-config to detect x11 and xv libs
+	  AC_PATH_XTRA macro unnecessarily pulls in libSM and libICE.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=731047
+
+2014-08-10 17:27:14 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/xvimage/xvimageallocator.c:
+	  xvimage: fix crash when outputting debug log
+	  Can't print a GstMemory via GST_PTR_FORMAT, it will crash
+	  inside GObject checking if it's a GObject, and we can't
+	  check generically whether it's a derived GstMemory type,
+	  as boxed types don't allowe derivation.
+
+2014-08-09 14:14:48 +0200  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* gst-libs/gst/audio/gstaudioencoder.c:
+	  audioencoder: Mark caps argument as not being transferred
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734540
+
+2014-08-09 14:20:32 +0200  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* ext/vorbis/gstvorbisenc.c:
+	  vorbisenc: Improve annotation of internal function
+	  https://bugzilla.gnome.org/show_bug.cgi?id=734541
+
+2014-08-06 13:41:46 +0200  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* tests/check/elements/appsrc.c:
+	* tests/examples/app/appsink-src.c:
+	* tests/examples/audio/audiomix.c:
+	* tests/examples/audio/volume.c:
+	* tests/examples/dynamic/codec-select.c:
+	* tests/examples/seek/scrubby.c:
+	* tests/examples/snapshot/snapshot.c:
+	* tests/icles/stress-videooverlay.c:
+	* tests/icles/test-textoverlay.c:
+	  tests: Add missing unrefs of objects after use
+	  Unreffing the objects returned by gst_bin_get_by_name() and
+	  gst_pipeline_get_use() were missing in several tests, so add these.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734359
+
+2014-08-06 13:22:56 +0200  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* ext/ogg/gstoggdemux.c:
+	  oggdemux: Unref peer pad after use in error case
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734350
+
+2014-08-06 10:07:42 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/app/gstappsrc.c:
+	  appsrc: Some minor fixes and cleanup
+
+2014-08-06 09:59:32 -0400  Wang Xin-yu (王昕宇) <comicfans44@gmail.com>
+
+	* gst-libs/gst/app/gstappsrc.c:
+	  appsrc: Make caps set action queued together with buffer
+	  https://bugzilla.gnome.org/show_bug.cgi?id=729760
+
+2014-08-01 15:00:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Keep a reference to the playsink sinkpads
+	  Otherwise playsink might get shut down without us noticing
+	  that our pad references are gone now.
+	  Probably fixes https://bugzilla.gnome.org/show_bug.cgi?id=733165
+
+2014-07-30 20:53:53 +0300  Mohammed Sameer <msameer@foolab.org>
+
+	* gst/playback/gststreamsynchronizer.c:
+	  streamsynchronizer: don't unset DISCONT flag
+	  Unsetting DISCONT flag means we need to copy the buffer. This copy operation
+	  mandates that all GstMemory should be copy-able which is not always the case
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727409
+
+2014-07-31 18:40:59 +0200  Edward Hervey <edward@collabora.com>
+
+	* Makefile.am:
+	* common:
+	  Makefile: Add usage of build-checks step
+	  Allows building checks without running them
+
+2014-07-31 16:09:41 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* tests/check/libs/rtpbasedepayload.c:
+	* tests/check/libs/rtpbasepayload.c:
+	  check: Fix include path of rtp checks
+	  Fixes make distcheck
+
+2014-07-30 15:23:39 +0200  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst-libs/gst/pbutils/gstdiscoverer.c:
+	  pbutils: discoverer: Always set the pipeline back to NULL after an error
+	  Otherwize the pipeline would be in an wrong state and on the next
+	  iteration any kind of error could happen
+	  Everytime an error happens in a pipeline the application has to set the
+	  pipeline back to NULL instead of READY.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733976
+
+2014-07-29 14:20:42 -0300  Thiago Santos <ts.santos@osg.sisa.samsung.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: add missing 'time' word to debug message
+	  It prints the buffers, bytes and time limits, but 'time' was missing
+	  from the string.
+
+2014-07-28 16:56:08 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Pass through NO_PREROLL state change returns
+	  Fixes playback of live pipelines.
+
+2014-07-28 16:55:17 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Pass through NO_PREROLL state change returns
+	  Fixes playback of live pipelines.
+
+2014-07-26 14:52:01 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: fix 'attempt to unlock mutex that was not locked' in error code path
+	  Fixes playbin unit test with latest GLib.
+
+2014-07-08 16:59:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/video/gstvideoencoder.c:
+	  videoencoder: Don't delay set_format
+	  This prevent implementing allocation query, as the format need to be
+	  known in order to determin the size and number of buffers needed.
+	  Note: This may lead to few regressions that will need fixing
+	  https://bugzilla.gnome.org/show_bug.cgi?id=732288
+
+2014-07-23 19:51:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Don't unref caps for which we don't own a reference... get one first
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733615
+
+2014-07-23 12:36:15 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Go asynchronously from READY to PAUSED
+	  We now add all our elements to uridecodebin *after*
+	  GstBin::change_state(READY->PAUSED), so we need to post async-start
+	  and async-done messages ourselves if we want to work async.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733495
+
+2014-07-23 12:27:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Go asynchronously from READY to PAUSED
+	  We now add all our elements to uridecodebin *after*
+	  GstBin::change_state(READY->PAUSED), so we need to post async-start
+	  and async-done messages ourselves if we want to work async.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733495
+
+2014-07-21 15:54:05 +0300  Vivia Nikolaidou <n.vivia@gmail.com>
+
+	* tools/gst-discoverer.c:
+	  discoverer: Pretty-print topology tags
+	  Call the code used in properties for topology tags too.
+	  Side-effect achieved: more tags printed, buffers (e.g. images) shortened.
+
+2014-07-21 13:53:17 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tools/gst-discoverer.c:
+	  discoverer: Fix code style a bit
+	  if (...)
+	  one_line;
+	  else if (...) {
+	  many_lines;
+	  } else
+	  one_line;
+	  looks a bit confusing.
+
+2014-07-21 13:48:31 +0300  Vivia Nikolaidou <n.vivia@gmail.com>
+
+	* tools/gst-discoverer.c:
+	  discoverer: prettier image tag printing
+	  Rather than dumping the serialized sample value, the code now
+	  prints the number of bytes in the buffer, then the caps in a
+	  human-readable format.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733482
+
+2014-07-10 12:39:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/audio/gstaudiodecoder.c:
+	  audiodecoder: Handle CAPS events immediately instead of delaying them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733147
+
+2014-07-11 21:51:05 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/video/gstvideodecoder.c:
+	  videodecoder: Handle CAPS events immediately instead of delaying them
+	  https://bugzilla.gnome.org/show_bug.cgi?id=733147
+
+2014-07-15 17:34:01 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/playbin.c:
+	  playbin: Fix unit test for last change
+	  It will successfully asynchronously go to PAUSED now and
+	  later fail.
+
+2014-07-15 17:23:24 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gsturidecodebin.c:
+	  uridecodebin: Create new sources after chaining up to the parent class
+	  Otherwise we start the new sources already before the parent class
+	  got ready to start.
+
+2014-07-15 17:20:05 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstplaybin2.c:
+	  playbin: Create new sources after chaining up to the parent class
+	  Otherwise we start the new sources already before the parent class
+	  got ready to start.
+
+2014-07-10 16:26:08 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/playbin-complex.c:
+	  playbin-complex: Change template name from %d to the more common %u
+
+2014-07-10 16:24:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Link Parser/Converter directly and already connect to pad-added and other signals before setting elements to PAUSED
+	  otherwise we're going to
+	  a) start Parser/Converter before they are linked to their capsfilter,
+	  breaking their negotiation of a proper stream format
+	  b) start demuxers without having connected to their pad-added signals. We
+	  miss pads and in the worst case don't link any pads at all
+
+2014-07-10 12:51:22 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Send sticky events to the new element after setting it to PAUSED
+	  ... and if this fails for whatever reason we skip the element and instead
+	  try with the next element. This allows us to handle elements that fail
+	  when setting caps on them by just skipping to the next alternative element.
+
+2014-07-10 12:50:17 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Only link elements further after setting them to PAUSED
+	  They might fail to go to PAUSED, and when connecting them further
+	  we might already expose their srcpads on decodebin if we're unlucky.
+	  This prevents us to handle failures going to PAUSED gracefully.
+
+2014-07-10 12:22:35 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Remove ERROR message filter after we set the element to PAUSED
+	  This allows us to catch more errors gracefully and switch to an alternative
+	  element instead.
+
+2014-07-10 12:17:52 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Only continue autoplugging once the pad has final caps
+	  If the caps query returned us fixed caps this doesn't mean yet
+	  that these caps are actually complete (fields might be missing).
+	  It allows to do us some decisions, but the selection of the next
+	  element should be delayed as only complete caps allow proper selection
+	  of the next element.
+
+2014-07-10 12:03:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/playback/gstdecodebin2.c:
+	  decodebin: Consider the caps after the capsfilter after parsers for autoplugging
+	  Otherwise we might try to continue autoplugging e.g. for a specific
+	  stream-format although the parser could convert to something else, thus giving
+	  us potentially less options for decoders.
+
+2014-07-21 00:17:38 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst-libs/gst/pbutils/missing-plugins.c:
+	  pbutils: fix missing plugin description for missing elements
+	  CID: 1226445
+
+2014-07-19 18:04:35 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
 === release 1.4.0 ===
 
-2014-07-19  Sebastian Dröge <slomo@coaxion.net>
+2014-07-19 17:04:57 +0200  Sebastian Dröge <sebastian@centricular.com>
 
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
 	* configure.ac:
-	  releasing 1.4.0
+	* docs/plugins/gst-plugins-base-plugins.args:
+	* docs/plugins/inspect/plugin-adder.xml:
+	* docs/plugins/inspect/plugin-alsa.xml:
+	* docs/plugins/inspect/plugin-app.xml:
+	* docs/plugins/inspect/plugin-audioconvert.xml:
+	* docs/plugins/inspect/plugin-audiorate.xml:
+	* docs/plugins/inspect/plugin-audioresample.xml:
+	* docs/plugins/inspect/plugin-audiotestsrc.xml:
+	* docs/plugins/inspect/plugin-cdparanoia.xml:
+	* docs/plugins/inspect/plugin-encoding.xml:
+	* docs/plugins/inspect/plugin-gio.xml:
+	* docs/plugins/inspect/plugin-ivorbisdec.xml:
+	* docs/plugins/inspect/plugin-libvisual.xml:
+	* docs/plugins/inspect/plugin-ogg.xml:
+	* docs/plugins/inspect/plugin-pango.xml:
+	* docs/plugins/inspect/plugin-playback.xml:
+	* docs/plugins/inspect/plugin-subparse.xml:
+	* docs/plugins/inspect/plugin-tcp.xml:
+	* docs/plugins/inspect/plugin-theora.xml:
+	* docs/plugins/inspect/plugin-typefindfunctions.xml:
+	* docs/plugins/inspect/plugin-videoconvert.xml:
+	* docs/plugins/inspect/plugin-videorate.xml:
+	* docs/plugins/inspect/plugin-videoscale.xml:
+	* docs/plugins/inspect/plugin-videotestsrc.xml:
+	* docs/plugins/inspect/plugin-volume.xml:
+	* docs/plugins/inspect/plugin-vorbis.xml:
+	* docs/plugins/inspect/plugin-ximagesink.xml:
+	* docs/plugins/inspect/plugin-xvimagesink.xml:
+	* gst-plugins-base.doap:
+	* win32/common/_stdint.h:
+	* win32/common/config.h:
+	  Release 1.4.0
+
+2014-07-19 16:27:43 +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:
+	  Update .po files
 
 2014-07-18 21:19:03 -0400  Youness Alaoui <kakaroto@kakaroto.homelinux.net>
 
diff --git a/NEWS b/NEWS
index 14b25e8df..710670b91 100644
--- a/NEWS
+++ b/NEWS
@@ -1,145 +1,2 @@
-This is GStreamer Base Plugins 1.4.0
+This is GStreamer Base Plugins 1.5.1
 
-Changes since 1.2:
-
-New API:
- • GstMessageType has GST_MESSAGE_EXTENDED added. All types before
-   that can be used together as a flags type as before, but from
-   that message onwards the types are just counted incrementally.
-   This was necessary to be able to add more message types.
-   In 2.0 GstMessageType will just become an enum and not a flags
-   type anymore.
- • GstDeviceMonitor for device probing, e.g. to list all available
-   audio or video capture devices. This is the replacement for
-   GstPropertyProbe from 0.10.
- • Events accumulate the running-time offset now when travelling
-   through pads, as set by the gst_pad_set_offset() function. This
-   allows to compensate for this in the QOS event for example.
- • GstBuffer has a new flag "tag-memory" that is set automatically
-   when memory is added or removed to a buffer. This allows buffer
-   pools to detect if they can recycle a buffer or need to reset
-   it first.
- • GstToc has new API to mark GstTocEntries as loops.
- • A not-authorized resource error has been defined to notify
-   applications that accessing the resource has failed because
-   of missing authorization and to distinguish this case from others.
-   This change is actually already in 1.2.4.
- • GstPad has a new flag "accept-intersect", that will let the default
-   ACCEPT_CAPS query handler do an intersection instead of subset check.
-   This is interesting for parser elements that can handle incomplete
-   caps.
- • GstCollectPads has support for flushing and a default handler for
-   SEEK events now.
- • New GstFlowAggregator helper object that simplifies handling of
-   flow returns in elements with multiple source pads. Additionally
-   GstPad now always stores the last flow return and provides an
-   API to retrieve it.
- • GstSegment has new API to offset the running time by a specific
-   value and this is used in GstPad to allow positive and negative
-   offsets in gst_pad_set_offset() in all situations.
- • Support for h265/HEVC and VP8 has been added to the codec utils and codec
-   parsers library, and was integrated into various elements.
- • API for adjusting the TLS validation of RTSP connection has been added.
- • The RTSP and SDP library has MIKEY (RFC 3830) support now, and
-   there is API to distinguish between the different RTSP profiles.
- • API to access RTP time information and statistics.
- • Support for auxiliary streams was added to rtpbin.
- • Support for tiled, raw video formats has been added.
- • GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
-   events and merge custom tags into them consistently.
- • GstBufferPool has support for flushing now.
- • playbin/playsink has support for application provided audio and video
-   filters.
- • GstDiscoverer has new and simplified API to get details about missing
-   plugins and information to pass to the plugin installer.
- • The GL library was merged from gst-plugins-gl to gst-plugins-bad,
-   providing a generic infrastructure for handling GL inside GStreamer
-   pipelines and a plugin with some elements using these, especially
-   a video sink. Supported platforms currently are Android, Cocoa (OS X),
-   DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
-   Wayland and EGL platforms.
-   This replaces eglglessink and also is supposed to replace osxvideosink.
- • New GstAggregator base class in gst-plugins-bad. This is supposed to
-   replace GstCollectPads in the future and fix long-known shortcomings
-   in its API. Together with the base class some elements are provided
-   already, like a videomixer (compositor).
-
-
-Major changes:
- • New plugins and elements:
-   ∘ v4l2videodec element for accessing hardware codecs on
-     platforms that make them accessible via V4L2, e.g.
-     Samsung Exynos. This comes together with major refactoring
-     of the existing V4L2 elements and the corresponding
-     infrastructure.
-     The v4l2videodec element replaces the mfcdec element.
-   ∘ New downloadbuffer element that replaces the download
-     buffering feature of queue2. Compared to queue2's code
-     it is much simpler and only for this single use case.
-     A noteworthy new feature is that it's downloading gaps
-     in the already downloaded stream parts when nothing else
-     is to be downloaded.
-     This is now used by playbin when download buffering is
-     enabled.
-   ∘ rtpstreampay and rtpstreamdepay elements for transmitting
-     RTP packets over a stream API (e.g. TCP) according to
-     RFC 4571.
-   ∘ rtprtx elements for standard compliant implementation of
-     retransmissions, integrated into the rtpmanager plugin.
-   ∘ audiomixer element that mixes multiple audio streams together
-     into a single one while keeping synchronization. This is
-     planned to become the replacement of the adder element.
-   ∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
-   ∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
-   ∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
-   ∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
-   ∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
-     are available on OS X and iOS now.
-
- • Other changes:
-   ∘ gst-libav now uses libav 10.2, and gained support for H265/HEVC.
-   ∘ Support for hardware codecs and special memory types has been
-     improved with bugfixes and feature additions in various plugins
-     and base classes.
-   ∘ Various bugfixes and improvements to buffering in queue2 and
-     multiqueue elements.
-   ∘ dvbsrc supports more delivery mechanisms and other features
-     now, including DVB S2 and T2 support.
-   ∘ The MPEGTS library has support for many more descriptors.
-   ∘ Major improvements to tsdemux and tsparse, especially time and
-     seeking related.
-   ∘ souphttpsrc now has support for keep-alive connections,
-     compression, configurable number of retries and configuration
-     for SSL certificate validation.
-   ∘ hlsdemux has undergone major refactoring and works more
-     reliable now and supports more HLS features like trick modes.
-     Also fragments are pushed downstream while they're downloaded
-     now instead of waiting for each fragment to finish.
-   ∘ dashdemux and mssdemux are now also pushing fragments downstream
-     while they're downloaded instead of waiting for each fragment to
-     finish.
-   ∘ videoflip can automatically flip based on the orientation tag.
-   ∘ openjpeg supports the OpenJPEG2 API.
-   ∘ waylandsink was refactored and should be more useful now. It also
-     includes a small library which most likely is going to be removed
-     in the future and will result in extensions to the GstVideoOverlay
-     interface.
-   ∘ gst-rtsp-server supports SRTP and MIKEY now.
-   ∘ gst-libav encoders are now negotiating any profile/level settings
-     with downstream via caps.
-   ∘ Lots of fixes for coverity warnings all over the place.
-   ∘ Negotiation related performance improvements.
-   ∘ 800+ fixed bug reports, and many other bug fixes and other
-     improvements everywhere that had no bug report.
-
-Things to look out for:
- • The eglglessink element was removed and replaced by the glimagesink
-   element.
- • The mfcdec element was removed and replaced by v4l2videodec.
- • osxvideosink is only available in OS X 10.6 or newer.
- • On Android the namespace of the automatically generated Java class
-   for initialization of GStreamer has changed from com.gstreamer to
-   org.freedesktop.gstreamer to prevent namespace pollution.
- • On iOS you have to update your gst_ios_init.h and gst_ios_init.m in
-   your projects from the one included in the binaries if you used the
-   GnuTLS GIO module before. The loading mechanism has slightly changed.
diff --git a/RELEASE b/RELEASE
index 86eeaf625..832bc8f38 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,23 +1,17 @@
 
-Release notes for GStreamer Base Plugins 1.4.0
+Release notes for GStreamer Base Plugins 1.5.1
 
 
-The GStreamer team is pleased to announce the first release of
-the stable 1.4 release series. The 1.4 release series is adding new
-features on top of the 1.0 and 1.2 series and is part of the API and
-ABI-stable 1.x release series of the GStreamer multimedia framework.
+The GStreamer team is pleased to announce the first release of the unstable
+1.5 release series. The 1.5 release series is adding new features on top of
+the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release
+series of the GStreamer multimedia framework. The unstable 1.5 release series
+will lead to the stable 1.6 release series in the next weeks, and newly added
+API can still change until that point.
 
 
-
-Binaries for Android, iOS, Mac OS X and Windows are provided together
-with this release.
-
-
-
-The stable 1.4 release series is API and ABI compatible with 1.0.x,
-1.2.x and any other 1.x release series in the future. Compared to 1.2.x
-it contains some new features and more intrusive changes that were
-considered too risky as a bugfix.
+Binaries for Android, iOS, Mac OS X and Windows will be provided separately
+during the unstable 1.5 release series.
 
 
 
@@ -67,10 +61,154 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
 
 Bugs fixed in this release
      
-      * 733012 : playbin: *-filter properties are settable, but not gettable
-      * 733207 : POTFILES.in is out of date
-      * 733349 : encodebin: Documentation fixes and updates for GstEncodingProfile
-      * 733386 : appsrc: Leaking callback user data
+      * 742924 : decodebin: Initial decoder negotiation will always fail
+      * 749676 : playbin: failed to get end-of-stream event when visualization flag is enabled
+      * 741355 : playbin: deadlock
+      * 650652 : encodebin: missing encoder error when trying to remux
+      * 673976 : pbutils: codec description should include profile
+      * 706066 : xvimagesink: Fails to allocate large xvimages but does not declare this limitation on the caps
+      * 722316 : playbin: flac playback broken
+      * 722442 : Internal data stream error in gstoggdemux.c
+      * 723252 : testsuite failure: libs/tag - exif tag: " Conversion from character set 'utf8' to 'latin1' is not supported " 
+      * 725383 : uridecodebin doesn't need to set the " iradio-mode " property in the source element any more
+      * 726709 : playback-test: Segment seeks do not work anymore
+      * 727409 : streamsynchronizer: Invalid memory accesses when using uncopyable memory
+      * 727955 : id3v2: ignore RVA2 tags with 0 peak bits
+      * 728379 : appsink: add push_sample() convenience function for easy appsrc - > appsink use
+      * 729198 : oggdemux: add non flushing time seeking to 0 in push mode
+      * 729314 : ogg: sample-accurate decoding/encoding is broken
+      * 729760 : appsrc: Changing caps and pushing buffers is not serialized
+      * 731047 : ximagesink, xvimagesink: configure checks pull in libSM and libICE even though they are not used
+      * 732186 : videoconvert optimization
+      * 733147 : audio/video decoder base classes needlessly delay caps events
+      * 733169 : decodebin: improve deadend pads handling
+      * 733405 : riff: wrong channel mask in wav should be ignored
+      * 733482 : discoverer: Prettify tags with samples
+      * 733495 : uridecodebin/playbin: Does not properly do async state changes
+      * 733524 : ges-launch crashes with SIGABRT when using h264 encoded assets
+      * 733615 : decodebin: Changing state of a playbin pipeline intensively segfault with several formats
+      * 733720 : videodecoder: output should not have DTS
+      * 734350 : oggdemux: Unref peer pad after use in error case
+      * 734359 : tests: Add missing unrefs of objects after use
+      * 734424 : videorate: produces bogus output when framerate=0/1
+      * 734441 : videodecoder: in reverse playback, flush the output queue after decoding each keyframe chain
+      * 734540 : audioencoder: Mark caps argument as not being transferred
+      * 734541 : vorbisenc: Improve annotation of internal function
+      * 734650 : videoscale: Does not support NV21 format
+      * 734666 : audiodecoder: Don't drain and flush on SEGMENT events.
+      * 735509 : oggdemux: should accumulate segment.base
+      * 735631 : riff: Recognize RF64 as RIFF file
+      * 735808 : oggdemux: should not set segment.base in gst_ogg_pad_submit_packet()
+      * 735879 : basetetxtoverlay: make shading_value a property
+      * 736028 : basetextoverlay: cairo transparence setting not needed
+      * 736267 : rtspconnection: Be more forgiving when parsing session header in requests
+      * 736797 : audio: correct condition for MPEG case in iec61937 / SPDIF payloader
+      * 736845 : videoscale: 4Tap resize support not present for NV format
+      * 737072 : videopool: add missing annotation for gst_video_buffer_pool_new()
+      * 737138 : audioencoder: weird error handling code path
+      * 737400 : videoscale: Lanczos resizing for NV image format
+      * 737757 : decodebin: memory leak
+      * 738018 : typefind: #define gst_type_find_peek is not needed any more
+      * 738026 : audioresample: struct GstAudioResample has unused variables
+      * 738131 : playbin: Bogus results from GST_STATE_NULL (audio-)sink
+      * 738242 : textoverlay: segfault when trying to position text outside of the video frame
+      * 738416 : decodebin: Don't plug multiple parsers one after another
+      * 738568 : videotestsrc: assertion failed error
+      * 738984 : basetextoverlay: segfault for min/max values of element properties
+      * 739346 : playback-test: correct the test apps
+      * 739433 : video: recent video-resampler addition causes build failures when building without orc
+      * 739446 : audiosink, audiosrc: fix silence for unsigned pcm formats
+      * 739536 : subtitleoverlay: return available factory caps instead of any on caps query
+      * 739545 : docs: gst_dmabuf_allocator_alloc: Improve documentation
+      * 739546 : New socketsrc element
+      * 739640 : tests : fix leaks in adder unit test
+      * 739689 : textoverlay: not rendering when x + text_width > frame_width &  & x < frame_width
+      * 740018 : videorate: Operate in a zero-latency mode if drop-only is set to TRUE
+      * 740214 : [API] encodebin: Add a way to disable caps renegotiation for output stream format
+      * 740422 : vorbisenc: Nothing encoded in some transcoding cases (regression)
+      * 740615 : alsa: warn on buffer underrun / overrun
+      * 740686 : audiodecoder: Error not handled in gst_audio_decoder_drain
+      * 740689 : decodebin/multiqueue/max-size-buffers is not set in playing state
+      * 740690 : Timeoverlay: add an option to choose between stream-time and running-time.
+      * 740693 : decodebin: Analyze source pad before setting to PAUSED for 'tag demuxers'
+      * 740694 : decodebin: Take STREAM_LOCK before sending sticky events.
+      * 740798 : videoscale: Videoscale test suite fails for 4-tap method
+      * 740834 : audiobasesink: racy clock jump when renegotiating
+      * 741015 : videoconvert: Tune quality setting to not degrade performance compared to 1.4
+      * 741030 : theoradec: Sets video-meta width/height from padded values
+      * 741097 : oggdemux: Fix seeking before the first frame.
+      * 741144 : id3demux: support UTF-16 - > UTF-8 conversion on systems with crippled iconv
+      * 741187 : [regression] ProRes files show up pink
+      * 741263 : videodecoder: implement caps query
+      * 741281 : audiorate: fill gap events
+      * 741501 : videopool: should update video alignment after change it
+      * 741640 : video-converter: support AYUV border
+      * 741879 : audio/videotestsrc: Report latency in live-mode
+      * 741987 : videoscale performance regression
+      * 742006 : discoverer: _get_missing_elements_installer_details() is documented to return a copy but doesn't
+      * 742110 : video: Add support for BT2020 colorspace (UHD)
+      * 742885 : decodebin: disable pad link checks as it has already been done
+      * 743687 : playback: gstreamer-vaapi doesn't work with Totem master
+      * 743834 : tcpserversink: fails with html5 < video > client
+      * 743900 : oggdemux gets first packet timestamp wrong - theora
+      * 743980 : decodebin2: crash in analyze_new_pad
+      * 744028 : video-converter: Converter doesn't work properly when offsets are specified
+      * 744070 : oggdemux: wrong duration for ogv file
+      * 744465 : install-plugins: add _set_desktop_id(), _set_startup_notification_id() and _set_confirm_search() API
+      * 744844 : playbin: forward template and ring buffer settings to existing decodebins
+      * 745006 : video-converter: Add frame 'alpha' property to video-converter
+      * 745073 : playbin, discoverer: criticals when switching from pull mode to push mode
+      * 745174 : gst-play: support play rate change
+      * 745207 : video-converter: sometimes crashes during ARGB - > BGRx conversion.
+      * 745337 : video: RGB15/16 pack/unpack unit test failure on big endian systems
+      * 745667 : volume: Unable to set the volume with gcc-4.9 on arm platform
+      * 745719 : video-converter: doesn't work properly with YUY2 and right border
+      * 745980 : ogg video file is unable to be seeked
+      * 746150 : multisocketsink: Map `GstMemory`s individually when sending
+      * 746457 : oggdemux: don't abuse GST_ERROR()
+      * 746466 : video: add NV61 format support
+      * 746480 : playbin: deadlock on PMT change in mpeg TS stream
+      * 746661 : audioconvert: slow dithering on architectures without 64-bit integer divide (e.g. armv7)
+      * 746865 : videoencoder: Keep sticky event when reset.
+      * 746908 : appsrc: allow sample with no caps or no buffer in push_sample()
+      * 747005 : audioconvert: avoid floating point calculations when mixing integer-formatted channels
+      * 747103 : discoverer: leak when handling toc messages
+      * 747190 : videodecoder: Sends GAP events before CAPS
+      * 747245 : navigation: Post navigation events as message on the bus
+      * 747283 : configure: playback and seek tests build error with gtk < 3.10.0
+      * 747293 : audiodecoder: Add sink and src query virtual method
+      * 747517 : appsrc: negotiates twice if caps are changed before pipeline starts
+      * 747602 : basetextoverlay: Leak in gst_base_text_overlay_text_chain
+      * 747624 : decodebin unit test fails: test environment not set up correctly with automake 1.11
+      * 747692 : check build error on osx: pipelines/tcp.c:161:34: error: use of undeclared identifier 'SOCK_CLOEXEC'
+      * 747790 : videoscale method=bilinear2 and UYVY/YUY2 distortion
+      * 747841 : gio: plugin dependencies wrong or insufficient
+      * 748021 : video-converter: unused variables n_taps max_taps
+      * 748027 : rtpbasedepayload: testcase crash
+      * 748247 : oggdemux: fix event leak
+      * 748289 : audio: " delay " virt-func mixes up samples and frames
+      * 748348 : video-converter: change data type of _GstLineCache::n_lines
+      * 748413 : xmptag: valgrind errors when printing debug output
+      * 748687 : video-converter: Remove unused macro
+      * 748814 : discoverer: add serialization/deserialization methods
+      * 748820 : oggdemux: remove unnecessary codes
+      * 748903 : fix navigation event leaks
+      * 748964 : oggdemux: fix chain leak
+      * 749104 : video-converter: Change some implicit string enums to  real enums
+      * 749105 : videoconvert: Expose some properties from the videoconverter API
+      * 749528 : playbin: need to avoid duplicated flag setting
+      * 749530 : xvimagesink: fix pool leak
+      * 749632 : FTBFS when srcdir != builddir since commit bfc13c8e
+      * 749673 : discoverer: Serialize the top level DiscovererInfo
+      * 749740 : tools: gst-play: print keyboard shortcuts help in interactive mode.
+      * 749824 : basetextoverlay: make deltax and deltay properties controllable
+      * 750032 : videorate: fails to renegotiate on streams with a variable framerate
+      * 750096 : sdp: prevent the sdp message parser from reading past the end of the buffer
+      * 750325 : rtcpbuffer: Update package validation to support reduced size rtcp packets
+      * 750406 : audioconvert: copy all metadata.
+      * 738302 : videorate: Should increase minimum buffer in allocation query
+      * 739281 : video-blend: fix blending of rectangles partially or fully outside of the video
+      * 740013 : rtspconnection: There is an warning by mismatch of parameter name in header and source files
 
 ==== Download ====
 
@@ -107,10 +245,76 @@ subscribe to the gstreamer-devel list.
         
 Contributors to this release
     
+      * Aleix Conchillo Flaqué
+      * Alessandro Decina
+      * Andreas Frisch
+      * Andrei Sarakeev
+      * Andres Gomez
+      * Anuj Jaiswal
+      * Arnaud Vrac
       * Arun Raghavan
+      * Aurélien Zanelli
+      * Bernhard Miller
+      * Branislav Katreniak
+      * Chad
+      * Chihyoung Kim
+      * Claudiu Florin Lazar
+      * Danny Song
+      * David Schleef
+      * Duncan Palmer
+      * Edward Hervey
+      * Garg
+      * George Kiagiadakis
+      * Guillaume Desmottes
+      * Göran Jönsson
+      * Heinrich Fink
+      * Hyunjun Ko
+      * Ilya Konstantinov
+      * Jan Alexander Steffens (heftig)
+      * Jan Schmidt
+      * Jonathan Matthew
+      * Jose Antonio Santos Cadenas
+      * Kalev Lember
+      * Kipp Cannon
+      * Luis de Bethencourt
+      * Mark Nauwelaerts
+      * Matej Knopp
+      * Mathieu Duponchelle
+      * Matthieu Bouron
+      * Matthieu Crapet
+      * Mohammed Sameer
+      * Nicola Murino
+      * Nicolas Dufresne
       * Nirbheek Chauhan
-      * Piotr Drąg
+      * Ognyan Tonchev
+      * Olivier Crete
+      * Olivier Crête
+      * Peter G. Baum
+      * Ramiro Polla
+      * Ravi Kiran K N
+      * Rico Tzschichholz
+      * Sam Thursfield
+      * Sanjay NM
       * Sebastian Dröge
+      * Sebastian Rasmussen
+      * Song Bing
+      * Sreerenj Balachandran
+      * Stefan Sauer
+      * Thiago Santos
+      * Thibault Saunier
+      * Thomas Klausner
+      * Thomas Roos
       * Tim-Philipp Müller
-      * Youness Alaoui
+      * Vincent Penquerc'h
+      * Vineeth T M
+      * Vivia Nikolaidou
+      * Víctor Manuel Jáquez Leal
+      * Wang Xin-yu (王昕宇)
+      * William Manley
+      * Wim Taymans
+      * Wonchul Lee
+      * Young Han Lee
+      * Zaheer Abbas Merali
+      * danny song
+      * eunhae choi
  
\ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 60c3ca3a0..dd993f1b9 100644
--- a/configure.ac
+++ b/configure.ac
@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
 dnl initialize autoconf
 dnl releases only do -Wall, git and prerelease does -Werror too
 dnl use a three digit version number for releases, and four for git/prerelease
-AC_INIT([GStreamer Base Plug-ins],[1.5.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
+AC_INIT([GStreamer Base Plug-ins],[1.5.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
 
 AG_GST_INIT
 
@@ -59,7 +59,7 @@ dnl sets GST_LT_LDFLAGS
 AS_LIBTOOL(GST, 501, 0, 501)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.5.0.1
+GST_REQ=1.5.1
 
 dnl *** autotools stuff ****
 
diff --git a/docs/plugins/gst-plugins-base-plugins.args b/docs/plugins/gst-plugins-base-plugins.args
index 7f2cac746..c048be673 100644
--- a/docs/plugins/gst-plugins-base-plugins.args
+++ b/docs/plugins/gst-plugins-base-plugins.args
@@ -45,7 +45,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>method</NICK>
 <BLURB>method.</BLURB>
-<DEFAULT>Bilinear</DEFAULT>
+<DEFAULT>Bilinear (2-tap)</DEFAULT>
 </ARG>
 
 <ARG>
@@ -69,6 +69,16 @@
 </ARG>
 
 <ARG>
+<NAME>GstVideoScale::gamma-decode</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Gamma Decode</NICK>
+<BLURB>Decode gamma before scaling.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstURIDecodeBin::buffer-duration</NAME>
 <TYPE>gint64</TYPE>
 <RANGE>>= G_MAXULONG</RANGE>
@@ -855,7 +865,87 @@
 <FLAGS>rw</FLAGS>
 <NICK>Dither</NICK>
 <BLURB>Apply dithering while converting.</BLURB>
-<DEFAULT>GST_VIDEO_DITHER_NONE</DEFAULT>
+<DEFAULT>GST_VIDEO_DITHER_BAYER</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::alpha-mode</NAME>
+<TYPE>GstVideoAlphaMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Alpha Mode</NICK>
+<BLURB>Alpha Mode to use.</BLURB>
+<DEFAULT>GST_VIDEO_ALPHA_MODE_COPY</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::alpha-value</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Alpha Value</NICK>
+<BLURB>Alpha Value to use.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::chroma-mode</NAME>
+<TYPE>GstVideoChromaMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Chroma Mode</NICK>
+<BLURB>Chroma Resampling Mode.</BLURB>
+<DEFAULT>GST_VIDEO_CHROMA_MODE_FULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::chroma-resampler</NAME>
+<TYPE>GstVideoResamplerMethod</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Chroma resampler</NICK>
+<BLURB>Chroma resampler method.</BLURB>
+<DEFAULT>GST_VIDEO_RESAMPLER_METHOD_LINEAR</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::dither-quantization</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Dither Quantize</NICK>
+<BLURB>Quantizer to use.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::gamma-mode</NAME>
+<TYPE>GstVideoGammaMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Gamma Mode</NICK>
+<BLURB>Gamma Conversion Mode.</BLURB>
+<DEFAULT>GST_VIDEO_GAMMA_MODE_NONE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::matrix-mode</NAME>
+<TYPE>GstVideoMatrixMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Matrix Mode</NICK>
+<BLURB>Matrix Conversion Mode.</BLURB>
+<DEFAULT>GST_VIDEO_MATRIX_MODE_FULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVideoConvert::primaries-mode</NAME>
+<TYPE>GstVideoPrimariesMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Primaries Mode</NICK>
+<BLURB>Primaries Conversion Mode.</BLURB>
+<DEFAULT>GST_VIDEO_PRIMARIES_MODE_NONE</DEFAULT>
 </ARG>
 
 <ARG>
@@ -2408,3 +2498,23 @@
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstSocketSrc::socket</NAME>
+<TYPE>GSocket*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Socket</NICK>
+<BLURB>The socket to receive packets from.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTimeOverlay::time-mode</NAME>
+<TYPE>GstTimeOverlayTimeLine</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Time Mode</NICK>
+<BLURB>What time to show.</BLURB>
+<DEFAULT>buffer-time</DEFAULT>
+</ARG>
+
diff --git a/docs/plugins/gst-plugins-base-plugins.hierarchy b/docs/plugins/gst-plugins-base-plugins.hierarchy
index c65cd17c4..df2fd05e1 100644
--- a/docs/plugins/gst-plugins-base-plugins.hierarchy
+++ b/docs/plugins/gst-plugins-base-plugins.hierarchy
@@ -57,6 +57,7 @@ GObject
                 GstAlsaSrc
             GstAudioCdSrc
               GstCdParanoiaSrc
+            GstSocketSrc
             GstTCPClientSrc
             GstTCPServerSrc
             GstVideoTestSrc
diff --git a/docs/plugins/gst-plugins-base-plugins.signals b/docs/plugins/gst-plugins-base-plugins.signals
index 5c6b76f8e..7daf891a1 100644
--- a/docs/plugins/gst-plugins-base-plugins.signals
+++ b/docs/plugins/gst-plugins-base-plugins.signals
@@ -518,3 +518,10 @@ GstCdParanoiaSrc *gstcdparanoiasrc
 gint  arg1
 </SIGNAL>
 
+<SIGNAL>
+<NAME>GstSocketSrc::connection-closed-by-peer</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>f</FLAGS>
+GstSocketSrc *gstsocketsrc
+</SIGNAL>
+
diff --git a/docs/plugins/inspect/plugin-adder.xml b/docs/plugins/inspect/plugin-adder.xml
index 8aa81fa10..4ab215e35 100644
--- a/docs/plugins/inspect/plugin-adder.xml
+++ b/docs/plugins/inspect/plugin-adder.xml
@@ -3,10 +3,10 @@
   <description>Adds multiple streams</description>
   <filename>../../gst/adder/.libs/libgstadder.so</filename>
   <basename>libgstadder.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-alsa.xml b/docs/plugins/inspect/plugin-alsa.xml
index 1ee1c5ce4..7874b9296 100644
--- a/docs/plugins/inspect/plugin-alsa.xml
+++ b/docs/plugins/inspect/plugin-alsa.xml
@@ -3,10 +3,10 @@
   <description>ALSA plugin library</description>
   <filename>../../ext/alsa/.libs/libgstalsa.so</filename>
   <basename>libgstalsa.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-app.xml b/docs/plugins/inspect/plugin-app.xml
index 01aab7291..1484b7e3d 100644
--- a/docs/plugins/inspect/plugin-app.xml
+++ b/docs/plugins/inspect/plugin-app.xml
@@ -3,10 +3,10 @@
   <description>Elements used to communicate with applications</description>
   <filename>../../gst/app/.libs/libgstapp.so</filename>
   <basename>libgstapp.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-audioconvert.xml b/docs/plugins/inspect/plugin-audioconvert.xml
index 885cae63b..5d0f92144 100644
--- a/docs/plugins/inspect/plugin-audioconvert.xml
+++ b/docs/plugins/inspect/plugin-audioconvert.xml
@@ -3,10 +3,10 @@
   <description>Convert audio to different formats</description>
   <filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
   <basename>libgstaudioconvert.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-audiorate.xml b/docs/plugins/inspect/plugin-audiorate.xml
index fe7e16d23..02ae0e399 100644
--- a/docs/plugins/inspect/plugin-audiorate.xml
+++ b/docs/plugins/inspect/plugin-audiorate.xml
@@ -3,10 +3,10 @@
   <description>Adjusts audio frames</description>
   <filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
   <basename>libgstaudiorate.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-audioresample.xml b/docs/plugins/inspect/plugin-audioresample.xml
index d31bc81bd..2ac5693c3 100644
--- a/docs/plugins/inspect/plugin-audioresample.xml
+++ b/docs/plugins/inspect/plugin-audioresample.xml
@@ -3,10 +3,10 @@
   <description>Resamples audio</description>
   <filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
   <basename>libgstaudioresample.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-audiotestsrc.xml b/docs/plugins/inspect/plugin-audiotestsrc.xml
index 3b6bbea1f..9ebc58fd4 100644
--- a/docs/plugins/inspect/plugin-audiotestsrc.xml
+++ b/docs/plugins/inspect/plugin-audiotestsrc.xml
@@ -3,10 +3,10 @@
   <description>Creates audio test signals of given frequency and volume</description>
   <filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
   <basename>libgstaudiotestsrc.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-cdparanoia.xml b/docs/plugins/inspect/plugin-cdparanoia.xml
index 86dfb1faa..3fec2c192 100644
--- a/docs/plugins/inspect/plugin-cdparanoia.xml
+++ b/docs/plugins/inspect/plugin-cdparanoia.xml
@@ -3,10 +3,10 @@
   <description>Read audio from CD in paranoid mode</description>
   <filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
   <basename>libgstcdparanoia.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-encoding.xml b/docs/plugins/inspect/plugin-encoding.xml
index 34e500f81..704e3eec1 100644
--- a/docs/plugins/inspect/plugin-encoding.xml
+++ b/docs/plugins/inspect/plugin-encoding.xml
@@ -3,10 +3,10 @@
   <description>various encoding-related elements</description>
   <filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
   <basename>libgstencodebin.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-gio.xml b/docs/plugins/inspect/plugin-gio.xml
index bcd2cc9cc..e88cde249 100644
--- a/docs/plugins/inspect/plugin-gio.xml
+++ b/docs/plugins/inspect/plugin-gio.xml
@@ -3,10 +3,10 @@
   <description>GIO elements</description>
   <filename>../../gst/gio/.libs/libgstgio.so</filename>
   <basename>libgstgio.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-libvisual.xml b/docs/plugins/inspect/plugin-libvisual.xml
index ef0c5563b..5e7e89232 100644
--- a/docs/plugins/inspect/plugin-libvisual.xml
+++ b/docs/plugins/inspect/plugin-libvisual.xml
@@ -3,10 +3,10 @@
   <description>libvisual visualization plugins</description>
   <filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
   <basename>libgstlibvisual.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-ogg.xml b/docs/plugins/inspect/plugin-ogg.xml
index 8c18eb92b..4a23ebfd5 100644
--- a/docs/plugins/inspect/plugin-ogg.xml
+++ b/docs/plugins/inspect/plugin-ogg.xml
@@ -3,10 +3,10 @@
   <description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
   <filename>../../ext/ogg/.libs/libgstogg.so</filename>
   <basename>libgstogg.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-pango.xml b/docs/plugins/inspect/plugin-pango.xml
index fb862419d..0b72cf7a2 100644
--- a/docs/plugins/inspect/plugin-pango.xml
+++ b/docs/plugins/inspect/plugin-pango.xml
@@ -3,10 +3,10 @@
   <description>Pango-based text rendering and overlay</description>
   <filename>../../ext/pango/.libs/libgstpango.so</filename>
   <basename>libgstpango.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
@@ -20,13 +20,13 @@
           <name>video_sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
@@ -47,13 +47,13 @@
           <name>video_sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
@@ -89,13 +89,13 @@
           <name>video_sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, RGBx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, I420, YV12, AYUV, YUY2, UYVY, v308, Y41B, Y42B, Y444, NV12, NV21, A420, YUV9, YVU9, IYU1, GRAY8 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-playback.xml b/docs/plugins/inspect/plugin-playback.xml
index 87000007f..1e2dcb6a7 100644
--- a/docs/plugins/inspect/plugin-playback.xml
+++ b/docs/plugins/inspect/plugin-playback.xml
@@ -3,10 +3,10 @@
   <description>various playback elements</description>
   <filename>../../gst/playback/.libs/libgstplayback.so</filename>
   <basename>libgstplayback.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-subparse.xml b/docs/plugins/inspect/plugin-subparse.xml
index ff085e162..ec90bdca7 100644
--- a/docs/plugins/inspect/plugin-subparse.xml
+++ b/docs/plugins/inspect/plugin-subparse.xml
@@ -3,10 +3,10 @@
   <description>Subtitle parsing</description>
   <filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
   <basename>libgstsubparse.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-tcp.xml b/docs/plugins/inspect/plugin-tcp.xml
index 1a24a301e..1bafb8e93 100644
--- a/docs/plugins/inspect/plugin-tcp.xml
+++ b/docs/plugins/inspect/plugin-tcp.xml
@@ -3,10 +3,10 @@
   <description>transfer data over the network via TCP</description>
   <filename>../../gst/tcp/.libs/libgsttcp.so</filename>
   <basename>libgsttcp.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
@@ -44,7 +44,7 @@
       <longname>socket source</longname>
       <class>Source/Network</class>
       <description>Receive data from a socket</description>
-      <author>William Manley &lt;will@williammanley.net&gt;</author>
+      <author>Thomas Vander Stichele &lt;thomas at apestaart dot org&gt;, William Manley &lt;will@williammanley.net&gt;</author>
       <pads>
         <caps>
           <name>src</name>
diff --git a/docs/plugins/inspect/plugin-theora.xml b/docs/plugins/inspect/plugin-theora.xml
index 577827b9c..6ead57df1 100644
--- a/docs/plugins/inspect/plugin-theora.xml
+++ b/docs/plugins/inspect/plugin-theora.xml
@@ -3,10 +3,10 @@
   <description>Theora plugin library</description>
   <filename>../../ext/theora/.libs/libgsttheora.so</filename>
   <basename>libgsttheora.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-typefindfunctions.xml b/docs/plugins/inspect/plugin-typefindfunctions.xml
index 92d675316..a85547862 100644
--- a/docs/plugins/inspect/plugin-typefindfunctions.xml
+++ b/docs/plugins/inspect/plugin-typefindfunctions.xml
@@ -3,10 +3,10 @@
   <description>default typefind functions</description>
   <filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
   <basename>libgsttypefindfunctions.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
   </elements>
diff --git a/docs/plugins/inspect/plugin-videoconvert.xml b/docs/plugins/inspect/plugin-videoconvert.xml
index 69b8014fa..99b7a757e 100644
--- a/docs/plugins/inspect/plugin-videoconvert.xml
+++ b/docs/plugins/inspect/plugin-videoconvert.xml
@@ -3,10 +3,10 @@
   <description>Colorspace conversion</description>
   <filename>../../gst/videoconvert/.libs/libgstvideoconvert.so</filename>
   <basename>libgstvideoconvert.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
@@ -20,13 +20,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-videorate.xml b/docs/plugins/inspect/plugin-videorate.xml
index 3bbf10c55..b8a4c55ec 100644
--- a/docs/plugins/inspect/plugin-videorate.xml
+++ b/docs/plugins/inspect/plugin-videorate.xml
@@ -3,10 +3,10 @@
   <description>Adjusts video frames</description>
   <filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
   <basename>libgstvideorate.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
@@ -20,13 +20,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw; image/jpeg; image/png</details>
+          <details>video/x-raw(ANY); image/jpeg(ANY); image/png(ANY)</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw; image/jpeg; image/png</details>
+          <details>video/x-raw(ANY); image/jpeg(ANY); image/png(ANY)</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-videoscale.xml b/docs/plugins/inspect/plugin-videoscale.xml
index 9443f1316..a35e3b59c 100644
--- a/docs/plugins/inspect/plugin-videoscale.xml
+++ b/docs/plugins/inspect/plugin-videoscale.xml
@@ -3,10 +3,10 @@
   <description>Resizes video</description>
   <filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
   <basename>libgstvideoscale.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
@@ -14,19 +14,19 @@
       <longname>Video scaler</longname>
       <class>Filter/Converter/Video/Scaler</class>
       <description>Resizes video</description>
-      <author>Wim Taymans &lt;wim.taymans@chello.be&gt;</author>
+      <author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
       <pads>
         <caps>
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-videotestsrc.xml b/docs/plugins/inspect/plugin-videotestsrc.xml
index 102ab020d..568d63020 100644
--- a/docs/plugins/inspect/plugin-videotestsrc.xml
+++ b/docs/plugins/inspect/plugin-videotestsrc.xml
@@ -3,10 +3,10 @@
   <description>Creates a test video stream</description>
   <filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
   <basename>libgstvideotestsrc.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
@@ -20,7 +20,7 @@
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-bayer, format=(string){ bggr, rggb, grbg, gbrg }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-bayer, format=(string){ bggr, rggb, grbg, gbrg }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-volume.xml b/docs/plugins/inspect/plugin-volume.xml
index 7da8e74bf..15cf98138 100644
--- a/docs/plugins/inspect/plugin-volume.xml
+++ b/docs/plugins/inspect/plugin-volume.xml
@@ -3,10 +3,10 @@
   <description>plugin for controlling audio volume</description>
   <filename>../../gst/volume/.libs/libgstvolume.so</filename>
   <basename>libgstvolume.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-vorbis.xml b/docs/plugins/inspect/plugin-vorbis.xml
index 77fba73d0..8eabc7ae2 100644
--- a/docs/plugins/inspect/plugin-vorbis.xml
+++ b/docs/plugins/inspect/plugin-vorbis.xml
@@ -3,10 +3,10 @@
   <description>Vorbis plugin library</description>
   <filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
   <basename>libgstvorbis.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-ximagesink.xml b/docs/plugins/inspect/plugin-ximagesink.xml
index aeed80d5a..c9bfe0bdb 100644
--- a/docs/plugins/inspect/plugin-ximagesink.xml
+++ b/docs/plugins/inspect/plugin-ximagesink.xml
@@ -3,10 +3,10 @@
   <description>X11 video output element based on standard Xlib calls</description>
   <filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
   <basename>libgstximagesink.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-xvimagesink.xml b/docs/plugins/inspect/plugin-xvimagesink.xml
index 5c1341697..d197a37c9 100644
--- a/docs/plugins/inspect/plugin-xvimagesink.xml
+++ b/docs/plugins/inspect/plugin-xvimagesink.xml
@@ -3,10 +3,10 @@
   <description>XFree86 video output plugin using Xv extension</description>
   <filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
   <basename>libgstxvimagesink.so</basename>
-  <version>1.5.0.1</version>
+  <version>1.5.1</version>
   <license>LGPL</license>
   <source>gst-plugins-base</source>
-  <package>GStreamer Base Plug-ins git</package>
+  <package>GStreamer Base Plug-ins source release</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap
index 5e04ca6b2..670d45300 100644
--- a/gst-plugins-base.doap
+++ b/gst-plugins-base.doap
@@ -36,6 +36,16 @@ A wide range of video and audio decoders, encoders, and filters are included.
 
  <release>
   <Version>
+   <revision>1.5.1</revision>
+   <branch>1.5</branch>
+   <name></name>
+   <created>2015-06-07</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.5.1.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.4.0</revision>
    <branch>1.4</branch>
    <name></name>
diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h
index a68e048e2..fc48babbd 100644
--- a/win32/common/_stdint.h
+++ b/win32/common/_stdint.h
@@ -1,8 +1,8 @@
 #ifndef _GST_PLUGINS_BASE__STDINT_H
 #define _GST_PLUGINS_BASE__STDINT_H 1
 #ifndef _GENERATED_STDINT_H
-#define _GENERATED_STDINT_H "gst-plugins-base 1.4.0"
-/* generated using gnu compiler Debian clang version 3.5.0-1 (trunk) (based on LLVM 3.5.0) */
+#define _GENERATED_STDINT_H "gst-plugins-base 1.5.1"
+/* generated using gnu compiler Debian clang version 3.6.1-1 (tags/RELEASE_361/final) (based on LLVM 3.6.1) */
 #define _STDINT_HAVE_STDINT_H 1
 #include <stdint.h>
 #endif
diff --git a/win32/common/config.h b/win32/common/config.h
index 2db8facb8..20fcdea00 100644
--- a/win32/common/config.h
+++ b/win32/common/config.h
@@ -50,6 +50,9 @@
 /* The GIO modules directory. */
 #undef GIO_MODULE_DIR
 
+/* The GIO install prefix. */
+#undef GIO_PREFIX
+
 /* major/minor version */
 #define GST_API_VERSION "1.0"
 
@@ -84,7 +87,7 @@
 #define GST_PACKAGE_ORIGIN "Unknown package origin"
 
 /* GStreamer package release date/time for plugins as YYYY-MM-DD */
-#define GST_PACKAGE_RELEASE_DATETIME "2014-07-19"
+#define GST_PACKAGE_RELEASE_DATETIME "2015-06-07"
 
 /* Define if static plugins should be built */
 #undef GST_PLUGIN_BUILD_STATIC
@@ -106,9 +109,15 @@
    the CoreFoundation framework. */
 #undef HAVE_CFPREFERENCESCOPYAPPVALUE
 
+/* Define if the target CPU is AARCH64 */
+#undef HAVE_CPU_AARCH64
+
 /* Define if the target CPU is an Alpha */
 #undef HAVE_CPU_ALPHA
 
+/* Define if the target CPU is an ARC */
+#undef HAVE_CPU_ARC
+
 /* Define if the target CPU is an ARM */
 #undef HAVE_CPU_ARM
 
@@ -176,6 +185,9 @@
 /* Define if the GNU gettext() function is already present or preinstalled. */
 #undef HAVE_GETTEXT
 
+/* Define to enable glib GIO unix (used by gio-unix-2.0). */
+#undef HAVE_GIO_UNIX_2_0
+
 /* Define to 1 if you have the `gmtime_r' function. */
 #undef HAVE_GMTIME_R
 
@@ -325,7 +337,7 @@
 #define PACKAGE_NAME "GStreamer Base Plug-ins"
 
 /* Define to the full name and version of this package. */
-#define PACKAGE_STRING "GStreamer Base Plug-ins 1.4.0"
+#define PACKAGE_STRING "GStreamer Base Plug-ins 1.5.1"
 
 /* Define to the one symbol short name of this package. */
 #define PACKAGE_TARNAME "gst-plugins-base"
@@ -334,7 +346,7 @@
 #undef PACKAGE_URL
 
 /* Define to the version of this package. */
-#define PACKAGE_VERSION "1.4.0"
+#define PACKAGE_VERSION "1.5.1"
 
 /* directory where plugins are located */
 #ifdef _DEBUG
@@ -368,7 +380,7 @@
 #undef USE_TREMOLO
 
 /* Version number of package */
-#define VERSION "1.4.0"
+#define VERSION "1.5.1"
 
 /* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most
    significant byte first (like Motorola and SPARC, unlike Intel). */
@@ -382,9 +394,6 @@
 # endif
 #endif
 
-/* Define to 1 if the X Window System is missing or not being used. */
-#undef X_DISPLAY_MISSING
-
 /* Enable large inode numbers on Mac OS X 10.5.  */
 #ifndef _DARWIN_USE_64_BIT_INODE
 # define _DARWIN_USE_64_BIT_INODE 1
diff --git a/win32/common/gstrtsp-enumtypes.c b/win32/common/gstrtsp-enumtypes.c
index 69ca5da87..85187f951 100644
--- a/win32/common/gstrtsp-enumtypes.c
+++ b/win32/common/gstrtsp-enumtypes.c
@@ -3,7 +3,98 @@
 
 #include "gstrtsp-enumtypes.h"
 
+#include "rtsp.h"
+#include "gstrtsp.h"
+#include "gstrtsptransport.h"
+#include "gstrtspurl.h"
+#include "gstrtspmessage.h"
+#include "gstrtspconnection.h"
 #include "gstrtspdefs.h"
+#include "gstrtspextension.h"
+#include "gstrtsprange.h"
+
+/* enumerations from "gstrtsptransport.h" */
+GType
+gst_rtsp_trans_mode_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GFlagsValue values[] = {
+      {GST_RTSP_TRANS_UNKNOWN, "GST_RTSP_TRANS_UNKNOWN", "unknown"},
+      {GST_RTSP_TRANS_RTP, "GST_RTSP_TRANS_RTP", "rtp"},
+      {GST_RTSP_TRANS_RDT, "GST_RTSP_TRANS_RDT", "rdt"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_flags_register_static ("GstRTSPTransMode", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_rtsp_profile_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GFlagsValue values[] = {
+      {GST_RTSP_PROFILE_UNKNOWN, "GST_RTSP_PROFILE_UNKNOWN", "unknown"},
+      {GST_RTSP_PROFILE_AVP, "GST_RTSP_PROFILE_AVP", "avp"},
+      {GST_RTSP_PROFILE_SAVP, "GST_RTSP_PROFILE_SAVP", "savp"},
+      {GST_RTSP_PROFILE_AVPF, "GST_RTSP_PROFILE_AVPF", "avpf"},
+      {GST_RTSP_PROFILE_SAVPF, "GST_RTSP_PROFILE_SAVPF", "savpf"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id = g_flags_register_static ("GstRTSPProfile", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_rtsp_lower_trans_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GFlagsValue values[] = {
+      {GST_RTSP_LOWER_TRANS_UNKNOWN, "GST_RTSP_LOWER_TRANS_UNKNOWN", "unknown"},
+      {GST_RTSP_LOWER_TRANS_UDP, "GST_RTSP_LOWER_TRANS_UDP", "udp"},
+      {GST_RTSP_LOWER_TRANS_UDP_MCAST, "GST_RTSP_LOWER_TRANS_UDP_MCAST",
+          "udp-mcast"},
+      {GST_RTSP_LOWER_TRANS_TCP, "GST_RTSP_LOWER_TRANS_TCP", "tcp"},
+      {GST_RTSP_LOWER_TRANS_HTTP, "GST_RTSP_LOWER_TRANS_HTTP", "http"},
+      {GST_RTSP_LOWER_TRANS_TLS, "GST_RTSP_LOWER_TRANS_TLS", "tls"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_flags_register_static ("GstRTSPLowerTrans", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+/* enumerations from "gstrtspmessage.h" */
+GType
+gst_rtsp_msg_type_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_RTSP_MESSAGE_INVALID, "GST_RTSP_MESSAGE_INVALID", "invalid"},
+      {GST_RTSP_MESSAGE_REQUEST, "GST_RTSP_MESSAGE_REQUEST", "request"},
+      {GST_RTSP_MESSAGE_RESPONSE, "GST_RTSP_MESSAGE_RESPONSE", "response"},
+      {GST_RTSP_MESSAGE_HTTP_REQUEST, "GST_RTSP_MESSAGE_HTTP_REQUEST",
+          "http-request"},
+      {GST_RTSP_MESSAGE_HTTP_RESPONSE, "GST_RTSP_MESSAGE_HTTP_RESPONSE",
+          "http-response"},
+      {GST_RTSP_MESSAGE_DATA, "GST_RTSP_MESSAGE_DATA", "data"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id = g_enum_register_static ("GstRTSPMsgType", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
 
 /* enumerations from "gstrtspdefs.h" */
 GType
@@ -392,3 +483,44 @@ gst_rtsp_status_code_get_type (void)
   }
   return g_define_type_id__volatile;
 }
+
+/* enumerations from "gstrtsprange.h" */
+GType
+gst_rtsp_range_unit_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_RTSP_RANGE_SMPTE, "GST_RTSP_RANGE_SMPTE", "smpte"},
+      {GST_RTSP_RANGE_SMPTE_30_DROP, "GST_RTSP_RANGE_SMPTE_30_DROP",
+          "smpte-30-drop"},
+      {GST_RTSP_RANGE_SMPTE_25, "GST_RTSP_RANGE_SMPTE_25", "smpte-25"},
+      {GST_RTSP_RANGE_NPT, "GST_RTSP_RANGE_NPT", "npt"},
+      {GST_RTSP_RANGE_CLOCK, "GST_RTSP_RANGE_CLOCK", "clock"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstRTSPRangeUnit", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_rtsp_time_type_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_RTSP_TIME_SECONDS, "GST_RTSP_TIME_SECONDS", "seconds"},
+      {GST_RTSP_TIME_NOW, "GST_RTSP_TIME_NOW", "now"},
+      {GST_RTSP_TIME_END, "GST_RTSP_TIME_END", "end"},
+      {GST_RTSP_TIME_FRAMES, "GST_RTSP_TIME_FRAMES", "frames"},
+      {GST_RTSP_TIME_UTC, "GST_RTSP_TIME_UTC", "utc"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id = g_enum_register_static ("GstRTSPTimeType", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
diff --git a/win32/common/gstrtsp-enumtypes.h b/win32/common/gstrtsp-enumtypes.h
index 3254324fc..c42ebdcba 100644
--- a/win32/common/gstrtsp-enumtypes.h
+++ b/win32/common/gstrtsp-enumtypes.h
@@ -8,6 +8,18 @@
 
 G_BEGIN_DECLS
 
+/* enumerations from "gstrtsptransport.h" */
+GType gst_rtsp_trans_mode_get_type (void);
+#define GST_TYPE_RTSP_TRANS_MODE (gst_rtsp_trans_mode_get_type())
+GType gst_rtsp_profile_get_type (void);
+#define GST_TYPE_RTSP_PROFILE (gst_rtsp_profile_get_type())
+GType gst_rtsp_lower_trans_get_type (void);
+#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
+
+/* enumerations from "gstrtspmessage.h" */
+GType gst_rtsp_msg_type_get_type (void);
+#define GST_TYPE_RTSP_MSG_TYPE (gst_rtsp_msg_type_get_type())
+
 /* enumerations from "gstrtspdefs.h" */
 GType gst_rtsp_result_get_type (void);
 #define GST_TYPE_RTSP_RESULT (gst_rtsp_result_get_type())
@@ -27,6 +39,12 @@ GType gst_rtsp_header_field_get_type (void);
 #define GST_TYPE_RTSP_HEADER_FIELD (gst_rtsp_header_field_get_type())
 GType gst_rtsp_status_code_get_type (void);
 #define GST_TYPE_RTSP_STATUS_CODE (gst_rtsp_status_code_get_type())
+
+/* enumerations from "gstrtsprange.h" */
+GType gst_rtsp_range_unit_get_type (void);
+#define GST_TYPE_RTSP_RANGE_UNIT (gst_rtsp_range_unit_get_type())
+GType gst_rtsp_time_type_get_type (void);
+#define GST_TYPE_RTSP_TIME_TYPE (gst_rtsp_time_type_get_type())
 G_END_DECLS
 
 #endif /* __gst_rtsp_ENUM_TYPES_H__ */
diff --git a/win32/common/pbutils-enumtypes.c b/win32/common/pbutils-enumtypes.c
index 2652ec066..99a6f1df0 100644
--- a/win32/common/pbutils-enumtypes.c
+++ b/win32/common/pbutils-enumtypes.c
@@ -68,3 +68,24 @@ gst_discoverer_result_get_type (void)
   }
   return g_define_type_id__volatile;
 }
+
+GType
+gst_discoverer_serialize_flags_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GFlagsValue values[] = {
+      {GST_DISCOVERER_SERIALIZE_BASIC, "GST_DISCOVERER_SERIALIZE_BASIC",
+          "basic"},
+      {GST_DISCOVERER_SERIALIZE_CAPS, "GST_DISCOVERER_SERIALIZE_CAPS", "caps"},
+      {GST_DISCOVERER_SERIALIZE_TAGS, "GST_DISCOVERER_SERIALIZE_TAGS", "tags"},
+      {GST_DISCOVERER_SERIALIZE_MISC, "GST_DISCOVERER_SERIALIZE_MISC", "misc"},
+      {GST_DISCOVERER_SERIALIZE_ALL, "GST_DISCOVERER_SERIALIZE_ALL", "all"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_flags_register_static ("GstDiscovererSerializeFlags", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
diff --git a/win32/common/pbutils-enumtypes.h b/win32/common/pbutils-enumtypes.h
index 30dadfd95..1ca0476db 100644
--- a/win32/common/pbutils-enumtypes.h
+++ b/win32/common/pbutils-enumtypes.h
@@ -15,6 +15,8 @@ GType gst_install_plugins_return_get_type (void);
 /* enumerations from "gstdiscoverer.h" */
 GType gst_discoverer_result_get_type (void);
 #define GST_TYPE_DISCOVERER_RESULT (gst_discoverer_result_get_type())
+GType gst_discoverer_serialize_flags_get_type (void);
+#define GST_TYPE_DISCOVERER_SERIALIZE_FLAGS (gst_discoverer_serialize_flags_get_type())
 G_END_DECLS
 
 #endif /* __PB_UTILS_ENUM_TYPES_H__ */
diff --git a/win32/common/video-enumtypes.c b/win32/common/video-enumtypes.c
index a1f32112a..77d8330f5 100644
--- a/win32/common/video-enumtypes.c
+++ b/win32/common/video-enumtypes.c
@@ -7,10 +7,13 @@
 #include "video-format.h"
 #include "video-color.h"
 #include "video-info.h"
+#include "video-dither.h"
 #include "colorbalance.h"
 #include "navigation.h"
 #include "video-chroma.h"
 #include "video-tile.h"
+#include "video-converter.h"
+#include "video-resampler.h"
 
 /* enumerations from "video-format.h" */
 GType
@@ -74,6 +77,13 @@ gst_video_format_get_type (void)
       {GST_VIDEO_FORMAT_NV24, "GST_VIDEO_FORMAT_NV24", "nv24"},
       {GST_VIDEO_FORMAT_NV12_64Z32, "GST_VIDEO_FORMAT_NV12_64Z32",
           "nv12-64z32"},
+      {GST_VIDEO_FORMAT_A420_10BE, "GST_VIDEO_FORMAT_A420_10BE", "a420-10be"},
+      {GST_VIDEO_FORMAT_A420_10LE, "GST_VIDEO_FORMAT_A420_10LE", "a420-10le"},
+      {GST_VIDEO_FORMAT_A422_10BE, "GST_VIDEO_FORMAT_A422_10BE", "a422-10be"},
+      {GST_VIDEO_FORMAT_A422_10LE, "GST_VIDEO_FORMAT_A422_10LE", "a422-10le"},
+      {GST_VIDEO_FORMAT_A444_10BE, "GST_VIDEO_FORMAT_A444_10BE", "a444-10be"},
+      {GST_VIDEO_FORMAT_A444_10LE, "GST_VIDEO_FORMAT_A444_10LE", "a444-10le"},
+      {GST_VIDEO_FORMAT_NV61, "GST_VIDEO_FORMAT_NV61", "nv61"},
       {0, NULL, NULL}
     };
     GType g_define_type_id = g_enum_register_static ("GstVideoFormat", values);
@@ -162,6 +172,8 @@ gst_video_color_matrix_get_type (void)
       {GST_VIDEO_COLOR_MATRIX_BT601, "GST_VIDEO_COLOR_MATRIX_BT601", "bt601"},
       {GST_VIDEO_COLOR_MATRIX_SMPTE240M, "GST_VIDEO_COLOR_MATRIX_SMPTE240M",
           "smpte240m"},
+      {GST_VIDEO_COLOR_MATRIX_BT2020, "GST_VIDEO_COLOR_MATRIX_BT2020",
+          "bt2020"},
       {0, NULL, NULL}
     };
     GType g_define_type_id =
@@ -189,6 +201,8 @@ gst_video_transfer_function_get_type (void)
       {GST_VIDEO_TRANSFER_GAMMA28, "GST_VIDEO_TRANSFER_GAMMA28", "gamma28"},
       {GST_VIDEO_TRANSFER_LOG100, "GST_VIDEO_TRANSFER_LOG100", "log100"},
       {GST_VIDEO_TRANSFER_LOG316, "GST_VIDEO_TRANSFER_LOG316", "log316"},
+      {GST_VIDEO_TRANSFER_BT2020_12, "GST_VIDEO_TRANSFER_BT2020_12",
+          "bt2020-12"},
       {0, NULL, NULL}
     };
     GType g_define_type_id =
@@ -218,6 +232,8 @@ gst_video_color_primaries_get_type (void)
           "GST_VIDEO_COLOR_PRIMARIES_SMPTE240M", "smpte240m"},
       {GST_VIDEO_COLOR_PRIMARIES_FILM, "GST_VIDEO_COLOR_PRIMARIES_FILM",
           "film"},
+      {GST_VIDEO_COLOR_PRIMARIES_BT2020, "GST_VIDEO_COLOR_PRIMARIES_BT2020",
+          "bt2020"},
       {0, NULL, NULL}
     };
     GType g_define_type_id =
@@ -270,6 +286,49 @@ gst_video_flags_get_type (void)
   return g_define_type_id__volatile;
 }
 
+/* enumerations from "video-dither.h" */
+GType
+gst_video_dither_method_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_DITHER_NONE, "GST_VIDEO_DITHER_NONE", "none"},
+      {GST_VIDEO_DITHER_VERTERR, "GST_VIDEO_DITHER_VERTERR", "verterr"},
+      {GST_VIDEO_DITHER_FLOYD_STEINBERG, "GST_VIDEO_DITHER_FLOYD_STEINBERG",
+          "floyd-steinberg"},
+      {GST_VIDEO_DITHER_SIERRA_LITE, "GST_VIDEO_DITHER_SIERRA_LITE",
+          "sierra-lite"},
+      {GST_VIDEO_DITHER_BAYER, "GST_VIDEO_DITHER_BAYER", "bayer"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoDitherMethod", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_video_dither_flags_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GFlagsValue values[] = {
+      {GST_VIDEO_DITHER_FLAG_NONE, "GST_VIDEO_DITHER_FLAG_NONE", "none"},
+      {GST_VIDEO_DITHER_FLAG_INTERLACED, "GST_VIDEO_DITHER_FLAG_INTERLACED",
+          "interlaced"},
+      {GST_VIDEO_DITHER_FLAG_QUANTIZE, "GST_VIDEO_DITHER_FLAG_QUANTIZE",
+          "quantize"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_flags_register_static ("GstVideoDitherFlags", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
 /* enumerations from "colorbalance.h" */
 GType
 gst_color_balance_type_get_type (void)
@@ -356,6 +415,7 @@ gst_navigation_message_type_get_type (void)
           "GST_NAVIGATION_MESSAGE_COMMANDS_CHANGED", "commands-changed"},
       {GST_NAVIGATION_MESSAGE_ANGLES_CHANGED,
           "GST_NAVIGATION_MESSAGE_ANGLES_CHANGED", "angles-changed"},
+      {GST_NAVIGATION_MESSAGE_EVENT, "GST_NAVIGATION_MESSAGE_EVENT", "event"},
       {0, NULL, NULL}
     };
     GType g_define_type_id =
@@ -494,3 +554,142 @@ gst_video_tile_mode_get_type (void)
   }
   return g_define_type_id__volatile;
 }
+
+/* enumerations from "video-converter.h" */
+GType
+gst_video_alpha_mode_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_ALPHA_MODE_COPY, "GST_VIDEO_ALPHA_MODE_COPY", "copy"},
+      {GST_VIDEO_ALPHA_MODE_SET, "GST_VIDEO_ALPHA_MODE_SET", "set"},
+      {GST_VIDEO_ALPHA_MODE_MULT, "GST_VIDEO_ALPHA_MODE_MULT", "mult"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoAlphaMode", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_video_chroma_mode_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_CHROMA_MODE_FULL, "GST_VIDEO_CHROMA_MODE_FULL", "full"},
+      {GST_VIDEO_CHROMA_MODE_UPSAMPLE_ONLY,
+          "GST_VIDEO_CHROMA_MODE_UPSAMPLE_ONLY", "upsample-only"},
+      {GST_VIDEO_CHROMA_MODE_DOWNSAMPLE_ONLY,
+          "GST_VIDEO_CHROMA_MODE_DOWNSAMPLE_ONLY", "downsample-only"},
+      {GST_VIDEO_CHROMA_MODE_NONE, "GST_VIDEO_CHROMA_MODE_NONE", "none"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoChromaMode", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_video_matrix_mode_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_MATRIX_MODE_FULL, "GST_VIDEO_MATRIX_MODE_FULL", "full"},
+      {GST_VIDEO_MATRIX_MODE_INPUT_ONLY, "GST_VIDEO_MATRIX_MODE_INPUT_ONLY",
+          "input-only"},
+      {GST_VIDEO_MATRIX_MODE_OUTPUT_ONLY, "GST_VIDEO_MATRIX_MODE_OUTPUT_ONLY",
+          "output-only"},
+      {GST_VIDEO_MATRIX_MODE_NONE, "GST_VIDEO_MATRIX_MODE_NONE", "none"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoMatrixMode", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_video_gamma_mode_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_GAMMA_MODE_NONE, "GST_VIDEO_GAMMA_MODE_NONE", "none"},
+      {GST_VIDEO_GAMMA_MODE_REMAP, "GST_VIDEO_GAMMA_MODE_REMAP", "remap"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoGammaMode", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_video_primaries_mode_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_PRIMARIES_MODE_NONE, "GST_VIDEO_PRIMARIES_MODE_NONE", "none"},
+      {GST_VIDEO_PRIMARIES_MODE_MERGE_ONLY,
+          "GST_VIDEO_PRIMARIES_MODE_MERGE_ONLY", "merge-only"},
+      {GST_VIDEO_PRIMARIES_MODE_FAST, "GST_VIDEO_PRIMARIES_MODE_FAST", "fast"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoPrimariesMode", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+/* enumerations from "video-resampler.h" */
+GType
+gst_video_resampler_method_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_RESAMPLER_METHOD_NEAREST, "GST_VIDEO_RESAMPLER_METHOD_NEAREST",
+          "nearest"},
+      {GST_VIDEO_RESAMPLER_METHOD_LINEAR, "GST_VIDEO_RESAMPLER_METHOD_LINEAR",
+          "linear"},
+      {GST_VIDEO_RESAMPLER_METHOD_CUBIC, "GST_VIDEO_RESAMPLER_METHOD_CUBIC",
+          "cubic"},
+      {GST_VIDEO_RESAMPLER_METHOD_SINC, "GST_VIDEO_RESAMPLER_METHOD_SINC",
+          "sinc"},
+      {GST_VIDEO_RESAMPLER_METHOD_LANCZOS, "GST_VIDEO_RESAMPLER_METHOD_LANCZOS",
+          "lanczos"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoResamplerMethod", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
+
+GType
+gst_video_resampler_flags_get_type (void)
+{
+  static volatile gsize g_define_type_id__volatile = 0;
+  if (g_once_init_enter (&g_define_type_id__volatile)) {
+    static const GEnumValue values[] = {
+      {GST_VIDEO_RESAMPLER_FLAG_NONE, "GST_VIDEO_RESAMPLER_FLAG_NONE", "none"},
+      {0, NULL, NULL}
+    };
+    GType g_define_type_id =
+        g_enum_register_static ("GstVideoResamplerFlags", values);
+    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);
+  }
+  return g_define_type_id__volatile;
+}
diff --git a/win32/common/video-enumtypes.h b/win32/common/video-enumtypes.h
index 04f14783a..48b980cbb 100644
--- a/win32/common/video-enumtypes.h
+++ b/win32/common/video-enumtypes.h
@@ -32,6 +32,12 @@ GType gst_video_interlace_mode_get_type (void);
 GType gst_video_flags_get_type (void);
 #define GST_TYPE_VIDEO_FLAGS (gst_video_flags_get_type())
 
+/* enumerations from "video-dither.h" */
+GType gst_video_dither_method_get_type (void);
+#define GST_TYPE_VIDEO_DITHER_METHOD (gst_video_dither_method_get_type())
+GType gst_video_dither_flags_get_type (void);
+#define GST_TYPE_VIDEO_DITHER_FLAGS (gst_video_dither_flags_get_type())
+
 /* enumerations from "colorbalance.h" */
 GType gst_color_balance_type_get_type (void);
 #define GST_TYPE_COLOR_BALANCE_TYPE (gst_color_balance_type_get_type())
@@ -59,6 +65,24 @@ GType gst_video_tile_type_get_type (void);
 #define GST_TYPE_VIDEO_TILE_TYPE (gst_video_tile_type_get_type())
 GType gst_video_tile_mode_get_type (void);
 #define GST_TYPE_VIDEO_TILE_MODE (gst_video_tile_mode_get_type())
+
+/* enumerations from "video-converter.h" */
+GType gst_video_alpha_mode_get_type (void);
+#define GST_TYPE_VIDEO_ALPHA_MODE (gst_video_alpha_mode_get_type())
+GType gst_video_chroma_mode_get_type (void);
+#define GST_TYPE_VIDEO_CHROMA_MODE (gst_video_chroma_mode_get_type())
+GType gst_video_matrix_mode_get_type (void);
+#define GST_TYPE_VIDEO_MATRIX_MODE (gst_video_matrix_mode_get_type())
+GType gst_video_gamma_mode_get_type (void);
+#define GST_TYPE_VIDEO_GAMMA_MODE (gst_video_gamma_mode_get_type())
+GType gst_video_primaries_mode_get_type (void);
+#define GST_TYPE_VIDEO_PRIMARIES_MODE (gst_video_primaries_mode_get_type())
+
+/* enumerations from "video-resampler.h" */
+GType gst_video_resampler_method_get_type (void);
+#define GST_TYPE_VIDEO_RESAMPLER_METHOD (gst_video_resampler_method_get_type())
+GType gst_video_resampler_flags_get_type (void);
+#define GST_TYPE_VIDEO_RESAMPLER_FLAGS (gst_video_resampler_flags_get_type())
 G_END_DECLS
 
 #endif /* __GST_VIDEO_ENUM_TYPES_H__ */