=== release 1.4.5 ===

2014-12-18  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.4.5

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.

2014-12-16 15:03:55 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideosink.c:
	* tests/check/libs/video.c:
	  Revert "video: Fix non-default usage of gst_video_sink_center_rect"
	  This reverts commit 899461d722e45f591eeddf33c405677170d63de4.
	  There seems to be a lot of code out there that does not properly initialize
	  the rectangles and then causes undefined behaviour. Including our video sinks.
	  Let's keep this out of 1.4, fix everything everywhere and keep it in 1.6

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 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 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

2014-11-28 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.

2014-12-12 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.

2014-12-11 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

2014-12-11 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

2014-11-28 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

2014-12-02 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

2014-12-01 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

2014-11-25 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

2014-11-22 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

2014-07-26 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.

2014-11-09 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

2014-11-06 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

=== release 1.4.4 ===

2014-11-06 12:52:52 +0100  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* 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:
	* gst-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/config.h:
	  Release 1.4.4

2014-11-06 12:36:18 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/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

2014-11-06 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

2014-10-01 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.

2014-10-13 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 12:14:17 +0300  Heinrich Fink <hfink@toolsonair.com>

	* gst/playback/gstplaysink.c:
	  playsink: Use correct property enum value for video-filter property installation

2014-10-07 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

2014-10-04 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

2014-10-01 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

2014-09-23 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

=== release 1.4.3 ===

2014-09-24 12:27:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* 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 1.4.3

2014-09-24 11:22:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/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

2014-09-23 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

2014-08-08 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

2014-09-11 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.

=== release 1.4.2 ===

2014-09-19 14:21:22 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* 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 1.4.2

2014-09-19 10:51:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/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

2014-09-19 09:58:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/da.po:
	* po/sr.po:
	  po: Update translations

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

2014-09-17 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 14:34:25 +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 14:11:21 +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

2014-09-18 12:39:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: Simplify code a bit

2014-09-17 14:08:17 +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 12:17:53 +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

2014-09-17 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

2014-09-05 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

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.

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

2014-09-15 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

2014-09-05 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

2014-09-12 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

2014-09-11 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)

2014-03-28 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

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

2014-09-03 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

2014-09-01 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 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

=== release 1.4.1 ===

2014-08-27 15:04:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* 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 1.4.1

2014-08-27 14:27:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/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

2014-08-27 12:30:10 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/de.po:
	* po/hu.po:
	* po/id.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  po: Update translations

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

2014-08-14 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

2014-08-12 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

2014-08-08 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

2014-08-01 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

2014-07-31 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

=== release 1.4.0 ===

2014-07-19 17:04:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* 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:
