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=== release 1.7.1 ===

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

	* configure.ac:
	  releasing 1.7.1

2015-12-24 12:22:04 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/nl.po:
	* po/sv.po:
	* po/zh_CN.po:
	  po: Update translations

2015-12-11 15:38:00 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encodebin: Implement an encoding profile serialization format
	  https://bugzilla.gnome.org/show_bug.cgi?id=759356

2015-12-21 00:43:49 +0100  Koop Mast <kwm@rainbow-runner.nl>

	* configure.ac:
	  configure: Make -Bsymbolic check work with clang.
	  Update the -Bsymbolic check with the version glib has. This version
	  works with clang.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759713

2015-12-03 11:53:05 +0900  Kazunori Kobayashi <kkobayas@igel.co.jp>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Clear is_eos flag when receiving the flush-stop event
	  The EOS event can be propagated to the downstream elements when
	  is_eos flag remains set even after leaving the flushing state.
	  This fix allows this element to normally restart the streaming
	  after receiving the flush event by clearing the is_eos flag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759110

2015-12-16 18:11:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/examples/playback/playback-test.c:
	  examples: playback-test: remove unused variables
	  audiosink and videosink string variables are unused

2015-11-30 10:28:55 +1100  Matthew Waters <matthew@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: only add the template caps when the result is empty
	  Unconditionally adding the template caps when proxying the caps query will play
	  havoc with decoders that attempt to choose an output format based on some caps
	  features.  Creating a sink that does not include those caps features and a
	  decoder/parser/etc that preferentially chooses some specific caps feature when
	  available, will always return the decoder/parser/etc template caps and choose a
	  feature that downstream will be unable to support.
	  Fix by limiting the addition of the template caps to when the result is actually
	  empty.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2015-12-17 13:39:01 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Don't use AG_GST_CHECK_FEATURE for checking for gio-unix-2.0
	  It's meant to be used for external plugins that can then all be disabled via
	  --disable-external. gio-unix-2.0 however is just an optional dependency for
	  the TCP unit test.
	  Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part
	  there needs to be an AM_CONDITIONAL for the feature with FALSE.

2015-12-16 17:07:54 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  Revert "decodebin2: fix deadlock on chain shutdown"
	  This reverts commit 77dc09c3a9a5e5e371e189f39b5557db440a8dc9.
	  It can cause the FLUSH_START/STOP events to go to the sink elements, which
	  then causes state changes and various other problems. We shouldn't really
	  flush downstream here, the idea is to do *draining*.
	  Apart from that the testcase for the original bug here works without this
	  commit now.

2015-12-16 11:12:00 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/tcp/gstmultifdsink.c:
	  multifdsink: fix typo in GST_WARNING_OBJECT
	  This should make easier to parse the debug logs.
	  s/fnctl/fcntl

2014-04-10 15:36:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: remove dead code
	  Since the loops increasing count from 0 are always run at least
	  once (if count < 1), count will always be at least one when
	  compared to the drop/dup conditions.
	  Coverity 1139674

2015-12-16 10:45:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* win32/common/libgstaudio.def:
	  audio-converter: rework the main processing loop
	  Rework the main processing loop. We now create an audio processing
	  chain from small core functions. This is very similar to how the
	  video-converter core works and allows us to statically calculate an
	  optimal allocation strategy for all possible combinations of operations.
	  Make sure we support non-interleaved data everywhere.
	  Add functions to calculate in and out frames and latency.

2015-12-16 10:44:16 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: clear convert object

2015-12-16 09:35:38 +0100  Sebastian Dröge <sebastian@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-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

2015-12-14 13:59:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/alsa/gstalsasrc.c:
	  Revert "alsasrc: Disable HW timestamp"
	  This reverts commit 3642e9a3913a35c00f379034780c27298d09929c.

2015-11-10 12:54:23 -0500  Xavier Claessens <xavier.claessens@collabora.com>

	* gst-libs/gst/allocators/gstfdmemory.h:
	* gst-libs/gst/app/gstappsink.h:
	* gst-libs/gst/app/gstappsrc.h:
	* gst-libs/gst/audio/audio-info.h:
	* gst-libs/gst/audio/gstaudiobasesink.h:
	* gst-libs/gst/audio/gstaudiobasesrc.h:
	* gst-libs/gst/audio/gstaudiocdsrc.h:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiodecoder.h:
	* gst-libs/gst/audio/gstaudioencoder.h:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst-libs/gst/audio/gstaudioringbuffer.h:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/pbutils/encoding-profile.h:
	* gst-libs/gst/pbutils/encoding-target.h:
	* gst-libs/gst/pbutils/gstdiscoverer.h:
	* gst-libs/gst/pbutils/install-plugins.h:
	* gst-libs/gst/rtp/gstrtpbaseaudiopayload.h:
	* gst-libs/gst/rtp/gstrtpbasedepayload.h:
	* gst-libs/gst/rtp/gstrtpbasepayload.h:
	* gst-libs/gst/rtsp/gstrtspurl.h:
	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/sdp/gstsdpmessage.h:
	* gst-libs/gst/tag/gsttagdemux.h:
	* gst-libs/gst/tag/gsttagmux.h:
	* gst-libs/gst/video/colorbalancechannel.h:
	* gst-libs/gst/video/gstvideodecoder.h:
	* gst-libs/gst/video/gstvideoencoder.h:
	* gst-libs/gst/video/gstvideofilter.h:
	* gst-libs/gst/video/gstvideopool.h:
	* gst-libs/gst/video/gstvideosink.h:
	* gst-libs/gst/video/gstvideoutils.h:
	* gst-libs/gst/video/video-info.h:
	* gst-libs/gst/video/video-overlay-composition.h:
	  base: Add g_autoptr() support to all types
	  https://bugzilla.gnome.org/show_bug.cgi?id=754464

2015-09-24 18:26:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

	* ext/alsa/gstalsasrc.c:
	  alsasrc: Disable HW timestamp
	  This is a workaround for broken pulse module.

2015-12-14 19:03:33 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes

2015-12-14 10:57:19 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Use relative URI for non-proxy tunneled requests
	  Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
	  are using a proxy server. Also, send Host header for compatability with
	  HTTP/1.1 and some HTTP/1.0 servers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758922

2015-12-14 09:10:16 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/rtsp/gstrtspconnection.h:
	* win32/common/libgstrtsp.def:
	  rtspconnection: Support authentication during tunneling setup
	  gst_rtsp_connection_connect_with_response accepts a response pointer
	  which it fills with the response from setup_tunneling if the
	  connection is configured to be tunneled.  The motivation for this is to
	  allow the caller to inspect the response header to determine if
	  additional authentication is required so that the connection can be
	  retried with the appropriate authentication headers.
	  The function prototype of gst_rtsp_connection_connect has been
	  preserved for compatability with existing code and wraps
	  gst_rtsp_connection_connect_with_response.
	  https://bugzilla.gnome.org/show_bug.cgi?id=749596

2015-12-14 13:11:21 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
	  CID 1139615

2015-12-10 17:46:26 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-channel-mix.h:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio: adapt API for non-interleaved formats
	  Allow an array of sample blocks to be passed to the channel mix and
	  quantizer functions to support non-interleaved formats.

2015-12-10 16:26:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	  audio-converter: improve API for non-interleaved formats
	  Make it possible to pass an array of sample blocks when dealing with
	  non-interleaved formats.

2015-12-12 17:49:28 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: add FourCC aliases
	  Support media using the aliases defined in http://www.fourcc.org/ that are
	  exact duplicates of already known codes.

2015-12-12 17:04:21 +0100  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: use defined FourCC
	  Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h,
	  like gst_riff_create_audio_caps() does.

2015-12-11 14:42:09 +0000  Julien Isorce <j.isorce@samsung.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: add some debug around pool negotiation
	  It lets us know easily which pool is activated or
	  inactivated during the negotiation.
	  https://bugzilla.gnome.org/show_bug.cgi?id=720597

2015-12-11 21:42:00 +0800  Song Bing <b06498@freescale.com>

	* gst-libs/gst/video/convertframe.c:
	  video/convertframe: Add crop meta support via videocrop
	  https://bugzilla.gnome.org/show_bug.cgi?id=759329

2015-12-11 11:01:53 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepay: when setting discont flag make sure rtpbuffer is current
	  Depayloaders will look at rtpbuffer->buffer for the discont flag.
	  When we set the discont flag on a buffer in the rtp base depayloader
	  and we have to make the buffer writable, make sure the rtpbuffer
	  actually contains the newly-flagged buffer, not the original input
	  buffer. This was introduced with the addition of the process_rtp_packet
	  vfunc, but would only trigger if the input buffer wasn't flagged
	  already and was not writable already.

2015-12-11 00:18:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/libs/rtpbasedepayload.c:
	  tests: rtpbasedepayload: add test for seqnum gap discont setting
	  The problem was triggered only when the input buffers were not
	  writable, so add extra ref to test this code path.

2015-12-11 10:25:00 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepay: fix possible refcounting issue when detecting a discont
	  When we detect a discont and the input buffer isn't already flagged
	  as discont, handle_buffer() does a gst_buffer_make_writable() on the
	  input buffer in order to set the flag. This assumed it had ownership
	  of the input buffer though, which it didn't. This would still work
	  fine in most scenarios, but could lead to crashes or mini object
	  unref criticals in some cases when a discont is detected, e.g. when
	  using pcapparse in front of a depayloader. This problem was
	  introduced in bc14cdf529e.

2015-12-10 12:18:04 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gstmultisocketsink.h:
	  multisocketsink: add GstNetworkMessage event
	  Add a property and logic to send a GstNetworkMessage event containing
	  the message that was received from a client. This can be used to
	  implement simply bidirectional communication.

2015-12-10 12:14:37 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gstmultisocketsink.h:
	  multisocketsink: add dispatched event
	  Add a property and logic to send a GstNetworkMessageDispatched
	  event upstream to notify that a buffer has been sent. This can be used
	  to keep track of what client received what buffers.

2015-12-04 11:17:37 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstsocketsrc.c:
	* gst/tcp/gstsocketsrc.h:
	  socketsrc: handle GstNetworkMessage events
	  Add a property to handle GstNetworkMessage events. These events contain
	  a buffer that is sent on the socket to allow for simple bidirectional
	  communication.

2015-12-09 17:16:26 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-convert: improve converter API
	  Improve the converter API to allow for an max input and output number of
	  samples and return the number of consumed/produced samples.

2015-12-08 11:15:34 +0100  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: duration query support based on the size property
	  https://bugzilla.gnome.org/show_bug.cgi?id=759126

2015-12-07 09:08:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>

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

2015-12-04 12:25:11 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	  multisocketsink: let downstream know we support metadata
	  Let downstream know that we support GstNetControlMessage metadata API.

2015-12-03 16:38:45 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Avoid pushing buffers before segment start
	  In the case where the stream doesn't have a framerate set and the frames
	  don't have a duration set, we still want to use the clipping path to
	  make sure we don't push buffers outside of the segment.
	  The problem was the previous iteration was setting a duration of 2s, which
	  meant that any buffer which was less than 2s before the segment start would
	  end up getting pushed.
	  Instead, use a saner 40ms (25fps single frame duration) to figure out whether
	  the frame could be within the segment or not

2015-12-02 20:19:43 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.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/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:
	  Drop usage of deprecated g-ir-scanner --strip-prefix flag

2015-12-02 18:16:05 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin2: fix "Attempt to unlock mutex that was not locked"
	  Introduced in commit ee44337f, caused the decodebin
	  test_text_plain_streams unit test to abort.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752651

2015-11-16 14:50:58 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstrawcaps.h:
	  playback: Expose XSUB formats by default
	  This is a workaround, we should remove this once we have a proper
	  decoder

2015-11-16 14:50:30 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Also consider XSUB as a subtitle format

2015-11-16 14:49:55 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: Add description for XSUB subpicture format

2015-11-16 14:49:19 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: 'DXSA' is the same as 'DXSB'
	  Which is subpicture/x-xsub

2015-07-21 09:58:56 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Rename GstStream => GstSyncStream
	  Avoid clashes with future GstStream from core

2015-12-02 09:00:31 -0500  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst-libs/gst/rtsp/gstrtspdefs.c:
	* gst-libs/gst/rtsp/gstrtspdefs.h:
	  rtspconnection: Update capitalization of x-sessioncookie
	  Some servers incorrectly parse header names with strict case-sensitivity.  For
	  compatibility with these systems change X-Sessioncookie to x-sessioncookie.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758921

2015-12-02 16:16:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Update buffering messages when removing an element that had buffering pending
	  Otherwise we'll remove that element while keeping its buffering message in our
	  list, and because of that never ever report buffering 100% as that element
	  will always be at a lower percentage.
	  This fixes e.g. seeking over Period boundaries in DASH and various other
	  issues when buffering happens between group switches.
	  Also use a new mutex for protecting the buffering messages. The object lock is
	  already used by gst_object_has_as_ancestor() and we need to use it now for
	  checking if the buffering message sender has the to-be-removed element as
	  ancestor.

2015-12-02 09:52:19 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/tcp/gstmultisocketsink.c:
	* gst/tcp/gstmultisocketsink.h:
	  multisocketsink: keep on reading when we stop sending
	  When we stop sending because we need more data, still keep a GSource
	  around to receive data from the clients.
	  Also handle read and write in the same go.

2015-12-01 19:57:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiobasesrc.c:
	  audiobasesrc: Post latency message on the bus after set_caps()
	  The latency is only known once the caps are known, and might change
	  whenever the caps are changing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758911

2015-09-25 14:47:48 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	  audiobasesink: Post latency message on the bus after set_caps()
	  Any latency query before this will not get the correct latency so a new
	  latency query should be triggered once the audio sink know its own latency.
	  Without this the initial latency query from the pipeline arrives too early
	  sometimes and the resulting latency is too short.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758911

2015-11-06 14:21:14 +0000  Thomas Bluemel <tbluemel@control4.com>

	* gst/playback/gstdecodebin2.c:
	  [PATCH] Fix a race condition accessing the decode_chain field.
	  Make sure that any access to the GstDecodeBin's decode_chain
	  field is protected using the EXPOSE_LOCK.  Also add a simple
	  reference counter to the GstDecodeChain structure so that when
	  the type_found signal fires it can hold onto the decode chain
	  even while the EXPOSE_LOCK is not held.  This should fix a
	  race condition if the type_found signal fires right in the
	  middle of a state change that messes with the same decode
	  chain.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755260

2015-08-20 17:30:38 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: early out on pad-added when the pad is inactive
	  The pad may be recently deactivated if the element is switched
	  back down very quickly.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752651

2015-08-20 17:29:36 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/playback/gstdecodebin2.c:
	  decodebin: lock the expose lock around decode_chain use
	  Helps with a crash in decodebin when quickly switching states.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752651

2015-11-28 14:24:55 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: accept wrong version field in OpusHead header
	  Some Opus files found on the wild have 0 in the version field of the
	  OpusHead header, instead of the correct value of 1. The files still
	  play, don't make this error fatal.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758754

2015-11-26 11:33:02 +0000  William Manley <will@williammanley.net>

	* gst-libs/gst/allocators/gstfdmemory.c:
	  allocators: add debug category for fd memory and allocator
	  Debugging can now be viewed by setting GST_DEBUG=fdmemory:9
	  https://bugzilla.gnome.org/show_bug.cgi?id=758744

2015-11-20 20:18:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/libs/tag.c:
	  tests: tags: add unit test for ID3v2 PRIVATE_DATA tag extraction
	  https://bugzilla.gnome.org/show_bug.cgi?id=730926

2014-09-29 14:17:39 +0530  Ravi Kiran K N <ravi.kiran@samsung.com>

	* gst-libs/gst/tag/gstid3tag.c:
	* gst-libs/gst/tag/id3v2frames.c:
	  id3v2frames: Handle private frames
	  Handle PRIV ID3 tag having owner information (string)
	  and binary data, add to tag messages list.
	  https://bugzilla.gnome.org/show_bug.cgi?id=730926

2015-11-20 19:15:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/tag/id3v2.c:
	  tags: id3: make sure to register private-id3v2-frame tag before using it

2015-11-17 17:07:37 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* tests/check/libs/rtspconnection.c:
	  rtspconnection: Add support for parsing custom headers
	  https://bugzilla.gnome.org/show_bug.cgi?id=758235

2015-11-15 02:58:54 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	* gst-libs/gst/pbutils/encoding-target.c:
	* gst-libs/gst/rtsp/gstrtspmessage.c:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	* tests/examples/encoding/encoding.c:
	  Remove unnecessary NULL checks before g_free()
	  g_free() is NULL-safe

2015-11-17 09:06:34 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* sys/ximage/ximagesink.c:
	* sys/xvimage/xvimagesink.c:
	  xvimagesink/ximagesink: Fix structure memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=758204

2015-11-12 14:39:17 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: guint8 can't hold value over 255
	  channels is a guint8, so the max value is 255 and checking if it value is
	  > 256 will never be false.
	  CID 1338687, CID 1338688

2015-11-12 14:18:03 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: remove unneeded check for unsigned < 0
	  Commit ff6d1a2a25b247688f38e117782a6b43d525706a changed sample's type from
	  gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
	  which means it can never be a negative value and the check making sure that
	  in_samples is >= 0 is never going to be false. Removing it.
	  CID 1338689

2015-11-11 14:44:55 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/check/libs/video.c:
	  tests:video: Fix overlay rectangle and buffer leak
	  Created overlay rectangle is not being freed in video tests
	  pix2 buffer is being created and not freed
	  https://bugzilla.gnome.org/show_bug.cgi?id=757927

2015-11-11 14:37:21 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst-libs/gst/pbutils/encoding-target.c:
	  pbutils:encoding-target: Fix string memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=757926

2015-11-11 15:02:39 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst-libs/gst/audio/audio-quantize.c:
	  audio-quantize: Fix dither_buffer memory leak
	  https://bugzilla.gnome.org/show_bug.cgi?id=757928

2015-11-11 00:59:16 +1100  Jan Schmidt <jan@centricular.com>

	* ext/vorbis/gstvorbisdec.c:
	  vorbisdec: Re-init on new caps
	  If we get new input caps, then reset the decoder
	  ready for new headers and fresh data. Makes
	  chained oggs work when reusing the decoder.

2015-11-02 23:12:19 +1100  Matthew Waters <matthew@centricular.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/video/Makefile.am:
	* gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
	* gst-libs/gst/video/gstvideoaffinetransformationmeta.h:
	* win32/common/libgstvideo.def:
	  videometa: add GstVideoAffineTransformationMeta
	  Adds a simple 4x4 affine transformations meta for passing arbitrary
	  transformations on buffers.
	  Based on patch by Matthieu Bouron
	  https://bugzilla.gnome.org/show_bug.cgi?id=731791

2015-11-10 09:52:24 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-converter: add output size argument
	  Make it possible to have a different number of output samples than input
	  samples when we, for example, want to add resampling later.

2015-11-07 00:43:55 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Check API arguments and assert if needed

2015-11-06 19:31:47 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Properly deactivate ghostpads
	  Just setting the ghostpad as flushing wasn't enough. It needs to be
	  consistent on the internal proxypad also, otherwise you end up in
	  situations where:
	  * a pending buffer on the target pad triggers the sticky event
	  propagation
	  * the default implementation sees that the proxypad is not flushing,
	  so it tries to push it to the other pad (the actual ghostpad)
	  * the ghostpad is flushing, so returns FALSE
	  * the push_event function sees that pushing the event failed...
	  * ... and pending buffer push returns GST_FLOW_ERROR, instead of
	  GST_FLOW_FLUSHING
	  By using gst_pad_set_active(FALSE), we ensure that both the ghostpad
	  and the proxypad are flushing/deactivated. The situation above will
	  no longer occur, and a GST_FLOW_FLUSHING will be returned.

2015-11-06 18:11:41 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioconvert/gstaudioconvertorc-dist.c:
	* gst/audioconvert/gstaudioconvertorc-dist.h:
	* gst/audioconvert/gstaudioconvertorc.orc:
	* gst/audioconvert/plugin.c:
	  audioconvert: fix build
	  Don't include file that is no longer generated, and remove some
	  files that are no longer needed because they have moved into the
	  lib. Fixes distcheck.

2015-11-06 18:00:41 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: require interleaved samples and no resampling
	  We can't yet do resampling or anything other than interleaved audio.

2015-11-06 17:54:21 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/gstaudiopack-dist.c:
	* gst-libs/gst/audio/gstaudiopack-dist.h:
	  audio: update ORC dist files

2015-11-06 17:49:00 +0100  Wim Taymans <wtaymans@redhat.com>

	* docs/plugins/Makefile.am:
	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiopack.orc:
	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.h:
	* tests/check/Makefile.am:
	* win32/common/libgstaudio.def:
	  audio-converter: move audio converter to audio libs
	  Move the audio-converter helper to the audio library.

2015-11-06 17:39:33 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-channel-mix.h:
	* gst-libs/gst/audio/audio.h:
	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	* win32/common/libgstaudio.def:
	  audio-channel-mix: move channel mixer to audio libs
	  Move the channel mixer code to the audio library

2015-11-06 17:29:22 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-info.c:
	* gst-libs/gst/audio/audio.c:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	  audio: add debug categories

2015-11-06 16:42:35 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: don't limit channelpositions
	  Don't set a limit on the channel positions, just like the metadata.

2015-11-06 16:03:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: simplify API a little
	  Remove the format and layout from the mix_samples function and use the
	  format when creating the channel mixer object. Also use a flag to handle
	  the unlikely case of non-interleaved samples like we do elsewhere.

2015-11-06 15:50:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	  channelmix: GstChannel -> GstAudioChannel
	  Rename GstChannel to GstAudioChannel

2015-11-06 13:02:19 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	  audio-quantize: update docs
	  Update docs
	  Add another flag for the quantizer

2015-11-06 12:46:36 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvertorc.orc:
	* gst/audioconvert/gstchannelmix.c:
	  audioconvert: cleanups and add some docs
	  Add docs for the internal audioconvert object before moving it to the
	  audio library.
	  Remove get_sizes and implement the trivial logic in the element.
	  Remove some unused orc functions

2015-11-06 12:46:12 +0100  Wim Taymans <wtaymans@redhat.com>

	* win32/common/libgstaudio.def:
	  defs: update defs

2015-11-06 12:37:14 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/gstaudiopack-dist.c:
	* gst-libs/gst/audio/gstaudiopack-dist.h:
	  audio: update orc files

2015-11-06 12:10:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiopack.orc:
	* gst/audioconvert/Makefile.am:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvert.h:
	* gst/audioconvert/gstaudioquantize.c:
	* gst/audioconvert/gstaudioquantize.h:
	* gst/audioconvert/gstfastrandom.h:
	  audioconvert: move audio quantize code to libs
	  Move the audio quantize code from audioconvert to the audio library.
	  work on making an audio converter helper function similar to the video
	  converter.
	  Fold fastrandom directly into the quantizer, add some ORC code to
	  optimize this later.

2015-11-05 12:42:56 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-channels.h:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* win32/common/libgstaudio.def:
	  audio-channels: rename get_default_mask
	  Rename _get_default_mask() to _get_fallback_mask() to make it more
	  clear that the function only provides a fallback if nothing else can be
	  done. Also clarify this in the documentation.
	  API: gst_audio_channel_get_fallback_mask()

2015-11-05 11:34:07 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/volume/gstvolume.c:
	  volume: Do not try to get binding value array if we are not processing any sample
	  In some conditions we might process empty buffers, calling
	  gst_control_binding_get_value_array in that case will lead
	  to the assertion:
	  (lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed

2015-11-05 10:40:18 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-channels.h:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst/audioconvert/gstaudioconvert.c:
	* win32/common/libgstaudio.def:
	  audio-channels: make method to get default channel-mask
	  Add a new method to get the default channel-mask.
	  Use the new method on audiodecoder and audioconvert.
	  API: gst_audio_channel_get_default_mask()

2014-11-10 11:11:37 +0100  Andreas Frisch <fraxinas@opendreambox.org>

	* tests/check/libs/video.c:
	  tests: Add a test for video blending over transparent frames
	  And fix the test_overlay_blend test where we blend over a
	  transparent frame and where expecting wrong results
	  https://bugzilla.gnome.org/show_bug.cgi?id=681447

2013-11-30 01:59:55 +0100  Arnaud Vrac <avrac@freebox.fr>

	* gst-libs/gst/video/video-blend.c:
	  video: blend using OVER operation
	  Also support all premultiplied/non-premultiplied source/destination
	  configurations
	  https://bugzilla.gnome.org/show_bug.cgi?id=681447

2015-11-03 16:51:47 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/gstoggstream.c:
	  oggdemux: Create full Opus caps with all fields
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 18:30:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/pbutils/Makefile.am:
	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/pbutils/codec-utils.h:
	* win32/common/libgstpbutils.def:
	  codec-utils: Add utilities for Opus caps and the OpusHead header
	  https://bugzilla.gnome.org/show_bug.cgi?id=757152

2015-11-03 11:11:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/gstoggmux.c:
	  oggmux: Use GstAudioClippingMeta for Opus for accurate end clipping
	  ... instead of relying on the segment. For the clipping at the start we assume
	  a proper value in the OpusHead, as generated by opusparse or opusenc.
	  Transmuxing in general is not guaranteed to produce the correct values, or
	  even have a OpusHead (e.g. when having RTP input).
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-03 10:58:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/Makefile.am:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggdemux: Add GstAudioClippingMeta for Opus for accurate start/end clipping
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-02 16:19:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/audio/audio.h:
	* gst-libs/gst/audio/gstaudiometa.c:
	* gst-libs/gst/audio/gstaudiometa.h:
	* win32/common/libgstaudio.def:
	  audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-02 11:19:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggstream.c:
	* ext/ogg/gstoggstream.h:
	  oggdemux: Allow start clipping for Opus
	  The granulepos does not have the pre-skip subtracted while timestamps do,
	  and the last granulepos will be shorter by the number of samples that should
	  be dropped because of padding in the end.
	  As such, extrapolating the granule of the beginning of the first frame will
	  lead to a negative value, which is not a problem but intentional.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757153

2015-11-03 16:38:09 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/audio/gstaudiopack-dist.c:
	* gst-libs/gst/audio/gstaudiopack-dist.h:
	  audio: update disted orc backup files

2015-11-03 14:08:25 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/audio/gstaudioclock.c:
	  audioclock: use GST_STIME_FORMAT for GstClockTimeDiff
	  GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
	  handle negative values better.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-03 13:44:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Print GstClockTimeDiff as a signed integer in debug logs

2015-11-03 11:59:09 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-format.c:
	* gst-libs/gst/audio/audio-format.h:
	* gst-libs/gst/audio/gstaudiopack.orc:
	* gst/audioconvert/audioconvert.c:
	  audio-format: add TRUNCATE_RANGE flag
	  Add a TRUNCATE_RANGE flag for unpack functions to fill the least
	  significate bits with 0 (as did the old code). Also add functions
	  that don't truncate. Use the TRUNC flag in audioconvert for
	  backwards compatibility for now.

2015-11-03 11:57:32 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/gstaudiopack.orc:
	  audiopack: improve pack functions
	  Avoid shifts by using convh functions.

2015-11-03 11:44:54 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioconvertorc.orc:
	* tests/check/elements/audioconvert.c:
	  audioconvert: change multiplier for int<->float conversion
	  Use (1 << 31) as the multiplier for int<->float conversions. This makes
	  sure that int->float conversions always end up with floats between
	  [-1.0, 1.0].
	  For the conversion from float to int, this multiplier will give the complete
	  int range after we perform clipping.
	  Change the unit test to take this into consideration.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301

2015-11-02 17:32:55 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

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

2015-11-02 16:36:35 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/ogg/gstoggmux.c:
	  oggmux: Print GstClockTimeDiff as a signed integer in debug logs

2015-11-02 16:09:52 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: Use GstClockTimeDiff and print signed integer in debug logs
	  Use GstClockTimeDiff and Clock macros to print signed integer time
	  differences in the debug logs.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-02 14:06:39 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* tests/examples/seek/scrubby.c:
	  examples: use GST_STIME_FORMAT for GstClockTimeDiff
	  GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
	  handle negative values better.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757480

2015-11-02 17:14:51 +0200  Sebastian Dröge <sebastian@centricular.com>
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