Commit 5f98203b authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.7.1

parent 1975bdcd
=== release 1.7.1 ===
2015-12-24 Sebastian Dröge <>
releasing 1.7.1
2015-12-24 12:22:04 +0100 Sebastian Dröge <>
* po/nl.po:
* po/sv.po:
* po/zh_CN.po:
po: Update translations
2015-12-11 15:38:00 +0100 Thibault Saunier <>
* gst-libs/gst/pbutils/encoding-profile.c:
encodebin: Implement an encoding profile serialization format
2015-12-21 00:43:49 +0100 Koop Mast <>
configure: Make -Bsymbolic check work with clang.
Update the -Bsymbolic check with the version glib has. This version
works with clang.
2015-12-03 11:53:05 +0900 Kazunori Kobayashi <>
* 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.
2015-12-16 18:11:05 -0300 Thiago Santos <>
* 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 <>
* 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
2015-12-17 13:39:01 +0100 Sebastian Dröge <>
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 <>
* 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 <>
* gst/tcp/gstmultifdsink.c:
multifdsink: fix typo in GST_WARNING_OBJECT
This should make easier to parse the debug logs.
2014-04-10 15:36:15 +0100 Vincent Penquerc'h <>
* 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 <>
* 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 <>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: clear convert object
2015-12-16 09:35:38 +0100 Sebastian Dröge <>
* 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 <>
* ext/alsa/gstalsasrc.c:
Revert "alsasrc: Disable HW timestamp"
This reverts commit 3642e9a3913a35c00f379034780c27298d09929c.
2015-11-10 12:54:23 -0500 Xavier Claessens <>
* 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
2015-09-24 18:26:51 -0400 Nicolas Dufresne <>
* 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 <>
* 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 <>
* 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.
2015-12-14 09:10:16 -0500 Evan Callaway <>
* 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
2015-12-14 13:11:21 +0100 Sebastian Dröge <>
* 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 <>
* 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 <>
* 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 <>
* gst-libs/gst/riff/riff-media.c:
riff: add FourCC aliases
Support media using the aliases defined in that are
exact duplicates of already known codes.
2015-12-12 17:04:21 +0100 Luis de Bethencourt <>
* 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 <>
* 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.
2015-12-11 21:42:00 +0800 Song Bing <>
* gst-libs/gst/video/convertframe.c:
video/convertframe: Add crop meta support via videocrop
2015-12-11 11:01:53 +0000 Tim-Philipp Müller <>
* 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 <>
* 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 <>
* 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 <>
* 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 <>
* 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 <>
* 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
2015-12-09 17:16:26 +0100 Wim Taymans <>
* 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 <>
* gst-libs/gst/app/gstappsrc.c:
appsrc: duration query support based on the size property
2015-12-07 09:08:05 -0500 Nicolas Dufresne <>
* common:
Automatic update of common submodule
From b319909 to 86e4663
2015-12-04 12:25:11 +0100 Wim Taymans <>
* 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 <>
* 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 <>
* gst-libs/gst/allocators/
* gst-libs/gst/app/
* gst-libs/gst/audio/
* gst-libs/gst/fft/
* gst-libs/gst/pbutils/
* gst-libs/gst/rtp/
* gst-libs/gst/rtsp/
* gst-libs/gst/sdp/
* gst-libs/gst/tag/
* gst-libs/gst/video/
Drop usage of deprecated g-ir-scanner --strip-prefix flag
2015-12-02 18:16:05 +0000 Tim-Philipp Müller <>
* 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.
2015-11-16 14:50:58 +0100 Edward Hervey <>
* gst/playback/gstrawcaps.h:
playback: Expose XSUB formats by default
This is a workaround, we should remove this once we have a proper
2015-11-16 14:50:30 +0100 Edward Hervey <>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Also consider XSUB as a subtitle format
2015-11-16 14:49:55 +0100 Edward Hervey <>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Add description for XSUB subpicture format
2015-11-16 14:49:19 +0100 Edward Hervey <>
* 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 <>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Rename GstStream => GstSyncStream
Avoid clashes with future GstStream from core
2015-12-02 09:00:31 -0500 Evan Callaway <>
* 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.
2015-12-02 16:16:22 +0200 Sebastian Dröge <>
* 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
2015-12-02 09:52:19 +0100 Wim Taymans <>
* 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 <>
* 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.
2015-09-25 14:47:48 +0200 Michael Olbrich <>
* 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.
2015-11-06 14:21:14 +0000 Thomas Bluemel <>
* 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
2015-08-20 17:30:38 +0100 Vincent Penquerc'h <>
* 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.
2015-08-20 17:29:36 +0100 Vincent Penquerc'h <>
* gst/playback/gstdecodebin2.c:
decodebin: lock the expose lock around decode_chain use
Helps with a crash in decodebin when quickly switching states.
2015-11-28 14:24:55 +0000 Luis de Bethencourt <>
* 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.
2015-11-26 11:33:02 +0000 William Manley <>
* 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
2015-11-20 20:18:34 +0000 Tim-Philipp Müller <>
* tests/check/libs/tag.c:
tests: tags: add unit test for ID3v2 PRIVATE_DATA tag extraction
2014-09-29 14:17:39 +0530 Ravi Kiran K N <>
* 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.
2015-11-20 19:15:22 +0000 Tim-Philipp Müller <>
* 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 <>
* gst-libs/gst/rtsp/gstrtspconnection.c:
* tests/check/libs/rtspconnection.c:
rtspconnection: Add support for parsing custom headers
2015-11-15 02:58:54 -0800 Reynaldo H. Verdejo Pinochet <>
* 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 <>
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
xvimagesink/ximagesink: Fix structure memory leak
2015-11-12 14:39:17 +0000 Luis de Bethencourt <>
* 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 <>
* 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 <>
* 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
2015-11-11 14:37:21 +0900 Vineeth TM <>
* gst-libs/gst/pbutils/encoding-target.c:
pbutils:encoding-target: Fix string memory leak
2015-11-11 15:02:39 +0900 Vineeth TM <>
* gst-libs/gst/audio/audio-quantize.c:
audio-quantize: Fix dither_buffer memory leak
2015-11-11 00:59:16 +1100 Jan Schmidt <>
* 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 <>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/
* 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
2015-11-10 09:52:24 +0100 Wim Taymans <>
* 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 <>
* gst-libs/gst/pbutils/gstdiscoverer.c:
discoverer: Check API arguments and assert if needed
2015-11-06 19:31:47 +0100 Edward Hervey <>
* 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
* 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
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 <>
* 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 <>
* 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 <>
* 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 <>
* docs/plugins/
* gst-libs/gst/audio/
* 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/
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* tests/check/
* 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 <>
* gst-libs/gst/audio/
* gst-libs/gst/audio/audio-channel-mix.c:
* gst-libs/gst/audio/audio-channel-mix.h:
* gst-libs/gst/audio/audio.h:
* gst/audioconvert/
* 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 <>
* 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: