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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.
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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.
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Add a property to handle GstNetworkMessage events. These events contain
a buffer that is sent on the socket to allow for simple bidirectional
communication.
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Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
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https://bugzilla.gnome.org/show_bug.cgi?id=759126
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From b319909 to 86e4663
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Let downstream know that we support GstNetControlMessage metadata API.
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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
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Introduced in commit ee44337f, caused the decodebin
test_text_plain_streams unit test to abort.
https://bugzilla.gnome.org/show_bug.cgi?id=752651
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This is a workaround, we should remove this once we have a proper
decoder
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Which is subpicture/x-xsub
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Avoid clashes with future GstStream from core
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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
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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.
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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.
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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
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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
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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
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The pad may be recently deactivated if the element is switched
back down very quickly.
https://bugzilla.gnome.org/show_bug.cgi?id=752651
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Helps with a crash in decodebin when quickly switching states.
https://bugzilla.gnome.org/show_bug.cgi?id=752651
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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
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Debugging can now be viewed by setting GST_DEBUG=fdmemory:9
https://bugzilla.gnome.org/show_bug.cgi?id=758744
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https://bugzilla.gnome.org/show_bug.cgi?id=730926
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=758235
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g_free() is NULL-safe
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https://bugzilla.gnome.org/show_bug.cgi?id=758204
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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
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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
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=757926
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https://bugzilla.gnome.org/show_bug.cgi?id=757928
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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.
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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
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Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
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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.
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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.
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We can't yet do resampling or anything other than interleaved audio.
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Move the audio-converter helper to the audio library.
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Move the channel mixer code to the audio library
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Don't set a limit on the channel positions, just like the metadata.
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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.
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Rename GstChannel to GstAudioChannel
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