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
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
Commit ff6d1a2a25 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
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.
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.
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
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.
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()
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
... 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
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
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.
