audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output. Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.
Fixes bug #665004.
After preroll the multiqueue limits are still set to the preroll
limits if use-buffering is set to TRUE. In that case we only want
time limits on the multiqueue if upstream is seekable.
Such streams were detected as seekable, as the query on the typefind
element was testing the m3u8 file listing the actual streams, and
not going through the demuxer(s).
We now check for seekability for each multiqueue following a demuxer,
so the query will flow through the elements which might prevent seeking.
https://bugzilla.gnome.org/show_bug.cgi?id=647769
API: GstVideoRate:force-fps
Changing the framerate during playback is not possible
with a capsfilter downstream if upstream is not using
gst_pad_alloc_buffer(). In that case there's no way in
0.10 to signal to videorate that the preferred framerate
has changed.
This new property will force the output framerate to
a specific value and can be changed during playback.
The ghostpad acceptcaps functions are not valid in this case because
we don't only accept the caps accepted by the target but could also
insert converters. Fixes bug #663892.
If we already saw the keyframes that we need to find,
we do not need to bisect to find them.
This will always be the case for streams with audio only,
where each frame acts as a keyframe, but will occasionally
also happen for streams with video.
https://bugzilla.gnome.org/show_bug.cgi?id=662475
Opus streams outside of Ogg may not have headers, and oggstream
may be used by oggmux to mux an Opus stream which does not come
from Ogg - thus without headers.
Determining headerness by packet count would strip the first two
packets from such an Opus stream, leading to a very small amount
of audio being clipped at the beginning of the stream.
