The time_position field of the stream is offset by the media_start
of its QtDemuxSegment compared to the start of the GstSegment of
the demuxer, take it into account when making comparisons.
If the conflict is detected when sending a packet, then also send an
upstream event to tell the source to reconfigure itself.
Also ignore the collision if we see more than one collision from the same
remote source to avoid problems on loops.
Add a new property "do-aggregate"* to the H.264 RTP payloader which
enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
packets are bundled instead of sent immediately, up until the MTU size.
Bundles also end at access unit boundaries or when packets have to be
fragmented.
*: The property-name is kept generic since it might apply more widely,
e.g. STAP-B or MTAP.
[1]: https://tools.ietf.org/html/rfc6184#section-5.7
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434
If an rtx packet arrives that hasn't been requested (it might
have been requested from prior to a reset), ignore it so that
it doesn't inadvertently trigger a clock skew.
In the case of reordered packets, calculating skew would cause
pts values to be off. Only calculate skew when packets come
in as expected. Also, late RTX packets should not trigger
clock skew adjustments.
Fixes#612
mpegaudioparse suggests MP3 needs 10 or 30 frames of lead-in (depending on
mpegaudioversion, which we don't know here), thus provide at least 30 frames
lead-in for such cases as a followup to commit cbfa4531ee.
The pre_push_frame default clipping behaviour was introduced in 2010
with commit 30be03004e and modified with commit 4163969a24 in 2011,
when most parsers didn't implement a pre_push_frame yet. Not having it
meant that clipping was done by default. Those that did implement a
pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag
adjusted as part of the 2011 refactor work.
All other parsers got a pre_push_frame vfunc implementation only in
2013, but seem to have forgot to keep the clipping behaviour, as
was done automatically when a pre_push_frame implementation doesn't
exist for the parser. aacparse lost it with commit 91d4abcea in
July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting
in commits 6f89b430e, d2ab5199b, 29f2cae12, 753d3c23a and 292780574.
Otherwise it can happen that we receive a caps event, then another caps
event and only then buffers. We would then send out the first caps event
in the stream but mark buffers with the caps version of the second caps
event.
Otherwise it can happen that we already collected 7 caps, miss the 8th
caps packet (packet loss) and then re-use the 1st caps for the following
buffers instead of the 8th caps which will likely cause errors further
downstream unless both caps are accidentally the same.
Keeping old caps around does not seem to have any value other than
potentially causing errors. We would always receive new caps whenever
they change (even if they were previous ones) and it's very unlikely
that they happen to be exactly the same as the previous ones.
Also after having received new caps or a buffer with a next caps
version, no buffers with old caps version will arrive anymore.
Make sure to clear any master clock on the media_clock
before unreffing it to release the timer callback that's
updating the clock and keeping it reffed.
Clear the mastering_display_info_present field explicitly
after reallocating the track context into a video context
to avoid uninitialised warnings in valgrind
