When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
also make sure that the clock is updated with the elapsed time so that it
alsways increments even when the ringbuffer goes back to 0. When this happened
we need to adjust the sample position for the reset ringbuffer.
Fixes#594136
Add a property to disable rendering of video frames during preroll. This
will only work for videosinks that use the new ::show_frame() vfunc instead
of overriding basesink's preroll and render vfuncs directly.
API: GstVideoSink:show-preroll-frame
The old one did the mistake of not actually advancing the ringbuffer, it just
adjusted the segbase, introducing the whole lenght of the ringbuffer as an
extra delay in the pipeline.
Also make sure that the resync can never go back in time, producing the same
timestamps that has already been produced, as this can cause severe problems
for sinks and other synching mechanisms.
Fixes#594256
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.
API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
... which is the default seed when creating a new GRand. Because
GLib in older versions used buffered IO this would take a lot of time.
Instead use the global GRand for getting random numbers and keep the
three instance GRand for backward compatibility with a simple seed.
Fixes bug #593284.
