If jackd changes the buffer size or sample rate, jackaudiosink hangs
and can't be stopped. This also happens if jack is configured as slave
and a gstreamer pipeline is started on the slave machine while the jack
master isn't running yet. If the the jack master is started it changes
the buffer size / sample rate and jackaudiosink can't be stopped.
This fix calls jack_shutdown_cb when jack_sample_rate_cb or
jack_buffer_size_cb is called.
https://bugzilla.gnome.org/show_bug.cgi?id=771272
And actually calculate the field duration instead of a frame duration so
that we can properly timestamp output frames in fields=all mode.
This is probably still broken for reverse playback in telecine mode.
When start qmlglsink app, it will set NULL buffer to GstQSGTexture
in which case that qt_context_ will be a random value and cause
gst_gl_context_activate() fail.
https://bugzilla.gnome.org/show_bug.cgi?id=770925
This may cause a few packets to be processed by the parser, but it's
better than never pushing out buffers from a slightly broken stream
where no marker bits are set.
QuickTime.h is no longer available on OS X 10.12 (Sierra),
and both the header and the framework seem unnecessary
for compilation - at least as of 10.11 (El Capitan).
https://bugzilla.gnome.org/show_bug.cgi?id=770526
To be able to cap the number of allowed streams for one session.
This is useful for preventing DoS attacks, where a sender can change
SSRC for every buffer, effectively bringing rtpbin to a halt.
https://bugzilla.gnome.org/show_bug.cgi?id=770292
Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
the payload. In this case the payload modifications will not affect the
rtp buffer. So instead of modifying the payload buffer directly we
should modify the buffer that actually gets pushed on the adapter.
The functionality of all the tests was kept exactly the same. Some tests
were renamed:
test_push_forward_seq -> test_rtxsend_rtxreceive
test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
test_rtxreceive_data_reconstruction was testing that retransmitted
buffer produced by rtxsend was correctly transformed to the original
buffer by rtxreceive. Now we are checking for this in all the tests
where both rtxsend & rtxreceive are involved. That's why the test was
removed.