Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.
A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.
RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=761947
The fomula, 'offset = time / rate', is correct only if
the rate is never changed. When the rate is changed,
the offset should be re-calculated based on the previous
offset.
https://bugzilla.gnome.org/show_bug.cgi?id=791269
We can either receive an element that is floating or not and need to
accomodate that in the signal return values. Do so by removing the
floating flag.
https://bugzilla.gnome.org/show_bug.cgi?id=792597
If timestamp goes forwards more than allowed, we consider that the
timestamp belongs to the previous counting, so the extended timestamp
is unwrapped.
https://bugzilla.gnome.org/show_bug.cgi?id=783443
Tests and documentation will follow separately.
The mixer elements in the opengl plugin need to stay
in -bad for now since they use GstVideoAggregator.
https://bugzilla.gnome.org/show_bug.cgi?id=754094
The videoscale test takes eternities to run, that's not
great. Split the test into multiple ones. That way they
can be run in parallel. Reduces time to run all tests in
-base from 29 secs to 12 secs when using meson/ninja.
Test that a pipeline can change from PLAYING to PAUSED and back in
the following scenarios:
1. One track reach EOS after pushed some buffers while another track
still pushes buffers
2. One track reach EOS without buffers while another track still pushes
buffers
https://bugzilla.gnome.org/show_bug.cgi?id=736655
There don't seem to be any unit tests for the socket handling elements. As
I am about to attempt some refactorings I've added some basic tests which
exercise some of the happy-paths in tcpclientsrc, tcpserversrc,
tcpserversink and tcpclientsink. They should let me know if I've caused
serious breakage.
They are far from exhaustive but are sufficient for me to have caught a few
memory-leaks in the existing code.
https://bugzilla.gnome.org/show_bug.cgi?id=739544
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.
See https://bugzilla.gnome.org/show_bug.cgi?id=732415
MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
Add a simple playback test that makes sure that video decoder pushes
buffers in the same order it receives and that it respects the
set timestamps and durations
