There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession. This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other. The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.
Demultiplexing happens internally instead of relying on separate
elements.
Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
Simplifies state tracking and potentially reduces latency as it's not
necessary to wait until all fragments of an OBU are received.
The last OBU of a TU is marked with the marker flag to allow parsers to
detect this without first seeing the beginning of the next TU.
Also use a simple `Vec` for collecting complete OBUs instead of a
`gst_base::Adapter` as this reduces the number of allocations.
And also handle invalid packets a little bit more gracefully.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
