gst-plugins-rs/net/rtp
Sebastian Dröge 31e836f4d6 rtp: basepay: Negotiate SSRC and PT with downstream if not set via property
This makes the new payloaders closer to the old ones, and makes usage in
webrtcbin easier.

Also properly configure default PT of subclasses. Previously any PT that
was set for these subclasses via g_object_new() would be overridden by
the default one during construction.

Additionally, do SSRC collision handling while queueing output packets.
This is the more natural place as that's where the SSRC is actually
used, it happens potentially earlier and also allows to drain any
pending packets before the SSRC change in the caps.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/557

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
..
src rtp: basepay: Negotiate SSRC and PT with downstream if not set via property 2024-08-10 08:06:40 +00:00
tests rtprecv: ensure that stopping the rtp src task does not critical 2024-06-24 13:13:28 +00:00
build.rs Rename rtpav1 plugin to just rtp 2022-10-23 20:04:43 +03:00
Cargo.toml rtp: Require bitstream-io < 2.4.0 2024-07-16 19:13:49 +03:00