[Feature request] Improvements of opus support in module-rtp
Dear community,
I am evaluating usage of module-rtp-sink in a setup to forward audio streams to clients via WebRTC. A really simple stack could consist of pipewire -> module-rtp-sink -> janus (WebRTC Server) -> clients.
Unfortunately, module-rtp-sink does not yet provide any means to control the opus encoder despite from sample rate (which I suspect might not be working, as in https://gitlab.freedesktop.org/pipewire/pipewire/-/blame/master/src/modules/module-rtp/opus.c#L326 there are hard coded frame counts that only match for 48 kHz sample rate (see https://opus-codec.org/docs/opus_api-1.2/group__opus__multistream.html#gaff832211e572536941b9d6094f9f42ce) ) and channel count.
In my application, I would like to enable FEC encoding.
Is it feasible to develop module-rtp further or should I rely on other means e.g. use gstreamer to forward audio from pipewire to RTP?