ESP-ADF support for Opus over RTP stream? (WebRTC)

[tpbedford]

I've been trying to use ESP-ADF's opus_decoder and raw_opus_decoder pipeline elements on a webrtc connection sending Opus audio frames. I've got nowhere - I never get any output from the Opus pipeline element.

• Is the opus_decoder expect Opus frames on Ogg or another container?
• Is the raw_opus_decoder expecting me to manually extract the frame size, bitrate, etc. from the TOC byte at the start of the RTP frame? Can this decoder adjust to dynamic changes to bitrate mid-stream?

I've had the G711 codec in the pipeline just fine, and it'd be nice to use the Opus codec in the pipeline.

I've also had the whole setup working ok using 78/esp-opus managed component, as this claimed support for RTP-style Opus frames and I can confirm this works if I just use this codec at the point of receiving RTP frames, and just read the decoded PCM into the pipeline.