Re: [webrtc-insertable-streams] How to handle transforms largely changing frame size (#50)

Correct, the audio RED encoder is wrapping the opus one, stores a packet (or two, three, four, ...) and then essentially concatenates them.

Since audio typically isn't split into multiple packets that could create a problem with insertable streams but I think we have a lot of margin here. The audio encoder does know the transport overhead which is calculated from RTP header size or RTP header extensions currently (see [here for details](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/audio_codecs/audio_encoder.h;l=231;drc=8ffe0a431f557f8f0b99820374fd9cfddab4af53;bpv=1;bpt=1))

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Received on Wednesday, 3 February 2021 10:53:12 UTC