WebRTC: non 10ms audio frames possible?

Question

Has anyone tried to change the audio frame size in WebRTC? It uses 10ms frames. I need a different size, and the code doesn't look promising...

Answer 1

Fundamentally there's no reason you can't use non-10ms frames, but much of the code is written with that assumption. It would indeed likely be a serious undertaking to change it.

Answer 2

On the device you can use other audio frame (e.g. 20，40ms). For the codec，you can use other audio frame because codec has a audio buffer. I used silk codec and ios device.