WebRTC: non 10ms audio frames possible?
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21-06-2021 - |
質問
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...
解決
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.
他のヒント
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.
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