Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)

Yes, an explainer would help.

The use case, as I understand it, is multiple speakers playing out the same source in the same (physical) room, with different network delays, and the desire to keep them in sync well enough to allow echo cancellation to cancel all the speakers at once.

One alternative design would be to timestamp the audio packets and have a local variable "timestamp to playout offset" that would hard-lock the playout of a given timestamp to the local system clock (choice of this value, and adjustment for things like clock drift, are left as an exercise for the reader); samples that arrived too late to make their scheduled playout time would be replaced with silence.

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Received on Monday, 8 April 2024 08:28:55 UTC