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

> It sounds like you want to add a separate operating mode for NetEq, which I'm hesitant to add (for this potentially narrow use case) unless there is a lot of interest.

Yes, from my perspective there is no preference if there is separate operating mode, maxDelay attribute or something else, which would allow to force jitter buffer to operate in a specific delay as precisely as possible. That is why I have proposed different options to find out what suits the best. First there was option to change behaviour of existing jitterBufferTarget to follow the target more closely. That indeed can break some user expectations who are after quality of not loosing a word.  Then mode vs. maxDelay. If to add a mode it could also be called something like 'low latency mode', which would disable delay jumps and keeping delay higher after under run. There could be an option for users to prefer latency vs. quality (risk of loosing some words). If I am not mistaken even the whole audio pipeline in webrtc is built with low latency in mind and eliminating the risk of under runs comes second.

> It also sounds like a potential footgun for developers where setting a low delay would result in terrible quality.

If that is about changing behaviour of current jitterBufferTarget then I agree. But if there is separate max delay attribute than I don't see how the potential impact can be misunderstood. There is no need to play with the max delay if you don't understand the consequences. Even today developers can set max number of packets the jitter buffer can hold to a low value. Would the possibility to do it dynamically during a call be more dangerous? If there is good documentation in place it should be fine I think.

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Received on Thursday, 21 March 2024 17:32:06 UTC