Re: [webrtc-extensions] How does a developer decide on a value for `playoutDelay` ? (#46)

> The problem is that the assumption was previously always that we wan to play out received media as soon as possible because that optimizes interactiveness, even if a shorter buffering necessarily entails a risk of reducing the quality when packets are dropped or don't arrive on time.

For what it's worth, this API has me excited because in our application, while interactivity is quite important, smoothness is also fairly important -- typically important enough that usually, we can tolerate between 100-500ms of latency if that's what it takes for smoothness, above which it's probably too much latency to be acceptable. At other, rarer moments, higher latencies up to 2 seconds are acceptable in the name of smoothness . At still other rare moments, interactivity is crucial above all-else, so playoutDelay = 0. This API seems to allow me to express that, and that maps well to what you described, so I think we're on the same page on that part, and I'm glad it supports that usecase.

I still think that application developers, including myself, want the more fine grained control that I described above. Ultimately, full control over the jitter buffer would be even better, but... baby steps. I think it's just key to remember that webrtc is being used for so many different things outside of video/audio conferencing, like remote rendering, gaming, AR/VR, animation playback, production work, etc... and developers out there are going to want to do things like grow their jitter buffer incredibly fast at the first sign of trouble (perhaps far faster than the browser's jitter buffer heuristic or enum presets would deem suitable) while still trying to keep it small if the connection is good, or things like that. While I understand it wasn't the ultimate goal of this API to allow for that kind of thing, personally I see it as a step in the right direction rather than sullying the API.

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Received on Friday, 24 July 2020 09:41:30 UTC