[webrtc-stats] Need metrics for playout glitches (#676)

henbos has just created a new issue for https://github.com/w3c/webrtc-stats:

== Need metrics for playout glitches ==
After audio has been produced (decoding, jitter buffer, concealments etc) it is pulled for playout but the playout client may not be able to provide samples to the device in time causing playout glitches. There is currently no way to measure this in getStats().

When this happens, samples are synthesized. Similar to concealedSamples, in practise in Chrome this would be silence, so silentConcealedSamples.

We need per-playout metrics. How about?
- RTCAudioPlayoutStats.synthesizedSamplesDuration: total duration of synthesized samples during playout.
- RTCAudioPlayoutStats.totalSamplesDuration: the total playout

Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/676 using your GitHub account

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Received on Tuesday, 20 September 2022 08:22:59 UTC