Re: [webrtc-stats] Add RTCAudioPlayoutStats, synthesizedSamplesDuration and totalSamplesDuration (#682)

> What happens to these metrics if the incoming media pipeline is unavailable and the audio device is trying to pull media from the pipeline?

Added a clarification that synthesization is when underperforming and does not cover inbound RTP jitter buffer metrics.

> What happens if there are two media devices that are trying to play a particular audio call on two different devices.

In practise this is not possible, you may have a virtual device, but it may be that these metrics are not applicable if you attempt to do this.


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Received on Tuesday, 20 September 2022 14:53:50 UTC