[webrtc-stats] Feedback on audio capture stats (#741)

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

== Feedback on audio capture stats ==
Being able to know the input latency is a welcome change, but there are a number of problems with the prose:

`totalSamplesCaptured`: this should be named `totalFramesCaptured`, to handle multichannel and to be consistent with the rest of the Web Platform. A frame can contain multiple samples (one per audio channel).

`droppedSamplesDuration`: it's unclear what this means. When does a capture device drop audio? It is because of a catastrophic real-time load problem? A bug in the implementation?

> The frequency of the media source is necessarily the same as the frequency of encoders later in pipeline.

This is using incorrect terminology, I think this is referring to the sampling-rate or sampling-frequency, not the frequency.

Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/741 using your GitHub account

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Received on Tuesday, 7 March 2023 14:04:48 UTC