Audio quality metrics similar to RTCAudioPlayoutStats

[hoch]

Would it be useful to have audio quality metrics in Web Audio API?

An example in WebRTC Stats Spec:
https://www.w3.org/TR/2023/CRD-webrtc-stats-20230303/#playoutstats-dict*

Although the current status of the example above seems a bit controversial (w3c/webrtc-stats#742, w3c/webrtc-stats#741), on 3/9 the Audio WG members had a general consensus that we need a way to inspect buffer under/overrun at the boundary between MediaStream (push model) and Web Audio API (pull model). The group also agreed that the MediaStreamTrack might be a better place to have such API.

This is a reference point of the group discussion, but ultimately we might want to voice our request to the relevant spec/WG.

[hoch]

Teleconference on 4/6:
We raised issues to the relevant WG (WebRTC WG) and there is no action item for the Audio WG.

[mehagar]

Would w3c/mediacapture-extensions#117 solve this?