You signed in with another tab or window. Reload to refresh your session.You signed out in another tab or window. Reload to refresh your session.You switched accounts on another tab or window. Reload to refresh your session.Dismiss alert
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.
The text was updated successfully, but these errors were encountered:
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.
The text was updated successfully, but these errors were encountered: