- From: Eldar Rello via GitHub <sysbot+gh@w3.org>
- Date: Thu, 22 Feb 2024 16:34:38 +0000
- To: public-webrtc-logs@w3.org
Under low latency streaming there is N38 "The application must be able to control the jitter buffer and rendering delay. This requirement is addressed by jitterBufferTarget, defined in [[WebRTC-Extensions](https://www.w3.org/TR/webrtc-nv-use-cases/#bib-webrtc-extensions)] Section 6." In my opinion it is only half addressed, as there is control over minimum jitter buffer delay but nothing to limit maximum delay. The use case, which I am interested to improve is when multiple participants join a meeting from their laptops in the same room then the remote audio playback has to be synchronised to avoid echo from that room. Current AEC in Webrtc is not able to cancel out playback signal from other laptops if that happens earlier than the reference playback signal. If the synchronisation of playbacks is in +/- 10ms the AEC can handle this pretty good. -- GitHub Notification of comment by eldarrello Please view or discuss this issue at https://github.com/w3c/webrtc-extensions/issues/199#issuecomment-1959825490 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Thursday, 22 February 2024 16:34:40 UTC