- From: youennf via GitHub <noreply@w3.org>
- Date: Tue, 07 Apr 2026 15:43:33 +0000
- To: public-webrtc-logs@w3.org
The end goal is to send audio transformed by the most-of-the-time-but-no-always-realtime transform via a peer connection. The end receiver will handle the variable processing time via its jitter buffer. WebTransport apps can already do this easily, but RTCPeerConnection cannot. For other audio sinks (local play back, web audio), AudioTrackGenerator seems to be a poor choice. Even for MediaRecorder, it would break existing implementations (at least WebKit one) and the processing model of AudioTrackGenerator with VideoTrackGenerator would not be aligned there. We could narrow down the scope of the issue to RTCPeerConnection, which allows for the possibility to look at RTCPeerConnection specific solutions, for instance: - RTCEncodedSource proposal - A dedicated audio push RTCPeerConnection source -- GitHub Notification of comment by youennf Please view or discuss this issue at https://github.com/w3c/mediacapture-transform/issues/124#issuecomment-4200305208 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Tuesday, 7 April 2026 15:43:34 UTC