- From: Harald Alvestrand via GitHub <noreply@w3.org>
- Date: Mon, 28 Jul 2025 06:56:59 +0000
- To: public-webrtc-logs@w3.org
Hmm... I think I got the wrong numbers; this is not about jitter buffer overflow. These numbers are useful when the sender's clock and the receiver's clock are not running at the same speed; if, for instance, the sender sends 8001 samples per second and the receiver plays them out at 7999 samples per second, we need to delete two samples from the input stream every second in order to ensure that the playout doesn't drift away from "real time". -- GitHub Notification of comment by alvestrand Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/809#issuecomment-3125803886 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Monday, 28 July 2025 06:57:00 UTC