- From: henbos via GitHub <sysbot+gh@w3.org>
- Date: Wed, 08 Feb 2017 13:38:35 +0000
- To: public-webrtc-logs@w3.org
Should this timestamp continously increasing between frames/audio samples, e.g. not discontinously jumping every time a frame (or audio buffer, how would that work?) arrives, that would complicate things if audio and video jumps by different amount of times or at different points in time. What happens if you fall behind because there is nothing to play? `currentPlayoutTimestamp` continues to increase but `HTMLMediaElement.currentTime` ("remoteVideo.currentTime") stops? Is the goal this? ``` // E2E delay let e2eVideoDelay = videoTrackStats.currentPlayoutTimestamp - remoteVideo.currentTime; let e2eAudioDelay = audioTrackStats.currentPlayoutTimestamp - remoteAudio.currentTime; // A/V sync let audioOffsetFromVideo = e2eVideoDelay - e2eAudioDelay; ``` Which unlike `remoteAudio.currentTime - remoteVideo.currentTime` works even if audio and video did not start sending at exactly the same time from the remote source. -- GitHub Notification of comment by henbos Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/158#issuecomment-278330845 using your GitHub account
Received on Wednesday, 8 February 2017 13:38:41 UTC