[webrtc-pc] Tweak MediaStreamTrack.onunmute: at decode, not reception (#3076)

henbos has just created a new issue for https://github.com/w3c/webrtc-pc:

== Tweak MediaStreamTrack.onunmute: at decode, not reception ==
The remote `MediaStreamTrack.onunmute` is supposed to unmute when "media arrives" such that you can listen to the event and display the track when ready for playout. The examples in the spec listens to this events and sets `srcObject` of an audio or video element in response to this.

But "[whenever an RTCRtpReceiver receives data on an RTP source](https://w3c.github.io/webrtc-pc/#mediastreamtrack-network-use)" actually needs to be tweaked for two reasons:
- The jitter buffer can be long for network reasons or because of `RTCRtpReceiver.jitterBufferTarget = 4000;`, we should not unmute the track 4 seconds earlier than the track being ready for playout.
- With Insertable Streams, RTP packets never actually arrive, but we would expect the track to unmute anyway.

I think we should align with [getSynchronizationSources()](https://w3c.github.io/webrtc-pc/#dom-rtcrtpcontributingsource) which talks about media being delivered to the MediaStreamTrack instead:
> When an audio or video frame from one or more RTP packets is delivered to the RTCRtpReceiver's MediaStreamTrack

This equivalent to "when a frame has been decoded", shall we just say that instead?

@fippo @jan-ivar @alvestrand 

Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/3076 using your GitHub account


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Received on Monday, 29 September 2025 07:50:57 UTC