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Re: [webrtc-pc] getSynchronizationSources (and getContriburingSources) should work for video too

From: Philipp Hancke via GitHub <sysbot+gh@w3.org>
Date: Wed, 12 Sep 2018 14:47:47 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-420676417-1536763666-sysbot+gh@w3.org>
> but while unmuting happens as a result of an RTP packet arriving, muting only happens as a result of negotiation (setRemoteDescription's muteTracks list).
I do not think that is true, see https://w3c.github.io/webrtc-pc/#mediastreamtrack-network-use
And I am not happy with muted flipflopping for both network and api events.

getSynchronizationSources() and getContributingSources() was built for audio-levels mainly (which is also the reason it does not make much sense for video -- i do not think any video rtp mixer uses csrcs...). If we want a timestamp for last-packet-received we should not use this API but expose something on the receiver.

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Received on Wednesday, 12 September 2018 14:47:48 UTC

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