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Re: [webrtc-pc] reducing audio packet rate while track is disabled

From: Harald Alvestrand via GitHub <sysbot+gh@w3.org>
Date: Tue, 06 Feb 2018 09:37:22 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-363364987-1517909841-sysbot+gh@w3.org>
The "one black frame per second" language was added very recently; discussion in the group revealed that some people felt uneasy about some implementations just sending a black frame and then no more video data for a long time, while others felt that sending at full frame rate was just silly. This got added both to advise senders on what to do and to advise receivers on what they can expect.

I don't think audio codecs have the idea of "frame rate" (they use "sample rate", which is something quite different), so the same language can't be used. Suggestions for language that is reasonable for all audio codecs?

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Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1764#issuecomment-363364987 using your GitHub account
Received on Tuesday, 6 February 2018 09:37:24 UTC

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