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Audio - Mixer conceptual documentation: fix footnote

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NOTRY=true

Bug: webrtc:12570
Change-Id: Ic413b302e0483dfc802e51d6766eb6504246405b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217523
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33934}
This commit is contained in:
Alessio Bazzica
2021-05-05 14:21:54 +02:00
committed by WebRTC LUCI CQ
parent a8b7b5d8a6
commit 7b1734a96b
1 changed files with 3 additions and 4 deletions
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7
modules/audio_mixer/g3doc/index.md
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@ -24,10 +24,9 @@ muted state) and it is identified by an SSRC[^1].
is used to retrieve the next 10 ms chunk of audio to be mixed. is used to retrieve the next 10 ms chunk of audio to be mixed.
[^1]: A synchronization source (SSRC) is the source of a stream of RTP packets, [^1]: A synchronization source (SSRC) is the source of a stream of RTP packets,
identified by a 32-bit numeric SSRC identifier carried in the RTP header
identified by a 32-bit numeric SSRC identifier carried in the RTP header so as so as not to be dependent upon the network address (see
not to be dependent upon the network address (see [RFC 3550](https://tools.ietf.org/html/rfc3550#section-3)).
[RFC 3550](https://tools.ietf.org/html/rfc3550#section-3)).
## AudioMixer ## AudioMixer