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As part of go/coil update code search links to not point to the "master" branch. Bug: chromium:1226942 Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081 Commit-Queue: Tony Herre <toprice@chromium.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34531}
54 lines
3.4 KiB
Markdown
54 lines
3.4 KiB
Markdown
<?% config.freshness.owner = 'alessiob' %?>
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<?% config.freshness.reviewed = '2021-04-21' %?>
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# The WebRTC Audio Mixer Module
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The WebRTC audio mixer module is responsible for mixing multiple incoming audio
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streams (sources) into a single audio stream (mix). It works with 10 ms frames,
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it supports sample rates up to 48 kHz and up to 8 audio channels. The API is
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defined in
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[`api/audio/audio_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/audio/audio_mixer.h)
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and it includes the definition of
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[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h),
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which describes an incoming audio stream, and the definition of
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[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h),
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which operates on a collection of
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[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
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objects to produce a mix.
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## AudioMixer::Source
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A source has different characteristic (e.g., sample rate, number of channels,
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muted state) and it is identified by an SSRC[^1].
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[`AudioMixer::Source::GetAudioFrameWithInfo()`](https://source.chromium.org/search?q=symbol:AudioMixer::Source::GetAudioFrameWithInfo%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
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is used to retrieve the next 10 ms chunk of audio to be mixed.
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[^1]: A synchronization source (SSRC) is the source of a stream of RTP packets,
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identified by a 32-bit numeric SSRC identifier carried in the RTP header
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so as not to be dependent upon the network address (see
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[RFC 3550](https://tools.ietf.org/html/rfc3550#section-3)).
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## AudioMixer
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The interface allows to add and remove sources and the
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[`AudioMixer::Mix()`](https://source.chromium.org/search?q=symbol:AudioMixer::Mix%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
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method allows to generates a mix with the desired number of channels.
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## WebRTC implementation
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The interface is implemented in different parts of WebRTC:
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* [`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h):
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[`audio/audio_receive_stream.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/audio_receive_stream.h)
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* [`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h):
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[`modules/audio_mixer/audio_mixer_impl.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_mixer/audio_mixer_impl.h)
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[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
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is thread-safe. The output sample rate of the generated mix is automatically
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assigned depending on the sample rate of the sources; whereas the number of
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output channels is defined by the caller[^2]. Samples from the non-muted sources
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are summed up and then a limiter is used to apply soft-clipping when needed.
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[^2]: [`audio/utility/channel_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/utility/channel_mixer.h)
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is used to mix channels in the non-trivial cases - i.e., if the number of
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channels for a source or the mix is greater than 3.
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