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+<!-- go/cmark -->
+<!--* freshness: {owner: 'alessiob' reviewed: '2021-04-21'} *-->
+
+# The WebRTC Audio Mixer Module
+
+The WebRTC audio mixer module is responsible for mixing multiple incoming audio
+streams (sources) into a single audio stream (mix). It works with 10 ms frames,
+it supports sample rates up to 48 kHz and up to 8 audio channels. The API is
+defined in
+[`api/audio/audio_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/audio/audio_mixer.h)
+and it includes the definition of
+[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h),
+which describes an incoming audio stream, and the definition of
+[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h),
+which operates on a collection of
+[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+objects to produce a mix.
+
+## AudioMixer::Source
+
+A source has different characteristic (e.g., sample rate, number of channels,
+muted state) and it is identified by an SSRC[^1].
+[`AudioMixer::Source::GetAudioFrameWithInfo()`](https://source.chromium.org/search?q=symbol:AudioMixer::Source::GetAudioFrameWithInfo%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+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,
+    identified by a 32-bit numeric SSRC identifier carried in the RTP header
+    so as not to be dependent upon the network address (see
+    [RFC 3550](https://tools.ietf.org/html/rfc3550#section-3)).
+
+## AudioMixer
+
+The interface allows to add and remove sources and the
+[`AudioMixer::Mix()`](https://source.chromium.org/search?q=symbol:AudioMixer::Mix%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+method allows to generates a mix with the desired number of channels.
+
+## WebRTC implementation
+
+The interface is implemented in different parts of WebRTC:
+
+*   [`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h):
+    [`audio/audio_receive_stream.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/audio_receive_stream.h)
+*   [`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h):
+    [`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)
+
+[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+is thread-safe. The output sample rate of the generated mix is automatically
+assigned depending on the sample rate of the sources; whereas the number of
+output channels is defined by the caller[^2]. Samples from the non-muted sources
+are summed up and then a limiter is used to apply soft-clipping when needed.
+
+[^2]: [`audio/utility/channel_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/utility/channel_mixer.h)
+    is used to mix channels in the non-trivial cases - i.e., if the number of
+    channels for a source or the mix is greater than 3.