1<?% config.freshness.owner = 'alessiob' %?> 2<?% config.freshness.reviewed = '2021-04-21' %?> 3 4# The WebRTC Audio Mixer Module 5 6The WebRTC audio mixer module is responsible for mixing multiple incoming audio 7streams (sources) into a single audio stream (mix). It works with 10 ms frames, 8it supports sample rates up to 48 kHz and up to 8 audio channels. The API is 9defined in 10[`api/audio/audio_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/audio/audio_mixer.h) 11and it includes the definition of 12[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h), 13which describes an incoming audio stream, and the definition of 14[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h), 15which operates on a collection of 16[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h) 17objects to produce a mix. 18 19## AudioMixer::Source 20 21A source has different characteristic (e.g., sample rate, number of channels, 22muted state) and it is identified by an SSRC[^1]. 23[`AudioMixer::Source::GetAudioFrameWithInfo()`](https://source.chromium.org/search?q=symbol:AudioMixer::Source::GetAudioFrameWithInfo%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h) 24is used to retrieve the next 10 ms chunk of audio to be mixed. 25 26[^1]: A synchronization source (SSRC) is the source of a stream of RTP packets, 27 identified by a 32-bit numeric SSRC identifier carried in the RTP header 28 so as not to be dependent upon the network address (see 29 [RFC 3550](https://tools.ietf.org/html/rfc3550#section-3)). 30 31## AudioMixer 32 33The interface allows to add and remove sources and the 34[`AudioMixer::Mix()`](https://source.chromium.org/search?q=symbol:AudioMixer::Mix%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h) 35method allows to generates a mix with the desired number of channels. 36 37## WebRTC implementation 38 39The interface is implemented in different parts of WebRTC: 40 41* [`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h): 42 [`audio/audio_receive_stream.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/audio_receive_stream.h) 43* [`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h): 44 [`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) 45 46[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h) 47is thread-safe. The output sample rate of the generated mix is automatically 48assigned depending on the sample rate of the sources; whereas the number of 49output channels is defined by the caller[^2]. Samples from the non-muted sources 50are summed up and then a limiter is used to apply soft-clipping when needed. 51 52[^2]: [`audio/utility/channel_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/utility/channel_mixer.h) 53 is used to mix channels in the non-trivial cases - i.e., if the number of 54 channels for a source or the mix is greater than 3. 55