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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 and it includes the definition of AudioMixer::Source, which describes an incoming audio stream, and the definition of AudioMixer, which operates on a collection of AudioMixer::Source 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 SSRC1. AudioMixer::Source::GetAudioFrameWithInfo() is used to retrieve the next 10 ms chunk of audio to be mixed.

AudioMixer

The interface allows to add and remove sources and the AudioMixer::Mix() method allows to generates a mix with the desired number of channels.

WebRTC implementation

The interface is implemented in different parts of WebRTC:

AudioMixer 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 caller2. Samples from the non-muted sources are summed up and then a limiter is used to apply soft-clipping when needed.

Footnotes

  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). ↩

  2. 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. ↩