dbcf8afbcd
Bug: None Change-Id: I8f52c08d14b826192830b395f8e63e24809224f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215972 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33817}
3.4 KiB
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.
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).
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.
-
A synchronization source (SSRC) is the source of a stream of RTP packets, ↩︎
-
audio/utility/channel_mixer.his 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. ↩︎