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webrtc/modules/audio_processing/g3doc/audio_processing_module.md
Artem Titov 3eceaf4669 Migrate WebRTC documentation to new renderer 
Bug: b/258408932
Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39205}
2023-01-26 14:58:00 +00:00

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Audio Processing Module (APM)

Overview

The APM is responsible for applying speech enhancements effects to the microphone signal. These effects are required for VoIP calling and some examples include echo cancellation (AEC), noise suppression (NS) and automatic gain control (AGC).

The API for APM resides in [/modules/audio_processing/include][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include]. APM is created using the [AudioProcessingBuilder][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include/audio_processing.h] builder that allows it to be customized and configured.

Some specific aspects of APM include that:

  • APM is fully thread-safe in that it can be accessed concurrently from different threads.
  • APM handles for any input sample rates < 384 kHz and achieves this by automatic reconfiguration whenever a new sample format is observed.
  • APM handles any number of microphone channels and loudspeaker channels, with the same automatic reconfiguration as for the sample rates.

APM can either be used as part of the WebRTC native pipeline, or standalone.