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webrtc/modules/audio_processing/g3doc/audio_processing_module.md
Per Åhgren dea5721efb Adding g3doc for AudioProcessingModule (APM) 
Bug: webrtc:12569
Change-Id: I8fa896a5afa9791ad6d8c2b5011d1e75ca068df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215141
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33726}
2021-04-14 09:40:25 +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/+/master:third_party/webrtc/modules/audio_processing/include]. APM is created using the [AudioProcessingBuilder][https://source.chromium.org/chromium/chromium/src/+/master: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.