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webrtc/modules/audio_processing/g3doc/audio_processing_module.md
Artem Titov a617867a45 Reland "Migrate WebRTC documentation to new renderer" 
This reverts commit 0f2ce5cc1c.

Reason for revert: Downstream infrastructure should be ready now

Original change's description:
> Revert "Migrate WebRTC documentation to new renderer"
>
> This reverts commit 3eceaf4669.
>
> Reason for revert:
>
> Original change's description:
> > 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}
>
> Bug: b/258408932
> Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39209}

Bug: b/258408932
Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39231}
2023-01-31 09:30:04 +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.