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Adding g3doc for AudioProcessingModule (APM)

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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}
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Per Åhgren 2021-04-14 09:19:37 +00:00 committed by Commit Bot
parent 9861f960c3
commit dea5721efb
2 changed files with 28 additions and 0 deletions
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g3doc/sitemap.md
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* AudioEngine
* [ADM](/modules/audio_device/g3doc/audio_device_module.md)
* [Audio Coding](/modules/audio_coding/g3doc/index.md)
* AudioProcessingModule
* [APM](/modules/audio_processing/g3doc/audio_processing_module.md)
* Video
* DataChannel
* PeerConnection

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modules/audio_processing/g3doc/audio_processing_module.md Normal file
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# Audio Processing Module (APM)
<?% config.freshness.owner = 'peah' %?>
<?% config.freshness.reviewed = '2021-04-13' %?>
## 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.