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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}
26 lines
1.3 KiB
Markdown
26 lines
1.3 KiB
Markdown
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<!--* freshness: {owner: 'peah' reviewed: '2021-04-13'} *-->
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# Audio Processing Module (APM)
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## Overview
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The APM is responsible for applying speech enhancements effects to the
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microphone signal. These effects are required for VoIP calling and some
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examples include echo cancellation (AEC), noise suppression (NS) and
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automatic gain control (AGC).
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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].
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APM is created using the [`AudioProcessingBuilder`][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include/audio_processing.h]
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builder that allows it to be customized and configured.
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Some specific aspects of APM include that:
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* APM is fully thread-safe in that it can be accessed concurrently from
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different threads.
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* APM handles for any input sample rates < 384 kHz and achieves this by
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automatic reconfiguration whenever a new sample format is observed.
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* APM handles any number of microphone channels and loudspeaker channels, with
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the same automatic reconfiguration as for the sample rates.
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APM can either be used as part of the WebRTC native pipeline, or standalone.
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