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webrtc_m130/modules/audio_processing/g3doc/audio_processing_module.md

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Reland "Migrate WebRTC documentation to new renderer" This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. > > 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}
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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 09:19:37 +00:00
Reland "Migrate WebRTC documentation to new renderer" This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. > > 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}
2023-01-30 10:51:01 +00:00
# Audio Processing Module (APM)
Revert "Migrate WebRTC documentation to new renderer" This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. 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}
2023-01-26 20:18:46 +00:00
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 09:19:37 +00:00
## 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).
Deprecate AudioProcessingBuilder in favor of the BuiltinAudioProcessingBuilder Update comments and doc mentioning AudioProcessingBuilder accordingly Bug: webrtc:369904700 Change-Id: If837ddace5fedce94853c80500c6a832de8db9c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43376}
2024-11-05 10:48:41 +01:00
The API for APM resides in [`/api/audio/audio_processing.h`][https://webrtc.googlesource.com/src/+/refs/heads/main/api/audio/audio_processing.h].
APM is created using the [`BuiltinAudioProcessingBuilder`][https://webrtc.googlesource.com/src/+/refs/heads/main/api/audio/builtin_audio_processing_builder.h]
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 09:19:37 +00:00
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.
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