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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
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parentInitial commit. (diff)
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Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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+<!-- go/cmark -->
+<!--* freshness: {owner: 'peah' reviewed: '2021-04-13'} *-->
+
+# 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.