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}
diff --git a/g3doc/sitemap.md b/g3doc/sitemap.md
index e58bf9d..5e0a9a8 100644
--- a/g3doc/sitemap.md
+++ b/g3doc/sitemap.md
@@ -17,6 +17,8 @@
* 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
diff --git a/modules/audio_processing/g3doc/audio_processing_module.md b/modules/audio_processing/g3doc/audio_processing_module.md
new file mode 100644
index 0000000..bb80dc9
--- /dev/null
+++ b/modules/audio_processing/g3doc/audio_processing_module.md
@@ -0,0 +1,26 @@
+# 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.
