diff --git a/g3doc/sitemap.md b/g3doc/sitemap.md
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@@ -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
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@@ -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.