Add g3doc for audio coding module.
Bug: webrtc:12567
Change-Id: I553ba45fe9d95f3471b2134c3631a74ed600dc3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215079
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33720}
diff --git a/g3doc/sitemap.md b/g3doc/sitemap.md
index d7445c5..81e3b0d 100644
--- a/g3doc/sitemap.md
+++ b/g3doc/sitemap.md
@@ -15,6 +15,7 @@
* Audio
* AudioEngine
* [ADM](/modules/audio_device/g3doc/audio_device_module.md)
+ * [Audio Coding](/modules/audio_coding/g3doc/index.md)
* Video
* DataChannel
* PeerConnection
diff --git a/modules/audio_coding/g3doc/index.md b/modules/audio_coding/g3doc/index.md
new file mode 100644
index 0000000..0be22f7
--- /dev/null
+++ b/modules/audio_coding/g3doc/index.md
@@ -0,0 +1,32 @@
+<?% config.freshness.owner = 'minyue' %?> <?% config.freshness.reviewed =
+'2021-04-13' %?>
+
+# The WebRTC Audio Coding Module
+
+WebRTC audio coding module can handle both audio sending and receiving. Folder
+[`acm2`][acm2] contains implementations of the APIs.
+
+* Audio Sending Audio frames, each of which should always contain 10 ms worth
+ of data, are provided to the audio coding module through
+ [`Add10MsData()`][Add10MsData]. The audio coding module uses a provided
+ audio encoder to encoded audio frames and deliver the data to a
+ pre-registered audio packetization callback, which is supposed to wrap the
+ encoded audio into RTP packets and send them over a transport. Built-in
+ audio codecs are included the [`codecs`][codecs] folder. The
+ [audio network adaptor][ANA] provides an add-on functionality to an audio
+ encoder (currently limited to Opus) to make the audio encoder adaptive to
+ network conditions (bandwidth, packet loss rate, etc).
+
+* Audio Receiving Audio packets are provided to the audio coding module
+ through [`IncomingPacket()`][IncomingPacket], and are processed by an audio
+ jitter buffer ([NetEq][NetEq]), which includes decoding of the packets.
+ Audio decoders are provided by an audio decoder factory. Decoded audio
+ samples should be queried by calling [`PlayoutData10Ms()`][PlayoutData10Ms].
+
+[acm2]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/acm2/;drc=854d59f7501aac9e9bccfa7b4d1f7f4db7842719
+[Add10MsData]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=136;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
+[codecs]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/codecs/;drc=883fea1548d58e0080f98d66fab2e0c744dfb556
+[ANA]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/audio_network_adaptor/;drc=1f99551775cd876c116d1d90cba94c8a4670d184
+[IncomingPacket]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=192;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
+[NetEq]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/neteq/;drc=213dc2cfc5f1b360b1c6fc51d393491f5de49d3d
+[PlayoutData10Ms]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=216;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
