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4 # The WebRTC Audio Coding Module
6 WebRTC audio coding module can handle both audio sending and receiving. Folder
9 * Audio Sending Audio frames, each of which should always contain 10 ms worth
10 of data, are provided to the audio coding module through
11 [`Add10MsData()`][Add10MsData]. The audio coding module uses a provided
12 audio encoder to encoded audio frames and deliver the data to a
13 pre-registered audio packetization callback, which is supposed to wrap the
14 encoded audio into RTP packets and send them over a transport. Built-in
15 audio codecs are included the [`codecs`][codecs] folder. The
16 [audio network adaptor][ANA] provides an add-on functionality to an audio
17 encoder (currently limited to Opus) to make the audio encoder adaptive to
20 * Audio Receiving Audio packets are provided to the audio coding module
21 through [`IncomingPacket()`][IncomingPacket], and are processed by an audio
23 Audio decoders are provided by an audio decoder factory. Decoded audio