0f2ce5cc1c
This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. Reason for revert: Original change's description: > Migrate WebRTC documentation to new renderer > > Bug: b/258408932 > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39205} Bug: b/258408932 Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Artem Titov <titovartem@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39209}
33 lines
2.5 KiB
Markdown
33 lines
2.5 KiB
Markdown
<?% config.freshness.owner = 'minyue' %?>
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<?% config.freshness.reviewed = '2021-04-13' %?>
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# The WebRTC Audio Coding Module
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WebRTC audio coding module can handle both audio sending and receiving. Folder
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[`acm2`][acm2] contains implementations of the APIs.
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* Audio Sending Audio frames, each of which should always contain 10 ms worth
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of data, are provided to the audio coding module through
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[`Add10MsData()`][Add10MsData]. The audio coding module uses a provided
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audio encoder to encoded audio frames and deliver the data to a
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pre-registered audio packetization callback, which is supposed to wrap the
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encoded audio into RTP packets and send them over a transport. Built-in
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audio codecs are included the [`codecs`][codecs] folder. The
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[audio network adaptor][ANA] provides an add-on functionality to an audio
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encoder (currently limited to Opus) to make the audio encoder adaptive to
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network conditions (bandwidth, packet loss rate, etc).
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* Audio Receiving Audio packets are provided to the audio coding module
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through [`IncomingPacket()`][IncomingPacket], and are processed by an audio
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jitter buffer ([NetEq][NetEq]), which includes decoding of the packets.
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Audio decoders are provided by an audio decoder factory. Decoded audio
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samples should be queried by calling [`PlayoutData10Ms()`][PlayoutData10Ms].
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[acm2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/acm2/;drc=854d59f7501aac9e9bccfa7b4d1f7f4db7842719
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[Add10MsData]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=136;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
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[codecs]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/codecs/;drc=883fea1548d58e0080f98d66fab2e0c744dfb556
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[ANA]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/audio_network_adaptor/;drc=1f99551775cd876c116d1d90cba94c8a4670d184
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[IncomingPacket]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=192;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
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[NetEq]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/;drc=213dc2cfc5f1b360b1c6fc51d393491f5de49d3d
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[PlayoutData10Ms]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=216;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
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