mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00

This reverts commit0f2ce5cc1c
. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit3eceaf4669
. > > 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} Bug: b/258408932 Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660 Owners-Override: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39231}
32 lines
2.5 KiB
Markdown
32 lines
2.5 KiB
Markdown
<!-- go/cmark -->
|
|
<!--* freshness: {owner: 'hlundin' 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/+/main:third_party/webrtc/modules/audio_coding/acm2/;drc=854d59f7501aac9e9bccfa7b4d1f7f4db7842719
|
|
[Add10MsData]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=136;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
|
|
[codecs]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/codecs/;drc=883fea1548d58e0080f98d66fab2e0c744dfb556
|
|
[ANA]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/audio_network_adaptor/;drc=1f99551775cd876c116d1d90cba94c8a4670d184
|
|
[IncomingPacket]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=192;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
|
|
[NetEq]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/;drc=213dc2cfc5f1b360b1c6fc51d393491f5de49d3d
|
|
[PlayoutData10Ms]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=216;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
|