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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/audio_coding/g3doc | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/g3doc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/g3doc/index.md | 32 |
1 files changed, 32 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/g3doc/index.md b/third_party/libwebrtc/modules/audio_coding/g3doc/index.md new file mode 100644 index 0000000000..d0f6b9f81b --- /dev/null +++ b/third_party/libwebrtc/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/+/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 |