This replaces the old AlrDetector used by the pacer with the one in GoogCC. This reduces the risk of accidentally changing only one version. Note that the pacer instance will be removed when moving over to the task queue based send side congestion controller. Bug: webrtc:8415 Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5 Reviewed-on: https://webrtc-review.googlesource.com/85363 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23791}
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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