This cl/ changes so that the RTCTransportStats bytes/packets sent/recevied is computed in P2PTransportChannel. Previously they were computed by aggregating over the Connections, but that does not work when Connections are created and destroyed. Bug: webrtc:13769 Change-Id: Ia97dfae70b5aced897d4813ec007ba61bc032f87 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253100 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36103}
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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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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