This is a reland of 444e04be6988fbdcc039d775481ac22481ff9ff4 Reason for reland: resolved the breaks from downstream project Original change's description: > ChannelStatistics used for RTP stats in VoipStatistics. > > - Added local and remote RTP statistics query API. > - Change includes simplifying remote SSRC change handling > via received RTP and RTCP packets. > > Bug: webrtc:11989 > Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Commit-Queue: Tim Na <natim@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32954} Bug: webrtc:11989 Change-Id: I88620a9f1c037b512821cac9d556905149666870 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201481 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32966}
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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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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