In order to add `RTCRemoteOutboundRtpStreamStats` (see [1]), the following stats must be added: - sender's packet count (see [2]) - sender's octet count (see [2]) - total number of RTCP SR blocks sent (see [3]) [1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats [2] https://tools.ietf.org/html/rfc3550#section-6.4.1 [3] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent Bug: webrtc:12529 Change-Id: I47ac2f79ba53631965d1cd7c1062f3d0f158d66e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210963 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33423}
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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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