9a9f18a736
Old way to produce this histogram was based on RtcpStatisticsCallback reporting sent RTCP messages, with some additional processing by the ReportBlockStats class. After this cl, to grand average fraction loss is computed by StreamStatistician, queried by VideoReceiveStream when the stream is closed down, and passed to ReceiveStatisticsProxy which produces histograms. This is a preparation for deleting the RtcpStatisticsCallback from ReceiveStatistics. Bug: webrtc:10679 Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28747}
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