1880c7162b
- Run analysis after all frames are processed. Before part of it was done at bitrate change points; - Analysis is done for whole stream as well as for each rate update interval; - Changed units from number of frames to time units for some metrics and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to 'time to reach target bitrate, sec'; - Changed data type of FrameStatistic::max_nalu_length (renamed to max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to use such advanced data type in such low level data structure. Bug: webrtc:8524 Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f Reviewed-on: https://webrtc-review.googlesource.com/31901 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21653}
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