d978cb43c2
For each SSRC report, record the number of bytes sent for that stream and expose them in analyzer stats. These numbers can be used to determine useful metrics such as total media throughput (by adding the bytes sent for all streams) and overhead (by subtracting that amount from the total bytes sent to the network). Bug: webrtc:9719 Change-Id: I977bbd40acdd0a1ec64763ddd55a642b9a50f309 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28637}
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