338f78ac95
Collected for current pairs, undefined for other pairs. This is the same as the old stats' VideoBwe.googAvailable[Send/Receive]Bandwidth. NOTE: The value this is based on for incoming bitrate is not set. This CL wires it up but has a TODO that the incoming bitrate needs to be collected properly. (Same problem for both old and new stats.) Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-availableoutgoingbitrate Discussion: https://github.com/w3c/webrtc-stats/issues/112#issuecomment-277167781 BUG=webrtc:7062 Review-Url: https://codereview.webrtc.org/2675923002 Cr-Commit-Position: refs/heads/master@{#16472}
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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