| commit | e2d83d6560272ee68cf99c4fd4f78a437adeb98c | [log] [tgz] |
|---|---|---|
| author | sprang <sprang@webrtc.org> | Fri Feb 19 17:03:26 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Feb 19 17:03:34 2016 |
| tree | d37ebd98f83a004a72f505e8042acba4aaccf7a2 | |
| parent | 45c44f0b94c3be9f35e351dda4d5b4ab17b44bd1 [diff] |
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() Also move some stats reporting from vie_channel to send stats proxy BUG= Review URL: https://codereview.webrtc.org/1669623004 Cr-Commit-Position: refs/heads/master@{#11688}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.