| commit | 66d4b3741487da22a7bd63d07160052883413245 | [log] [tgz] |
|---|---|---|
| author | asapersson <asapersson@webrtc.org> | Mon Dec 19 14:50:53 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Dec 19 14:50:53 2016 |
| tree | 7973b101a13c36bdbf8c3c5c100a96b200bfdd7d | |
| parent | 0cd27ba088766b7822df1bfe7c21c3dfd57d368c [diff] |
Move histogram for number of pause events to per stream: "WebRTC.Call.NumberOfPauseEvents" -> "WebRTC.Video.NumberOfPauseEvents" Recorded if a certain time has passed (10 sec) since the first media packet was sent. Moved to per stream to know when media has started and to prevent logging stats for calls that was never in use. Add histogram for percentage of paused video time for sent video streams: "WebRTC.Video.PausedTimeInPercent" BUG=b/32659204 Review-Url: https://codereview.webrtc.org/2530393003 Cr-Commit-Position: refs/heads/master@{#15681}
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.