| commit | 0c43f779f8ed38e02fd8bf7bef6f8bf15e8f1aad | [log] [tgz] |
|---|---|---|
| author | asapersson <asapersson@webrtc.org> | Wed Nov 30 01:42:26 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Nov 30 09:42:32 2016 +0000 |
| tree | 30c5c111fd6e54d1c69032681924895553d0976e | |
| parent | 759e0b7241edbaebf9a4506c32e82f25c8accf10 [diff] |
Update video histograms that do not have a minimum lifetime limit before being recorded. Updated histograms: "WebRTC.Video.ReceivedPacketsLostInPercent" (two RTCP RR previously needed) "WebRTC.Video.ReceivedFecPacketsInPercent" (one received packet previously needed) "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec" (one received FEC packet previously needed) Prevents logging stats if call was shortly in use. BUG=b/32659204 Review-Url: https://codereview.webrtc.org/2536653002 Cr-Commit-Position: refs/heads/master@{#15315}
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.