commit | 93e1e23537431893c34b5c4fd879532c7bba839a | [log] [tgz] |
---|---|---|
author | asapersson <asapersson@webrtc.org> | Mon Feb 06 13:18:35 2017 |
committer | Commit bot <commit-bot@chromium.org> | Mon Feb 06 13:18:35 2017 |
tree | 844b2153535b3cd46f88fc07e1b026cf4c067a37 | |
parent | 447dba958681dc3054c3fd92ef571176ab52c279 [diff] |
Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. Intervals when video is paused is no longer included in the stats: "WebRTC.Video.BitrateSentInKbps" "WebRTC.Video.MediaBitrateSentInKbps" "WebRTC.Video.PaddingBitrateSentInKbps" "WebRTC.Video.RetransmittedBitrateSentInKbps" "WebRTC.Video.RtxBitrateSentInKbps" "WebRTC.Video.FecBitrateSentInKbps" BUG=webrtc:5283 Review-Url: https://codereview.webrtc.org/2536613002 Cr-Commit-Position: refs/heads/master@{#16447}
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.