ce2e13602e
Integrate AvgCounter to be used for BWE stats in call. Fixes for stats regression in: WebRTC.Call.EstimatedSendBitrateInKbps WebRTC.Call.PacerBitrateInKbps Example: BWE for a 15 seconds long call (with intervals of 1 sec): |300|400|500|600|600|600|600| 0 | 0 | 0 | 0 | 0 |800|800|800| // 0 - network state down Reported via OnNetworkChanged: |300|400|500|600| x | x | x | 0 | x | x | x | x |800| x | x | // x - empty interval, 0 -> pauses stats Stats: |300|400|500|600|600|600|600| - | - | - | - | - |800|800|800| // x -> last value used (intervals during pause ignored) AvgCounter uses the average of samples within an interval (interval length is 2 sec). BUG=webrtc:6244 Review-Url: https://codereview.webrtc.org/2307913002 Cr-Commit-Position: refs/heads/master@{#14147}
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