| commit | ce2e13602e06c5f7df7245f8d0e66e7070c0be34 | [log] [tgz] |
|---|---|---|
| author | asapersson <asapersson@webrtc.org> | Fri Sep 09 07:13:35 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Sep 09 07:13:39 2016 |
| tree | f54b555d3ace9b59a40cc3ee0f90e356ab518e88 | |
| parent | 2a5f371df3c5211ea4e1dd7f155e490b2e4bcc7b [diff] |
Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats). 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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.