| commit | e80f4c91d0a2854a339e419162fdcd1d916f7de0 | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <holmer@chromium.org> | Thu Jun 01 14:29:28 2017 |
| committer | Stefan Holmer <holmer@chromium.org> | Thu Jun 01 14:29:30 2017 |
| tree | d5af07ed57900d547cfc4a744b0c31842d17e984 | |
| parent | 26f833fe33f4d635c189e6be6cc67fe443f66b07 [diff] |
Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have. BUG=webrtc:5079 R=deadbeef@webrtc.org, hbos@webrtc.org Review-Url: https://codereview.webrtc.org/2863123002 . Cr-Commit-Position: refs/heads/master@{#18384}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.