| commit | 0d7f04daa0ddc621a7e3d6a0373b3ae458511030 | [log] [tgz] |
|---|---|---|
| author | saza <saza@webrtc.org> | Tue Jul 04 11:05:06 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Jul 04 11:05:06 2017 |
| tree | 02e07bb0994d16f17346188a35bf5ad2e0546213 | |
| parent | 38fecafa48ff6abfdf0279ce3c04df7e26401e08 [diff] |
Reland of Add received audio/video call duration metrics based on packets. Original issue: https://codereview.webrtc.org/2957073002/ Reason for reland: Failed Android unit tests and failed Windows compile. The tests seemed related at the time, but not after more consideration. Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps. BUG=webrtc:7882 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2970793003 Cr-Commit-Position: refs/heads/master@{#18886}
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.