| commit | 64e739aeae5629cbbebf2a19e1d3e6b452bb6d0b | [log] [tgz] |
|---|---|---|
| author | ilnik <ilnik@webrtc.org> | Tue Apr 11 08:46:04 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Apr 11 08:46:04 2017 |
| tree | 391aea9ba95bc313f9a97df879bfc94c524fb8ec | |
| parent | 93cda2ebde909ad9cc424690f44f59a6fb84b149 [diff] |
Add content type information to Encoded Images and add corresponding RTP extension header. Use it to separate UMA e2e delay metric between screenshare from video. Content type extension is set based on encoder settings and processed and decoders. Also, Fix full-stack-tests to calculate RTT correctly, so new metric could be tested. BUG=webrtc:7420 Review-Url: https://codereview.webrtc.org/2772033002 Cr-Commit-Position: refs/heads/master@{#17640}
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.