| commit | bec70ab0fd3613c532359a849bfbc67749da6d24 | [log] [tgz] |
|---|---|---|
| author | fippo <fippo@andyet.net> | Thu Jan 28 09:27:15 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Jan 28 09:27:20 2016 |
| tree | 563500c882b9da99819cdc05ea955dfcf81aa6aa | |
| parent | 6a062bd7afe0ab088a9ac37eafb8fc70b58fe855 [diff] |
https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type. This is somewhat easier than looking up the media type by iterating pc.getLocalStreams / pc.getRemoteStreams and all tracks. Temporary until stats get revamped to conform to the spec obviously. BUG=webrtc:4117 Review URL: https://codereview.webrtc.org/1307633007 Cr-Commit-Position: refs/heads/master@{#11412}
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.