| commit | 84abeb1d37ee27147674cd94a5606bd4ac725d9a | [log] [tgz] |
|---|---|---|
| author | hbos <hbos@webrtc.org> | Mon Jan 16 06:16:44 2017 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Jan 16 14:16:44 2017 +0000 |
| tree | 876983f73bb253887e373ec68106243528569eea | |
| parent | 93f16d74fc1249d574ae3504945985970a10975c [diff] |
RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. Based on the mapping between [Audio/Video]TrackInterface and [Voice/Video][Sender/Receiver]Info. The IDs of RTCMediaStreamTrackStats are updated to distinguish between local and remote cases. Previously, if local and remote cases had the same label only one of them would be included in the report (bug). BUG=webrtc:6758, chromium:657854, chromium:657855, chromium:657856, chromium:627816 Review-Url: https://codereview.webrtc.org/2610843003 Cr-Commit-Position: refs/heads/master@{#16095}
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.