commit | 5dfde18c77d96918ec0c18c9496a2253a049a253 | [log] [tgz] |
---|---|---|
author | Steve Anton <steveanton@webrtc.org> | Tue Feb 06 18:34:40 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Feb 06 19:02:44 2018 |
tree | e937a5e96cbe946e8b01018b56ddd76f54181f5c | |
parent | 8234ead6d9671ba01861c0e11042a02c7335c8dd [diff] |
Change PeerConnection stats interface to be more flexible This removes the SessionStats object and replaces it with two methods on PeerConnection: GetTransportNamesByMid and GetTransportStatsByNames for use by the stats collectors. These methods are more flexible and can cover cases where there are more than one video/audio channel. Bug: webrtc:8764 Change-Id: Id400cc548fc43675462ff6175a7fa9c9f4fd5948 Reviewed-on: https://webrtc-review.googlesource.com/47244 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21921}
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.
Authoritative list of directories that contain the native API header files.