commit | 189849fa0feca530781aa20f95de93539bbe12f0 | [log] [tgz] |
---|---|---|
author | Henrik Boström <hbos@webrtc.org> | Fri Feb 07 15:12:16 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Feb 07 15:14:38 2020 |
tree | 9be81b0905b139dfd46d7a815ff24025eb373b8d | |
parent | 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1 [diff] |
[Stats] Remove jitterBufferDelay TODO; it's already implemented. This TODO says this metric is only available for audio and should also be implemented for video, but ever since M76 this has been implemented for both audio and video (https://crbug.com/webrtc/10450). TBR=guido@webrtc.org, hta@webrtc.org NOTRY=True Bug: webrtc:10450 Change-Id: Icf2b60fdacae606c66f9d03492f107df9e32ba33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168343 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30485}
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.