commit | 44125faba5bf646116c5a6335ac46c3b544808e8 | [log] [tgz] |
---|---|---|
author | Henrik Lundin <henrik.lundin@webrtc.org> | Mon Apr 29 15:00:46 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Apr 29 15:39:50 2019 |
tree | 2d0bb66823440034b0101f5606c07e7e0f70f3d1 | |
parent | 7cca042dd409c08a2a6e2383c39c692626f8a2d9 [diff] |
Reland "Piping audio interruption metrics to API layer" The metrics are now added as RTCNonStandardStatsMember objects in RTCMediaStreamTrackStats. Unit tests are updated. This is a reland of https://webrtc-review.googlesource.com/c/src/+/134303, with fixes. TBR=kwiberg@webrtc.org Bug: webrtc:10549 Change-Id: I29dcc6fbfc69156715664e71acfa054c1b2d9038 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134500 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27806}
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.