commit | 299c4e68461f1c4428b2a919913b161115025dff | [log] [tgz] |
---|---|---|
author | Henrik Lundin <henrik.lundin@webrtc.org> | Fri Apr 26 12:41:13 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Apr 26 13:32:34 2019 |
tree | 2a8b068905bdc07df43ee6ff9e8564a396b07c23 | |
parent | c35b6e675a878996c61e89f2b472f6f71bd74d67 [diff] |
Piping audio interruption metrics to API layer Bug: webrtc:10549 Change-Id: Ie6abe5819c5b73dc5f5f89bdc375bad77f44ce97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134303 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27788}
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.