commit | 3d976f60666c0d800f9112edbae7c93bee99acd7 | [log] [tgz] |
---|---|---|
author | Harald Alvestrand <hta@webrtc.org> | Mon Mar 19 18:05:06 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Mar 19 18:39:01 2018 |
tree | d7335fdd1f3077b3995aa404a8b64bcb754b19a9 | |
parent | bb60a3a5fa999d0dcf12319068bb6ea01f4a64c3 [diff] |
Discard link to media channel when audio sender stopped. Bug: chromium:822799 Change-Id: Ib863cf048318b04369cc51ed1b1c8b03010a2fd2 Reviewed-on: https://webrtc-review.googlesource.com/62941 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22503}
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.