| commit | 06953bac6d61c4e24169c0ad4711648472bcab25 | [log] [tgz] |
|---|---|---|
| author | Sam Zackrisson <saza@webrtc.org> | Thu Feb 01 15:53:16 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 01 16:49:39 2018 |
| tree | 5d362801855b2d24813a062aea0f3cc7ead99aef | |
| parent | b90a64a449dcc911f66413cfe154762cb2ac344d [diff] |
Move AudioSendStream lifetime reporting into destructor This avoids a data race in which the lifetime TimeInterval is accessed by the owning Call objects concurrently with SendRtp calls on the underlying Channel object. Bug: webrtc:8794 Change-Id: If53d5680095c0177656b659162457287cb8e45dd Reviewed-on: https://webrtc-review.googlesource.com/46525 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21853}
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.