| commit | 17048012261406343c4e3f82418720b7cf0f4dd7 | [log] [tgz] |
|---|---|---|
| author | Yves Gerey <yvesg@webrtc.org> | Fri Jul 26 15:49:52 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Jul 26 17:03:29 2019 |
| tree | 5ef3f21b33b78427a57d09ac7bced31ea0fe1f13 | |
| parent | 378cae2543f5ea7d0a95a1c15516da0c5cd15f6c [diff] |
Prevent concurrent access to AudioSendStream's configuration. By design: * OnPacketAdded() is meant to be called on pacer thread. * Reconfigure() is meant to be called on worker thread. Thus we guard against race condition on config_ member. Possible downside: packet filtering based on ssrc might be slowed down. Bug: webrtc:9849 Change-Id: I734bb9b34b01db160705897adb1b58e866e12639 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146980 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28691}
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.