| commit | 0ce3bf9cc4d7b8acb4b61556840756504e11f8a4 | [log] [tgz] |
|---|---|---|
| author | tkchin <tkchin@webrtc.org> | Sun Mar 13 00:52:04 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Sun Mar 13 00:52:13 2016 |
| tree | 41c87a34464918b821a52bee08f5c135c1c6fa33 | |
| parent | b25345ee3f2f27a6814495f7f617f954fb4b192a [diff] |
Fix lock behavior on RTCAudioSession. In addition: - Introduces RTCAudioSessionTest - iOS/Mac gtests now have an autoreleasepool - Moves ScopedAutoreleasePool to rtc_base_approved - Introduces route change button in AppRTCDemo BUG=webrtc:5649 Review URL: https://codereview.webrtc.org/1782363002 Cr-Commit-Position: refs/heads/master@{#11971}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
