| commit | 3355f6d6f5bdcd529705228c6f48612ecc0f0a62 | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@chromium.org> | Fri Oct 21 10:45:25 2016 |
| committer | henrika <henrika@chromium.org> | Fri Oct 21 10:45:31 2016 |
| tree | 29e200d4f7831bcc9c5c249b2ad718860887d38e | |
| parent | c4d2dc4e021cc676cc93be2c3278da115197e8c2 [diff] |
Avoids invalid copy of audio buffer to task queue. Now does level estimate on the audio threads to avoid complex copying of audio data to task queue. The old implementation could also crash due to unclear ownership of the audio buffer. BUG=webrtc:6569 R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/2433393002 . Cr-Commit-Position: refs/heads/master@{#14720}
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.