| commit | 5588a13fe7c2768bf6a6b9dc6492e8076db02369 | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@webrtc.org> | Tue Oct 18 12:14:30 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Oct 18 12:14:35 2016 |
| tree | 6cdb82d667794420491f58ae1c44f333dbb22648 | |
| parent | 44666997ca912705f8f96c9bd211e719525a3ccc [diff] |
Now uses rtc::Buffer in AudioDeviceBuffer. The main goal of this CL is to remove old buffer handling using static arrays and switch to the improved rtc::Buffer class instead. By doing so, we can remove some members (since Buffer maintains them instead) and do some additional cleanup. This CL also fixes some minor style issues and improves the locking mechanism. Finally, AudioDeviceBuffer::SetRecordingChannel() is deprecated since it has never been used and is not included in any test. BUG=NONE Review-Url: https://codereview.webrtc.org/2333273002 Cr-Commit-Position: refs/heads/master@{#14661}
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.