| commit | f50222244371fdcb00190017a40afa2c0796fed1 | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@chromium.org> | Mon Nov 07 14:56:59 2016 |
| committer | henrika <henrika@chromium.org> | Mon Nov 07 14:57:11 2016 |
| tree | 9a2a8e44cbfe4c653316c5e8842c4ab4b2424f93 | |
| parent | 41b8ca042065d2f1bec632b80d0e1c1ef8f452fc [diff] |
- Adds thread safety annotations to the AudioDeviceBuffer class. - Removes the lock that was used to protect the audio transport object. It is now protected "by design" instead. - Removes rec/play_bytes_per_sample_ since we only support 16-bit samples. BUG=webrtc:6560 R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/2466613002 . Cr-Commit-Position: refs/heads/master@{#14950}
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.