| commit | bb6f7524bab7c839f8281e1db1d0e7f6cfaeeba6 | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@webrtc.org> | Tue May 30 02:01:30 2017 -0700 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue May 30 09:01:30 2017 +0000 |
| tree | 5c6c60f08c0252e270c973293810b532e1775a63 | |
| parent | d7620582342bb896b766bb306135b12aefbeb12c [diff] |
Improved audio buffer handling for iOS. This change: Reduces complexity for audio playout by removing a redundant memcopy in the output audio path. Adds support for iOS simulator for playout since we now allow the audio layer to ask for different sizes of audio buffers at each callback. Real iOS devices always asks for the same size, simulators does not. This change comes without any new cost for real devices. BUG=b/37580746 Review-Url: https://codereview.webrtc.org/2894873002 Cr-Commit-Position: refs/heads/master@{#18321}
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.