| commit | 40cbec54154da73b9d10195799894b861796f965 | [log] [tgz] |
|---|---|---|
| author | Alejandro Luebs <aluebs@webrtc.org> | Wed Apr 06 00:29:19 2016 |
| committer | Alejandro Luebs <aluebs@webrtc.org> | Wed Apr 06 00:29:29 2016 |
| tree | c45e58d127b8b8a9e47a2e3ac293256d0e70208d | |
| parent | faed4ab24bf76fbcaa84f231b9cf0cc019e3df8a [diff] |
Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo() R=henrik.lundin@webrtc.org, peah@webrtc.org TBR=tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1862553002 . Cr-Commit-Position: refs/heads/master@{#12249}
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.