commit | 4b6c8b7bf7d2489c3cda13a5b93f03043fd76584 | [log] [tgz] |
---|---|---|
author | aluebs <aluebs@webrtc.org> | Mon Jun 20 11:02:30 2016 -0700 |
committer | Commit bot <commit-bot@chromium.org> | Mon Jun 20 18:02:38 2016 +0000 |
tree | 236f238133d0da7df03f2995e62ec355f6880ced | |
parent | 884c336c345d988974c2a69cea402b0fb8b07a63 [diff] |
Fix ProcessReverseStream usage in audioproc_f Also added IntelligibilityEnhancer setting to aecdump simulator in audioproc_f Review-Url: https://codereview.webrtc.org/2075093003 Cr-Commit-Position: refs/heads/master@{#13220}
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.