commit | 65881de6c81cf3f33b0f2f5f8f4a19716ee93fdc | [log] [tgz] |
---|---|---|
author | henrik.lundin <henrik.lundin@webrtc.org> | Wed Apr 26 15:23:35 2017 |
committer | Commit bot <commit-bot@chromium.org> | Wed Apr 26 15:23:35 2017 |
tree | 960e4a5c29f34efe28c1cd72acb63593d17fc5f9 | |
parent | afcf7f5591de47063f7da3507ea83b48970a6e1f [diff] |
NetEq: Limit payload size for replacement audio input With this fix, the size of the fake encoded payload is limited to 120 ms at 48000 samples/second. BUG=webrtc:7467 Review-Url: https://codereview.webrtc.org/2838353002 Cr-Commit-Position: refs/heads/master@{#17891}
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.