commit | d0fa82055947b76554e9b3dd11fea7c1c61650a8 | [log] [tgz] |
---|---|---|
author | Per Åhgren <peah@webrtc.org> | Wed Apr 18 09:35:13 2018 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Apr 18 09:05:54 2018 +0000 |
tree | c9ab45321b2c53de6e638a461fa7c0adaa9c1402 | |
parent | c841d18d257ba8e4ed7d77d105e3c46006bb1e7e [diff] |
Allow AEC3 to use any externally reported audio buffer delay in AEC3 This CL adds support for using any externally reported audio buffer delay to set the initial alignment in AEC3 which is used before the AEC has been able to detect the delay. Bug: chromium:834182,webrtc:9163 Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb Reviewed-on: https://webrtc-review.googlesource.com/70580 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22917}
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.
Authoritative list of directories that contain the native API header files.