| commit | 701d628f5f392a80436bd53cb118800b7845cb9d | [log] [tgz] |
|---|---|---|
| author | peah <peah@webrtc.org> | Tue Oct 25 05:42:20 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Oct 25 12:42:25 2016 +0000 |
| tree | 9d87f199ac8248d3c566f3ddbfa04817decabcd2 | |
| parent | d8872c590709501bc327546284204e6b85692166 [diff] |
Moved the AGC render sample queue into the audio processing module Several subcomponents inside APM copy render audio from the render side to the capture side using the same pattern. Currently this is done independently for the submodules. This CL moves the the AGC functionality for this into APM. BUG=webrtc:5298, webrtc:6540 Review-Url: https://codereview.webrtc.org/2444283002 Cr-Commit-Position: refs/heads/master@{#14770}
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.