| commit | ee84d39fce1ec4f000b86138a84e56293a9c126a | [log] [tgz] |
|---|---|---|
| author | Gustaf Ullberg <gustaf@webrtc.org> | Tue Sep 10 09:36:43 2019 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Sep 10 08:16:07 2019 +0000 |
| tree | cf4fd9577bb78e7f722f8515e5f090dc1e790253 | |
| parent | d181ee798da57ce5b955f09e8dcb755fba70b51b [diff] |
AEC3: Downmix multichannel signals before delay estimation Multichannel signals are downmixed to mono before decimation and delay estimation. This is useful when not all channels play audio content. The feature can be toggled in the AEC3 configuration. Bug: webrtc:10913 Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29126}
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.