| commit | 3f6077d22f97a951424c915c0148904a1f7975cf | [log] [tgz] |
|---|---|---|
| author | Gustaf Ullberg <gustaf@webrtc.org> | Mon Sep 24 12:54:32 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Sep 24 13:44:21 2018 |
| tree | a09400515119cb2f5616aaa162114336dbc80889 | |
| parent | c5744b8b21b627213286f1b6f2c65da5df9ce8d0 [diff] |
AEC3: Delay estimator adapts even when estimated echo saturates Speeds up adaptation of the matched filter of the delay estimator by allowing the estimated echo and the error signal (microphone minus estimated echo) to be saturated. Only microphone saturation pauses the filter adaptation. Bug: webrtc:9773 Change-Id: I8b8400539fde3ee821f36a95818bece02ddd626b Reviewed-on: https://webrtc-review.googlesource.com/101341 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24802}
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.