This CL significantly improves the response time of the AEC3 delay estimator to audio buffer issues. The CL adds ensures that the delay estimator correlators reacts to buffer issues from the zero state which is much faster than if it has already achieved a state matching a previous alignment. The CL has been extensively tested on offline recordings. Bug: webrtc:9023, chromium:822245 Change-Id: Ic149b9429e592d4c3535eb8432582f435a1b4745 Reviewed-on: https://webrtc-review.googlesource.com/62081 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22461}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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