b20b93796f
This CL refactors the code in AEC3 that analyzes how well the adaptive filter performs. The purpose of this is both to simplify code that is more complex than needed and also to pave the wave for the upcoming CLs that softens the echo suppression during doubletalk. The main changes are that: -The shadow adaptive filter is now never analyzed. This turned out to never affect the output in the recordings it was tested on. -The convergence analysis was moved to the aec state code. The changes are bitexact on all testcases where they have been tested on. Bug: webrtc:8671 Change-Id: If76b669565325c8eb4d11d1178a7e20306da9a26 Reviewed-on: https://webrtc-review.googlesource.com/87430 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23958}
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