790da37b72f96feb8d5309b4485ef3df54ffa2d1

The fuzzer data is used to configure the field trials of the AEC. This increases fuzzer coverage of modules/audio_processing/aec3/ by roughly 500 lines of code, ~ 3 % points increase in APM coverage for desktop Chrome. Bug: webrtc:9413 Change-Id: Iea9059747a8492a7ca2091a359e7883750c45b27 Reviewed-on: https://webrtc-review.googlesource.com/83732 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23650}
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
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