59f1d1e36d262cc3dfbadcd8d0214fffb4664676

This CL adds and wires up the following parameters: - VAD probability attack used in `VadLevelAnalyzer` - Adjacent spech frames threshold used in `AdaptiveModeLevelEstimator` - Initial saturation margin used in `AdaptiveModeLevelEstimator` The deprecated ctor in `AdaptiveModeLevelEstimator` is removed. Tested: bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: Idf94aaadba1476757f845e696bfb47ff6252d5f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186048 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32265}
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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 here 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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