It does: -Handle saturations in a better manner by adding different gain change step sizes for upwards and downwards changes, as well as when there is saturation. -Handle conditions with initial noise-only regions in a better way by setting a high initial peak level estimate which is gradually reduced until certainty about the peak level is achieved. -Limit the maximum gain to limit noise amplification, and to reflect that it initially is intended to be used in cascade with the fixed digital AGC mode. -Lower the maximum allowed stationary noise floor to reduce the risk of excessive noise amplification. -Lower the target gain to reduce the risk of causing the AEC on the other end to fail due to high playout levels triggering nonlinearities. This also reduces the risk for saturation. -Handle the noise-only regions in a better manner. NOTRY=true TBR=aleloi BUG=webrtc:5920 Review-Url: https://codereview.webrtc.org/2111553002 Cr-Commit-Position: refs/heads/master@{#13350}
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
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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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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