The old value was 170, but experiments have shown that 70 is better. This will let the AGC reduce the gain further when input clipping is detected. The effect should be less clipping, but sometimes slightly lower signals. In Chrome, the value 70 has already been used since June (see https://codereview.chromium.org/2928133002). Bug: webrtc:6622, chromium:672476 Change-Id: Ie5a60bb875eef71f303b28e096b22a8cd4b449d4 Reviewed-on: https://webrtc-review.googlesource.com/20222 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20563}
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.
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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