4bb1e4a1d5
The AdaptiveAgc often boosts the signal outside of Float S16 range. It is expected, which is why we have a limiter after it in the process chain. But it turns out that this happens regularly even for simple input examples. The output signal peaks can be as high as +4 dBFs for a single speaker example (which should be easy). It leads to excessive gain modulation by the limiter. This CL is a new tuning designed to produce a safer gain. After this, we shouldn't hit the saturation region of the limiter as often. But we will still maintain a high gain. We have a 'configurable kill-switch': the settings can be changed via field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin. Bug: webrtc:7494, chromium:892043 Change-Id: I5014377050c74c32ae8998282991141eae31cf58 Reviewed-on: https://webrtc-review.googlesource.com/c/102922 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25006}
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