This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.
Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
1. Level Estimator - it gets the energy and a speech probability
and updates a speech level estimate.
2. Noise Estimator - it gets an immutable view of the speech frame
and updates the noise level estimate
3. Gain applier - it gets the speech frame, the current speech and
noise estimates, and the speech probability. It finds a gain to
apply and applies it.
4. AdaptiveAgc - sets up and controls the sub-modules described
above.
Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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