5c71e74331
The AGC submodule of APM changes analog gain. These gain changes are typically ignored by the test tool audioproc_f. There is an option of the test tool to take action on the gain changes. It's the '--simulate_mic_gain' option. The option converts the analog gain to a digital gain. The digital gain is applied to the capture stream. This change adds a new simulated microphone kind. The new microphone has a gain curve defined by modules/audio_processing/agc/gain_map_internal.h. That gain curve defines how AGC1 expects a microphone to behave. Bug: webrtc:7494 Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780 Reviewed-on: https://webrtc-review.googlesource.com/86128 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23801}
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