| commit | 5c71e74331bd57dd94be6d66d7490c1a95154e48 | [log] [tgz] |
|---|---|---|
| author | Alex Loiko <aleloi@webrtc.org> | Mon Jul 02 11:37:47 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Jul 02 12:29:36 2018 +0000 |
| tree | 1c193132465ae003d7f65b83d4d885dfe1ba9ce9 | |
| parent | c167673c4df8b5c20adfe33061f25db4c841949b [diff] |
Add AGC1-compliant fake recording device. 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.
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.