Compromise solution where WebRtcAudioUtils.setWebRtcBasedAutomaticGainControl() is marked as deprecated and where as many APIs as possible that touches the HW AGC are removed. Some basic architecture is saved to ensure that we can restore usage of the HW AGC if ever required for future devices. The AppRTCMobile demo does still contain an AGC check box but it is now grayed out. BUG=b/30387905 Review-Url: https://codereview.webrtc.org/2402883003 Cr-Commit-Position: refs/heads/master@{#14596}
Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ )
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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. This page is maintained by the Google Chrome team.
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://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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