435f279433
The AGC2 params must be exposed via `AudioProcessing::Config::GainController2` and the Finch params must be parsed in blink (see [1]). Note: this CL breaks the chain of 3 CLs titled "AGC2 AdaptiveModeLevelEstimator min consecutive speech frames". [1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/blink/renderer/modules/mediastream/media_stream_audio_processor.cc;l=593-596?q=HybridAgc&start=11 Bug: webrtc:7494 Change-Id: Ie7bd1bef1d6caf7d2b20600a1626c12171b67c82 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185044 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32230}
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 here 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