529131d3e4
Implement AnalogGainStatsReporter and add it in AudioProcessingImpl. This class computes statistics for analog gain updates and periodically reports them into a histogram. The added histograms for analog gain update statistics: - WebRTC.Audio.ApmAnalogGainDecreaseRate - WebRTC.Audio.ApmAnalogGainIncreaseRate - WebRTC.Audio.ApmAnalogGainUpdateRate - WebRTC.Audio.ApmAnalogGainDecreaseAverage - WebRTC.Audio.ApmAnalogGainIncreaseAverage - WebRTC.Audio.ApmAnalogGainUpdateAverage The histograms are defined in https://chromium-review.googlesource.com/c/chromium/src/+/3207987 Bug: webrtc:12774 Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35301}
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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation