8845f7e32b
This is the last CL needed to add a new `AdaptiveModeLevelEstimator` feature that makes AGC2 more robus to VAD mistakes: the level estimator discards estimation updates when too few consecutive speech frames are observed. This CL adds a second state property to hold temporary updates and a counter for consecutive speech frames. When enough speech frames are observed, the reliable state is updated; otherwise, the temporary state is discarded. The default for `AdaptiveModeLevelEstimator::min_consecutive_speech_frames_` is 1, which means that the new feature is disabled. Tested: - Bit-exactness verified with audioproc_f - Not bit-exact if `min_consecutive_speech_frames_` set to 10 Bug: webrtc:7494 No-Try: True Change-Id: I0daa00e90c27c418c00baec39fb8eacd26eed858 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185125 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32250}
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