64e5830969
Not passing the sample rate to the `VoiceActivityDetectorWrapper` ctor yet since that would require an unnecessary refactoring of `AdaptiveAgc` which will soon be removed. Instead, to ensure correct initialization until the child CL [1] lands, `VoiceActivityDetectorWrapper::initialized_` is temporarily added. Bit exactness verified with audioproc_f on a collection of AEC dumps and Wav files (42 recordings in total). [1] https://webrtc-review.googlesource.com/c/src/+/234583 Bug: webrtc:7494 Change-Id: I4b4be7b8106ba36c958d91bf263a7b30271a1ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234587 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35213}
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