736ff83e69
- Passing the speech peak power instead of VAD data - The private class SaturationProtector::PeakEnveloper has been removed - Added `initial_saturation_margin_db_` parameter to correctly initialize `last_margin_` (renamed to `margin_db_`) - Member names have been fixed and/or shortened for better readability Tested: Bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: I6cad2974397319737c8ac201d44311bf16275f28 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184925 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32168}
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