d5e6f413ab
This CL was written in preparation for the next CL in the chain and it contains the following changes: - SignalWithLevels -> AdaptiveDigitalGainApplier::FrameInfo - Frame view removed from AdaptiveDigitalGainApplier::FrameInfo - AdaptiveDigitalGainApplier::Process now gets side info as const& to avoid unnecessary copies - AdaptiveAgc::Process: `last_audio_level` renamed to `limiter_envelope` to better reflect what that actually is - Missing class/method docstrings added Tested: bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: Ie25dcd389d6eed74ea9a65f0720eeb8f20f0096b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186040 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32251}
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