389010438d
When `AudioProcessingImpl::ApplyConfig()` is called, AGC2 is initialized and then the new config is applied. That is error prone and for example breaks bit exactness in [1]. Changes: - `GainController2` must be created by passing configuration, sample rate and number of channels - `GainController2::ApplyConfig()` removed 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/+/234587. Bug: webrtc:7494 Change-Id: I251e03603394a4fc8769b9b5c197a157893676a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235060 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35206}
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