623472219f
Also read and apply settings when parsing and replaying dumps. The implementation contains * an extra field in debug.proto for the runtime settings * code in AudioProcessingImpl to initiate the logging of the RS to the AecDump * code in aec_dump/ to log the RS in the AecDump * code in test/ for re-playing the RS. E.g. for APM simulation with audioproc_f. Bug: webrtc:9138 Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f Reviewed-on: https://webrtc-review.googlesource.com/70502 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24647}
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 http://www.webrtc.org/native-code/development 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