879d33b9f8
This CL adds the ability in audioproc_f and unpack_aecdump to: -Clearly identify the Init events and when those occur. -Optionally only process a specific Init section of an aecdump. -Optionally selectively turn on dumping of internal data for a specific init section, and a specific time interval. -Optionally let unpack_aecdump produce file names based on inits. Bug: webrtc:5298 Change-Id: Id654b7175407a23ef634fca832994d87d1073239 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33181}
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
- Reporting bugs