8efafdf84b
This change moves the configuration parser that converts a JSON representation of the VideoStreamReceiver::Config structure into a native object into the test directory so that it can be shared with the new corpus_generator utility that is being built. This rtc_source_set will have an additional utility function added in a subsequent CL that will allow the generation of a VideoStreamSender::Config from a given VideoStreamReceiver::Config and visa versa. Bug: webrtc:10117 Change-Id: I3035826f799f8d1fcdeaa76997391f030c855a5c Reviewed-on: https://webrtc-review.googlesource.com/c/116880 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26252}
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