f9f448b32d
RTCAudioSession and RTCAudioSessionConfiguration allow users to handle audio manually and is used by the AppRTCMobile example. RTCVideoFrameBuffer exposes a protocol that users can implement to create their own frame buffer formats, as long as they can be converted into i420. RTCVideoCapturer and RTCVideoViewShading are imported by other headers already included by the umbrella header, so they were always accessible to users. Added them to the umbrella header to make it explicit. BUG=webrtc:7351, webrtc:8027 Review-Url: https://codereview.webrtc.org/2994253002 Cr-Commit-Position: refs/heads/master@{#19379}
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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