a77e6bbd30
This layer takes in a simplified "options" struct and the current local description, and generates a new offer/answer. Previously the options struct assumed there would only be one media description per media type (audio/video), but it now supports N number of audio/video descriptions. The |add_legacy_stream| options is removed from the mediasession.cc/.h in this CL. The next step is to add the ability for PeerConnection/WebRtcSession to create "options" to represent multiple RtpTransceivers, and apply the Unified Plan descriptions correctly. Right now, only Plan B descriptions will be generated in unit tests. BUG=chromium:465349 Review-Url: https://codereview.webrtc.org/2991693002 Cr-Commit-Position: refs/heads/master@{#19343}
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