| commit | 1c378ed83b5b32a00368bbfc3ce2ee7687691abe | [log] [tgz] |
|---|---|---|
| author | zhihuang <zhihuang@webrtc.org> | Thu Aug 17 21:10:50 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Aug 17 21:10:50 2017 |
| tree | a5e8af9afc219a0c5e3f8adb7a74af974801f274 | |
| parent | 825f65e9d2285e69467acd6449549f10f9e1df6b [diff] |
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. 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-Original-Commit-Position: refs/heads/master@{#19343} Committed: https://chromium.googlesource.com/external/webrtc/+/a77e6bbd30276bdc5b30f2cbc1e92ca181ae76f0 Review-Url: https://codereview.webrtc.org/2991693002 Cr-Commit-Position: refs/heads/master@{#19394}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.