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(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) Relanding after fixing CallAndModifyStream to account for new procedures for adding/removing a track from a stream. Original issue's description: > Adding the ability to create an RtpSender without a track. > > This CL also changes AddStream to immediately create a sender, rather > than waiting until the track is seen in SDP. And the PeerConnection now > builds the list of "send streams" from the list of senders, rather than > the collection of local media streams. > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > Cr-Commit-Position: refs/heads/master@{#10414} Review URL: https://codereview.webrtc.org/1468113002 Cr-Commit-Position: refs/heads/master@{#10790}
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. This page is maintained by the Google Chrome team.
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