| commit | ac9d92ccbe2b29590c53f702e11dc625820480d5 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Mon Oct 26 18:48:22 2015 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Oct 26 18:48:26 2015 |
| tree | e29f178c6072bd418a4f8eda7b4b6ac467cf3bfa | |
| parent | 4cba4eba596706f2238d14f96f4e181f47e5034c [diff] |
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. Review URL: https://codereview.webrtc.org/1413713003 Cr-Commit-Position: refs/heads/master@{#10414}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.