| commit | 4171afb1865adb374e7529f41292347cd4deb0ab | [log] [tgz] |
|---|---|---|
| author | Steve Anton <steveanton@webrtc.org> | Mon Nov 20 10:20:22 2017 -0800 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Nov 20 22:41:44 2017 +0000 |
| tree | 678906fde277f957c373509d5277fe6f37ce5313 | |
| parent | cada60193dd77c62568e05cf4f0d130bf372655a [diff] |
Use RtpTransceivers in PeerConnection Moves ownership of the RtpSenders/RtpReceivers/BaseChannels to RtpTransceiver objects. For now, there can only be one RtpTransceiver for audio and one for video. Future work to implement Unified Plan will relax this restriction. Bug: webrtc:7600 Change-Id: I9dfe324de61e2b363948858da72624396e27fc1a Reviewed-on: https://webrtc-review.googlesource.com/21461 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20802}
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.
Authoritative list of directories that contain the native API header files.