| commit | bc5831999d3354509d75357b659b4bb8e39f8a6c | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Jun 24 21:06:35 2016 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Jun 24 21:06:42 2016 |
| tree | fbff19820ab72b8d452b4ca9cf65c9aff2b26bfd | |
| parent | ba8d4337b7751e57161c8b0dfa77d3e4faeb953b [diff] |
Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. This eliminates the need for the extra layer of indirection provided by mediastreamprovider.h. It will thus make it easier to implement new functionality in RtpSender/RtpReceiver. It also brings us one step closer to the end goal of combining "senders" and "send streams". Currently the sender still needs to go through the BaseChannel and MediaChannel, using an SSRC as a key. R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/2046173002 . Cr-Commit-Position: refs/heads/master@{#13285}
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.