| commit | 6eca7e3c371383020095ba346e1ac70f38a8c0fd | [log] [tgz] |
|---|---|---|
| author | tommi <tommi@webrtc.org> | Tue Dec 15 04:27:11 2015 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Dec 15 12:27:20 2015 +0000 |
| tree | 507dadd06870705d53861cb0bc2f8787060067d4 | |
| parent | 6db6cdc604f9a866991ecf8454eb7f7aa69918ea [diff] |
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( Additionally: * Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack. * AddSink/RemoveSink are now on all audio sources (like they are for video sources). While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state. BUG=chromium:569526 Review URL: https://codereview.webrtc.org/1522903002 Cr-Commit-Position: refs/heads/master@{#11026}
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.