| Index: talk/app/webrtc/mediacontroller.h
|
| diff --git a/talk/app/webrtc/mediacontroller.h b/talk/app/webrtc/mediacontroller.h
|
| index 1191206e18dbf9a7f035c9b95761641db2bcf7f3..d31ce4dc0ae338896285d500d45e0dad4b64526b 100644
|
| --- a/talk/app/webrtc/mediacontroller.h
|
| +++ b/talk/app/webrtc/mediacontroller.h
|
| @@ -25,5 +25,25 @@
|
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
| */
|
|
|
| -// Place holder file to be able to update Chrome's libjingle.gyp before the real
|
| -// implementation goes in.
|
| +#ifndef TALK_APP_WEBRTC_MEDIACONTROLLER_H_
|
| +#define TALK_APP_WEBRTC_MEDIACONTROLLER_H_
|
| +
|
| +#include "webrtc/base/thread.h"
|
| +
|
| +namespace webrtc {
|
| +class Call;
|
| +class VoiceEngine;
|
| +
|
| +// MediaController currently owns shared state between media channels, but in
|
| +// the future will create and own RtpSenders and RtpReceivers.
|
| +class MediaController {
|
| + public:
|
| + static MediaController* Create(rtc::Thread* worker_thread,
|
| + webrtc::VoiceEngine* voice_engine);
|
| +
|
| + virtual ~MediaController() {}
|
| + virtual webrtc::Call* call_w() = 0;
|
| +};
|
| +} // namespace webrtc
|
| +
|
| +#endif // TALK_APP_WEBRTC_MEDIACONTROLLER_H_
|
|
|