Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/mediastreamprovider.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
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  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 
+#include "webrtc/base/basictypes.h"
+
 namespace cricket {
 
 class AudioRenderer;
 class VideoCapturer;
 class VideoRenderer;
 struct AudioOptions;
 struct VideoOptions;
 
 }  // namespace cricket
 
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   virtual void SetVideoSend(uint32 ssrc, bool enable,
                             const cricket::VideoOptions* options) = 0;
 
  protected:
   virtual ~VideoProviderInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_