Move talk/app/webrtc to webrtc/api

talk/app/webrtc/rtpreceiverinterface.h

Index: talk/app/webrtc/rtpreceiverinterface.h
diff --git a/talk/app/webrtc/rtpreceiverinterface.h b/talk/app/webrtc/rtpreceiverinterface.h
deleted file mode 100644
index 099699efc4be7a6d729558144131a4163eeed865..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/rtpreceiverinterface.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 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,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * 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.
- */
-
-// This file contains interfaces for RtpReceivers
-// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
-
-#ifndef TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_
-#define TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_
-
-#include <string>
-
-#include "talk/app/webrtc/proxy.h"
-#include "talk/app/webrtc/mediastreaminterface.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-
-namespace webrtc {
-
-class RtpReceiverInterface : public rtc::RefCountInterface {
- public:
-  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
-
-  // Not to be confused with "mid", this is a field we can temporarily use
-  // to uniquely identify a receiver until we implement Unified Plan SDP.
-  virtual std::string id() const = 0;
-
-  virtual void Stop() = 0;
-
- protected:
-  virtual ~RtpReceiverInterface() {}
-};
-
-// Define proxy for RtpReceiverInterface.
-BEGIN_PROXY_MAP(RtpReceiver)
-PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
-PROXY_CONSTMETHOD0(std::string, id)
-PROXY_METHOD0(void, Stop)
-END_PROXY()
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_