/*
 * 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 RtpSenders
// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface

#ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
#define TALK_APP_WEBRTC_RTPSENDERINTERFACE_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 RtpSenderInterface : public rtc::RefCountInterface {
 public:
  // Returns true if successful in setting the track.
  // Fails if an audio track is set on a video RtpSender, or vice-versa.
  virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
  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 ~RtpSenderInterface() {}
};

// Define proxy for RtpSenderInterface.
BEGIN_PROXY_MAP(RtpSender)
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_METHOD0(void, Stop)
END_PROXY()

}  // namespace webrtc

#endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_