talk/app/webrtc/rtpsenderinterface.h

/*
 * libjingle
 * Copyright 2015 Google Inc.
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// 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_