Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/peerconnectioninterface.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,
@@ skipping 59 matching lines @@
 
 #include <string>
 #include <vector>
 
 #include "talk/app/webrtc/datachannelinterface.h"
 #include "talk/app/webrtc/dtlsidentitystore.h"
 #include "talk/app/webrtc/dtmfsenderinterface.h"
 #include "talk/app/webrtc/dtlsidentitystore.h"
 #include "talk/app/webrtc/jsep.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/app/webrtc/rtpreceiverinterface.h"
+#include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/statstypes.h"
 #include "talk/app/webrtc/umametrics.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/socketaddress.h"
 
 namespace rtc {
 class SSLIdentity;
@@ skipping 49 matching lines @@
                                   int value) = 0;
   // TODO(jbauch): Make method abstract when it is implemented by Chromium.
   virtual void AddHistogramSample(PeerConnectionMetricsName type,
                                   const std::string& value) {}
 
  protected:
   virtual ~MetricsObserverInterface() {}
 };
 
 typedef MetricsObserverInterface UMAObserver;
+typedef std::vector<rtc::scoped_refptr<RtpSenderInterface>> RtpSenderRefptrs;
+typedef std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
+    RtpReceiverRefptrs;
 
 class PeerConnectionInterface : public rtc::RefCountInterface {
  public:
   // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
   enum SignalingState {
     kStable,
     kHaveLocalOffer,
     kHaveLocalPrAnswer,
     kHaveRemoteOffer,
     kHaveRemotePrAnswer,
@@ skipping 156 matching lines @@
   // Remove a MediaStream from this PeerConnection.
   // Note that a SessionDescription negotiation is need before the
   // remote peer is notified.
   virtual void RemoveStream(MediaStreamInterface* stream) = 0;
 
   // Returns pointer to the created DtmfSender on success.
   // Otherwise returns NULL.
   virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) = 0;
 
+  virtual RtpSenderRefptrs GetSenders() const = 0;
+
+  virtual RtpReceiverRefptrs GetReceivers() const = 0;
+
   virtual bool GetStats(StatsObserver* observer,
                         MediaStreamTrackInterface* track,
                         StatsOutputLevel level) = 0;
 
   virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
       const std::string& label,
       const DataChannelInit* config) = 0;
 
   virtual const SessionDescriptionInterface* local_description() const = 0;
   virtual const SessionDescriptionInterface* remote_description() const = 0;
@@ skipping 263 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_