Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/p2p/base/packettransportinternal.h

 /*
  *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
-#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
+#ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
+#define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_
 
 #include <string>
 #include <vector>
 
+#include "webrtc/api/packettransportinterface.h"
 // This is included for PacketOptions.
 #include "webrtc/base/asyncpacketsocket.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 
 namespace cricket {
 class TransportChannel;
 }
 
 namespace rtc {
 struct PacketOptions;
 struct PacketTime;
 struct SentPacket;
 
-class PacketTransportInterface : public sigslot::has_slots<> {
+class PacketTransportInternal : public virtual webrtc::PacketTransportInterface,
+                                public sigslot::has_slots<> {
  public:
-  virtual ~PacketTransportInterface() {}
-
   // Identify the object for logging and debug purpose.
   virtual std::string debug_name() const = 0;
 
   // The transport has been established.
   virtual bool writable() const = 0;
 
   // The transport has received a packet in the last X milliseconds, here X is
   // configured by each implementation.
   virtual bool receiving() const = 0;
 
@@ skipping 15 matching lines @@
   virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
 
   // TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements
   // this, remove the default implementation.
   virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; }
 
   // Returns the most recent error that occurred on this channel.
   virtual int GetError() = 0;
 
   // Emitted when the writable state, represented by |writable()|, changes.
-  sigslot::signal1<PacketTransportInterface*> SignalWritableState;
+  sigslot::signal1<PacketTransportInternal*> SignalWritableState;
 
-  //  Emitted when the PacketTransportInterface is ready to send packets. "Ready
+  //  Emitted when the PacketTransportInternal is ready to send packets. "Ready
   //  to send" is more sensitive than the writable state; a transport may be
   //  writable, but temporarily not able to send packets. For example, the
   //  underlying transport's socket buffer may be full, as indicated by
   //  SendPacket's return code and/or GetError.
-  sigslot::signal1<PacketTransportInterface*> SignalReadyToSend;
+  sigslot::signal1<PacketTransportInternal*> SignalReadyToSend;
 
   // Emitted when receiving state changes to true.
-  sigslot::signal1<PacketTransportInterface*> SignalReceivingState;
+  sigslot::signal1<PacketTransportInternal*> SignalReceivingState;
 
   // Signalled each time a packet is received on this channel.
-  sigslot::signal5<PacketTransportInterface*,
+  sigslot::signal5<PacketTransportInternal*,
                    const char*,
                    size_t,
                    const rtc::PacketTime&,
                    int>
       SignalReadPacket;
 
   // Signalled each time a packet is sent on this channel.
-  sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&>
+  sigslot::signal2<PacketTransportInternal*, const rtc::SentPacket&>
       SignalSentPacket;
+
+ protected:
+  PacketTransportInternal* GetInternal() { return this; }
 };
 
 }  // namespace rtc
 
-#endif  // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_
+#endif  // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERNAL_H_