Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.

webrtc/p2p/base/p2ptransportchannel.h

 /*
  *  Copyright 2004 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.
  */
 
@@ skipping 214 matching lines @@
                         ProtocolType proto,
                         IceMessage* stun_msg,
                         const std::string& remote_username,
                         bool port_muxed);
   void OnPortDestroyed(PortInterface* port);
   void OnRoleConflict(PortInterface* port);
 
   void OnConnectionStateChange(Connection* connection);
   void OnReadPacket(Connection *connection, const char *data, size_t len,
                     const rtc::PacketTime& packet_time);
-  void OnSentPacket(PortInterface* port, const rtc::SentPacket& sent_packet);
+  void OnSentPacket(const rtc::SentPacket& sent_packet);
   void OnReadyToSend(Connection* connection);
   void OnConnectionDestroyed(Connection *connection);
 
   void OnNominated(Connection* conn);
 
   void OnMessage(rtc::Message* pmsg) override;
   void OnSort();
   void OnCheckAndPing();
 
   void PruneConnections();
@@ skipping 50 matching lines @@
   bool gather_continually_ = false;
   int weak_ping_delay_ = WEAK_PING_DELAY;
   TransportChannelState state_ = TransportChannelState::STATE_INIT;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_