Refactor TestClient to use std::unique_ptr, and fix VirtualSocketServerTest leaks.

webrtc/base/testclient.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.
  */
 
 #ifndef WEBRTC_BASE_TESTCLIENT_H_
 #define WEBRTC_BASE_TESTCLIENT_H_
 
+#include <memory>
 #include <vector>
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 
 namespace rtc {
 
 // A simple client that can send TCP or UDP data and check that it receives
 // what it expects to receive. Useful for testing server functionality.
 class TestClient : public sigslot::has_slots<> {
@@ skipping 11 matching lines @@
     char*  buf;
     size_t size;
     PacketTime packet_time;
   };
 
   // Default timeout for NextPacket reads.
   static const int kTimeoutMs = 5000;
 
   // Creates a client that will send and receive with the given socket and
   // will post itself messages with the given thread.
-  explicit TestClient(AsyncPacketSocket* socket);
+  explicit TestClient(std::unique_ptr<AsyncPacketSocket> socket);
   ~TestClient() override;
 
   SocketAddress address() const { return socket_->GetLocalAddress(); }
   SocketAddress remote_address() const { return socket_->GetRemoteAddress(); }
 
   // Checks that the socket moves to the specified connect state.
   bool CheckConnState(AsyncPacketSocket::State state);
 
   // Checks that the socket is connected to the remote side.
   bool CheckConnected() {
     return CheckConnState(AsyncPacketSocket::STATE_CONNECTED);
   }
 
   // Sends using the clients socket.
   int Send(const char* buf, size_t size);
 
   // Sends using the clients socket to the given destination.
   int SendTo(const char* buf, size_t size, const SocketAddress& dest);
 
-  // Returns the next packet received by the client or 0 if none is received
-  // within the specified timeout. The caller must delete the packet
-  // when done with it.
-  Packet* NextPacket(int timeout_ms);
+  // Returns the next packet received by the client or null if none is received
+  // within the specified timeout.
+  std::unique_ptr<Packet> NextPacket(int timeout_ms);
 
   // Checks that the next packet has the given contents. Returns the remote
   // address that the packet was sent from.
   bool CheckNextPacket(const char* buf, size_t len, SocketAddress* addr);
 
   // Checks that no packets have arrived or will arrive in the next second.
   bool CheckNoPacket();
 
   int GetError();
   int SetOption(Socket::Option opt, int value);
 
   bool ready_to_send() const { return ready_to_send_count() > 0; }
 
   // How many times SignalReadyToSend has been fired.
   int ready_to_send_count() const { return ready_to_send_count_; }
 
  private:
   // Timeout for reads when no packet is expected.
   static const int kNoPacketTimeoutMs = 1000;
   // Workaround for the fact that AsyncPacketSocket::GetConnState doesn't exist.
   Socket::ConnState GetState();
   // Slot for packets read on the socket.
   void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t len,
                 const SocketAddress& remote_addr,
                 const PacketTime& packet_time);
   void OnReadyToSend(AsyncPacketSocket* socket);
   bool CheckTimestamp(int64_t packet_timestamp);
 
   CriticalSection crit_;
-  AsyncPacketSocket* socket_;
-  std::vector<Packet*>* packets_;
+  std::unique_ptr<AsyncPacketSocket> socket_;
+  std::vector<std::unique_ptr<Packet>> packets_;
   int ready_to_send_count_ = 0;
   int64_t prev_packet_timestamp_;
   RTC_DISALLOW_COPY_AND_ASSIGN(TestClient);
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_TESTCLIENT_H_