Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.

BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/base/testclient.h

@@ -0,0 +1,93 @@
+/*
+ *  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 <vector>
+#include "webrtc/base/asyncudpsocket.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<> {
+ public:
+  // Records the contents of a packet that was received.
+  struct Packet {
+    Packet(const SocketAddress& a, const char* b, size_t s);
+    Packet(const Packet& p);
+    virtual ~Packet();
+
+    SocketAddress addr;
+    char*  buf;
+    size_t size;
+  };
+
+  // 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);
+  ~TestClient();
+
+  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 a reasonable amount of time.  The caller must delete the packet
+  // when done with it.
+  Packet* NextPacket();
+
+  // 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;
+
+ private:
+  static const int kTimeout = 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);
+
+  CriticalSection crit_;
+  AsyncPacketSocket* socket_;
+  std::vector<Packet*>* packets_;
+  bool ready_to_send_;
+  DISALLOW_EVIL_CONSTRUCTORS(TestClient);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_TESTCLIENT_H_