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=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/base/testclient.h b/webrtc/base/testclient.h
new file mode 100644
index 0000000..d56f948
--- /dev/null
+++ b/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_
