|  | /* | 
|  | *  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/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<> { | 
|  | 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; | 
|  | }; | 
|  |  | 
|  | // 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); | 
|  | ~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); | 
|  |  | 
|  | // 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: | 
|  | // 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); | 
|  |  | 
|  | CriticalSection crit_; | 
|  | AsyncPacketSocket* socket_; | 
|  | std::vector<Packet*>* packets_; | 
|  | bool ready_to_send_; | 
|  | RTC_DISALLOW_COPY_AND_ASSIGN(TestClient); | 
|  | }; | 
|  |  | 
|  | }  // namespace rtc | 
|  |  | 
|  | #endif  // WEBRTC_BASE_TESTCLIENT_H_ |