| /* |
| * 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, |
| const PacketTime& packet_time); |
| Packet(const Packet& p); |
| virtual ~Packet(); |
| |
| SocketAddress addr; |
| 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); |
| ~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 { 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_; |
| int ready_to_send_count_ = 0; |
| int64_t prev_packet_timestamp_; |
| RTC_DISALLOW_COPY_AND_ASSIGN(TestClient); |
| }; |
| |
| } // namespace rtc |
| |
| #endif // WEBRTC_BASE_TESTCLIENT_H_ |