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/*
* 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 RTC_BASE_TEST_CLIENT_H_
#define RTC_BASE_TEST_CLIENT_H_
#include <cstddef>
#include <memory>
#include <optional>
#include <vector>
#include "api/units/timestamp.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/buffer.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/network/received_packet.h"
#include "rtc_base/socket.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace webrtc {
// 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 ReceivedIpPacket& received_packet);
Packet(const Packet& p);
SocketAddress addr;
Buffer buf;
std::optional<Timestamp> 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(std::unique_ptr<AsyncPacketSocket> socket);
// Create a test client that will use a fake clock. NextPacket needs to wait
// for a packet to be received, and thus it needs to advance the fake clock
// if the test is using one, rather than just sleeping.
TestClient(std::unique_ptr<AsyncPacketSocket> socket,
ThreadProcessingFakeClock* fake_clock);
~TestClient() override;
TestClient(const TestClient&) = delete;
TestClient& operator=(const TestClient&) = delete;
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 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();
void OnPacket(AsyncPacketSocket* socket,
const ReceivedIpPacket& received_packet);
void OnReadyToSend(AsyncPacketSocket* socket);
bool CheckTimestamp(std::optional<Timestamp> packet_timestamp);
void AdvanceTime(int ms);
ThreadProcessingFakeClock* fake_clock_ = nullptr;
Mutex mutex_;
std::unique_ptr<AsyncPacketSocket> socket_;
std::vector<std::unique_ptr<Packet>> packets_;
int ready_to_send_count_ = 0;
std::optional<Timestamp> prev_packet_timestamp_;
};
} // namespace webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace rtc {
using ::webrtc::TestClient;
} // namespace rtc
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // RTC_BASE_TEST_CLIENT_H_
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