Repo created

TMessagesProj/jni/voip/webrtc/rtc_base/test_client.h

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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 <memory>
+#include <vector>
+
+#include "api/units/timestamp.h"
+#include "rtc_base/async_udp_socket.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/network/received_packet.h"
+#include "rtc_base/synchronization/mutex.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 rtc::ReceivedPacket& received_packet);
+    Packet(const Packet& p);
+
+    SocketAddress addr;
+    Buffer buf;
+    absl::optional<webrtc::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 rtc::ReceivedPacket& received_packet);
+  void OnReadyToSend(AsyncPacketSocket* socket);
+  bool CheckTimestamp(absl::optional<webrtc::Timestamp> packet_timestamp);
+  void AdvanceTime(int ms);
+
+  ThreadProcessingFakeClock* fake_clock_ = nullptr;
+  webrtc::Mutex mutex_;
+  std::unique_ptr<AsyncPacketSocket> socket_;
+  std::vector<std::unique_ptr<Packet>> packets_;
+  int ready_to_send_count_ = 0;
+  absl::optional<webrtc::Timestamp> prev_packet_timestamp_;
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_TEST_CLIENT_H_