Move webrtc/{base => rtc_base}

webrtc/base/testclient.h

Index: webrtc/base/testclient.h
diff --git a/webrtc/base/testclient.h b/webrtc/base/testclient.h
index bd78d83b355eeeaec99fb7e19f02b2cacee29a84..378e2b81d461dd014c90dd52b96d86479060bc44 100644
--- a/webrtc/base/testclient.h
+++ b/webrtc/base/testclient.h
@@ -11,104 +11,9 @@
 #ifndef WEBRTC_BASE_TESTCLIENT_H_
 #define WEBRTC_BASE_TESTCLIENT_H_
 
-#include <memory>
-#include <vector>
-#include "webrtc/base/asyncudpsocket.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/fakeclock.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(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, FakeClock* fake_clock);
-  ~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 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();
-  // 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);
-  void AdvanceTime(int ms);
-
-  FakeClock* fake_clock_ = nullptr;
-  CriticalSection crit_;
-  std::unique_ptr<AsyncPacketSocket> socket_;
-  std::vector<std::unique_ptr<Packet>> packets_;
-  int ready_to_send_count_ = 0;
-  int64_t prev_packet_timestamp_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(TestClient);
-};
-
-}  // namespace rtc
+// This header is deprecated and is just left here temporarily during
+// refactoring. See https://bugs.webrtc.org/7634 for more details.
+#include "webrtc/rtc_base/testclient.h"
 
 #endif  // WEBRTC_BASE_TESTCLIENT_H_