Move webrtc/{base => rtc_base}

webrtc/base/asynctcpsocket.h

Index: webrtc/base/asynctcpsocket.h
diff --git a/webrtc/base/asynctcpsocket.h b/webrtc/base/asynctcpsocket.h
index 2e4ff9aaa7a0f9d20a708a5f629a08b9319b3fa0..d64927bcd562a1b52c55f944c7226cd8b522aa2b 100644
--- a/webrtc/base/asynctcpsocket.h
+++ b/webrtc/base/asynctcpsocket.h
@@ -11,98 +11,9 @@
 #ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_
 #define WEBRTC_BASE_ASYNCTCPSOCKET_H_
 
-#include <memory>
 
-#include "webrtc/base/asyncpacketsocket.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/socketfactory.h"
-
-namespace rtc {
-
-// Simulates UDP semantics over TCP.  Send and Recv packet sizes
-// are preserved, and drops packets silently on Send, rather than
-// buffer them in user space.
-class AsyncTCPSocketBase : public AsyncPacketSocket {
- public:
-  AsyncTCPSocketBase(AsyncSocket* socket, bool listen, size_t max_packet_size);
-  ~AsyncTCPSocketBase() override;
-
-  // Pure virtual methods to send and recv data.
-  int Send(const void *pv, size_t cb,
-                   const rtc::PacketOptions& options) override = 0;
-  virtual void ProcessInput(char* data, size_t* len) = 0;
-  // Signals incoming connection.
-  virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
-
-  SocketAddress GetLocalAddress() const override;
-  SocketAddress GetRemoteAddress() const override;
-  int SendTo(const void* pv,
-             size_t cb,
-             const SocketAddress& addr,
-             const rtc::PacketOptions& options) override;
-  int Close() override;
-
-  State GetState() const override;
-  int GetOption(Socket::Option opt, int* value) override;
-  int SetOption(Socket::Option opt, int value) override;
-  int GetError() const override;
-  void SetError(int error) override;
-
- protected:
-  // Binds and connects |socket| and creates AsyncTCPSocket for
-  // it. Takes ownership of |socket|. Returns null if bind() or
-  // connect() fail (|socket| is destroyed in that case).
-  static AsyncSocket* ConnectSocket(AsyncSocket* socket,
-                                    const SocketAddress& bind_address,
-                                    const SocketAddress& remote_address);
-  virtual int SendRaw(const void* pv, size_t cb);
-  int FlushOutBuffer();
-  // Add data to |outbuf_|.
-  void AppendToOutBuffer(const void* pv, size_t cb);
-
-  // Helper methods for |outpos_|.
-  bool IsOutBufferEmpty() const { return outbuf_.size() == 0; }
-  void ClearOutBuffer() { outbuf_.Clear(); }
-
- private:
-  // Called by the underlying socket
-  void OnConnectEvent(AsyncSocket* socket);
-  void OnReadEvent(AsyncSocket* socket);
-  void OnWriteEvent(AsyncSocket* socket);
-  void OnCloseEvent(AsyncSocket* socket, int error);
-
-  std::unique_ptr<AsyncSocket> socket_;
-  bool listen_;
-  Buffer inbuf_;
-  Buffer outbuf_;
-  size_t max_insize_;
-  size_t max_outsize_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocketBase);
-};
-
-class AsyncTCPSocket : public AsyncTCPSocketBase {
- public:
-  // Binds and connects |socket| and creates AsyncTCPSocket for
-  // it. Takes ownership of |socket|. Returns null if bind() or
-  // connect() fail (|socket| is destroyed in that case).
-  static AsyncTCPSocket* Create(AsyncSocket* socket,
-                                const SocketAddress& bind_address,
-                                const SocketAddress& remote_address);
-  AsyncTCPSocket(AsyncSocket* socket, bool listen);
-  ~AsyncTCPSocket() override {}
-
-  int Send(const void* pv,
-           size_t cb,
-           const rtc::PacketOptions& options) override;
-  void ProcessInput(char* data, size_t* len) override;
-  void HandleIncomingConnection(AsyncSocket* socket) override;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocket);
-};
-
-}  // 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/asynctcpsocket.h"
 
 #endif  // WEBRTC_BASE_ASYNCTCPSOCKET_H_