| /* |
| * 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_ASYNCTCPSOCKET_H_ |
| #define WEBRTC_BASE_ASYNCTCPSOCKET_H_ |
| |
| #include "webrtc/base/asyncpacketsocket.h" |
| #include "webrtc/base/scoped_ptr.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 outpos_ == 0; } |
| void ClearOutBuffer() { outpos_ = 0; } |
| |
| private: |
| // Called by the underlying socket |
| void OnConnectEvent(AsyncSocket* socket); |
| void OnReadEvent(AsyncSocket* socket); |
| void OnWriteEvent(AsyncSocket* socket); |
| void OnCloseEvent(AsyncSocket* socket, int error); |
| |
| scoped_ptr<AsyncSocket> socket_; |
| bool listen_; |
| char* inbuf_, * outbuf_; |
| size_t insize_, inpos_, outsize_, outpos_; |
| |
| 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: |
| DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocket); |
| }; |
| |
| } // namespace rtc |
| |
| #endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_ |