f048872e91
migrating talk/base to webrtc/base. BUG=N/A R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
101 lines
3.6 KiB
C++
101 lines
3.6 KiB
C++
/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_
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#define WEBRTC_BASE_ASYNCTCPSOCKET_H_
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#include "webrtc/base/asyncpacketsocket.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/socketfactory.h"
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namespace rtc {
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// Simulates UDP semantics over TCP. Send and Recv packet sizes
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// are preserved, and drops packets silently on Send, rather than
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// buffer them in user space.
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class AsyncTCPSocketBase : public AsyncPacketSocket {
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public:
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AsyncTCPSocketBase(AsyncSocket* socket, bool listen, size_t max_packet_size);
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virtual ~AsyncTCPSocketBase();
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// Pure virtual methods to send and recv data.
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virtual int Send(const void *pv, size_t cb,
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const rtc::PacketOptions& options) = 0;
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virtual void ProcessInput(char* data, size_t* len) = 0;
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// Signals incoming connection.
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virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
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virtual SocketAddress GetLocalAddress() const;
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virtual SocketAddress GetRemoteAddress() const;
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virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
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const rtc::PacketOptions& options);
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virtual int Close();
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virtual State GetState() const;
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virtual int GetOption(Socket::Option opt, int* value);
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virtual int SetOption(Socket::Option opt, int value);
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virtual int GetError() const;
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virtual void SetError(int error);
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protected:
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// Binds and connects |socket| and creates AsyncTCPSocket for
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// it. Takes ownership of |socket|. Returns NULL if bind() or
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// connect() fail (|socket| is destroyed in that case).
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static AsyncSocket* ConnectSocket(AsyncSocket* socket,
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const SocketAddress& bind_address,
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const SocketAddress& remote_address);
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virtual int SendRaw(const void* pv, size_t cb);
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int FlushOutBuffer();
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// Add data to |outbuf_|.
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void AppendToOutBuffer(const void* pv, size_t cb);
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// Helper methods for |outpos_|.
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bool IsOutBufferEmpty() const { return outpos_ == 0; }
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void ClearOutBuffer() { outpos_ = 0; }
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private:
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// Called by the underlying socket
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void OnConnectEvent(AsyncSocket* socket);
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void OnReadEvent(AsyncSocket* socket);
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void OnWriteEvent(AsyncSocket* socket);
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void OnCloseEvent(AsyncSocket* socket, int error);
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scoped_ptr<AsyncSocket> socket_;
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bool listen_;
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char* inbuf_, * outbuf_;
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size_t insize_, inpos_, outsize_, outpos_;
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DISALLOW_EVIL_CONSTRUCTORS(AsyncTCPSocketBase);
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};
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class AsyncTCPSocket : public AsyncTCPSocketBase {
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public:
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// Binds and connects |socket| and creates AsyncTCPSocket for
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// it. Takes ownership of |socket|. Returns NULL if bind() or
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// connect() fail (|socket| is destroyed in that case).
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static AsyncTCPSocket* Create(AsyncSocket* socket,
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const SocketAddress& bind_address,
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const SocketAddress& remote_address);
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AsyncTCPSocket(AsyncSocket* socket, bool listen);
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virtual ~AsyncTCPSocket() {}
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virtual int Send(const void* pv, size_t cb,
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const rtc::PacketOptions& options);
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virtual void ProcessInput(char* data, size_t* len);
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virtual void HandleIncomingConnection(AsyncSocket* socket);
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private:
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DISALLOW_EVIL_CONSTRUCTORS(AsyncTCPSocket);
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};
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} // namespace rtc
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#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_
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