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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 WEBRTC_RTC_BASE_ASYNCTCPSOCKET_H_
#define WEBRTC_RTC_BASE_ASYNCTCPSOCKET_H_
#include <memory>
#include "webrtc/rtc_base/asyncpacketsocket.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_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
#endif // WEBRTC_RTC_BASE_ASYNCTCPSOCKET_H_