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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 RTC_BASE_ASYNC_TCP_SOCKET_H_
#define RTC_BASE_ASYNC_TCP_SOCKET_H_
#include <stddef.h>
#include <cstdint>
#include <memory>
#include "api/array_view.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/buffer.h"
#include "rtc_base/socket.h"
#include "rtc_base/socket_address.h"
namespace webrtc {
// 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(Socket* socket, size_t max_packet_size);
~AsyncTCPSocketBase() override;
AsyncTCPSocketBase(const AsyncTCPSocketBase&) = delete;
AsyncTCPSocketBase& operator=(const AsyncTCPSocketBase&) = delete;
// Pure virtual methods to send and recv data.
int Send(const void* pv,
size_t cb,
const AsyncSocketPacketOptions& options) override = 0;
// Must return the number of bytes processed.
virtual size_t ProcessInput(ArrayView<const uint8_t> data) = 0;
SocketAddress GetLocalAddress() const override;
SocketAddress GetRemoteAddress() const override;
int SendTo(const void* pv,
size_t cb,
const SocketAddress& addr,
const AsyncSocketPacketOptions& 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 Socket* ConnectSocket(Socket* socket,
const SocketAddress& bind_address,
const SocketAddress& remote_address);
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(Socket* socket);
void OnReadEvent(Socket* socket);
void OnWriteEvent(Socket* socket);
void OnCloseEvent(Socket* socket, int error);
std::unique_ptr<Socket> socket_;
Buffer inbuf_;
Buffer outbuf_;
size_t max_insize_;
size_t max_outsize_;
};
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(Socket* socket,
const SocketAddress& bind_address,
const SocketAddress& remote_address);
explicit AsyncTCPSocket(Socket* socket);
~AsyncTCPSocket() override {}
AsyncTCPSocket(const AsyncTCPSocket&) = delete;
AsyncTCPSocket& operator=(const AsyncTCPSocket&) = delete;
int Send(const void* pv,
size_t cb,
const AsyncSocketPacketOptions& options) override;
size_t ProcessInput(ArrayView<const uint8_t>) override;
};
class AsyncTcpListenSocket : public AsyncListenSocket {
public:
explicit AsyncTcpListenSocket(std::unique_ptr<Socket> socket);
State GetState() const override;
SocketAddress GetLocalAddress() const override;
virtual void HandleIncomingConnection(Socket* socket);
private:
// Called by the underlying socket
void OnReadEvent(Socket* socket);
std::unique_ptr<Socket> socket_;
};
} // namespace webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace rtc {
using ::webrtc::AsyncTcpListenSocket;
using ::webrtc::AsyncTCPSocket;
using ::webrtc::AsyncTCPSocketBase;
} // namespace rtc
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // RTC_BASE_ASYNC_TCP_SOCKET_H_
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