chromium/third_party/webrtc/rtc_base/async_tcp_socket.h

/*
 *  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 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 { … };

class AsyncTCPSocket : public AsyncTCPSocketBase { … };

class AsyncTcpListenSocket : public AsyncListenSocket { … };

}  // namespace rtc

#endif  // RTC_BASE_ASYNC_TCP_SOCKET_H_