chromium/third_party/webrtc/rtc_base/async_packet_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_PACKET_SOCKET_H_
#define RTC_BASE_ASYNC_PACKET_SOCKET_H_

#include <cstdint>
#include <vector>

#include "api/sequence_checker.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/dscp.h"
#include "rtc_base/network/received_packet.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/socket.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/system/rtc_export.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/time_utils.h"

namespace rtc {

// This structure holds the info needed to update the packet send time header
// extension, including the information needed to update the authentication tag
// after changing the value.
struct PacketTimeUpdateParams { … };

// This structure holds meta information for the packet which is about to send
// over network.
struct RTC_EXPORT PacketOptions { … };

// Provides the ability to receive packets asynchronously. Sends are not
// buffered since it is acceptable to drop packets under high load.
class RTC_EXPORT AsyncPacketSocket : public sigslot::has_slots<> { … };

// Listen socket, producing an AsyncPacketSocket when a peer connects.
class RTC_EXPORT AsyncListenSocket : public sigslot::has_slots<> { … };

void CopySocketInformationToPacketInfo(size_t packet_size_bytes,
                                       const AsyncPacketSocket& socket_from,
                                       bool is_connectionless,
                                       rtc::PacketInfo* info);

}  // namespace rtc

#endif  // RTC_BASE_ASYNC_PACKET_SOCKET_H_