chromium/third_party/webrtc/net/dcsctp/public/types.h

/*
 *  Copyright 2019 The Chromium Authors. All rights reserved.
 *  Copyright (c) 2021 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 NET_DCSCTP_PUBLIC_TYPES_H_
#define NET_DCSCTP_PUBLIC_TYPES_H_

#include <cstdint>
#include <limits>

#include "api/units/time_delta.h"
#include "rtc_base/strong_alias.h"

namespace dcsctp {

// Stream Identifier
StreamID;

// Payload Protocol Identifier (PPID)
PPID;

// Timeout Identifier
TimeoutID;

// Indicates if a message is allowed to be received out-of-order compared to
// other messages on the same stream.
IsUnordered;

// Stream priority, where higher values indicate higher priority. The meaning of
// this value and how it's used depends on the stream scheduler.
StreamPriority;

// Duration, as milliseconds. Overflows after 24 days.
class DurationMs : public webrtc::StrongAlias<class DurationMsTag, int32_t> { … };

constexpr inline DurationMs operator+(DurationMs lhs, DurationMs rhs) { … }
constexpr inline DurationMs operator-(DurationMs lhs, DurationMs rhs) { … }
template <typename T>
constexpr inline DurationMs operator*(DurationMs lhs, T rhs) { … }
template <typename T>
constexpr inline DurationMs operator*(T lhs, DurationMs rhs) { … }
constexpr inline int32_t operator/(DurationMs lhs, DurationMs rhs) { … }

// Represents time, in milliseconds since a client-defined epoch.
class TimeMs : public webrtc::StrongAlias<class TimeMsTag, int64_t> { … };

constexpr inline TimeMs operator+(TimeMs lhs, DurationMs rhs) { … }
constexpr inline TimeMs operator+(DurationMs lhs, TimeMs rhs) { … }
constexpr inline TimeMs operator-(TimeMs lhs, DurationMs rhs) { … }
constexpr inline DurationMs operator-(TimeMs lhs, TimeMs rhs) { … }

// The maximum number of times the socket should attempt to retransmit a
// message which fails the first time in unreliable mode.
class MaxRetransmits
    : public webrtc::StrongAlias<class MaxRetransmitsTag, uint16_t> { … };

// An identifier that can be set on sent messages, and picked by the sending
// client. If different from `::NotSet()`, lifecycle events will be generated,
// and eventually `DcSctpSocketCallbacks::OnLifecycleEnd` will be called to
// indicate that the lifecycle isn't tracked any longer. The value zero (0) is
// not a valid lifecycle identifier, and will be interpreted as not having it
// set.
class LifecycleId : public webrtc::StrongAlias<class LifecycleIdTag, uint64_t> { … };

// To enable zero checksum feature, both peers must agree on which alternate
// error detection method that is used. See
// https://www.ietf.org/archive/id/draft-ietf-tsvwg-sctp-zero-checksum-06.html.
class ZeroChecksumAlternateErrorDetectionMethod
    : public webrtc::StrongAlias<
          class ZeroChecksumAlternateErrorDetectionMethodTag,
          uint32_t> { … };
}  // namespace dcsctp

#endif  // NET_DCSCTP_PUBLIC_TYPES_H_