| /* |
| * 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 "rtc_base/strong_alias.h" |
| |
| namespace dcsctp { |
| |
| // Stream Identifier |
| using StreamID = webrtc::StrongAlias<class StreamIDTag, uint16_t>; |
| |
| // Payload Protocol Identifier (PPID) |
| using PPID = webrtc::StrongAlias<class PPIDTag, uint32_t>; |
| |
| // Timeout Identifier |
| using TimeoutID = webrtc::StrongAlias<class TimeoutTag, uint64_t>; |
| |
| // Indicates if a message is allowed to be received out-of-order compared to |
| // other messages on the same stream. |
| using IsUnordered = webrtc::StrongAlias<class IsUnorderedTag, bool>; |
| |
| // Stream priority, where higher values indicate higher priority. The meaning of |
| // this value and how it's used depends on the stream scheduler. |
| using StreamPriority = webrtc::StrongAlias<class StreamPriorityTag, uint16_t>; |
| |
| // Duration, as milliseconds. Overflows after 24 days. |
| class DurationMs : public webrtc::StrongAlias<class DurationMsTag, int32_t> { |
| public: |
| constexpr explicit DurationMs(const UnderlyingType& v) |
| : webrtc::StrongAlias<class DurationMsTag, int32_t>(v) {} |
| |
| // Convenience methods for working with time. |
| constexpr DurationMs& operator+=(DurationMs d) { |
| value_ += d.value_; |
| return *this; |
| } |
| constexpr DurationMs& operator-=(DurationMs d) { |
| value_ -= d.value_; |
| return *this; |
| } |
| template <typename T> |
| constexpr DurationMs& operator*=(T factor) { |
| value_ *= factor; |
| return *this; |
| } |
| }; |
| |
| constexpr inline DurationMs operator+(DurationMs lhs, DurationMs rhs) { |
| return lhs += rhs; |
| } |
| constexpr inline DurationMs operator-(DurationMs lhs, DurationMs rhs) { |
| return lhs -= rhs; |
| } |
| template <typename T> |
| constexpr inline DurationMs operator*(DurationMs lhs, T rhs) { |
| return lhs *= rhs; |
| } |
| template <typename T> |
| constexpr inline DurationMs operator*(T lhs, DurationMs rhs) { |
| return rhs *= lhs; |
| } |
| constexpr inline int32_t operator/(DurationMs lhs, DurationMs rhs) { |
| return lhs.value() / rhs.value(); |
| } |
| |
| // Represents time, in milliseconds since a client-defined epoch. |
| class TimeMs : public webrtc::StrongAlias<class TimeMsTag, int64_t> { |
| public: |
| constexpr explicit TimeMs(const UnderlyingType& v) |
| : webrtc::StrongAlias<class TimeMsTag, int64_t>(v) {} |
| |
| // Convenience methods for working with time. |
| constexpr TimeMs& operator+=(DurationMs d) { |
| value_ += *d; |
| return *this; |
| } |
| constexpr TimeMs& operator-=(DurationMs d) { |
| value_ -= *d; |
| return *this; |
| } |
| |
| static constexpr TimeMs InfiniteFuture() { |
| return TimeMs(std::numeric_limits<int64_t>::max()); |
| } |
| }; |
| |
| constexpr inline TimeMs operator+(TimeMs lhs, DurationMs rhs) { |
| return lhs += rhs; |
| } |
| constexpr inline TimeMs operator+(DurationMs lhs, TimeMs rhs) { |
| return rhs += lhs; |
| } |
| constexpr inline TimeMs operator-(TimeMs lhs, DurationMs rhs) { |
| return lhs -= rhs; |
| } |
| constexpr inline DurationMs operator-(TimeMs lhs, TimeMs rhs) { |
| return DurationMs(*lhs - *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> { |
| public: |
| constexpr explicit MaxRetransmits(const UnderlyingType& v) |
| : webrtc::StrongAlias<class MaxRetransmitsTag, uint16_t>(v) {} |
| |
| // There should be no limit - the message should be sent reliably. |
| static constexpr MaxRetransmits NoLimit() { |
| return MaxRetransmits(std::numeric_limits<uint16_t>::max()); |
| } |
| }; |
| |
| // 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> { |
| public: |
| constexpr explicit LifecycleId(const UnderlyingType& v) |
| : webrtc::StrongAlias<class LifecycleIdTag, uint64_t>(v) {} |
| |
| constexpr bool IsSet() const { return value_ != 0; } |
| |
| static constexpr LifecycleId NotSet() { return LifecycleId(0); } |
| }; |
| } // namespace dcsctp |
| |
| #endif // NET_DCSCTP_PUBLIC_TYPES_H_ |