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

/*
 *  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_DCSCTP_HANDOVER_STATE_H_
#define NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_H_

#include <cstdint>
#include <string>
#include <vector>

#include "rtc_base/strong_alias.h"

namespace dcsctp {

// Stores state snapshot of a dcSCTP socket. The snapshot can be used to
// recreate the socket - possibly in another process. This state should be
// treaded as opaque - the calling client should not inspect or alter it except
// for serialization. Serialization is not provided by dcSCTP. If needed it has
// to be implemented in the calling client.
struct DcSctpSocketHandoverState { … };

// A list of possible reasons for a socket to be not ready for handover.
enum class HandoverUnreadinessReason : uint32_t { … };

// Return value of `DcSctpSocketInterface::GetHandoverReadiness`. Set of
// `HandoverUnreadinessReason` bits. When no bit is set, the socket is in the
// state in which a snapshot of the state can be made by
// `GetHandoverStateAndClose()`.
class HandoverReadinessStatus
    : public webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t> { … };

}  // namespace dcsctp

#endif  // NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_H_