/*
 *  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_TX_SEND_QUEUE_H_
#define NET_DCSCTP_TX_SEND_QUEUE_H_

#include <cstddef>
#include <optional>
#include <utility>
#include <vector>

#include "api/units/timestamp.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/data.h"
#include "net/dcsctp/public/dcsctp_handover_state.h"
#include "net/dcsctp/public/types.h"

namespace dcsctp {

class SendQueue {
 public:
  // Container for a data chunk that is produced by the SendQueue
  struct DataToSend {
    DataToSend(OutgoingMessageId message_id, Data data)
        : message_id(message_id), data(std::move(data)) {}

    OutgoingMessageId message_id;

    // The data to send, including all parameters.
    Data data;

    // Partial reliability - RFC3758
    MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit();
    webrtc::Timestamp expires_at = webrtc::Timestamp::PlusInfinity();

    // Lifecycle - set for the last fragment, and `LifecycleId::NotSet()` for
    // all other fragments.
    LifecycleId lifecycle_id = LifecycleId::NotSet();
  };

  virtual ~SendQueue() = default;

  // TODO(boivie): This interface is obviously missing an "Add" function, but
  // that is postponed a bit until the story around how to model message
  // prioritization, which is important for any advanced stream scheduler, is
  // further clarified.

  // Produce a chunk to be sent.
  //
  // `max_size` refers to how many payload bytes that may be produced, not
  // including any headers.
  virtual std::optional<DataToSend> Produce(webrtc::Timestamp now,
                                            size_t max_size) = 0;

  // Discards a partially sent message identified by the parameters
  // `stream_id` and `message_id`. The `message_id` comes from the returned
  // information when having called `Produce`. A partially sent message means
  // that it has had at least one fragment of it returned when `Produce` was
  // called prior to calling this method).
  //
  // This is used when a message has been found to be expired (by the partial
  // reliability extension), and the retransmission queue will signal the
  // receiver that any partially received message fragments should be skipped.
  // This means that any remaining fragments in the Send Queue must be removed
  // as well so that they are not sent.
  //
  // This function returns true if this message had unsent fragments still in
  // the queue that were discarded, and false if there were no such fragments.
  virtual bool Discard(StreamID stream_id, OutgoingMessageId message_id) = 0;

  // Prepares the stream to be reset. This is used to close a WebRTC data
  // channel and will be signaled to the other side.
  //
  // Concretely, it discards all whole (not partly sent) messages in the given
  // stream and pauses that stream so that future added messages aren't
  // produced until `ResumeStreams` is called.
  //
  // TODO(boivie): Investigate if it really should discard any message at all.
  // RFC8831 only mentions that "[RFC6525] also guarantees that all the messages
  // are delivered (or abandoned) before the stream is reset."
  //
  // This method can be called multiple times to add more streams to be
  // reset, and paused while they are resetting. This is the first part of the
  // two-phase commit protocol to reset streams, where the caller completes the
  // procedure by either calling `CommitResetStreams` or `RollbackResetStreams`.
  virtual void PrepareResetStream(StreamID stream_id) = 0;

  // Indicates if there are any streams that are ready to be reset.
  virtual bool HasStreamsReadyToBeReset() const = 0;

  // Returns a list of streams that are ready to be included in an outgoing
  // stream reset request. Any streams that are returned here must be included
  // in an outgoing stream reset request, and there must not be concurrent
  // requests. Before calling this method again, you must have called
  virtual std::vector<StreamID> GetStreamsReadyToBeReset() = 0;

  // Called to commit to reset the streams returned by
  // `GetStreamsReadyToBeReset`. It will reset the stream sequence numbers
  // (SSNs) and message identifiers (MIDs) and resume the paused streams.
  virtual void CommitResetStreams() = 0;

  // Called to abort the resetting of streams returned by
  // `GetStreamsReadyToBeReset`. Will resume the paused streams without
  // resetting the stream sequence numbers (SSNs) or message identifiers (MIDs).
  // Note that the non-partial messages that were discarded when calling
  // `PrepareResetStreams` will not be recovered, to better match the intention
  // from the sender to "close the channel".
  virtual void RollbackResetStreams() = 0;

  // Resets all message identifier counters (MID, SSN) and makes all partially
  // messages be ready to be re-sent in full. This is used when the peer has
  // been detected to have restarted and is used to try to minimize the amount
  // of data loss. However, data loss cannot be completely guaranteed when a
  // peer restarts.
  virtual void Reset() = 0;

  // Returns the amount of buffered data. This doesn't include packets that are
  // e.g. inflight.
  virtual size_t buffered_amount(StreamID stream_id) const = 0;

  // Returns the total amount of buffer data, for all streams.
  virtual size_t total_buffered_amount() const = 0;

  // Returns the limit for the `OnBufferedAmountLow` event. Default value is 0.
  virtual size_t buffered_amount_low_threshold(StreamID stream_id) const = 0;

  // Sets a limit for the `OnBufferedAmountLow` event.
  virtual void SetBufferedAmountLowThreshold(StreamID stream_id,
                                             size_t bytes) = 0;

  // Configures the send queue to support interleaved message sending as
  // described in RFC8260. Every send queue starts with this value set as
  // disabled, but can later change it when the capabilities of the connection
  // have been negotiated. This affects the behavior of the `Produce` method.
  virtual void EnableMessageInterleaving(bool enabled) = 0;

  // Adds handover state to `state`.
  virtual void AddHandoverState(webrtc::Timestamp now,
                                DcSctpSocketHandoverState& state) const = 0;

  // Restores state from `state`.
  virtual void RestoreFromState(webrtc::Timestamp now,
                                const DcSctpSocketHandoverState& state) = 0;
};
}  // namespace dcsctp

#endif  // NET_DCSCTP_TX_SEND_QUEUE_H_