text: /*
   *  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 <cstdint>
  #include <limits>
  #include <utility>
  #include <vector>
  
  #include "absl/types/optional.h"
  #include "api/array_view.h"
  #include "net/dcsctp/common/internal_types.h"
  #include "net/dcsctp/packet/data.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 {
      explicit DataToSend(Data data) : data(std::move(data)) {}
      // The data to send, including all parameters.
      Data data;
  
      // Partial reliability - RFC3758
      MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit();
      TimeMs expires_at = TimeMs::InfiniteFuture();
    };
  
    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 absl::optional<DataToSend> Produce(TimeMs now, size_t max_size) = 0;
  
    // Discards a partially sent message identified by the parameters `unordered`,
    // `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(IsUnordered unordered,
                         StreamID stream_id,
                         MID message_id) = 0;
  
    // Prepares the streams 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
    // streams and pauses those streams 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 PrepareResetStreams(rtc::ArrayView<const StreamID> streams) = 0;
  
    // Returns true if all non-discarded messages during `PrepareResetStreams`
    // (which are those that was partially sent before that method was called)
    // have been sent.
    virtual bool CanResetStreams() const = 0;
  
    // Called to commit to reset the streams provided to `PrepareResetStreams`.
    // 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 provided to `PrepareResetStreams`.
    // 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;
  };
  }  // namespace dcsctp
  
  #endif  // NET_DCSCTP_TX_SEND_QUEUE_H_
  