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/*
* 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
absl::optional<int> max_retransmissions;
absl::optional<TimeMs> expires_at;
};
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_
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