143 lines
6.2 KiB
C++
143 lines
6.2 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef NET_DCSCTP_TX_SEND_QUEUE_H_
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#define NET_DCSCTP_TX_SEND_QUEUE_H_
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#include <cstdint>
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#include <limits>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "net/dcsctp/common/internal_types.h"
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#include "net/dcsctp/packet/data.h"
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#include "net/dcsctp/public/types.h"
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namespace dcsctp {
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class SendQueue {
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public:
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// Container for a data chunk that is produced by the SendQueue
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struct DataToSend {
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explicit DataToSend(Data data) : data(std::move(data)) {}
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// The data to send, including all parameters.
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Data data;
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// Partial reliability - RFC3758
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MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit();
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TimeMs expires_at = TimeMs::InfiniteFuture();
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// Lifecycle - set for the last fragment, and `LifecycleId::NotSet()` for
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// all other fragments.
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LifecycleId lifecycle_id = LifecycleId::NotSet();
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};
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virtual ~SendQueue() = default;
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// TODO(boivie): This interface is obviously missing an "Add" function, but
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// that is postponed a bit until the story around how to model message
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// prioritization, which is important for any advanced stream scheduler, is
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// further clarified.
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// Produce a chunk to be sent.
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//
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// `max_size` refers to how many payload bytes that may be produced, not
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// including any headers.
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virtual absl::optional<DataToSend> Produce(TimeMs now, size_t max_size) = 0;
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// Discards a partially sent message identified by the parameters `unordered`,
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// `stream_id` and `message_id`. The `message_id` comes from the returned
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// information when having called `Produce`. A partially sent message means
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// that it has had at least one fragment of it returned when `Produce` was
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// called prior to calling this method).
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//
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// This is used when a message has been found to be expired (by the partial
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// reliability extension), and the retransmission queue will signal the
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// receiver that any partially received message fragments should be skipped.
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// This means that any remaining fragments in the Send Queue must be removed
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// as well so that they are not sent.
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//
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// This function returns true if this message had unsent fragments still in
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// the queue that were discarded, and false if there were no such fragments.
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virtual bool Discard(IsUnordered unordered,
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StreamID stream_id,
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MID message_id) = 0;
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// Prepares the stream to be reset. This is used to close a WebRTC data
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// channel and will be signaled to the other side.
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//
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// Concretely, it discards all whole (not partly sent) messages in the given
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// stream and pauses that stream so that future added messages aren't
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// produced until `ResumeStreams` is called.
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//
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// TODO(boivie): Investigate if it really should discard any message at all.
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// RFC8831 only mentions that "[RFC6525] also guarantees that all the messages
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// are delivered (or abandoned) before the stream is reset."
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//
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// This method can be called multiple times to add more streams to be
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// reset, and paused while they are resetting. This is the first part of the
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// two-phase commit protocol to reset streams, where the caller completes the
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// procedure by either calling `CommitResetStreams` or `RollbackResetStreams`.
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virtual void PrepareResetStream(StreamID stream_id) = 0;
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// Indicates if there are any streams that are ready to be reset.
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virtual bool HasStreamsReadyToBeReset() const = 0;
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// Returns a list of streams that are ready to be included in an outgoing
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// stream reset request. Any streams that are returned here must be included
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// in an outgoing stream reset request, and there must not be concurrent
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// requests. Before calling this method again, you must have called
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virtual std::vector<StreamID> GetStreamsReadyToBeReset() = 0;
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// Called to commit to reset the streams returned by
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// `GetStreamsReadyToBeReset`. It will reset the stream sequence numbers
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// (SSNs) and message identifiers (MIDs) and resume the paused streams.
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virtual void CommitResetStreams() = 0;
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// Called to abort the resetting of streams returned by
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// `GetStreamsReadyToBeReset`. Will resume the paused streams without
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// resetting the stream sequence numbers (SSNs) or message identifiers (MIDs).
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// Note that the non-partial messages that were discarded when calling
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// `PrepareResetStreams` will not be recovered, to better match the intention
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// from the sender to "close the channel".
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virtual void RollbackResetStreams() = 0;
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// Resets all message identifier counters (MID, SSN) and makes all partially
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// messages be ready to be re-sent in full. This is used when the peer has
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// been detected to have restarted and is used to try to minimize the amount
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// of data loss. However, data loss cannot be completely guaranteed when a
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// peer restarts.
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virtual void Reset() = 0;
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// Returns the amount of buffered data. This doesn't include packets that are
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// e.g. inflight.
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virtual size_t buffered_amount(StreamID stream_id) const = 0;
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// Returns the total amount of buffer data, for all streams.
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virtual size_t total_buffered_amount() const = 0;
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// Returns the limit for the `OnBufferedAmountLow` event. Default value is 0.
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virtual size_t buffered_amount_low_threshold(StreamID stream_id) const = 0;
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// Sets a limit for the `OnBufferedAmountLow` event.
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virtual void SetBufferedAmountLowThreshold(StreamID stream_id,
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size_t bytes) = 0;
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// Configures the send queue to support interleaved message sending as
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// described in RFC8260. Every send queue starts with this value set as
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// disabled, but can later change it when the capabilities of the connection
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// have been negotiated. This affects the behavior of the `Produce` method.
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virtual void EnableMessageInterleaving(bool enabled) = 0;
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};
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} // namespace dcsctp
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#endif // NET_DCSCTP_TX_SEND_QUEUE_H_
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