blob: b6c359dc1e3629203f84197c707b59de077982b3 [file] [log] [blame]
* 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.
#include <cstdint>
#include <deque>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/public/dcsctp_message.h"
#include "net/dcsctp/public/dcsctp_socket.h"
#include "net/dcsctp/public/types.h"
#include "net/dcsctp/tx/send_queue.h"
#include "net/dcsctp/tx/stream_scheduler.h"
namespace dcsctp {
// The Round Robin SendQueue holds all messages that the client wants to send,
// but that haven't yet been split into chunks and fully sent on the wire.
// As defined in,
// it will cycle to send messages from different streams. It will send all
// fragments from one message before continuing with a different message on
// possibly a different stream, until support for message interleaving has been
// implemented.
// As messages can be (requested to be) sent before the connection is properly
// established, this send queue is always present - even for closed connections.
// The send queue may trigger callbacks:
// * `OnBufferedAmountLow`, `OnTotalBufferedAmountLow`
// These will be triggered as defined in their documentation.
// * `OnLifecycleMessageExpired(/*maybe_delivered=*/false)`, `OnLifecycleEnd`
// These will be triggered when messages have been expired, abandoned or
// discarded from the send queue. If a message is fully produced, meaning
// that the last fragment has been produced, the responsibility to send
// lifecycle events is then transferred to the retransmission queue, which
// is the one asking to produce the message.
class RRSendQueue : public SendQueue {
RRSendQueue(absl::string_view log_prefix,
DcSctpSocketCallbacks* callbacks,
size_t buffer_size,
size_t mtu,
StreamPriority default_priority,
size_t total_buffered_amount_low_threshold);
// Indicates if the buffer is full. Note that it's up to the caller to ensure
// that the buffer is not full prior to adding new items to it.
bool IsFull() const;
// Indicates if the buffer is empty.
bool IsEmpty() const;
// Adds the message to be sent using the `send_options` provided. The current
// time should be in `now`. Note that it's the responsibility of the caller to
// ensure that the buffer is not full (by calling `IsFull`) before adding
// messages to it.
void Add(webrtc::Timestamp now,
DcSctpMessage message,
const SendOptions& send_options = {});
// Implementation of `SendQueue`.
absl::optional<DataToSend> Produce(webrtc::Timestamp now,
size_t max_size) override;
bool Discard(StreamID stream_id, OutgoingMessageId message_id) override;
void PrepareResetStream(StreamID streams) override;
bool HasStreamsReadyToBeReset() const override;
std::vector<StreamID> GetStreamsReadyToBeReset() override;
void CommitResetStreams() override;
void RollbackResetStreams() override;
void Reset() override;
size_t buffered_amount(StreamID stream_id) const override;
size_t total_buffered_amount() const override {
return total_buffered_amount_.value();
size_t buffered_amount_low_threshold(StreamID stream_id) const override;
void SetBufferedAmountLowThreshold(StreamID stream_id, size_t bytes) override;
void EnableMessageInterleaving(bool enabled) override {
void SetStreamPriority(StreamID stream_id, StreamPriority priority);
StreamPriority GetStreamPriority(StreamID stream_id) const;
HandoverReadinessStatus GetHandoverReadiness() const;
void AddHandoverState(DcSctpSocketHandoverState& state);
void RestoreFromState(const DcSctpSocketHandoverState& state);
struct MessageAttributes {
IsUnordered unordered;
MaxRetransmits max_retransmissions;
webrtc::Timestamp expires_at;
LifecycleId lifecycle_id;
// Represents a value and a "low threshold" that when the value reaches or
// goes under the "low threshold", will trigger `on_threshold_reached`
// callback.
class ThresholdWatcher {
explicit ThresholdWatcher(std::function<void()> on_threshold_reached)
: on_threshold_reached_(std::move(on_threshold_reached)) {}
// Increases the value.
void Increase(size_t bytes) { value_ += bytes; }
// Decreases the value and triggers `on_threshold_reached` if it's at or
// below `low_threshold()`.
void Decrease(size_t bytes);
size_t value() const { return value_; }
size_t low_threshold() const { return low_threshold_; }
void SetLowThreshold(size_t low_threshold);
const std::function<void()> on_threshold_reached_;
size_t value_ = 0;
size_t low_threshold_ = 0;
// Per-stream information.
class OutgoingStream : public StreamScheduler::StreamProducer {
RRSendQueue* parent,
StreamScheduler* scheduler,
StreamID stream_id,
StreamPriority priority,
std::function<void()> on_buffered_amount_low,
const DcSctpSocketHandoverState::OutgoingStream* state = nullptr)
: parent_(*parent),
scheduler_stream_(scheduler->CreateStream(this, stream_id, priority)),
next_unordered_mid_(MID(state ? state->next_unordered_mid : 0)),
next_ordered_mid_(MID(state ? state->next_ordered_mid : 0)),
next_ssn_(SSN(state ? state->next_ssn : 0)),
buffered_amount_(std::move(on_buffered_amount_low)) {}
StreamID stream_id() const { return scheduler_stream_->stream_id(); }
// Enqueues a message to this stream.
void Add(DcSctpMessage message, MessageAttributes attributes);
// Implementing `StreamScheduler::StreamProducer`.
absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
size_t max_size) override;
size_t bytes_to_send_in_next_message() const override;
const ThresholdWatcher& buffered_amount() const { return buffered_amount_; }
ThresholdWatcher& buffered_amount() { return buffered_amount_; }
// Discards a partially sent message, see `SendQueue::Discard`.
bool Discard(OutgoingMessageId message_id);
// Pauses this stream, which is used before resetting it.
void Pause();
// Resumes a paused stream.
void Resume();
bool IsReadyToBeReset() const {
return pause_state_ == PauseState::kPaused;
bool IsResetting() const { return pause_state_ == PauseState::kResetting; }
void SetAsResetting() {
RTC_DCHECK(pause_state_ == PauseState::kPaused);
pause_state_ = PauseState::kResetting;
// Resets this stream, meaning MIDs and SSNs are set to zero.
void Reset();
// Indicates if this stream has a partially sent message in it.
bool has_partially_sent_message() const;
StreamPriority priority() const { return scheduler_stream_->priority(); }
void SetPriority(StreamPriority priority) {
void AddHandoverState(
DcSctpSocketHandoverState::OutgoingStream& state) const;
// Streams are paused before they can be reset. To reset a stream, the
// socket sends an outgoing stream reset command with the TSN of the last
// fragment of the last message, so that receivers and senders can agree on
// when it stopped. And if the send queue is in the middle of sending a
// message, and without fragments not yet sent and without TSNs allocated to
// them, it will keep sending data until that message has ended.
enum class PauseState {
// The stream is not paused, and not scheduled to be reset.
// The stream has requested to be reset/paused but is still producing
// fragments of a message that hasn't ended yet. When it does, it will
// transition to the `kPaused` state.
// The stream is fully paused and can be reset.
// The stream has been added to an outgoing stream reset request and a
// response from the peer hasn't been received yet.
// An enqueued message and metadata.
struct Item {
explicit Item(OutgoingMessageId message_id,
DcSctpMessage msg,
MessageAttributes attributes)
: message_id(message_id),
remaining_size(message.payload().size()) {}
OutgoingMessageId message_id;
DcSctpMessage message;
MessageAttributes attributes;
// The remaining payload (offset and size) to be sent, when it has been
// fragmented.
size_t remaining_offset;
size_t remaining_size;
// If set, an allocated Message ID and SSN. Will be allocated when the
// first fragment is sent.
absl::optional<MID> mid = absl::nullopt;
absl::optional<SSN> ssn = absl::nullopt;
// The current Fragment Sequence Number, incremented for each fragment.
FSN current_fsn = FSN(0);
bool IsConsistent() const;
void HandleMessageExpired(OutgoingStream::Item& item);
RRSendQueue& parent_;
const std::unique_ptr<StreamScheduler::Stream> scheduler_stream_;
PauseState pause_state_ = PauseState::kNotPaused;
// MIDs are different for unordered and ordered messages sent on a stream.
MID next_unordered_mid_;
MID next_ordered_mid_;
SSN next_ssn_;
// Enqueued messages, and metadata.
std::deque<Item> items_;
// The current amount of buffered data.
ThresholdWatcher buffered_amount_;
bool IsConsistent() const;
OutgoingStream& GetOrCreateStreamInfo(StreamID stream_id);
absl::optional<DataToSend> Produce(
std::map<StreamID, OutgoingStream>::iterator it,
webrtc::Timestamp now,
size_t max_size);
const absl::string_view log_prefix_;
DcSctpSocketCallbacks& callbacks_;
const size_t buffer_size_;
const StreamPriority default_priority_;
OutgoingMessageId current_message_id = OutgoingMessageId(0);
StreamScheduler scheduler_;
// The total amount of buffer data, for all streams.
ThresholdWatcher total_buffered_amount_;
// All streams, and messages added to those.
std::map<StreamID, OutgoingStream> streams_;
} // namespace dcsctp