Repo created

TMessagesProj/jni/voip/webrtc/net/dcsctp/tx/stream_scheduler.h

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+/*
+ *  Copyright (c) 2022 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_STREAM_SCHEDULER_H_
+#define NET_DCSCTP_TX_STREAM_SCHEDULER_H_
+
+#include <algorithm>
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <memory>
+#include <queue>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.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 "rtc_base/containers/flat_set.h"
+#include "rtc_base/strong_alias.h"
+
+namespace dcsctp {
+
+// A parameterized stream scheduler. Currently, it implements the round robin
+// scheduling algorithm using virtual finish time. It is to be used as a part of
+// a send queue and will track all active streams (streams that have any data
+// that can be sent).
+//
+// The stream scheduler works with the concept of associating active streams
+// with a "virtual finish time", which is the time when a stream is allowed to
+// produce data. Streams are ordered by their virtual finish time, and the
+// "current virtual time" will advance to the next following virtual finish time
+// whenever a chunk is to be produced.
+//
+// When message interleaving is enabled, the WFQ - Weighted Fair Queueing -
+// scheduling algorithm will be used. And when it's not, round-robin scheduling
+// will be used instead.
+//
+// In the round robin scheduling algorithm, a stream's virtual finish time will
+// just increment by one (1) after having produced a chunk, which results in a
+// round-robin scheduling.
+//
+// In WFQ scheduling algorithm, a stream's virtual finish time will be defined
+// as the number of bytes in the next fragment to be sent, multiplied by the
+// inverse of the stream's priority, meaning that a high priority - or a smaller
+// fragment - results in a closer virtual finish time, compared to a stream with
+// either a lower priority or a larger fragment to be sent.
+class StreamScheduler {
+ private:
+  class VirtualTime : public webrtc::StrongAlias<class VirtualTimeTag, double> {
+   public:
+    constexpr explicit VirtualTime(const UnderlyingType& v)
+        : webrtc::StrongAlias<class VirtualTimeTag, double>(v) {}
+
+    static constexpr VirtualTime Zero() { return VirtualTime(0); }
+  };
+  class InverseWeight
+      : public webrtc::StrongAlias<class InverseWeightTag, double> {
+   public:
+    constexpr explicit InverseWeight(StreamPriority priority)
+        : webrtc::StrongAlias<class InverseWeightTag, double>(
+              1.0 / std::max(static_cast<double>(*priority), 0.000001)) {}
+  };
+
+ public:
+  class StreamProducer {
+   public:
+    virtual ~StreamProducer() = default;
+
+    // Produces a fragment of data to send. The current wall time is specified
+    // as `now` and should be used to skip chunks with expired limited lifetime.
+    // The parameter `max_size` specifies the maximum amount of actual payload
+    // that may be returned. If these constraints prevents the stream from
+    // sending some data, `absl::nullopt` should be returned.
+    virtual absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+                                                          size_t max_size) = 0;
+
+    // Returns the number of payload bytes that is scheduled to be sent in the
+    // next enqueued message, or zero if there are no enqueued messages or if
+    // the stream has been actively paused.
+    virtual size_t bytes_to_send_in_next_message() const = 0;
+  };
+
+  class Stream {
+   public:
+    StreamID stream_id() const { return stream_id_; }
+
+    StreamPriority priority() const { return priority_; }
+    void SetPriority(StreamPriority priority);
+
+    // Will activate the stream _if_ it has any data to send. That is, if the
+    // callback to `bytes_to_send_in_next_message` returns non-zero. If the
+    // callback returns zero, the stream will not be made active.
+    void MaybeMakeActive();
+
+    // Will remove the stream from the list of active streams, and will not try
+    // to produce data from it. To make it active again, call `MaybeMakeActive`.
+    void MakeInactive();
+
+    // Make the scheduler move to another message, or another stream. This is
+    // used to abort the scheduler from continuing producing fragments for the
+    // current message in case it's deleted.
+    void ForceReschedule() { parent_.ForceReschedule(); }
+
+   private:
+    friend class StreamScheduler;
+
+    Stream(StreamScheduler* parent,
+           StreamProducer* producer,
+           StreamID stream_id,
+           StreamPriority priority)
+        : parent_(*parent),
+          producer_(*producer),
+          stream_id_(stream_id),
+          priority_(priority),
+          inverse_weight_(priority) {}
+
+    // Produces a message from this stream. This will only be called on streams
+    // that have data.
+    absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+                                                  size_t max_size);
+
+    void MakeActive(size_t bytes_to_send_next);
+    void ForceMarkInactive();
+
+    VirtualTime current_time() const { return current_virtual_time_; }
+    VirtualTime next_finish_time() const { return next_finish_time_; }
+    size_t bytes_to_send_in_next_message() const {
+      return producer_.bytes_to_send_in_next_message();
+    }
+
+    VirtualTime CalculateFinishTime(size_t bytes_to_send_next) const;
+
+    StreamScheduler& parent_;
+    StreamProducer& producer_;
+    const StreamID stream_id_;
+    StreamPriority priority_;
+    InverseWeight inverse_weight_;
+    // This outgoing stream's "current" virtual_time.
+    VirtualTime current_virtual_time_ = VirtualTime::Zero();
+    VirtualTime next_finish_time_ = VirtualTime::Zero();
+  };
+
+  // The `mtu` parameter represents the maximum SCTP packet size, which should
+  // be the same as `DcSctpOptions::mtu`.
+  StreamScheduler(absl::string_view log_prefix, size_t mtu)
+      : log_prefix_(log_prefix),
+        max_payload_bytes_(mtu - SctpPacket::kHeaderSize -
+                           IDataChunk::kHeaderSize) {}
+
+  std::unique_ptr<Stream> CreateStream(StreamProducer* producer,
+                                       StreamID stream_id,
+                                       StreamPriority priority) {
+    return absl::WrapUnique(new Stream(this, producer, stream_id, priority));
+  }
+
+  void EnableMessageInterleaving(bool enabled) {
+    enable_message_interleaving_ = enabled;
+  }
+
+  // Makes the scheduler stop producing message from the current stream and
+  // re-evaluates which stream to produce from.
+  void ForceReschedule() { currently_sending_a_message_ = false; }
+
+  // Produces a fragment of data to send. The current wall time is specified as
+  // `now` and will be used to skip chunks with expired limited lifetime. The
+  // parameter `max_size` specifies the maximum amount of actual payload that
+  // may be returned. If no data can be produced, `absl::nullopt` is returned.
+  absl::optional<SendQueue::DataToSend> Produce(webrtc::Timestamp now,
+                                                size_t max_size);
+
+  std::set<StreamID> ActiveStreamsForTesting() const;
+
+ private:
+  struct ActiveStreamComparator {
+    // Ordered by virtual finish time (primary), stream-id (secondary).
+    bool operator()(Stream* a, Stream* b) const {
+      VirtualTime a_vft = a->next_finish_time();
+      VirtualTime b_vft = b->next_finish_time();
+      if (a_vft == b_vft) {
+        return a->stream_id() < b->stream_id();
+      }
+      return a_vft < b_vft;
+    }
+  };
+
+  bool IsConsistent() const;
+
+  const absl::string_view log_prefix_;
+  const size_t max_payload_bytes_;
+
+  // The current virtual time, as defined in the WFQ algorithm.
+  VirtualTime virtual_time_ = VirtualTime::Zero();
+
+  // The current stream to send chunks from.
+  Stream* current_stream_ = nullptr;
+
+  bool enable_message_interleaving_ = false;
+
+  // Indicates if the streams is currently sending a message, and should then
+  // - if message interleaving is not enabled - continue sending from this
+  // stream until that message has been sent in full.
+  bool currently_sending_a_message_ = false;
+
+  // The currently active streams, ordered by virtual finish time.
+  webrtc::flat_set<Stream*, ActiveStreamComparator> active_streams_;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_STREAM_SCHEDULER_H_