modules/pacing/round_robin_packet_queue.h

/*
 *  Copyright (c) 2017 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 MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
#define MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_

#include <map>
#include <queue>
#include <set>

#include "modules/pacing/packet_queue_interface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"

namespace webrtc {

class RoundRobinPacketQueue : public PacketQueueInterface {
 public:
  explicit RoundRobinPacketQueue(const Clock* clock);
  ~RoundRobinPacketQueue() override;

  using Packet = PacketQueueInterface::Packet;

  void Push(const Packet& packet) override;
  const Packet& BeginPop() override;
  void CancelPop(const Packet& packet) override;
  void FinalizePop(const Packet& packet) override;

  bool Empty() const override;
  size_t SizeInPackets() const override;
  uint64_t SizeInBytes() const override;

  int64_t OldestEnqueueTimeMs() const override;
  int64_t AverageQueueTimeMs() const override;
  void UpdateQueueTime(int64_t timestamp_ms) override;
  void SetPauseState(bool paused, int64_t timestamp_ms) override;

 private:
  struct StreamPrioKey {
    StreamPrioKey() = default;
    StreamPrioKey(RtpPacketSender::Priority priority, int64_t bytes)
        : priority(priority), bytes(bytes) {}

    bool operator<(const StreamPrioKey& other) const {
      if (priority != other.priority)
        return priority < other.priority;
      return bytes < other.bytes;
    }

    const RtpPacketSender::Priority priority;
    const size_t bytes;
  };

  struct Stream {
    Stream();

    virtual ~Stream();

    size_t bytes;
    uint32_t ssrc;
    std::priority_queue<Packet> packet_queue;

    // Whenever a packet is inserted for this stream we check if |priority_it|
    // points to an element in |stream_priorities_|, and if it does it means
    // this stream has already been scheduled, and if the scheduled priority is
    // lower than the priority of the incoming packet we reschedule this stream
    // with the higher priority.
    std::multimap<StreamPrioKey, uint32_t>::iterator priority_it;
  };

  static constexpr size_t kMaxLeadingBytes = 1400;

  Stream* GetHighestPriorityStream();

  // Just used to verify correctness.
  bool IsSsrcScheduled(uint32_t ssrc) const;

  const Clock* const clock_;
  int64_t time_last_updated_;
  rtc::Optional<Packet> pop_packet_;
  rtc::Optional<Stream*> pop_stream_;

  bool paused_ = false;
  size_t size_packets_ = 0;
  size_t size_bytes_ = 0;
  size_t max_bytes_ = kMaxLeadingBytes;
  int64_t queue_time_sum_ms_ = 0;
  int64_t pause_time_sum_ms_ = 0;

  // A map of streams used to prioritize from which stream to send next. We use
  // a multimap instead of a priority_queue since the priority of a stream can
  // change as a new packet is inserted, and a multimap allows us to remove and
  // then reinsert a StreamPrioKey if the priority has increased.
  std::multimap<StreamPrioKey, uint32_t> stream_priorities_;

  // A map of SSRCs to Streams.
  std::map<uint32_t, Stream> streams_;

  // The enqueue time of every packet currently in the queue. Used to figure out
  // the age of the oldest packet in the queue.
  std::multiset<int64_t> enqueue_times_;
};
}  // namespace webrtc

#endif  // MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_