modules/audio_coding/neteq/delay_manager.h

/*
 *  Copyright (c) 2012 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_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
#define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_

#include <string.h>  // Provide access to size_t.

#include <deque>
#include <memory>

#include "absl/types/optional.h"
#include "api/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/histogram.h"
#include "rtc_base/constructor_magic.h"

namespace webrtc {

class DelayManager {
 public:
  DelayManager(int max_packets_in_buffer,
               int base_minimum_delay_ms,
               int histogram_quantile,
               absl::optional<int> resample_interval_ms,
               int max_history_ms,
               const TickTimer* tick_timer,
               std::unique_ptr<Histogram> histogram);

  // Create a DelayManager object. Notify the delay manager that the packet
  // buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
  // is the number of packet slots in the buffer) and that the target delay
  // should be greater than or equal to |base_minimum_delay_ms|. Supply a
  // PeakDetector object to the DelayManager.
  static std::unique_ptr<DelayManager> Create(int max_packets_in_buffer,
                                              int base_minimum_delay_ms,
                                              const TickTimer* tick_timer);

  virtual ~DelayManager();

  // Updates the delay manager with a new incoming packet, with |timestamp| from
  // the RTP header. This updates the statistics and a new target buffer level
  // is calculated. Returns the relative delay if it can be calculated. If
  // |reset| is true, restarts the relative arrival delay calculation from this
  // packet.
  virtual absl::optional<int> Update(uint32_t timestamp,
                                     int sample_rate_hz,
                                     bool reset = false);

  // Resets all state.
  virtual void Reset();

  // Gets the target buffer level in milliseconds.
  virtual int TargetDelayMs() const;

  // Notifies the DelayManager of how much audio data is carried in each packet.
  virtual int SetPacketAudioLength(int length_ms);

  // Accessors and mutators.
  // Assuming |delay| is in valid range.
  virtual bool SetMinimumDelay(int delay_ms);
  virtual bool SetMaximumDelay(int delay_ms);
  virtual bool SetBaseMinimumDelay(int delay_ms);
  virtual int GetBaseMinimumDelay() const;

  // These accessors are only intended for testing purposes.
  int effective_minimum_delay_ms_for_test() const {
    return effective_minimum_delay_ms_;
  }
  int histogram_quantile() const { return histogram_quantile_; }
  Histogram* histogram() const { return histogram_.get(); }

 private:
  // Provides value which minimum delay can't exceed based on current buffer
  // size and given |maximum_delay_ms_|. Lower bound is a constant 0.
  int MinimumDelayUpperBound() const;

  // Updates |delay_history_|.
  void UpdateDelayHistory(int iat_delay_ms,
                          uint32_t timestamp,
                          int sample_rate_hz);

  // Calculate relative packet arrival delay from |delay_history_|.
  int CalculateRelativePacketArrivalDelay() const;

  // Updates |effective_minimum_delay_ms_| delay based on current
  // |minimum_delay_ms_|, |base_minimum_delay_ms_| and |maximum_delay_ms_|
  // and buffer size.
  void UpdateEffectiveMinimumDelay();

  // Makes sure that |delay_ms| is less than maximum delay, if any maximum
  // is set. Also, if possible check |delay_ms| to be less than 75% of
  // |max_packets_in_buffer_|.
  bool IsValidMinimumDelay(int delay_ms) const;

  bool IsValidBaseMinimumDelay(int delay_ms) const;

  bool first_packet_received_;
  // TODO(jakobi): set maximum buffer delay instead of number of packets.
  const int max_packets_in_buffer_;
  std::unique_ptr<Histogram> histogram_;
  const int histogram_quantile_;
  const TickTimer* tick_timer_;
  const absl::optional<int> resample_interval_ms_;
  const int max_history_ms_;

  int base_minimum_delay_ms_;
  int effective_minimum_delay_ms_;  // Used as lower bound for target delay.
  int minimum_delay_ms_;            // Externally set minimum delay.
  int maximum_delay_ms_;            // Externally set maximum allowed delay.

  int packet_len_ms_ = 0;
  std::unique_ptr<TickTimer::Stopwatch>
      packet_iat_stopwatch_;  // Time elapsed since last packet.
  int target_level_ms_;       // Currently preferred buffer level.
  uint32_t last_timestamp_;   // Timestamp for the last received packet.
  int num_reordered_packets_ = 0;
  int max_delay_in_interval_ms_ = 0;
  std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;

  struct PacketDelay {
    int iat_delay_ms;
    uint32_t timestamp;
  };
  std::deque<PacketDelay> delay_history_;

  RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
};

}  // namespace webrtc
#endif  // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_