underrun_optimizer.h

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/*

 *  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.

*/

#ifndef MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_

#define MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_

#include <memory>

#include "absl/types/optional.h"

#include "api/neteq/tick_timer.h"

#include "modules/audio_coding/neteq/histogram.h"

namespace webrtc {

// Estimates probability of buffer underrun due to late packet arrival.

// The optimal delay is decided such that the probability of underrun is lower

// than 1 - `histogram_quantile`.

class UnderrunOptimizer {

 public:

  UnderrunOptimizer(const TickTimer* tick_timer,

                    int histogram_quantile,

                    int forget_factor,

                    absl::optional<int> start_forget_weight,

                    absl::optional<int> resample_interval_ms);

  void Update(int relative_delay_ms);

  absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }

  void Reset();

 private:

  const TickTimer* tick_timer_;

  Histogram histogram_;

  const int histogram_quantile_;  // In Q30.

  const absl::optional<int> resample_interval_ms_;

  std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;

  int max_delay_in_interval_ms_ = 0;

  absl::optional<int> optimal_delay_ms_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_