/*
 *  Copyright (c) 2016 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 WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_
#define WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_

#include "webrtc/api/optional.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/system_wrappers/include/clock.h"

namespace webrtc {

class SmoothingFilter {
 public:
  virtual ~SmoothingFilter() = default;
  virtual void AddSample(float sample) = 0;
  virtual rtc::Optional<float> GetAverage() = 0;
  virtual bool SetTimeConstantMs(int time_constant_ms) = 0;
};

// SmoothingFilterImpl applies an exponential filter
//   alpha = exp(-1.0 / time_constant_ms);
//   y[t] = alpha * y[t-1] + (1 - alpha) * sample;
// This implies a sample rate of 1000 Hz, i.e., 1 sample / ms.
// But SmoothingFilterImpl allows sparse samples. All missing samples will be
// assumed to equal the last received sample.
class SmoothingFilterImpl final : public SmoothingFilter {
 public:
  // |init_time_ms| is initialization time. It defines a period starting from
  // the arriving time of the first sample. During this period, the exponential
  // filter uses a varying time constant so that a smaller time constant will be
  // applied to the earlier samples. This is to allow the the filter to adapt to
  // earlier samples quickly. After the initialization period, the time constant
  // will be set to |init_time_ms| first and can be changed through
  // |SetTimeConstantMs|.
  explicit SmoothingFilterImpl(int init_time_ms);
  ~SmoothingFilterImpl() override;

  void AddSample(float sample) override;
  rtc::Optional<float> GetAverage() override;
  bool SetTimeConstantMs(int time_constant_ms) override;

  // Methods used for unittests.
  float alpha() const { return alpha_; }

 private:
  void UpdateAlpha(int time_constant_ms);
  void ExtrapolateLastSample(int64_t time_ms);

  const int init_time_ms_;
  const float init_factor_;
  const float init_const_;

  rtc::Optional<int64_t> init_end_time_ms_;
  float last_sample_;
  float alpha_;
  float state_;
  int64_t last_state_time_ms_;

  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
};

}  // namespace webrtc

#endif  // WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_