| /* | 
 |  *  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 COMMON_AUDIO_SMOOTHING_FILTER_H_ | 
 | #define COMMON_AUDIO_SMOOTHING_FILTER_H_ | 
 |  | 
 | #include <stdint.h> | 
 |  | 
 | #include "absl/types/optional.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | class SmoothingFilter { | 
 |  public: | 
 |   virtual ~SmoothingFilter() = default; | 
 |   virtual void AddSample(float sample) = 0; | 
 |   virtual absl::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() = delete; | 
 |   SmoothingFilterImpl(const SmoothingFilterImpl&) = delete; | 
 |   SmoothingFilterImpl& operator=(const SmoothingFilterImpl&) = delete; | 
 |  | 
 |   ~SmoothingFilterImpl() override; | 
 |  | 
 |   void AddSample(float sample) override; | 
 |   absl::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_; | 
 |  | 
 |   absl::optional<int64_t> init_end_time_ms_; | 
 |   float last_sample_; | 
 |   float alpha_; | 
 |   float state_; | 
 |   int64_t last_state_time_ms_; | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // COMMON_AUDIO_SMOOTHING_FILTER_H_ |