1 /* 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_ 12 #define COMMON_AUDIO_SMOOTHING_FILTER_H_ 13 14 #include <stdint.h> 15 16 #include "absl/types/optional.h" 17 #include "rtc_base/constructor_magic.h" 18 19 namespace webrtc { 20 21 class SmoothingFilter { 22 public: 23 virtual ~SmoothingFilter() = default; 24 virtual void AddSample(float sample) = 0; 25 virtual absl::optional<float> GetAverage() = 0; 26 virtual bool SetTimeConstantMs(int time_constant_ms) = 0; 27 }; 28 29 // SmoothingFilterImpl applies an exponential filter 30 // alpha = exp(-1.0 / time_constant_ms); 31 // y[t] = alpha * y[t-1] + (1 - alpha) * sample; 32 // This implies a sample rate of 1000 Hz, i.e., 1 sample / ms. 33 // But SmoothingFilterImpl allows sparse samples. All missing samples will be 34 // assumed to equal the last received sample. 35 class SmoothingFilterImpl final : public SmoothingFilter { 36 public: 37 // |init_time_ms| is initialization time. It defines a period starting from 38 // the arriving time of the first sample. During this period, the exponential 39 // filter uses a varying time constant so that a smaller time constant will be 40 // applied to the earlier samples. This is to allow the the filter to adapt to 41 // earlier samples quickly. After the initialization period, the time constant 42 // will be set to |init_time_ms| first and can be changed through 43 // |SetTimeConstantMs|. 44 explicit SmoothingFilterImpl(int init_time_ms); 45 ~SmoothingFilterImpl() override; 46 47 void AddSample(float sample) override; 48 absl::optional<float> GetAverage() override; 49 bool SetTimeConstantMs(int time_constant_ms) override; 50 51 // Methods used for unittests. alpha()52 float alpha() const { return alpha_; } 53 54 private: 55 void UpdateAlpha(int time_constant_ms); 56 void ExtrapolateLastSample(int64_t time_ms); 57 58 const int init_time_ms_; 59 const float init_factor_; 60 const float init_const_; 61 62 absl::optional<int64_t> init_end_time_ms_; 63 float last_sample_; 64 float alpha_; 65 float state_; 66 int64_t last_state_time_ms_; 67 68 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl); 69 }; 70 71 } // namespace webrtc 72 73 #endif // COMMON_AUDIO_SMOOTHING_FILTER_H_ 74