|  | /* | 
|  | *  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_ |