| /* |
| * 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_ |