74 lines
2.4 KiB
C
74 lines
2.4 KiB
C
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/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_
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#define COMMON_AUDIO_SMOOTHING_FILTER_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class SmoothingFilter {
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public:
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virtual ~SmoothingFilter() = default;
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virtual void AddSample(float sample) = 0;
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virtual absl::optional<float> GetAverage() = 0;
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virtual bool SetTimeConstantMs(int time_constant_ms) = 0;
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};
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// SmoothingFilterImpl applies an exponential filter
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// alpha = exp(-1.0 / time_constant_ms);
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// y[t] = alpha * y[t-1] + (1 - alpha) * sample;
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// This implies a sample rate of 1000 Hz, i.e., 1 sample / ms.
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// But SmoothingFilterImpl allows sparse samples. All missing samples will be
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// assumed to equal the last received sample.
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class SmoothingFilterImpl final : public SmoothingFilter {
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public:
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// |init_time_ms| is initialization time. It defines a period starting from
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// the arriving time of the first sample. During this period, the exponential
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// filter uses a varying time constant so that a smaller time constant will be
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// applied to the earlier samples. This is to allow the the filter to adapt to
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// earlier samples quickly. After the initialization period, the time constant
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// will be set to |init_time_ms| first and can be changed through
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// |SetTimeConstantMs|.
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explicit SmoothingFilterImpl(int init_time_ms);
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~SmoothingFilterImpl() override;
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void AddSample(float sample) override;
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absl::optional<float> GetAverage() override;
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bool SetTimeConstantMs(int time_constant_ms) override;
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// Methods used for unittests.
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float alpha() const { return alpha_; }
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private:
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void UpdateAlpha(int time_constant_ms);
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void ExtrapolateLastSample(int64_t time_ms);
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const int init_time_ms_;
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const float init_factor_;
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const float init_const_;
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absl::optional<int64_t> init_end_time_ms_;
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float last_sample_;
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float alpha_;
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float state_;
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int64_t last_state_time_ms_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_SMOOTHING_FILTER_H_
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