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/*
* 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_
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