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| Original file line number | Diff line number | Diff line change |
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| @@ -0,0 +1,56 @@ | ||
| #pragma once | ||
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| #include "iir.hpp" | ||
| #include <vector> | ||
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| /** | ||
| High-order filter to be used for anti-aliasing or anti-imaging. | ||
| The template parameter N should be 1/2 the desired filter order. | ||
| Currently uses an 2*N-th order Butterworth filter. | ||
| @TODO: implement Chebyshev, Elliptic filter options. | ||
| */ | ||
| template<int N> | ||
| class AAFilter | ||
| { | ||
| public: | ||
| AAFilter() = default; | ||
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| /** Calculate Q values for a Butterworth filter of a given order */ | ||
| static std::vector<float> calculateButterQs(int order) { | ||
| const int lim = int (order / 2); | ||
| std::vector<float> Qs; | ||
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| for(int k = 1; k <= lim; ++k) { | ||
| auto b = -2.0f * std::cos((2.0f * k + order - 1) * 3.14159 / (2.0f * order)); | ||
| Qs.push_back(1.0f / b); | ||
| } | ||
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| std::reverse(Qs.begin(), Qs.end()); | ||
| return Qs; | ||
| } | ||
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| /** | ||
| * Resets the filter to process at a new sample rate. | ||
| * | ||
| * @param sampleRate: The base (i.e. pre-oversampling) sample rate of the audio being processed | ||
| * @param osRatio: The oversampling ratio at which the filter is being used | ||
| */ | ||
| void reset(float sampleRate, int osRatio) { | ||
| float fc = 0.98f * (sampleRate / 2.0f); | ||
| auto Qs = calculateButterQs(2*N); | ||
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| for(int i = 0; i < N; ++i) | ||
| filters[i].setParameters(BiquadFilter::Type::LOWPASS, fc / (osRatio * sampleRate), Qs[i], 1.0f); | ||
| } | ||
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| inline float process(float x) noexcept { | ||
| for(int i = 0; i < N; ++i) | ||
| x = filters[i].process(x); | ||
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| return x; | ||
| } | ||
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| private: | ||
| BiquadFilter filters[N]; | ||
| }; |
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