FilterModelConfig.h

/*
 * This file is part of libsidplayfp, a SID player engine.
 *
 * Copyright 2011-2024 Leandro Nini <drfiemost@users.sourceforge.net>
 * Copyright 2007-2010 Antti Lankila
 * Copyright 2004,2010 Dag Lem
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
 */
 
#ifndef FILTERMODELCONFIG_H
#define FILTERMODELCONFIG_H
 
#include <algorithm>
#include <random>
#include <cassert>
#include <climits>
 
#include "OpAmp.h"
#include "Spline.h"
 
#include "sidcxx11.h"
 
namespace reSIDfp
{
 
class FilterModelConfig
{
public:
    // The highpass summer has 2 - 6 inputs (bandpass, lowpass, and 0 - 4 voices).
    template<int i>
    struct summer_offset
    {
        static constexpr int value = summer_offset<i - 1>::value + ((2 + i - 1) << 16);
    };
 
    // The mixer has 0 - 7 inputs (0 - 4 voices and 0 - 3 filter outputs).
    template<int i>
    struct mixer_offset
    {
        static constexpr int value = mixer_offset<i - 1>::value + ((i - 1) << 16);
    };
 
protected:
    static inline unsigned short to_ushort_dither(double x, double d_noise)
    {
        const int tmp = static_cast<int>(x + d_noise);
        assert((tmp >= 0) && (tmp <= USHRT_MAX));
        return static_cast<unsigned short>(tmp);
    }
 
    static inline unsigned short to_ushort(double x)
    {
        return to_ushort_dither(x, 0.5);
    }
 
private:
    /*
     * Hack to add quick dither when converting values from float to int
     * and avoid quantization noise.
     * Hopefully this can be removed the day we move all the analog part
     * processing to floats.
     *
     * Not sure about the effect of using such small buffer of numbers
     * since the random sequence repeats every 1024 values but for
     * now it seems to do the job.
     */
    class Randomnoise
    {
    private:
        double buffer[1024];
        mutable int index = 0;
    public:
        Randomnoise()
        {
            std::uniform_real_distribution<double> unif(0., 1.);
            std::default_random_engine re;
            for (int i=0; i<1024; i++)
                buffer[i] = unif(re);
        }
        double getNoise() const { index = (index + 1) & 0x3ff; return buffer[index]; }
    };
 
protected:
    const double C;
 
 
    static constexpr double Ut = 26.0e-3;
 
    const double Vdd;           
    const double Vth;           
    const double Vddt;          
    double uCox;                
 
    // Derived stuff
    const double vmin, vmax;
    const double denorm, norm;
 
    const double N16;
 
    const double voice_voltage_range;
 
    double currFactorCoeff;
 
 
    unsigned short* mixer;          //-V730_NOINIT this is initialized in the derived class constructor
    unsigned short* summer;         //-V730_NOINIT this is initialized in the derived class constructor
    unsigned short* volume;         //-V730_NOINIT this is initialized in the derived class constructor
    unsigned short* resonance;      //-V730_NOINIT this is initialized in the derived class constructor
 
    unsigned short opamp_rev[1 << 16]; //-V730_NOINIT this is initialized in the derived class constructor
 
private:
    Randomnoise rnd;
 
private:
    FilterModelConfig(const FilterModelConfig&) = delete;
    FilterModelConfig& operator= (const FilterModelConfig&) = delete;
 
    inline double getVoiceVoltage(float value, unsigned int env) const
    {
        return value * voice_voltage_range + getVoiceDC(env);
    }
 
protected:
    FilterModelConfig(
        double vvr,
        double c,
        double vdd,
        double vth,
        double ucox,
        const Spline::Point *opamp_voltage,
        int opamp_size
    );
 
    ~FilterModelConfig();
 
    void setUCox(double new_uCox);
 
    virtual double getVoiceDC(unsigned int env) const = 0;
 
    inline void buildSummerTable(const OpAmp& opampModel)
    {
        const double r_N16 = 1. / N16;
 
        int idx = 0;
        for (int i = 0; i < 5; i++)
        {
            const int idiv = 2 + i;        // 2 - 6 input "resistors".
            const int size = idiv << 16;
            const double n = idiv;
            const double r_idiv = 1. / idiv;
            opampModel.reset();
 
            for (int vi = 0; vi < size; vi++)
            {
                const double vin = vmin + vi * r_N16 * r_idiv; /* vmin .. vmax */
                summer[idx++] = getNormalizedValue(opampModel.solve(n, vin));
            }
        }
    }
 
    inline void buildMixerTable(const OpAmp& opampModel, double nRatio)
    {
        const double r_N16 = 1. / N16;
 
        int idx = 0;
        for (int i = 0; i < 8; i++)
        {
            const int idiv = (i == 0) ? 1 : i;
            const int size = (i == 0) ? 1 : i << 16;
            const double n = i * nRatio;
            const double r_idiv = 1. / idiv;
            opampModel.reset();
 
            for (int vi = 0; vi < size; vi++)
            {
                const double vin = vmin + vi * r_N16 * r_idiv; /* vmin .. vmax */
                mixer[idx++] = getNormalizedValue(opampModel.solve(n, vin));
            }
        }
    }
 
    inline void buildVolumeTable(const OpAmp& opampModel, double nDivisor)
    {
        const double r_N16 = 1. / N16;
 
        int idx = 0;
        for (int n8 = 0; n8 < 16; n8++)
        {
            constexpr int size = 1 << 16;
            const double n = n8 / nDivisor;
            opampModel.reset();
 
            for (int vi = 0; vi < size; vi++)
            {
                const double vin = vmin + vi * r_N16; /* vmin .. vmax */
                volume[idx++] = getNormalizedValue(opampModel.solve(n, vin));
            }
        }
    }
 
    inline void buildResonanceTable(const OpAmp& opampModel, const double resonance_n[16])
    {
        const double r_N16 = 1. / N16;
 
        int idx = 0;
        for (int n8 = 0; n8 < 16; n8++)
        {
            constexpr int size = 1 << 16;
            opampModel.reset();
 
            for (int vi = 0; vi < size; vi++)
            {
                const double vin = vmin + vi * r_N16; /* vmin .. vmax */
                resonance[idx++] = getNormalizedValue(opampModel.solve(resonance_n[n8], vin));
            }
        }
    }
 
public:
    unsigned short* getVolume() { return volume; }
    unsigned short* getResonance() { return resonance; }
    unsigned short* getSummer() { return summer; }
    unsigned short* getMixer() { return mixer; }
 
    inline unsigned short getOpampRev(int i) const { return opamp_rev[i]; }
    inline double getVddt() const { return Vddt; }
    inline double getVth() const { return Vth; }
 
    // helper functions
 
    inline unsigned short getNormalizedValue(double value) const
    {
        return to_ushort_dither(N16 * (value - vmin), rnd.getNoise());
    }
 
    template<int N>
    inline unsigned short getNormalizedCurrentFactor(double wl) const
    {
        return to_ushort((1 << N) * currFactorCoeff * wl);
    }
 
    inline unsigned short getNVmin() const
    {
        return to_ushort(N16 * vmin);
    }
 
    inline int getNormalizedVoice(float value, unsigned int env) const
    {
        return static_cast<int>(getNormalizedValue(getVoiceVoltage(value, env)));
    }
};
 
template<>
struct FilterModelConfig::summer_offset<0>
{
    static constexpr int value = 0;
};
 
template<>
struct FilterModelConfig::mixer_offset<1>
{
    static constexpr int value = 1;
};
 
template<>
struct FilterModelConfig::mixer_offset<0>
{
    static constexpr int value = 0;
};
 
} // namespace reSIDfp
 
#endif