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AnalogFilter.C

/*
  ZynAddSubFX - a software synthesizer
 
  AnalogFilter.C - Several analog filters (lowpass, highpass...)
  Copyright (C) 2002-2005 Nasca Octavian Paul
  Author: Nasca Octavian Paul

  This program is free software; you can redistribute it and/or modify
  it under the terms of version 2 of the GNU General Public License 
  as published by the Free Software Foundation.

  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 (version 2) for more details.

  You should have received a copy of the GNU General Public License (version 2)
  along with this program; if not, write to the Free Software Foundation,
  Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA

*/

#include <math.h>
#include <stdio.h>
#include "AnalogFilter.h"


AnalogFilter::AnalogFilter (unsigned char Ftype, REALTYPE Ffreq, REALTYPE Fq,
                      unsigned char Fstages)
{
  stages = Fstages;
  for (int i = 0; i < 3; i++)
    {
      oldc[i] = 0.0;
      oldd[i] = 0.0;
      c[i] = 0.0;
      d[i] = 0.0;
    };
  type = Ftype;
  freq = Ffreq;
  q = Fq;
  gain = 1.0;
  if (stages >= MAX_FILTER_STAGES)
    stages = MAX_FILTER_STAGES;
  cleanup ();
  firsttime = 0;
  abovenq = 0;
  oldabovenq = 0;
  setfreq_and_q (Ffreq, Fq);
  firsttime = 1;
  d[0] = 0;             //this is not used
  outgain = 1.0;


};

AnalogFilter::~AnalogFilter ()
{
};

void
AnalogFilter::cleanup ()
{
  for (int i = 0; i < MAX_FILTER_STAGES + 1; i++)
    {
      x[i].c1 = 0.0;
      x[i].c2 = 0.0;
      y[i].c1 = 0.0;
      y[i].c2 = 0.0;
      oldx[i] = x[i];
      oldy[i] = y[i];
    };
  needsinterpolation = 0;


};

void
AnalogFilter::computefiltercoefs ()
{
  REALTYPE tmp;
  REALTYPE omega, sn, cs, alpha, beta;
  int zerocoefs = 0;          //this is used if the freq is too high

  //do not allow frequencies bigger than samplerate/2
  REALTYPE freq = this->freq;
  if (freq > (SAMPLE_RATE / 2 - 500.0))
    {
      freq = (float)SAMPLE_RATE * .5f - 500.0f;
      zerocoefs = 1;
    };
  if (freq < 0.1)
    freq = 0.1f;
  //do not allow bogus Q
  if (q < 0.0)
    q = 0.0;
  REALTYPE tmpq, tmpgain;
  if (stages == 0)
    {
      tmpq = q;
      tmpgain = gain;
    }
  else
    {
      tmpq = (q > 1.0 ? powf (q, 1.0f /(float) (stages + 1)) : q);
      tmpgain = powf (gain, 1.0f / (float)(stages + 1));
    };

  //most of theese are implementations of 
  //the "Cookbook formulae for audio EQ" by Robert Bristow-Johnson
  //The original location of the Cookbook is:
  //http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt
  switch (type)
    {
    case 0:             //LPF 1 pole         
      if (zerocoefs == 0)
      tmp = expf (-D_PI * freq / (float)SAMPLE_RATE);
      else
      tmp = 0.0f;
      c[0] = 1.0f - tmp;
      c[1] = 0.0;
      c[2] = 0.0;
      d[1] = tmp;
      d[2] = 0.0;
      order = 1;
      break;
    case 1:             //HPF 1 pole
      if (zerocoefs == 0)
      tmp = expf (-D_PI * freq / (float)SAMPLE_RATE);
      else
      tmp = 0.0f;
      c[0] = (1.0f + tmp) * .5f;
      c[1] = -(1.0f + tmp) * .5f;
      c[2] = 0.0f;
      d[1] = tmp;
      d[2] = 0.0f;
      order = 1;
      break;
    case 2:             //LPF 2 poles 
      if (zerocoefs == 0)
      {
        omega = D_PI * freq / (float)SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        alpha = sn / (2.0f * tmpq);
        tmp = 1 + alpha;
        c[0] = (1.0f - cs) * .5f / tmp;
        c[1] = (1.0f - cs) / tmp;
        c[2] = (1.0f - cs) * .5f / tmp;
        d[1] = -2.0f * cs / tmp * (-1.0f);
        d[2] = (1.0f - alpha) / tmp * (-1.0f);
      }
      else
      {
        c[0] = 1.0;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    case 3:             //HPF 2 poles 
      if (zerocoefs == 0)
      {
        omega =  D_PI * freq / (float)SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        alpha = sn / (2.0f * tmpq);
        tmp = 1.0f + alpha;
        c[0] = (1.0f + cs) / 2.0f / tmp;
        c[1] = -(1.0f + cs) / tmp;
        c[2] = (1.0f + cs) / 2.0f / tmp;
        d[1] = -2.0f * cs / tmp * (-1.0f);
        d[2] = (1.0f - alpha) / tmp * (-1.0f);
      }
      else
      {
        c[0] = 0.0;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    case 4:             //BPF 2 poles 
      if (zerocoefs == 0)
      {
        omega = D_PI * freq / (float)SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        alpha = sn / (2.0f * tmpq);
        tmp = 1.0f + alpha;
        c[0] = alpha / tmp * sqrtf (tmpq + 1.0f);
        c[1] = 0.0f;
        c[2] = -alpha / tmp * sqrtf (tmpq + 1.0f);
        d[1] = -2.0f * cs / tmp * (-1.0f);
        d[2] = (1.0f - alpha) / tmp * (-1.0f);
      }
      else
      {
        c[0] = 0.0;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    case 5:             //NOTCH 2 poles 
      if (zerocoefs == 0)
      {
        omega = D_PI * freq /(float) SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        alpha = sn / (2.0f * sqrtf (tmpq));
        tmp = 1.0f + alpha;
        c[0] = 1.0f / tmp;
        c[1] = -2.0f * cs / tmp;
        c[2] = 1.0f / tmp;
        d[1] = -2.0f * cs / tmp * (-1.0f);
        d[2] = (1.0f - alpha) / tmp * (-1.0f);
      }
      else
      {
        c[0] = 1.0;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    case 6:             //PEAK (2 poles)
      if (zerocoefs == 0)
      {
        omega = D_PI * freq / (float)SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        tmpq *= 3.0f;
        alpha = sn / (2.0f * tmpq);
        tmp = 1.0f + alpha / tmpgain;
        c[0] = (1.0f + alpha * tmpgain) / tmp;
        c[1] = (-2.0f * cs) / tmp;
        c[2] = (1.0f - alpha * tmpgain) / tmp;
        d[1] = -2.0f * cs / tmp * (-1.0f);
        d[2] = (1.0f - alpha / tmpgain) / tmp * (-1.0f);
      }
      else
      {
        c[0] = 1.0;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    case 7:             //Low Shelf - 2 poles
      if (zerocoefs == 0)
      {
        omega = D_PI * freq / (float)SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        tmpq = sqrtf (tmpq);
        alpha = sn / (2.0f * tmpq);
        beta = sqrtf (tmpgain) / tmpq;
        tmp = (tmpgain + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn;

        c[0] =
          tmpgain * ((tmpgain + 1.0f) - (tmpgain - 1.0f) * cs +
                   beta * sn) / tmp;
        c[1] =
          2.0f * tmpgain * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp;
        c[2] =
          tmpgain * ((tmpgain + 1.0f) - (tmpgain - 1.0f) * cs -
                   beta * sn) / tmp;
        d[1] = -2.0f * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp * (-1.0f);
        d[2] =
          ((tmpgain + 1.0f) + (tmpgain - 1.0f) * cs - beta * sn) / tmp * (-1.0f);
      }
      else
      {
        c[0] = tmpgain;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    case 8:             //High Shelf - 2 poles
      if (zerocoefs == 0)
      {
        omega = D_PI * freq / (float)SAMPLE_RATE;
        sn = sinf (omega);
        cs = cosf (omega);
        tmpq = sqrtf (tmpq);
        alpha = sn / (2.0f * tmpq);
        beta = sqrtf (tmpgain) / tmpq;
        tmp = (tmpgain + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn;

        c[0] =
          tmpgain * ((tmpgain + 1.0f) + (tmpgain - 1.0f) * cs +
                   beta * sn) / tmp;
        c[1] =
          -2.0f * tmpgain * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp;
        c[2] =
          tmpgain * ((tmpgain + 1.0f) + (tmpgain - 1.0f) * cs -
                   beta * sn) / tmp;
        d[1] = 2.0f * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp * (-1.0f);
        d[2] =
          ((tmpgain + 1.0f) - (tmpgain - 1.0f) * cs - beta * sn) / tmp * (-1.0f);
      }
      else
      {
        c[0] = 1.0;
        c[1] = 0.0;
        c[2] = 0.0;
        d[1] = 0.0;
        d[2] = 0.0;
      };
      order = 2;
      break;
    default:                  //wrong type
      type = 0;
      computefiltercoefs ();
      break;
    };
};


void
AnalogFilter::setfreq (REALTYPE frequency)
{
  if (frequency < 0.1)
    frequency = 0.1f;
  REALTYPE rap = freq / frequency;
  if (rap < 1.0)
    rap = 1.0f / rap;

  oldabovenq = abovenq;
  abovenq = frequency > (SAMPLE_RATE / 2 - 500.0);

  int nyquistthresh = (abovenq ^ oldabovenq);


  if ((rap > 3.0) || (nyquistthresh != 0))
    {                   //if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
      for (int i = 0; i < 3; i++)
      {
        oldc[i] = c[i];
        oldd[i] = d[i];
      };
      for (int i = 0; i < MAX_FILTER_STAGES + 1; i++)
      {
        oldx[i] = x[i];
        oldy[i] = y[i];
      };
      if (firsttime == 0)
      needsinterpolation = 1;
    };
  freq = frequency;
  computefiltercoefs ();
  firsttime = 0;

};

void
AnalogFilter::setfreq_and_q (REALTYPE frequency, REALTYPE q_)
{
  q = q_;
  setfreq (frequency);
};

void
AnalogFilter::setq (REALTYPE q_)
{
  q = q_;
  computefiltercoefs ();
};

void
AnalogFilter::settype (int type_)
{
  type = type_;
  computefiltercoefs ();
};

void
AnalogFilter::setgain (REALTYPE dBgain)
{
  gain = dB2rap(dBgain);
  computefiltercoefs ();
};

void
AnalogFilter::setstages (int stages_)
{
  if (stages_ >= MAX_FILTER_STAGES)
    stages_ = MAX_FILTER_STAGES - 1;
  stages = stages_;
  cleanup ();
  computefiltercoefs ();
};

void
AnalogFilter::singlefilterout (REALTYPE * smp, fstage & x, fstage & y,
                         REALTYPE * c, REALTYPE * d)
{
  int i;
  REALTYPE y0;
  if (order == 1)
    {                   //First order filter
      for (i = 0; i < PERIOD; i++)
      {

        y0 = smp[i] * c[0] + x.c1 * c[1] + y.c1 * d[1];
        y.c1 = y0;
        x.c1 = smp[i];
        //output
        smp[i] = y0 + DENORMAL_GUARD;
      };
    };
  if (order == 2)
    {                   //Second order filter
      for (i = 0; i < PERIOD; i++)
      {
        y0 =
          (smp[i] * c[0]) + (x.c1 * c[1]) + (x.c2 * c[2]) + (y.c1 * d[1]) +
          (y.c2 * d[2]);
        y.c2 = y.c1;
        y.c1 = y0;
        x.c2 = x.c1;
        x.c1 = smp[i];
        //output
        smp[i] = y0 + DENORMAL_GUARD;
      };
    };
};

void
AnalogFilter::filterout (REALTYPE * smp)
{
  REALTYPE *ismp = NULL;      //used if it needs interpolation
  int i;
  if (needsinterpolation != 0)
    {
      ismp = new REALTYPE[PERIOD];
      for (i = 0; i < PERIOD; i++)
      ismp[i] = smp[i];
      for (i = 0; i < stages + 1; i++)
      singlefilterout (ismp, oldx[i], oldy[i], oldc, oldd);
    };

  for (i = 0; i < stages + 1; i++)
    singlefilterout (smp, x[i], y[i], c, d);

  if (needsinterpolation != 0)
    {
      for (i = 0; i < PERIOD; i++)
      {
        REALTYPE x = (float) i / (float) PERIOD;
        smp[i] = ismp[i] * (1.0f - x) + smp[i] * x;
      };
      delete (ismp);
      needsinterpolation = 0;
    };

};

REALTYPE AnalogFilter::H (REALTYPE freq)
{
  REALTYPE
    fr = freq / (float)SAMPLE_RATE * PI * 2.0f;
  REALTYPE
    x = c[0], y = 0.0;
  for (int n = 1; n < 3; n++)
    {
      x += cosf ((float)n * fr) * c[n];
      y -= sinf ((float)n * fr) * c[n];
    };
  REALTYPE
    h = x * x + y * y;
  x = 1.0;
  y = 0.0;
  for (int n = 1; n < 3; n++)
    {
      x -= cosf ((float)n * fr) * d[n];
      y += sinf ((float)n * fr) * d[n];
    };
  h = h / (x * x + y * y);
  return (powf (h, (float)(stages + 1) / 2.0f));
};



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