StateVariable.h

/*
 * StateVariable.h
 *
 * This file is part of Mozzi.
 *
 * Copyright 2012-2024 Tim Barrass and the Mozzi Team
 *
 * Mozzi is licensed under the GNU Lesser General Public Licence (LGPL) Version 2.1 or later.
 *
 */
 
/**
State Variable Filter (approximation of Chamberlin version)
Informed by pseudocode at http://www.musicdsp.org/showone.php?id=23 and
http://www.musicdsp.org/showone.php?id=142. Here's the original:
---------------------
cutoff = cutoff freq in Hz
fs = sampling frequency //(e.g. 44100Hz)
f = 2 sin (pi * cutoff / fs) //[approximately]
q = resonance/bandwidth [0 < q <= 1]  most res: q=1, less: q=0
low = lowpass output
high = highpass output
band = bandpass output
notch = notch output
 
scale = q
 
low=high=band=0;
 
//--beginloop
low = low + f * band;
high = scale * input - low - q*band;
band = f * high + band;
notch = high + low;
//--endloop
----------------------
References :
Hal Chamberlin, Musical Applications of Microprocessors, 2nd Ed, Hayden Book
Company 1985. pp 490-492. Jon Dattorro, Effect Design Part 1, J. Audio Eng.
Soc., Vol 45, No. 9, 1997 September
*/
 
#ifndef STATEVARIABLE_H_
#define STATEVARIABLE_H_
 
#include "Arduino.h"
#include "math.h"
#include "meta.h"
#include "mozzi_fixmath.h"
#include "mozzi_utils.h"
#include "ResonantFilter.h"
 
//enum filter_types { LOWPASS, BANDPASS, HIGHPASS, NOTCH };
 
/** A State Variable filter which offers 12db resonant low, high, bandpass and
notch modes.
@tparam FILTER_TYPE choose between LOWPASS, BANDPASS, HIGHPASS and NOTCH.
@note To save processing time, this version of the filter does not saturate
internally, so any resonant peaks are unceremoniously truncated.  It may be
worth adding code to constrain the internal variables to enable resonant
saturating effects.
@todo Try adding code to constrain the internal variables to enable resonant
saturating effects.
*/
template <int8_t FILTER_TYPE> class StateVariable {
 
public:
  /** Constructor.
   */
  StateVariable() {}
 
  /** Set how resonant the filter will be.
  @param resonance a byte value between 1 and 255.
  The lower this value is, the more resonant the filter.
  At very low values, the filter can output loud peaks which can exceed
  Mozzi's output range, so you may need to attenuate the output in your sketch.
  @note Timing < 500 ns
  */
  void setResonance(Q0n8 resonance) {
    // qvalue goes from 255 to 0, representing .999 to 0 in fixed point
    // lower q, more resonance
    q = resonance;
    scale = (Q0n8)sqrt((unsigned int)resonance << 8);
  }
 
  /** Set the centre or corner frequency of the filter.
  @param centre_freq 20 - 4096 Hz (MOZZI_AUDIO_RATE/4).
  This will be the cut-off frequency for LOWPASS and HIGHPASS, and the
  centre frequency to pass or reduce for BANDPASS and NOTCH.
  @note Timing 25-30us
  @note The frequency calculation is VERY "approximate".  This really needs to
  be fixed.
  */
  void setCentreFreq(unsigned int centre_freq) {
    // simple frequency tuning with error towards nyquist (reference?  where did
    // this come from?)
    // f = (Q1n15)(((Q16n16_2PI*centre_freq)>>AUDIO_RATE_AS_LSHIFT)>>1);
    f = (Q15n16)((Q16n16_2PI * centre_freq) >>
                 (AUDIO_RATE_AS_LSHIFT)); // this works best for now
    // f = (Q15n16)(((Q16n16_2PI*centre_freq)<<(16-AUDIO_RATE_AS_LSHIFT))>>16);
    // // a small shift left and a round 16 right is faster than big
    // non-byte-aligned right in one go float ff =
    // Q16n16_to_float(((Q16n16_PI*centre_freq))>>AUDIO_RATE_AS_LSHIFT); f =
    // float_to_Q15n16(2.0f *sin(ff));
  }
 
  /** Calculate the next sample, given an input signal.
  @param input the signal input.
  @return the signal output.
  @note Timing: 16 - 20 us
  */
  inline int next(int input) {
    // chooses a different next() function depending on whether the
    // filter is declared as LOWPASS, BANDPASS, HIGHPASS or NOTCH.
    // See meta.h.
    return next(input, Int2Type<FILTER_TYPE>());
  }
 
private:
  int low, band;
  Q0n8 q, scale;
  volatile Q15n16 f;
 
  /** Calculate the next sample, given an input signal.
  @param in the signal input.
  @return the signal output.
  @note Timing: 16 - 20 us
  */
  inline int next(int input, Int2Type<LOWPASS>) {
    // setPin13High();
    low += ((f * band) >> 16);
    int high = (((long)input - low - (((long)band * q) >> 8)) * scale) >> 8;
    band += ((f * high) >> 16);
    // int notch = high + low;
    // setPin13Low();
    return low;
  }
 
  /** Calculate the next sample, given an input signal.
  @param input the signal input.
  @return the signal output.
  @note Timing:
  */
  inline int next(int input, Int2Type<BANDPASS>) {
    // setPin13High();
    low += ((f * band) >> 16);
    int high = (((long)input - low - (((long)band * q) >> 8)) * scale) >> 8;
    band += ((f * high) >> 16);
    // int notch = high + low;
    // setPin13Low();
    return band;
  }
 
  /** Calculate the next sample, given an input signal.
  @param input the signal input.
  @return the signal output.
  @note Timing:
  */
  inline int next(int input, Int2Type<HIGHPASS>) {
    // setPin13High();
    low += ((f * band) >> 16);
    int high = (((long)input - low - (((long)band * q) >> 8)) * scale) >> 8;
    band += ((f * high) >> 16);
    // int notch = high + low;
    // setPin13Low();
    return high;
  }
 
  /** Calculate the next sample, given an input signal.
  @param input the signal input.
  @return the signal output.
  @note Timing: 16 - 20 us
  */
  inline int next(int input, Int2Type<NOTCH>) {
    // setPin13High();
    low += ((f * band) >> 16);
    int high = (((long)input - low - (((long)band * q) >> 8)) * scale) >> 8;
    band += ((f * high) >> 16);
    int notch = high + low;
    // setPin13Low();
    return notch;
  }
};
 
/**
@example 11.Audio_Filters/StateVariableFilter/StateVariableFilter.ino
This example demonstrates the StateVariable class.
*/
 
#endif /* STATEVARIABLE_H_ */