/
PhaserPatch.hpp
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/
PhaserPatch.hpp
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////////////////////////////////////////////////////////////////////////////////////////////////////
/*
LICENSE:
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 3 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, see <http://www.gnu.org/licenses/>.
*/
/*
class: Phaser
implemented by: Ross Bencina <rossb@kagi.com>
date: 24/8/98
Phaser is a six stage phase shifter, intended to reproduce the
sound of a traditional analogue phaser effect.
This implementation uses six first order all-pass filters in
series, with delay time modulated by a sinusoidal.
This implementation was created to be clear, not efficient.
Obvious modifications include using a table lookup for the lfo,
not updating the filter delay times every sample, and not
tuning all of the filters to the same delay time.
Thanks to:
The nice folks on the music-dsp mailing list, including...
Chris Towsend and Marc Lindahl
...and Scott Lehman's Phase Shifting page at harmony central:
http://www.harmony-central.com/Effects/Articles/Phase_Shifting/
*/
////////////////////////////////////////////////////////////////////////////////////////////////////
#ifndef __PhaserPatch_hpp__
#define __PhaserPatch_hpp__
#include "StompBox.h"
class PhaserPatch : public Patch {
public:
//initialise to some usefull defaults...
PhaserPatch() : _lfoPhase( 0.f ), depth( 1.f ), feedback( .7f ), _zm1( 0.f ){
registerParameter(PARAMETER_A, "Rate");
registerParameter(PARAMETER_B, "Depth");
registerParameter(PARAMETER_C, "Feedback");
registerParameter(PARAMETER_D, "");
Range( 440.f, 1600.f );
Rate( .5f );
}
void Range( float fMin, float fMax ){ // Hz
_dmin = fMin / (getSampleRate()/2.f);
_dmax = fMax / (getSampleRate()/2.f);
}
float Rate( float rate ){ // cps
_lfoInc = 2.f * M_PI * (rate / getSampleRate());
return _lfoInc * 1000.f;
}
void processAudio(AudioBuffer &buffer) {
int size = buffer.getSize();
float y;
rate = Rate(getParameterValue(PARAMETER_A));
depth = getParameterValue(PARAMETER_B);
feedback = getParameterValue(PARAMETER_C);
//calculate and update phaser sweep lfo...
float d = _dmin + (_dmax-_dmin) * ((sin( _lfoPhase ) + 1.f)/2.f);
_lfoPhase += rate;
if( _lfoPhase >= M_PI * 2.f )
_lfoPhase -= M_PI * 2.f;
//update filter coeffs
for( int i=0; i<6; i++ )
_alps[i].Delay( d );
// for (int ch = 0; ch<buffer.getChannels(); ++ch) {
float* buf = buffer.getSamples(0);
for (int i = 0; i < size; i++) {
//calculate output
y = _alps[0].Update(_alps[1].Update(_alps[2].Update(_alps[3].Update(_alps[4].Update(
_alps[5].Update( buf[i] + _zm1 * feedback ))))));
_zm1 = y;
buf[i] = buf[i] + y * depth;
// }
}
}
private:
class AllpassDelay{
public:
AllpassDelay()
: _a1( 0.f )
, _zm1( 0.f )
{}
void Delay( float delay ){ //sample delay time
_a1 = (1.f - delay) / (1.f + delay);
}
float Update( float inSamp ){
float y = inSamp * -_a1 + _zm1;
_zm1 = y * _a1 + inSamp;
return y;
}
private:
float _a1, _zm1;
};
AllpassDelay _alps[6];
float _dmin, _dmax; //range
float _lfoPhase;
float _lfoInc;
float depth, rate, feedback;
float _zm1;
};
#endif /* __PhaserPatch_hpp__ */