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//===========================================================================
//
// simplechorus
//
// Version 0.0.1
//
//
//
//
// Copyright (c) 2006 Nil Geisweiller
//
//
//
// 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., 59 Temple Place - Suite 330, Boston, MA
// 02111-1307, USA or point your web browser to http://www.gnu.org.
//===========================================================================
#include "simplechorusmodel.h"
#include <math.h>
#include <stdio.h>
#define ABS(x) (x>=0?x:-x)
// Linearly interpolate [ = a * (1 - f) + b * f]
inline float lin_interp(float f, float a, float b) {
return a + f * (b - a);
}
// Cubic interpolation function
inline float cube_interp(const float fr,
const float inm1,
const float in,
const float inp1,
const float inp2) {
return in + 0.5f * fr * (inp1 - inm1 +
fr * (4.0f * inp1 + 2.0f * inm1 - 5.0f * in - inp2 +
fr * (3.0f * (in - inp1) - inm1 + inp2)));
}
float SimpleChorusModel::sinus[MAXSINUSRESOLUTION];
int SimpleChorusModel::useCount = 0;
SimpleChorusModel::SimpleChorusModel(float samplerate) {
_sampleRate = samplerate;
//sinus
if (useCount++ == 0)
for(int i = 0; i < MAXSINUSRESOLUTION; i++)
sinus[i] = (float)(sin(((double)i * 2.0 * M_PI) /
(double)MAXSINUSRESOLUTION));
_index = 0.0;
//init buffer
for(int i = 0; i < MAXBUFFERLENGTH; i++) {
_leftBuffer[i] = 0.0;
_rightBuffer[i] = 0.0;
}
_position = 0;
//initial parameters
_pan = 0.5;
_LFOFreq = 1.0;
_depth = 0.5;
setChorus();
}
SimpleChorusModel::~SimpleChorusModel() {
}
void SimpleChorusModel::process_chorus(float leftInput, float rightInput,
float* leftOutput, float* rightOutput) {
float ocsDiff;
_ocsDistance = _depthAmp * sinus[(int)_index];
ocsDiff = _ocsDistance - floorf(_ocsDistance);
_past_position_left = MAXBUFFERLENGTH //to be sure that _past_position_left>0
+ _position - _leftMidDistance + (int)_ocsDistance;
_past_position_right = MAXBUFFERLENGTH
+ _position - _rightMidDistance + (int)_ocsDistance;
*leftOutput = _leftAmp *
lin_interp(ocsDiff, _leftBuffer[_past_position_left%MAXBUFFERLENGTH],
_leftBuffer[(_past_position_left+1)%MAXBUFFERLENGTH]);
*rightOutput = _rightAmp *
lin_interp(ocsDiff, _rightBuffer[_past_position_right%MAXBUFFERLENGTH],
_rightBuffer[(_past_position_right+1)%MAXBUFFERLENGTH]);
_leftBuffer[_position] = leftInput;
_rightBuffer[_position] = rightInput;
_position++;
_position %= MAXBUFFERLENGTH;
_index += _inct;
_index = (_index<MAXSINUSRESOLUTION?_index:_index-MAXSINUSRESOLUTION);
}
void SimpleChorusModel::setPan(float p) {
_pan = p;
setChorus();
}
void SimpleChorusModel::setLFOFreq(float l) {
_LFOFreq = l;
setChorus();
}
void SimpleChorusModel::setDepth(float d) {
_depth = d;
setChorus();
}
void SimpleChorusModel::setSampleRate(float s) {
_sampleRate = s;
setChorus();
}
float SimpleChorusModel::getPan() {
return _pan;
}
float SimpleChorusModel::getLFOFreq() {
return _LFOFreq;
}
float SimpleChorusModel::getDepth() {
return _depth;
}
void SimpleChorusModel::setChorus() {
//inct
_inct = (float)MAXSINUSRESOLUTION/_sampleRate * _LFOFreq;
//left & right amp
_leftAmp = lin_interp(1.0 - _pan, 1.0 - PANAMP, 1.0 + PANAMP);
_rightAmp = lin_interp(_pan, 1.0 - PANAMP, 1.0 + PANAMP);
//left & right midDistance
float leftmdm; //left mid distance in meter
float rightmdm; //right mid distance in meter
leftmdm = MIDSOURCEDISTANCE - EARSDISTANCE * (0.5 - _pan);
rightmdm = MIDSOURCEDISTANCE + EARSDISTANCE * (0.5 - _pan);
_leftMidDistance = (int)(_sampleRate * leftmdm / SOUNDSPEED);
_rightMidDistance = (int)(_sampleRate * rightmdm / SOUNDSPEED);
//depthAmp
_depthAmp =
_sampleRate * (MAXDEPTH * _depth) /SOUNDSPEED;
//filter coef
_filterCoef1 = 1 - COEFFILTER;
_filterCoef2 = COEFFILTER;
}
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