/* ========================================
* VoiceTrick - VoiceTrick.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __VoiceTrick_H
#include "VoiceTrick.h"
#endif
void VoiceTrick::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
lowpassChase = pow(A,2);
//should not scale with sample rate, because values reaching 1 are important
//to its ability to bypass when set to max
double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0);
lastLowpass = lowpassChase;
double invLowpass;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37;
if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37;
lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount;
//setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control,
//but if I say it's the lowpass out of Capacitor it should literally be that in every behavior.
long double inputSample = (inputSampleL + inputSampleR) * 0.5;
//this is now our mono audio
count++; if (count > 5) count = 0; switch (count)
{
case 0:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
break;
case 1:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
break;
case 2:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
break;
case 3:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
break;
case 4:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
break;
case 5:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
break;
}
//Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively
//steepens the filter after minimizing artifacts.
inputSampleL = -inputSample;
inputSampleR = inputSample;
//and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone.
//The purpose of all this is to allow for recording of lead vocals without use of headphones:
//or at least sealed headphones. You should be able to use this to record vocals with either
//open-back headphones, or literally speakers in the room so long as the mic is exactly
//equidistant from each speaker/headphone side.
//You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor
//only reverb and echo for vibe. Direct sound is the singer's direct sound.
//The filtering is because, if you use open-back headphones and move your head, highs will
//bleed through first like a through-zero flange coming out of cancellation (which it is).
//Therefore, you can filter off highs until the bleed isn't annoying.
//Or just run with it, it shouldn't be that loud.
//Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :)
//begin 32 bit stereo floating point dither
int expon; frexpf((float)inputSampleL, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
frexpf((float)inputSampleR, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
//end 32 bit stereo floating point dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
void VoiceTrick::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
lowpassChase = pow(A,2);
//should not scale with sample rate, because values reaching 1 are important
//to its ability to bypass when set to max
double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0);
lastLowpass = lowpassChase;
double invLowpass;
while (--sampleFrames >= 0)
{
long double inputSampleL = *in1;
long double inputSampleR = *in2;
if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43;
if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43;
lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount;
//setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control,
//but if I say it's the lowpass out of Capacitor it should literally be that in every behavior.
long double inputSample = (inputSampleL + inputSampleR) * 0.5;
//this is now our mono audio
count++; if (count > 5) count = 0; switch (count)
{
case 0:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
break;
case 1:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
break;
case 2:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
break;
case 3:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
break;
case 4:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
break;
case 5:
iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
break;
}
//Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively
//steepens the filter after minimizing artifacts.
inputSampleL = -inputSample;
inputSampleR = inputSample;
//and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone.
//The purpose of all this is to allow for recording of lead vocals without use of headphones:
//or at least sealed headphones. You should be able to use this to record vocals with either
//open-back headphones, or literally speakers in the room so long as the mic is exactly
//equidistant from each speaker/headphone side.
//You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor
//only reverb and echo for vibe. Direct sound is the singer's direct sound.
//The filtering is because, if you use open-back headphones and move your head, highs will
//bleed through first like a through-zero flange coming out of cancellation (which it is).
//Therefore, you can filter off highs until the bleed isn't annoying.
//Or just run with it, it shouldn't be that loud.
//Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :)
//begin 64 bit stereo floating point dither
int expon; frexp((double)inputSampleL, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
frexp((double)inputSampleR, &expon);
fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
//end 64 bit stereo floating point dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
