/* ========================================
* PurestEcho - PurestEcho.h
* Copyright (c) 2016 airwindows, All rights reserved
* ======================================== */
#ifndef __PurestEcho_H
#include "PurestEcho.h"
#endif
void PurestEcho::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
{
float* in1 = inputs[0];
float* in2 = inputs[1];
float* out1 = outputs[0];
float* out2 = outputs[1];
int loopLimit = (int)(totalsamples * 0.499);
//this is a double buffer so we will be splitting it in two
double time = pow(A,2) * 0.999;
double tap1 = B;
double tap2 = C;
double tap3 = D;
double tap4 = E;
double gainTrim = 1.0 / (1.0 + tap1 + tap2 + tap3 + tap4);
//this becomes our equal-loudness mechanic. 0.2 to 1.0 gain on all things.
double tapsTrim = gainTrim * 0.5;
//the taps interpolate and require half that gain: 0.1 to 0.5 on all taps.
int position1 = (int)(loopLimit * time * 0.25);
int position2 = (int)(loopLimit * time * 0.5);
int position3 = (int)(loopLimit * time * 0.75);
int position4 = (int)(loopLimit * time);
//basic echo information: we're taking four equally spaced echoes and setting their levels as desired.
//position4 is what you'd have for 'just set a delay time'
double volAfter1 = (loopLimit * time * 0.25) - position1;
double volAfter2 = (loopLimit * time * 0.5) - position2;
double volAfter3 = (loopLimit * time * 0.75) - position3;
double volAfter4 = (loopLimit * time) - position4;
//these are 0-1: casting to an (int) truncates fractional numbers towards zero (and is faster than floor() )
//so, when we take the integer number (all above zero) and subtract it from the real value, we get 0-1
double volBefore1 = (1.0 - volAfter1) * tap1;
double volBefore2 = (1.0 - volAfter2) * tap2;
double volBefore3 = (1.0 - volAfter3) * tap3;
double volBefore4 = (1.0 - volAfter4) * tap4;
//and if we are including a bit of the previous/next sample to interpolate, then if the sample position is 1.0001
//we'll be leaning most heavily on the 'before' sample which is nearer to us, and the 'after' sample is almost not used.
//if the sample position is 1.9999, the 'after' sample is strong and 'before' is almost not used.
volAfter1 *= tap1;
volAfter2 *= tap2;
volAfter3 *= tap3;
volAfter4 *= tap4;
//and like with volBefore, we also want to scale this 'interpolate' to the loudness of this tap.
//We do it here because we can do it only once per audio buffer, not on every sample. This assumes we're
//not moving the tap every sample: if so we'd have to do this every sample as well.
int oneBefore1 = position1 - 1;
int oneBefore2 = position2 - 1;
int oneBefore3 = position3 - 1;
int oneBefore4 = position4 - 1;
if (oneBefore1 < 0) oneBefore1 = 0;
if (oneBefore2 < 0) oneBefore2 = 0;
if (oneBefore3 < 0) oneBefore3 = 0;
if (oneBefore4 < 0) oneBefore4 = 0;
int oneAfter1 = position1 + 1;
int oneAfter2 = position2 + 1;
int oneAfter3 = position3 + 1;
int oneAfter4 = position4 + 1;
//this is setting up the way we interpolate samples: we're doing an echo-darkening thing
//to make it sound better. Pretty much no acoustic delay in human-breathable air will give
//you zero attenuation at 22 kilohertz: forget this at your peril ;)
double delaysBufferL;
double delaysBufferR;
long double inputSampleL;
long double inputSampleR;
while (--sampleFrames >= 0)
{
inputSampleL = *in1;
inputSampleR = *in2;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
if (gcount < 0 || gcount > loopLimit) gcount = loopLimit;
dL[gcount+loopLimit] = dL[gcount] = inputSampleL * tapsTrim;
dR[gcount+loopLimit] = dR[gcount] = inputSampleR * tapsTrim; //this is how the double buffer works:
//we can look for delay taps without ever having to 'wrap around' within our calculation.
//As long as the delay tap is less than our loop limit we can always just add it to where we're
//at, and get a valid sample back right away, no matter where we are in the buffer.
//The 0.5 is taking into account the interpolation, by padding down the whole buffer.
delaysBufferL = (dL[gcount+oneBefore4]*volBefore4);
delaysBufferL += (dL[gcount+oneAfter4]*volAfter4);
delaysBufferL += (dL[gcount+oneBefore3]*volBefore3);
delaysBufferL += (dL[gcount+oneAfter3]*volAfter3);
delaysBufferL += (dL[gcount+oneBefore2]*volBefore2);
delaysBufferL += (dL[gcount+oneAfter2]*volAfter2);
delaysBufferL += (dL[gcount+oneBefore1]*volBefore1);
delaysBufferL += (dL[gcount+oneAfter1]*volAfter1);
delaysBufferR = (dR[gcount+oneBefore4]*volBefore4);
delaysBufferR += (dR[gcount+oneAfter4]*volAfter4);
delaysBufferR += (dR[gcount+oneBefore3]*volBefore3);
delaysBufferR += (dR[gcount+oneAfter3]*volAfter3);
delaysBufferR += (dR[gcount+oneBefore2]*volBefore2);
delaysBufferR += (dR[gcount+oneAfter2]*volAfter2);
delaysBufferR += (dR[gcount+oneBefore1]*volBefore1);
delaysBufferR += (dR[gcount+oneAfter1]*volAfter1);
//These are the interpolated samples. We're adding them first, because we know they're smaller
//and while the value of delaysBuffer is small we'll add similarly small values to it. Note the order.
delaysBufferL += (dL[gcount+position4]*tap4);
delaysBufferL += (dL[gcount+position3]*tap3);
delaysBufferL += (dL[gcount+position2]*tap2);
delaysBufferL += (dL[gcount+position1]*tap1);
delaysBufferR += (dR[gcount+position4]*tap4);
delaysBufferR += (dR[gcount+position3]*tap3);
delaysBufferR += (dR[gcount+position2]*tap2);
delaysBufferR += (dR[gcount+position1]*tap1);
//These are the primary samples for the echo, and we're adding them last. As before we're starting with the
//most delayed echoes, and ending with what we think might be the loudest echo. We're building this delaybuffer
//from the faintest noises to the loudest, to avoid adding a bunch of teeny values at the end.
//You can of course put the last echo as loudest, but with diminishing echo volumes this is optimal.
//This technique is also present in other plugins such as Iron Oxide.
inputSampleL = (inputSampleL * gainTrim) + delaysBufferL;
inputSampleR = (inputSampleR * gainTrim) + delaysBufferR;
//this could be just inputSample += d[gcount+position1];
//for literally a single, full volume echo combined with dry.
//What I'm doing is making the echoes more interesting.
gcount--;
//stereo 32 bit dither, made small and tidy.
int expon; frexpf((float)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexpf((float)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 32 bit dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
void PurestEcho::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
{
double* in1 = inputs[0];
double* in2 = inputs[1];
double* out1 = outputs[0];
double* out2 = outputs[1];
int loopLimit = (int)(totalsamples * 0.499);
//this is a double buffer so we will be splitting it in two
double time = pow(A,2) * 0.999;
double tap1 = B;
double tap2 = C;
double tap3 = D;
double tap4 = E;
double gainTrim = 1.0 / (1.0 + tap1 + tap2 + tap3 + tap4);
//this becomes our equal-loudness mechanic. 0.2 to 1.0 gain on all things.
double tapsTrim = gainTrim * 0.5;
//the taps interpolate and require half that gain: 0.1 to 0.5 on all taps.
int position1 = (int)(loopLimit * time * 0.25);
int position2 = (int)(loopLimit * time * 0.5);
int position3 = (int)(loopLimit * time * 0.75);
int position4 = (int)(loopLimit * time);
//basic echo information: we're taking four equally spaced echoes and setting their levels as desired.
//position4 is what you'd have for 'just set a delay time'
double volAfter1 = (loopLimit * time * 0.25) - position1;
double volAfter2 = (loopLimit * time * 0.5) - position2;
double volAfter3 = (loopLimit * time * 0.75) - position3;
double volAfter4 = (loopLimit * time) - position4;
//these are 0-1: casting to an (int) truncates fractional numbers towards zero (and is faster than floor() )
//so, when we take the integer number (all above zero) and subtract it from the real value, we get 0-1
double volBefore1 = (1.0 - volAfter1) * tap1;
double volBefore2 = (1.0 - volAfter2) * tap2;
double volBefore3 = (1.0 - volAfter3) * tap3;
double volBefore4 = (1.0 - volAfter4) * tap4;
//and if we are including a bit of the previous/next sample to interpolate, then if the sample position is 1.0001
//we'll be leaning most heavily on the 'before' sample which is nearer to us, and the 'after' sample is almost not used.
//if the sample position is 1.9999, the 'after' sample is strong and 'before' is almost not used.
volAfter1 *= tap1;
volAfter2 *= tap2;
volAfter3 *= tap3;
volAfter4 *= tap4;
//and like with volBefore, we also want to scale this 'interpolate' to the loudness of this tap.
//We do it here because we can do it only once per audio buffer, not on every sample. This assumes we're
//not moving the tap every sample: if so we'd have to do this every sample as well.
int oneBefore1 = position1 - 1;
int oneBefore2 = position2 - 1;
int oneBefore3 = position3 - 1;
int oneBefore4 = position4 - 1;
if (oneBefore1 < 0) oneBefore1 = 0;
if (oneBefore2 < 0) oneBefore2 = 0;
if (oneBefore3 < 0) oneBefore3 = 0;
if (oneBefore4 < 0) oneBefore4 = 0;
int oneAfter1 = position1 + 1;
int oneAfter2 = position2 + 1;
int oneAfter3 = position3 + 1;
int oneAfter4 = position4 + 1;
//this is setting up the way we interpolate samples: we're doing an echo-darkening thing
//to make it sound better. Pretty much no acoustic delay in human-breathable air will give
//you zero attenuation at 22 kilohertz: forget this at your peril ;)
double delaysBufferL;
double delaysBufferR;
long double inputSampleL;
long double inputSampleR;
while (--sampleFrames >= 0)
{
inputSampleL = *in1;
inputSampleR = *in2;
if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) {
static int noisesource = 0;
//this declares a variable before anything else is compiled. It won't keep assigning
//it to 0 for every sample, it's as if the declaration doesn't exist in this context,
//but it lets me add this denormalization fix in a single place rather than updating
//it in three different locations. The variable isn't thread-safe but this is only
//a random seed and we can share it with whatever.
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleL = applyresidue;
}
if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) {
static int noisesource = 0;
noisesource = noisesource % 1700021; noisesource++;
int residue = noisesource * noisesource;
residue = residue % 170003; residue *= residue;
residue = residue % 17011; residue *= residue;
residue = residue % 1709; residue *= residue;
residue = residue % 173; residue *= residue;
residue = residue % 17;
double applyresidue = residue;
applyresidue *= 0.00000001;
applyresidue *= 0.00000001;
inputSampleR = applyresidue;
//this denormalization routine produces a white noise at -300 dB which the noise
//shaping will interact with to produce a bipolar output, but the noise is actually
//all positive. That should stop any variables from going denormal, and the routine
//only kicks in if digital black is input. As a final touch, if you save to 24-bit
//the silence will return to being digital black again.
}
if (gcount < 0 || gcount > loopLimit) gcount = loopLimit;
dL[gcount+loopLimit] = dL[gcount] = inputSampleL * tapsTrim;
dR[gcount+loopLimit] = dR[gcount] = inputSampleR * tapsTrim; //this is how the double buffer works:
//we can look for delay taps without ever having to 'wrap around' within our calculation.
//As long as the delay tap is less than our loop limit we can always just add it to where we're
//at, and get a valid sample back right away, no matter where we are in the buffer.
//The 0.5 is taking into account the interpolation, by padding down the whole buffer.
delaysBufferL = (dL[gcount+oneBefore4]*volBefore4);
delaysBufferL += (dL[gcount+oneAfter4]*volAfter4);
delaysBufferL += (dL[gcount+oneBefore3]*volBefore3);
delaysBufferL += (dL[gcount+oneAfter3]*volAfter3);
delaysBufferL += (dL[gcount+oneBefore2]*volBefore2);
delaysBufferL += (dL[gcount+oneAfter2]*volAfter2);
delaysBufferL += (dL[gcount+oneBefore1]*volBefore1);
delaysBufferL += (dL[gcount+oneAfter1]*volAfter1);
delaysBufferR = (dR[gcount+oneBefore4]*volBefore4);
delaysBufferR += (dR[gcount+oneAfter4]*volAfter4);
delaysBufferR += (dR[gcount+oneBefore3]*volBefore3);
delaysBufferR += (dR[gcount+oneAfter3]*volAfter3);
delaysBufferR += (dR[gcount+oneBefore2]*volBefore2);
delaysBufferR += (dR[gcount+oneAfter2]*volAfter2);
delaysBufferR += (dR[gcount+oneBefore1]*volBefore1);
delaysBufferR += (dR[gcount+oneAfter1]*volAfter1);
//These are the interpolated samples. We're adding them first, because we know they're smaller
//and while the value of delaysBuffer is small we'll add similarly small values to it. Note the order.
delaysBufferL += (dL[gcount+position4]*tap4);
delaysBufferL += (dL[gcount+position3]*tap3);
delaysBufferL += (dL[gcount+position2]*tap2);
delaysBufferL += (dL[gcount+position1]*tap1);
delaysBufferR += (dR[gcount+position4]*tap4);
delaysBufferR += (dR[gcount+position3]*tap3);
delaysBufferR += (dR[gcount+position2]*tap2);
delaysBufferR += (dR[gcount+position1]*tap1);
//These are the primary samples for the echo, and we're adding them last. As before we're starting with the
//most delayed echoes, and ending with what we think might be the loudest echo. We're building this delaybuffer
//from the faintest noises to the loudest, to avoid adding a bunch of teeny values at the end.
//You can of course put the last echo as loudest, but with diminishing echo volumes this is optimal.
//This technique is also present in other plugins such as Iron Oxide.
inputSampleL = (inputSampleL * gainTrim) + delaysBufferL;
inputSampleR = (inputSampleR * gainTrim) + delaysBufferR;
//this could be just inputSample += d[gcount+position1];
//for literally a single, full volume echo combined with dry.
//What I'm doing is making the echoes more interesting.
gcount--;
//stereo 64 bit dither, made small and tidy.
int expon; frexp((double)inputSampleL, &expon);
long double dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleL += (dither-fpNShapeL); fpNShapeL = dither;
frexp((double)inputSampleR, &expon);
dither = (rand()/(RAND_MAX*7.737125245533627e+25))*pow(2,expon+62);
dither /= 536870912.0; //needs this to scale to 64 bit zone
inputSampleR += (dither-fpNShapeR); fpNShapeR = dither;
//end 64 bit dither
*out1 = inputSampleL;
*out2 = inputSampleR;
*in1++;
*in2++;
*out1++;
*out2++;
}
}
