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Diffstat (limited to 'plugins/WinVST/Crystal/CrystalProc.cpp')
| -rwxr-xr-x | plugins/WinVST/Crystal/CrystalProc.cpp | 434 |
1 files changed, 434 insertions, 0 deletions
diff --git a/plugins/WinVST/Crystal/CrystalProc.cpp b/plugins/WinVST/Crystal/CrystalProc.cpp new file mode 100755 index 0000000..9c62355 --- /dev/null +++ b/plugins/WinVST/Crystal/CrystalProc.cpp @@ -0,0 +1,434 @@ +/* ======================================== + * Crystal - Crystal.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Crystal_H +#include "Crystal.h" +#endif + +void Crystal::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double threshold = A; + double hardness; + double breakup = (1.0-(threshold/2.0))*3.14159265358979; + double bridgerectifier; + double sqdrive = B; + if (sqdrive > 1.0) sqdrive *= sqdrive; + sqdrive = sqrt(sqdrive); + double indrive = C*3.0; + if (indrive > 1.0) indrive *= indrive; + indrive *= (1.0-(0.1695*sqdrive)); + //no gain loss of convolution for APIcolypse + //calibrate this to match noise level with character at 1.0 + //you get for instance 0.819 and 1.0-0.819 is 0.181 + double randy; + double outlevel = D; + + if (threshold < 1) hardness = 1.0 / (1.0-threshold); + else hardness = 999999999999999999999.0; + //set up hardness to exactly fill gap between threshold and 0db + //if threshold is literally 1 then hardness is infinite, so we make it very big + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + static int noisesourceL = 0; + static int noisesourceR = 850010; + int residue; + double applyresidue; + + noisesourceL = noisesourceL % 1700021; noisesourceL++; + residue = noisesourceL * noisesourceL; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL += applyresidue; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + inputSampleL -= applyresidue; + } + + noisesourceR = noisesourceR % 1700021; noisesourceR++; + residue = noisesourceR * noisesourceR; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR += applyresidue; + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + inputSampleR -= applyresidue; + } + //for live air, we always apply the dither noise. Then, if our result is + //effectively digital black, we'll subtract it again. We want a 'air' hiss + inputSampleL *= indrive; + inputSampleR *= indrive; + + //calibrated to match gain through convolution and -0.3 correction + if (sqdrive > 0.0){ + bL[23] = bL[22]; bL[22] = bL[21]; bL[21] = bL[20]; bL[20] = bL[19]; bL[19] = bL[18]; bL[18] = bL[17]; bL[17] = bL[16]; bL[16] = bL[15]; + bL[15] = bL[14]; bL[14] = bL[13]; bL[13] = bL[12]; bL[12] = bL[11]; bL[11] = bL[10]; bL[10] = bL[9]; bL[9] = bL[8]; bL[8] = bL[7]; + bL[7] = bL[6]; bL[6] = bL[5]; bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; bL[1] = bL[0]; bL[0] = inputSampleL * sqdrive; + inputSampleL += (bL[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bL[1])))); + inputSampleL -= (bL[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bL[2])))); + inputSampleL += (bL[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bL[3])))); + inputSampleL -= (bL[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bL[4])))); + inputSampleL += (bL[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bL[5])))); + inputSampleL -= (bL[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bL[6])))); + inputSampleL += (bL[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bL[7])))); + inputSampleL -= (bL[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bL[8])))); + inputSampleL += (bL[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bL[9])))); + inputSampleL -= (bL[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bL[10])))); + inputSampleL += (bL[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bL[11])))); + inputSampleL -= (bL[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bL[12])))); + inputSampleL += (bL[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bL[13])))); + inputSampleL -= (bL[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bL[14])))); + inputSampleL += (bL[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bL[15])))); + inputSampleL -= (bL[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bL[16])))); + inputSampleL += (bL[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bL[17])))); + inputSampleL -= (bL[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bL[18])))); + inputSampleL += (bL[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bL[19])))); + inputSampleL -= (bL[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bL[20])))); + inputSampleL += (bL[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bL[21])))); + inputSampleL -= (bL[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bL[22])))); + inputSampleL += (bL[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bL[23])))); + } //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce + // more of a consistent variation between soft and loud convolutions. For years I thought this code was a + //mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single + //convolution kernel location: this will be true of the Character individual plugins as well. + + //calibrated to match gain through convolution and -0.3 correction + if (sqdrive > 0.0){ + bR[23] = bR[22]; bR[22] = bR[21]; bR[21] = bR[20]; bR[20] = bR[19]; bR[19] = bR[18]; bR[18] = bR[17]; bR[17] = bR[16]; bR[16] = bR[15]; + bR[15] = bR[14]; bR[14] = bR[13]; bR[13] = bR[12]; bR[12] = bR[11]; bR[11] = bR[10]; bR[10] = bR[9]; bR[9] = bR[8]; bR[8] = bR[7]; + bR[7] = bR[6]; bR[6] = bR[5]; bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; bR[1] = bR[0]; bR[0] = inputSampleR * sqdrive; + inputSampleR += (bR[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bR[1])))); + inputSampleR -= (bR[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bR[2])))); + inputSampleR += (bR[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bR[3])))); + inputSampleR -= (bR[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bR[4])))); + inputSampleR += (bR[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bR[5])))); + inputSampleR -= (bR[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bR[6])))); + inputSampleR += (bR[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bR[7])))); + inputSampleR -= (bR[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bR[8])))); + inputSampleR += (bR[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bR[9])))); + inputSampleR -= (bR[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bR[10])))); + inputSampleR += (bR[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bR[11])))); + inputSampleR -= (bR[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bR[12])))); + inputSampleR += (bR[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bR[13])))); + inputSampleR -= (bR[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bR[14])))); + inputSampleR += (bR[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bR[15])))); + inputSampleR -= (bR[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bR[16])))); + inputSampleR += (bR[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bR[17])))); + inputSampleR -= (bR[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bR[18])))); + inputSampleR += (bR[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bR[19])))); + inputSampleR -= (bR[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bR[20])))); + inputSampleR += (bR[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bR[21])))); + inputSampleR -= (bR[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bR[22])))); + inputSampleR += (bR[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bR[23])))); + } //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce + // more of a consistent variation between soft and loud convolutions. For years I thought this code was a + //mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single + //convolution kernel location: this will be true of the Character individual plugins as well. + + if (fabs(inputSampleL) > threshold) + { + bridgerectifier = (fabs(inputSampleL)-threshold)*hardness; + //skip flat area if any, scale to distortion limit + if (bridgerectifier > breakup) bridgerectifier = breakup; + //max value for sine function, 'breakup' modeling for trashed console tone + //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag + bridgerectifier = sin(bridgerectifier)/hardness; + //do the sine factor, scale back to proper amount + if (inputSampleL > 0) inputSampleL = bridgerectifier+threshold; + else inputSampleL = -(bridgerectifier+threshold); + } //otherwise we leave it untouched by the overdrive stuff + //this is the notorious New Channel Density algorithm. It's much less popular than the original Density, + //because it introduces a point where the saturation 'curve' changes from straight to curved. + //People don't like these discontinuities, but you can use them for effect or to grit up the sound. + + if (fabs(inputSampleR) > threshold) + { + bridgerectifier = (fabs(inputSampleR)-threshold)*hardness; + //skip flat area if any, scale to distortion limit + if (bridgerectifier > breakup) bridgerectifier = breakup; + //max value for sine function, 'breakup' modeling for trashed console tone + //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag + bridgerectifier = sin(bridgerectifier)/hardness; + //do the sine factor, scale back to proper amount + if (inputSampleR > 0) inputSampleR = bridgerectifier+threshold; + else inputSampleR = -(bridgerectifier+threshold); + } //otherwise we leave it untouched by the overdrive stuff + //this is the notorious New Channel Density algorithm. It's much less popular than the original Density, + //because it introduces a point where the saturation 'curve' changes from straight to curved. + //People don't like these discontinuities, but you can use them for effect or to grit up the sound. + + randy = ((rand()/(double)RAND_MAX)*0.022); + bridgerectifier = ((inputSampleL*(1-randy))+(lastSampleL*randy)) * outlevel; + lastSampleL = inputSampleL; + inputSampleL = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal. + + randy = ((rand()/(double)RAND_MAX)*0.022); + bridgerectifier = ((inputSampleR*(1-randy))+(lastSampleR*randy)) * outlevel; + lastSampleR = inputSampleR; + inputSampleR = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal. + + //This is akin to the old Chrome Oxide plugin, applying a fuzz to only the slews. The noise only appears + //when current and old samples are different from each other, otherwise you can't tell it's there. + //This is not only during silence but the tops of low frequency waves: it scales down to affect lows more gently. + + //noise shaping to 32-bit floating point + float fpTemp = inputSampleL; + fpNShapeL += (inputSampleL-fpTemp); + inputSampleL += fpNShapeL; + //if this confuses you look at the wordlength for fpTemp :) + fpTemp = inputSampleR; + fpNShapeR += (inputSampleR-fpTemp); + inputSampleR += fpNShapeR; + //for deeper space and warmth, we try a non-oscillating noise shaping + //that is kind of ruthless: it will forever retain the rounding errors + //except we'll dial it back a hair at the end of every buffer processed + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } + fpNShapeL *= 0.999999; + fpNShapeR *= 0.999999; + //we will just delicately dial back the FP noise shaping, not even every sample + //this is a good place to put subtle 'no runaway' calculations, though bear in mind + //that it will be called more often when you use shorter sample buffers in the DAW. + //So, very low latency operation will call these calculations more often. +} + +void Crystal::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double threshold = A; + double hardness; + double breakup = (1.0-(threshold/2.0))*3.14159265358979; + double bridgerectifier; + double sqdrive = B; + if (sqdrive > 1.0) sqdrive *= sqdrive; + sqdrive = sqrt(sqdrive); + double indrive = C*3.0; + if (indrive > 1.0) indrive *= indrive; + indrive *= (1.0-(0.1695*sqdrive)); + //no gain loss of convolution for APIcolypse + //calibrate this to match noise level with character at 1.0 + //you get for instance 0.819 and 1.0-0.819 is 0.181 + double randy; + double outlevel = D; + + if (threshold < 1) hardness = 1.0 / (1.0-threshold); + else hardness = 999999999999999999999.0; + //set up hardness to exactly fill gap between threshold and 0db + //if threshold is literally 1 then hardness is infinite, so we make it very big + + while (--sampleFrames >= 0) + { + long double inputSampleL = *in1; + long double inputSampleR = *in2; + + static int noisesourceL = 0; + static int noisesourceR = 850010; + int residue; + double applyresidue; + + noisesourceL = noisesourceL % 1700021; noisesourceL++; + residue = noisesourceL * noisesourceL; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL += applyresidue; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + inputSampleL -= applyresidue; + } + + noisesourceR = noisesourceR % 1700021; noisesourceR++; + residue = noisesourceR * noisesourceR; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR += applyresidue; + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + inputSampleR -= applyresidue; + } + //for live air, we always apply the dither noise. Then, if our result is + //effectively digital black, we'll subtract it again. We want a 'air' hiss + inputSampleL *= indrive; + inputSampleR *= indrive; + + //calibrated to match gain through convolution and -0.3 correction + if (sqdrive > 0.0){ + bL[23] = bL[22]; bL[22] = bL[21]; bL[21] = bL[20]; bL[20] = bL[19]; bL[19] = bL[18]; bL[18] = bL[17]; bL[17] = bL[16]; bL[16] = bL[15]; + bL[15] = bL[14]; bL[14] = bL[13]; bL[13] = bL[12]; bL[12] = bL[11]; bL[11] = bL[10]; bL[10] = bL[9]; bL[9] = bL[8]; bL[8] = bL[7]; + bL[7] = bL[6]; bL[6] = bL[5]; bL[5] = bL[4]; bL[4] = bL[3]; bL[3] = bL[2]; bL[2] = bL[1]; bL[1] = bL[0]; bL[0] = inputSampleL * sqdrive; + inputSampleL += (bL[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bL[1])))); + inputSampleL -= (bL[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bL[2])))); + inputSampleL += (bL[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bL[3])))); + inputSampleL -= (bL[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bL[4])))); + inputSampleL += (bL[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bL[5])))); + inputSampleL -= (bL[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bL[6])))); + inputSampleL += (bL[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bL[7])))); + inputSampleL -= (bL[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bL[8])))); + inputSampleL += (bL[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bL[9])))); + inputSampleL -= (bL[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bL[10])))); + inputSampleL += (bL[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bL[11])))); + inputSampleL -= (bL[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bL[12])))); + inputSampleL += (bL[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bL[13])))); + inputSampleL -= (bL[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bL[14])))); + inputSampleL += (bL[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bL[15])))); + inputSampleL -= (bL[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bL[16])))); + inputSampleL += (bL[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bL[17])))); + inputSampleL -= (bL[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bL[18])))); + inputSampleL += (bL[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bL[19])))); + inputSampleL -= (bL[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bL[20])))); + inputSampleL += (bL[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bL[21])))); + inputSampleL -= (bL[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bL[22])))); + inputSampleL += (bL[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bL[23])))); + } //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce + // more of a consistent variation between soft and loud convolutions. For years I thought this code was a + //mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single + //convolution kernel location: this will be true of the Character individual plugins as well. + + //calibrated to match gain through convolution and -0.3 correction + if (sqdrive > 0.0){ + bR[23] = bR[22]; bR[22] = bR[21]; bR[21] = bR[20]; bR[20] = bR[19]; bR[19] = bR[18]; bR[18] = bR[17]; bR[17] = bR[16]; bR[16] = bR[15]; + bR[15] = bR[14]; bR[14] = bR[13]; bR[13] = bR[12]; bR[12] = bR[11]; bR[11] = bR[10]; bR[10] = bR[9]; bR[9] = bR[8]; bR[8] = bR[7]; + bR[7] = bR[6]; bR[6] = bR[5]; bR[5] = bR[4]; bR[4] = bR[3]; bR[3] = bR[2]; bR[2] = bR[1]; bR[1] = bR[0]; bR[0] = inputSampleR * sqdrive; + inputSampleR += (bR[1] * (0.38856694371895023 + (0.14001177830115491*fabs(bR[1])))); + inputSampleR -= (bR[2] * (0.17469488984546111 + (0.05204541941091459*fabs(bR[2])))); + inputSampleR += (bR[3] * (0.11643521461774288 - (0.01193121216518472*fabs(bR[3])))); + inputSampleR -= (bR[4] * (0.08874416268268183 - (0.05867502375036486*fabs(bR[4])))); + inputSampleR += (bR[5] * (0.07222999223073785 - (0.08519974113692971*fabs(bR[5])))); + inputSampleR -= (bR[6] * (0.06103207678880003 - (0.09230674983449150*fabs(bR[6])))); + inputSampleR += (bR[7] * (0.05277389277465404 - (0.08487342372497046*fabs(bR[7])))); + inputSampleR -= (bR[8] * (0.04631144388636078 - (0.06976851898821038*fabs(bR[8])))); + inputSampleR += (bR[9] * (0.04102721072495113 - (0.05337974329110802*fabs(bR[9])))); + inputSampleR -= (bR[10] * (0.03656047655964371 - (0.03990914278458497*fabs(bR[10])))); + inputSampleR += (bR[11] * (0.03268677450573373 - (0.03090433934018759*fabs(bR[11])))); + inputSampleR -= (bR[12] * (0.02926012259262895 - (0.02585223214266682*fabs(bR[12])))); + inputSampleR += (bR[13] * (0.02618257163789973 - (0.02326667039588473*fabs(bR[13])))); + inputSampleR -= (bR[14] * (0.02338568277879992 - (0.02167067760829789*fabs(bR[14])))); + inputSampleR += (bR[15] * (0.02082142324645262 - (0.02013392273267951*fabs(bR[15])))); + inputSampleR -= (bR[16] * (0.01845525966656259 - (0.01833038930966512*fabs(bR[16])))); + inputSampleR += (bR[17] * (0.01626113504980445 - (0.01631893218593511*fabs(bR[17])))); + inputSampleR -= (bR[18] * (0.01422084088669267 - (0.01427828125219885*fabs(bR[18])))); + inputSampleR += (bR[19] * (0.01231993595709338 - (0.01233991521342998*fabs(bR[19])))); + inputSampleR -= (bR[20] * (0.01054774630451013 - (0.01054774630542346*fabs(bR[20])))); + inputSampleR += (bR[21] * (0.00889548162355088 - (0.00889548162263755*fabs(bR[21])))); + inputSampleR -= (bR[22] * (0.00735749099304526 - (0.00735749099395860*fabs(bR[22])))); + inputSampleR += (bR[23] * (0.00592812350468000 - (0.00592812350376666*fabs(bR[23])))); + } //the Character plugins as individual processors did this. BussColors applies an averaging factor to produce + // more of a consistent variation between soft and loud convolutions. For years I thought this code was a + //mistake and did nothing, but in fact what it's doing is producing slightly different curves for every single + //convolution kernel location: this will be true of the Character individual plugins as well. + + if (fabs(inputSampleL) > threshold) + { + bridgerectifier = (fabs(inputSampleL)-threshold)*hardness; + //skip flat area if any, scale to distortion limit + if (bridgerectifier > breakup) bridgerectifier = breakup; + //max value for sine function, 'breakup' modeling for trashed console tone + //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag + bridgerectifier = sin(bridgerectifier)/hardness; + //do the sine factor, scale back to proper amount + if (inputSampleL > 0) inputSampleL = bridgerectifier+threshold; + else inputSampleL = -(bridgerectifier+threshold); + } //otherwise we leave it untouched by the overdrive stuff + //this is the notorious New Channel Density algorithm. It's much less popular than the original Density, + //because it introduces a point where the saturation 'curve' changes from straight to curved. + //People don't like these discontinuities, but you can use them for effect or to grit up the sound. + + if (fabs(inputSampleR) > threshold) + { + bridgerectifier = (fabs(inputSampleR)-threshold)*hardness; + //skip flat area if any, scale to distortion limit + if (bridgerectifier > breakup) bridgerectifier = breakup; + //max value for sine function, 'breakup' modeling for trashed console tone + //more hardness = more solidness behind breakup modeling. more softness, more 'grunge' and sag + bridgerectifier = sin(bridgerectifier)/hardness; + //do the sine factor, scale back to proper amount + if (inputSampleR > 0) inputSampleR = bridgerectifier+threshold; + else inputSampleR = -(bridgerectifier+threshold); + } //otherwise we leave it untouched by the overdrive stuff + //this is the notorious New Channel Density algorithm. It's much less popular than the original Density, + //because it introduces a point where the saturation 'curve' changes from straight to curved. + //People don't like these discontinuities, but you can use them for effect or to grit up the sound. + + randy = ((rand()/(double)RAND_MAX)*0.022); + bridgerectifier = ((inputSampleL*(1-randy))+(lastSampleL*randy)) * outlevel; + lastSampleL = inputSampleL; + inputSampleL = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal. + + randy = ((rand()/(double)RAND_MAX)*0.022); + bridgerectifier = ((inputSampleR*(1-randy))+(lastSampleR*randy)) * outlevel; + lastSampleR = inputSampleR; + inputSampleR = bridgerectifier; //applies a tiny 'fuzz' to highs: from original Crystal. + + //This is akin to the old Chrome Oxide plugin, applying a fuzz to only the slews. The noise only appears + //when current and old samples are different from each other, otherwise you can't tell it's there. + //This is not only during silence but the tops of low frequency waves: it scales down to affect lows more gently. + + //noise shaping to 64-bit floating point + double fpTemp = inputSampleL; + fpNShapeL += (inputSampleL-fpTemp); + inputSampleL += fpNShapeL; + //if this confuses you look at the wordlength for fpTemp :) + fpTemp = inputSampleR; + fpNShapeR += (inputSampleR-fpTemp); + inputSampleR += fpNShapeR; + //for deeper space and warmth, we try a non-oscillating noise shaping + //that is kind of ruthless: it will forever retain the rounding errors + //except we'll dial it back a hair at the end of every buffer processed + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } + fpNShapeL *= 0.999999; + fpNShapeR *= 0.999999; + //we will just delicately dial back the FP noise shaping, not even every sample + //this is a good place to put subtle 'no runaway' calculations, though bear in mind + //that it will be called more often when you use shorter sample buffers in the DAW. + //So, very low latency operation will call these calculations more often. +} |