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-rwxr-xr-xplugins/LinuxVST/src/Compresaturator/CompresaturatorProc.cpp438
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diff --git a/plugins/LinuxVST/src/Compresaturator/CompresaturatorProc.cpp b/plugins/LinuxVST/src/Compresaturator/CompresaturatorProc.cpp
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+++ b/plugins/LinuxVST/src/Compresaturator/CompresaturatorProc.cpp
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+/* ========================================
+ * Compresaturator - Compresaturator.h
+ * Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Compresaturator_H
+#include "Compresaturator.h"
+#endif
+
+void Compresaturator::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
+{
+ float* in1 = inputs[0];
+ float* in2 = inputs[1];
+ float* out1 = outputs[0];
+ float* out2 = outputs[1];
+
+ double inputgain = pow(10.0,((A*24.0)-12.0)/20.0);
+ double satComp = B*2.0;
+ int widestRange = C*C*C*5000;
+ if (widestRange < 50) widestRange = 50;
+ satComp += (((double)widestRange/3000.0)*satComp);
+ //set the max wideness of comp zone, minimum range boosted (too much?)
+ double output = D;
+ double wet = E;
+
+ 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
+ long double drySampleL = inputSampleL;
+ long double drySampleR = inputSampleR;
+
+ if (dCount < 1 || dCount > 5000) {dCount = 5000;}
+
+ //begin drive L
+ long double temp = inputSampleL;
+ double variSpeed = 1.0 + ((padFactorL/lastWidthL)*satComp);
+ if (variSpeed < 1.0) variSpeed = 1.0;
+ double totalgain = inputgain / variSpeed;
+ if (totalgain != 1.0) {
+ inputSampleL *= totalgain;
+ if (totalgain < 1.0) {
+ temp *= totalgain;
+ //no boosting beyond unity please
+ }
+ }
+
+ long double bridgerectifier = fabs(inputSampleL);
+ double overspill = 0;
+ int targetWidth = widestRange;
+ //we now have defaults and an absolute input value to work with
+ if (bridgerectifier < 0.01) padFactorL *= 0.9999;
+ //in silences we bring back padFactor if it got out of hand
+ if (bridgerectifier > 1.57079633) {
+ bridgerectifier = 1.57079633;
+ targetWidth = 8;
+ }
+ //if our output's gone beyond saturating to distorting, we begin chasing the
+ //buffer size smaller. Anytime we don't have that, we expand (smoothest sound, only adding to an increasingly subdivided buffer)
+
+ bridgerectifier = sin(bridgerectifier);
+ if (inputSampleL > 0) {
+ inputSampleL = bridgerectifier;
+ overspill = temp - bridgerectifier;
+ }
+
+ if (inputSampleL < 0) {
+ inputSampleL = -bridgerectifier;
+ overspill = (-temp) - bridgerectifier;
+ }
+ //drive section L
+
+ //begin drive R
+ temp = inputSampleR;
+ variSpeed = 1.0 + ((padFactorR/lastWidthR)*satComp);
+ if (variSpeed < 1.0) variSpeed = 1.0;
+ totalgain = inputgain / variSpeed;
+ if (totalgain != 1.0) {
+ inputSampleR *= totalgain;
+ if (totalgain < 1.0) {
+ temp *= totalgain;
+ //no boosting beyond unity please
+ }
+ }
+
+ bridgerectifier = fabs(inputSampleR);
+ overspill = 0;
+ targetWidth = widestRange;
+ //we now have defaults and an absolute input value to work with
+ if (bridgerectifier < 0.01) padFactorR *= 0.9999;
+ //in silences we bring back padFactor if it got out of hand
+ if (bridgerectifier > 1.57079633) {
+ bridgerectifier = 1.57079633;
+ targetWidth = 8;
+ }
+ //if our output's gone beyond saturating to distorting, we begin chasing the
+ //buffer size smaller. Anytime we don't have that, we expand (smoothest sound, only adding to an increasingly subdivided buffer)
+
+ bridgerectifier = sin(bridgerectifier);
+ if (inputSampleR > 0) {
+ inputSampleR = bridgerectifier;
+ overspill = temp - bridgerectifier;
+ }
+
+ if (inputSampleR < 0) {
+ inputSampleR = -bridgerectifier;
+ overspill = (-temp) - bridgerectifier;
+ }
+ //drive section R
+
+
+ dL[dCount + 5000] = dL[dCount] = overspill * satComp;
+ dR[dCount + 5000] = dR[dCount] = overspill * satComp;
+ dCount--;
+ //we now have a big buffer to draw from, which is always positive amount of overspill
+
+ //begin pad L
+ padFactorL += dL[dCount];
+ double randy = (rand()/(double)RAND_MAX);
+ if ((targetWidth*randy) > lastWidthL) {
+ //we are expanding the buffer so we don't remove this trailing sample
+ lastWidthL += 1;
+ } else {
+ padFactorL -= dL[dCount+lastWidthL];
+ //zero change, or target is smaller and we are shrinking
+ if (targetWidth < lastWidthL) {
+ lastWidthL -= 1;
+ if (lastWidthL < 2) lastWidthL = 2;
+ //sanity check as randy can give us target zero
+ padFactorL -= dL[dCount+lastWidthL];
+ }
+ }
+ //variable attack/release speed more rapid as comp intensity increases
+ //implemented in a way where we're repeatedly not altering the buffer as it expands, which makes the comp artifacts smoother
+ if (padFactorL < 0) padFactorL = 0;
+ //end pad L
+
+ //begin pad R
+ padFactorR += dR[dCount];
+ randy = (rand()/(double)RAND_MAX);
+ if ((targetWidth*randy) > lastWidthR) {
+ //we are expanding the buffer so we don't remove this trailing sample
+ lastWidthR += 1;
+ } else {
+ padFactorR -= dR[dCount+lastWidthR];
+ //zero change, or target is smaller and we are shrinking
+ if (targetWidth < lastWidthR) {
+ lastWidthR -= 1;
+ if (lastWidthR < 2) lastWidthR = 2;
+ //sanity check as randy can give us target zero
+ padFactorR -= dR[dCount+lastWidthR];
+ }
+ }
+ //variable attack/release speed more rapid as comp intensity increases
+ //implemented in a way where we're repeatedly not altering the buffer as it expands, which makes the comp artifacts smoother
+ if (padFactorR < 0) padFactorR = 0;
+ //end pad R
+
+ if (output < 1.0) {
+ inputSampleL *= output;
+ inputSampleR *= output;
+ }
+
+ if (wet < 1.0) {
+ inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
+ inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
+ }
+
+ //begin 32 bit stereo floating point dither
+ int expon; frexpf((float)inputSampleL, &expon);
+ fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+ inputSampleL += static_cast<int32_t>(fpd) * 5.960464655174751e-36L * pow(2,expon+62);
+ frexpf((float)inputSampleR, &expon);
+ fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+ inputSampleR += static_cast<int32_t>(fpd) * 5.960464655174751e-36L * pow(2,expon+62);
+ //end 32 bit stereo floating point dither
+
+ *out1 = inputSampleL;
+ *out2 = inputSampleR;
+
+ *in1++;
+ *in2++;
+ *out1++;
+ *out2++;
+ }
+}
+
+void Compresaturator::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
+{
+ double* in1 = inputs[0];
+ double* in2 = inputs[1];
+ double* out1 = outputs[0];
+ double* out2 = outputs[1];
+
+ double inputgain = pow(10.0,((A*24.0)-12.0)/20.0);
+ double satComp = B*2.0;
+ int widestRange = C*C*5000;
+ if (widestRange < 50) widestRange = 50;
+ satComp += (((double)widestRange/3000.0)*satComp);
+ //set the max wideness of comp zone, minimum range boosted (too much?)
+ double output = D;
+ double wet = E;
+
+ 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
+ long double drySampleL = inputSampleL;
+ long double drySampleR = inputSampleR;
+
+ if (dCount < 1 || dCount > 5000) {dCount = 5000;}
+
+ //begin L
+ long double temp = inputSampleL;
+ double variSpeed = 1.0 + ((padFactorL/lastWidthL)*satComp);
+ if (variSpeed < 1.0) variSpeed = 1.0;
+ double totalgain = inputgain / variSpeed;
+ if (totalgain != 1.0) {
+ inputSampleL *= totalgain;
+ if (totalgain < 1.0) {
+ temp *= totalgain;
+ //no boosting beyond unity please
+ }
+ }
+
+ long double bridgerectifier = fabs(inputSampleL);
+ double overspill = 0;
+ int targetWidth = widestRange;
+ //we now have defaults and an absolute input value to work with
+ if (bridgerectifier < 0.01) padFactorL *= 0.9999;
+ //in silences we bring back padFactor if it got out of hand
+ if (bridgerectifier > 1.57079633) {
+ bridgerectifier = 1.57079633;
+ targetWidth = 8;
+ }
+ //if our output's gone beyond saturating to distorting, we begin chasing the
+ //buffer size smaller. Anytime we don't have that, we expand (smoothest sound, only adding to an increasingly subdivided buffer)
+
+ bridgerectifier = sin(bridgerectifier);
+ if (inputSampleL > 0) {
+ inputSampleL = bridgerectifier;
+ overspill = temp - bridgerectifier;
+ }
+
+ if (inputSampleL < 0) {
+ inputSampleL = -bridgerectifier;
+ overspill = (-temp) - bridgerectifier;
+ }
+ //drive section
+
+ dL[dCount + 5000] = dL[dCount] = overspill * satComp;
+ //we now have a big buffer to draw from, which is always positive amount of overspill
+
+ padFactorL += dL[dCount];
+ double randy = (rand()/(double)RAND_MAX);
+ if ((targetWidth*randy) > lastWidthL) {
+ //we are expanding the buffer so we don't remove this trailing sample
+ lastWidthL += 1;
+ } else {
+ padFactorL -= dL[dCount+lastWidthL];
+ //zero change, or target is smaller and we are shrinking
+ if (targetWidth < lastWidthL) {
+ lastWidthL -= 1;
+ if (lastWidthL < 2) lastWidthL = 2;
+ //sanity check as randy can give us target zero
+ padFactorL -= dL[dCount+lastWidthL];
+ }
+ }
+ //variable attack/release speed more rapid as comp intensity increases
+ //implemented in a way where we're repeatedly not altering the buffer as it expands, which makes the comp artifacts smoother
+ if (padFactorL < 0) padFactorL = 0;
+ //end L
+
+ //begin R
+ temp = inputSampleR;
+ variSpeed = 1.0 + ((padFactorR/lastWidthR)*satComp);
+ if (variSpeed < 1.0) variSpeed = 1.0;
+ totalgain = inputgain / variSpeed;
+ if (totalgain != 1.0) {
+ inputSampleR *= totalgain;
+ if (totalgain < 1.0) {
+ temp *= totalgain;
+ //no boosting beyond unity please
+ }
+ }
+
+ bridgerectifier = fabs(inputSampleR);
+ overspill = 0;
+ targetWidth = widestRange;
+ //we now have defaults and an absolute input value to work with
+ if (bridgerectifier < 0.01) padFactorR *= 0.9999;
+ //in silences we bring back padFactor if it got out of hand
+ if (bridgerectifier > 1.57079633) {
+ bridgerectifier = 1.57079633;
+ targetWidth = 8;
+ }
+ //if our output's gone beyond saturating to distorting, we begin chasing the
+ //buffer size smaller. Anytime we don't have that, we expand (smoothest sound, only adding to an increasingly subdivided buffer)
+
+ bridgerectifier = sin(bridgerectifier);
+ if (inputSampleR > 0) {
+ inputSampleR = bridgerectifier;
+ overspill = temp - bridgerectifier;
+ }
+
+ if (inputSampleR < 0) {
+ inputSampleR = -bridgerectifier;
+ overspill = (-temp) - bridgerectifier;
+ }
+ //drive section
+
+ dR[dCount + 5000] = dR[dCount] = overspill * satComp;
+ //we now have a big buffer to draw from, which is always positive amount of overspill
+
+ padFactorR += dR[dCount];
+ randy = (rand()/(double)RAND_MAX);
+ if ((targetWidth*randy) > lastWidthR) {
+ //we are expanding the buffer so we don't remove this trailing sample
+ lastWidthR += 1;
+ } else {
+ padFactorR -= dR[dCount+lastWidthR];
+ //zero change, or target is smaller and we are shrinking
+ if (targetWidth < lastWidthR) {
+ lastWidthR -= 1;
+ if (lastWidthR < 2) lastWidthR = 2;
+ //sanity check as randy can give us target zero
+ padFactorR -= dR[dCount+lastWidthR];
+ }
+ }
+ //variable attack/release speed more rapid as comp intensity increases
+ //implemented in a way where we're repeatedly not altering the buffer as it expands, which makes the comp artifacts smoother
+ if (padFactorR < 0) padFactorR = 0;
+ //end R
+
+ dCount--;
+
+ if (output < 1.0) {
+ inputSampleL *= output;
+ inputSampleR *= output;
+ }
+
+ if (wet < 1.0) {
+ inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet));
+ inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet));
+ }
+
+ //begin 64 bit stereo floating point dither
+ int expon; frexp((double)inputSampleL, &expon);
+ fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+ inputSampleL += static_cast<int32_t>(fpd) * 1.110223024625156e-44L * pow(2,expon+62);
+ frexp((double)inputSampleR, &expon);
+ fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5;
+ inputSampleR += static_cast<int32_t>(fpd) * 1.110223024625156e-44L * pow(2,expon+62);
+ //end 64 bit stereo floating point dither
+
+ *out1 = inputSampleL;
+ *out2 = inputSampleR;
+
+ *in1++;
+ *in2++;
+ *out1++;
+ *out2++;
+ }
+}