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280 lines
11 KiB
C++
Executable file
280 lines
11 KiB
C++
Executable file
/* ========================================
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* ConsoleMCBuss - ConsoleMCBuss.h
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* Copyright (c) airwindows, Airwindows uses the MIT license
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* ======================================== */
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#ifndef __ConsoleMCBuss_H
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#include "ConsoleMCBuss.h"
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#endif
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void ConsoleMCBuss::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
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{
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float* in1 = inputs[0];
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float* in2 = inputs[1];
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float* out1 = outputs[0];
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float* out2 = outputs[1];
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VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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gainA = gainB;
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gainB = sqrt(A); //smoothed master fader from Z2 filters
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//this will be applied three times: this is to make the various tone alterations
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//hit differently at different master fader drive levels.
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//in particular, backing off the master fader tightens the super lows
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//but opens up the modified Sinew, because more of the attentuation happens before
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//you even get to slew clipping :) and if the fader is not active, it bypasses completely.
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double threshSinew = 0.5171104/overallscale;
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double subTrim = 0.001 / overallscale;
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while (--sampleFrames >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
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if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
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double temp = (double)sampleFrames/inFramesToProcess;
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double gain = (gainA*temp)+(gainB*(1.0-temp));
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//setting up smoothed master fader
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//begin SubTight section
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double subSampleL = inputSampleL * subTrim;
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double subSampleR = inputSampleR * subTrim;
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double scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
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subAL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
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subAR = subSampleR*scale;
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scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
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subBL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
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subBR = subSampleR*scale;
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scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subCL+(sin(subCL-subSampleL)*scale));
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subCL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subCR+(sin(subCR-subSampleR)*scale));
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subCR = subSampleR*scale;
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scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subDL+(sin(subDL-subSampleL)*scale));
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subDL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subDR+(sin(subDR-subSampleR)*scale));
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subDR = subSampleR*scale;
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if (subSampleL > 0.25) subSampleL = 0.25;
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if (subSampleL < -0.25) subSampleL = -0.25;
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if (subSampleR > 0.25) subSampleR = 0.25;
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if (subSampleR < -0.25) subSampleR = -0.25;
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inputSampleL -= (subSampleL*16.0);
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inputSampleR -= (subSampleR*16.0);
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//end SubTight section
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if (gain < 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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} //if using the master fader, we are going to attenuate three places.
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//subtight is always fully engaged: tighten response when restraining full console
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//begin Console7Buss which is the one we choose for ConsoleMC
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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inputSampleL = ((asin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.618033988749894848204586)+(asin(inputSampleL)*0.381966011250105);
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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inputSampleR = ((asin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.618033988749894848204586)+(asin(inputSampleR)*0.381966011250105);
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//this is an asin version of Spiral blended with regular asin ConsoleBuss.
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//It's blending between two different harmonics in the overtones of the algorithm
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if (gain < 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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} //if using the master fader, we are going to attenuate three places.
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//after C7Buss but before EverySlew: allow highs to come out a bit more
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//when pulling back master fader. Less drive equals more open
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temp = inputSampleL;
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double clamp = inputSampleL - lastSinewL;
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if (lastSinewL > 1.0) lastSinewL = 1.0;
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if (lastSinewL < -1.0) lastSinewL = -1.0;
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double sinew = threshSinew * cos(lastSinewL);
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if (clamp > sinew) temp = lastSinewL + sinew;
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if (-clamp > sinew) temp = lastSinewL - sinew;
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inputSampleL = lastSinewL = temp;
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temp = inputSampleR;
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clamp = inputSampleR - lastSinewR;
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if (lastSinewR > 1.0) lastSinewR = 1.0;
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if (lastSinewR < -1.0) lastSinewR = -1.0;
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sinew = threshSinew * cos(lastSinewR);
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if (clamp > sinew) temp = lastSinewR + sinew;
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if (-clamp > sinew) temp = lastSinewR - sinew;
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inputSampleR = lastSinewR = temp;
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if (gain < 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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} //if using the master fader, we are going to attenuate three places.
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//after EverySlew fades the total output sound: least change in tone here.
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//begin 32 bit stereo floating point dither
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int expon; frexpf((float)inputSampleL, &expon);
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fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
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inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
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frexpf((float)inputSampleR, &expon);
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fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
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inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
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//end 32 bit stereo floating point dither
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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in1++;
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in2++;
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out1++;
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out2++;
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}
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}
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void ConsoleMCBuss::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
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{
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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VstInt32 inFramesToProcess = sampleFrames; //vst doesn't give us this as a separate variable so we'll make it
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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gainA = gainB;
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gainB = sqrt(A); //smoothed master fader from Z2 filters
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//this will be applied three times: this is to make the various tone alterations
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//hit differently at different master fader drive levels.
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//in particular, backing off the master fader tightens the super lows
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//but opens up the modified Sinew, because more of the attentuation happens before
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//you even get to slew clipping :) and if the fader is not active, it bypasses completely.
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double threshSinew = 0.5171104/overallscale;
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double subTrim = 0.001 / overallscale;
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while (--sampleFrames >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
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if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
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double temp = (double)sampleFrames/inFramesToProcess;
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double gain = (gainA*temp)+(gainB*(1.0-temp));
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//setting up smoothed master fader
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//begin SubTight section
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double subSampleL = inputSampleL * subTrim;
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double subSampleR = inputSampleR * subTrim;
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double scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subAL+(sin(subAL-subSampleL)*scale));
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subAL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subAR+(sin(subAR-subSampleR)*scale));
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subAR = subSampleR*scale;
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scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subBL+(sin(subBL-subSampleL)*scale));
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subBL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subBR+(sin(subBR-subSampleR)*scale));
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subBR = subSampleR*scale;
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scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subCL+(sin(subCL-subSampleL)*scale));
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subCL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subCR+(sin(subCR-subSampleR)*scale));
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subCR = subSampleR*scale;
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scale = 0.5+fabs(subSampleL*0.5);
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subSampleL = (subDL+(sin(subDL-subSampleL)*scale));
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subDL = subSampleL*scale;
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scale = 0.5+fabs(subSampleR*0.5);
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subSampleR = (subDR+(sin(subDR-subSampleR)*scale));
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subDR = subSampleR*scale;
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if (subSampleL > 0.25) subSampleL = 0.25;
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if (subSampleL < -0.25) subSampleL = -0.25;
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if (subSampleR > 0.25) subSampleR = 0.25;
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if (subSampleR < -0.25) subSampleR = -0.25;
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inputSampleL -= (subSampleL*16.0);
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inputSampleR -= (subSampleR*16.0);
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//end SubTight section
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if (gain < 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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} //if using the master fader, we are going to attenuate three places.
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//subtight is always fully engaged: tighten response when restraining full console
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//begin Console7Buss which is the one we choose for ConsoleMC
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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inputSampleL = ((asin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.618033988749894848204586)+(asin(inputSampleL)*0.381966011250105);
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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inputSampleR = ((asin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.618033988749894848204586)+(asin(inputSampleR)*0.381966011250105);
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//this is an asin version of Spiral blended with regular asin ConsoleBuss.
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//It's blending between two different harmonics in the overtones of the algorithm
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if (gain < 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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} //if using the master fader, we are going to attenuate three places.
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//after C7Buss but before EverySlew: allow highs to come out a bit more
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//when pulling back master fader. Less drive equals more open
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temp = inputSampleL;
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double clamp = inputSampleL - lastSinewL;
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if (lastSinewL > 1.0) lastSinewL = 1.0;
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if (lastSinewL < -1.0) lastSinewL = -1.0;
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double sinew = threshSinew * cos(lastSinewL);
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if (clamp > sinew) temp = lastSinewL + sinew;
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if (-clamp > sinew) temp = lastSinewL - sinew;
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inputSampleL = lastSinewL = temp;
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temp = inputSampleR;
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clamp = inputSampleR - lastSinewR;
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if (lastSinewR > 1.0) lastSinewR = 1.0;
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if (lastSinewR < -1.0) lastSinewR = -1.0;
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sinew = threshSinew * cos(lastSinewR);
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if (clamp > sinew) temp = lastSinewR + sinew;
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if (-clamp > sinew) temp = lastSinewR - sinew;
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inputSampleR = lastSinewR = temp;
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if (gain < 1.0) {
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inputSampleL *= gain;
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inputSampleR *= gain;
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} //if using the master fader, we are going to attenuate three places.
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//after EverySlew fades the total output sound: least change in tone here.
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//begin 64 bit stereo floating point dither
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//int expon; frexp((double)inputSampleL, &expon);
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fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
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//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//frexp((double)inputSampleR, &expon);
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fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
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//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//end 64 bit stereo floating point dither
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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in1++;
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in2++;
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out1++;
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out2++;
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}
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}
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