mirror of
https://github.com/airwindows/airwindows.git
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218 lines
8.4 KiB
C++
Executable file
218 lines
8.4 KiB
C++
Executable file
/* ========================================
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* Tube - Tube.h
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* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
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* ======================================== */
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#ifndef __Tube_H
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#include "Tube.h"
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#endif
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void Tube::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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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double gain = 1.0+(A*0.2246161992650486);
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//this maxes out at +1.76dB, which is the exact difference between what a triangle/saw wave
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//would be, and a sine (the fullest possible wave at the same peak amplitude). Why do this?
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//Because the nature of this plugin is the 'more FAT TUUUUBE fatness!' knob, and because
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//sticking to this amount maximizes that effect on a 'normal' sound that is itself unclipped
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//while confining the resulting 'clipped' area to what is already 'fattened' into a flat
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//and distorted region. You can always put a gain trim in front of it for more distortion,
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//or cascade them in the DAW for more distortion.
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double iterations = 1.0-A;
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int powerfactor = (5.0*iterations)+1;
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double gainscaling = 1.0/(double)(powerfactor+1);
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double outputscaling = 1.0 + (1.0/(double)(powerfactor));
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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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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleA; previousSampleA = stored; inputSampleL *= 0.5;
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stored = inputSampleR;
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inputSampleR += previousSampleB; previousSampleB = stored; inputSampleR *= 0.5;
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} //for high sample rates on this plugin we are going to do a simple average
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inputSampleL *= gain;
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inputSampleR *= gain;
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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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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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double factor = inputSampleL; //Left channel
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for (int x = 0; x < powerfactor; x++) factor *= inputSampleL;
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//this applies more and more of a 'curve' to the transfer function
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if ((powerfactor % 2 == 1) && (inputSampleL != 0.0)) factor = (factor/inputSampleL)*fabs(inputSampleL);
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//if we would've got an asymmetrical effect this undoes the last step, and then
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//redoes it using an absolute value to make the effect symmetrical again
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factor *= gainscaling;
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inputSampleL -= factor;
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inputSampleL *= outputscaling;
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factor = inputSampleR; //Right channel
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for (int x = 0; x < powerfactor; x++) factor *= inputSampleR;
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//this applies more and more of a 'curve' to the transfer function
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if ((powerfactor % 2 == 1) && (inputSampleR != 0.0)) factor = (factor/inputSampleR)*fabs(inputSampleR);
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//if we would've got an asymmetrical effect this undoes the last step, and then
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//redoes it using an absolute value to make the effect symmetrical again
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factor *= gainscaling;
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inputSampleR -= factor;
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inputSampleR *= outputscaling;
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/* This is the simplest raw form of the 'fattest' TUBE boost between -1.0 and 1.0
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if (inputSample > 1.0) inputSample = 1.0; if (inputSample < -1.0) inputSample = -1.0;
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inputSample = (inputSample-(inputSample*fabs(inputSample)*0.5))*2.0;
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*/
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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleC; previousSampleC = stored; inputSampleL *= 0.5;
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stored = inputSampleR;
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inputSampleR += previousSampleD; previousSampleD = stored; inputSampleR *= 0.5;
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} //for high sample rates on this plugin we are going to do a simple average
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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 Tube::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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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double gain = 1.0+(A*0.2246161992650486);
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//this maxes out at +1.76dB, which is the exact difference between what a triangle/saw wave
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//would be, and a sine (the fullest possible wave at the same peak amplitude). Why do this?
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//Because the nature of this plugin is the 'more FAT TUUUUBE fatness!' knob, and because
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//sticking to this amount maximizes that effect on a 'normal' sound that is itself unclipped
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//while confining the resulting 'clipped' area to what is already 'fattened' into a flat
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//and distorted region. You can always put a gain trim in front of it for more distortion,
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//or cascade them in the DAW for more distortion.
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double iterations = 1.0-A;
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int powerfactor = (5.0*iterations)+1;
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double gainscaling = 1.0/(double)(powerfactor+1);
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double outputscaling = 1.0 + (1.0/(double)(powerfactor));
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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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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleA; previousSampleA = stored; inputSampleL *= 0.5;
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stored = inputSampleR;
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inputSampleR += previousSampleB; previousSampleB = stored; inputSampleR *= 0.5;
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} //for high sample rates on this plugin we are going to do a simple average
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inputSampleL *= gain;
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inputSampleR *= gain;
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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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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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double factor = inputSampleL; //Left channel
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for (int x = 0; x < powerfactor; x++) factor *= inputSampleL;
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//this applies more and more of a 'curve' to the transfer function
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if ((powerfactor % 2 == 1) && (inputSampleL != 0.0)) factor = (factor/inputSampleL)*fabs(inputSampleL);
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//if we would've got an asymmetrical effect this undoes the last step, and then
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//redoes it using an absolute value to make the effect symmetrical again
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factor *= gainscaling;
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inputSampleL -= factor;
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inputSampleL *= outputscaling;
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factor = inputSampleR; //Right channel
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for (int x = 0; x < powerfactor; x++) factor *= inputSampleR;
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//this applies more and more of a 'curve' to the transfer function
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if ((powerfactor % 2 == 1) && (inputSampleR != 0.0)) factor = (factor/inputSampleR)*fabs(inputSampleR);
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//if we would've got an asymmetrical effect this undoes the last step, and then
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//redoes it using an absolute value to make the effect symmetrical again
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factor *= gainscaling;
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inputSampleR -= factor;
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inputSampleR *= outputscaling;
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/* This is the simplest raw form of the 'fattest' TUBE boost between -1.0 and 1.0
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if (inputSample > 1.0) inputSample = 1.0; if (inputSample < -1.0) inputSample = -1.0;
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inputSample = (inputSample-(inputSample*fabs(inputSample)*0.5))*2.0;
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*/
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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleC; previousSampleC = stored; inputSampleL *= 0.5;
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stored = inputSampleR;
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inputSampleR += previousSampleD; previousSampleD = stored; inputSampleR *= 0.5;
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} //for high sample rates on this plugin we are going to do a simple average
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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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