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Diffstat (limited to 'plugins/WinVST/NC-17/NCSeventeenProc.cpp')
| -rwxr-xr-x | plugins/WinVST/NC-17/NCSeventeenProc.cpp | 721 |
1 files changed, 721 insertions, 0 deletions
diff --git a/plugins/WinVST/NC-17/NCSeventeenProc.cpp b/plugins/WinVST/NC-17/NCSeventeenProc.cpp new file mode 100755 index 0000000..e124cb5 --- /dev/null +++ b/plugins/WinVST/NC-17/NCSeventeenProc.cpp @@ -0,0 +1,721 @@ +/* ======================================== + * NCSeventeen - NCSeventeen.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __NCSeventeen_H +#include "NCSeventeen.h" +#endif + +void NCSeventeen::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double inP2; + double chebyshev; + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + + double IIRscaleback = 0.0004716; + double bassScaleback = 0.0002364; + double trebleScaleback = 0.0005484; + double addBassBuss = 0.000243; + double addTrebBuss = 0.000407; + double addShortBuss = 0.000326; + IIRscaleback /= overallscale; + bassScaleback /= overallscale; + trebleScaleback /= overallscale; + addBassBuss /= overallscale; + addTrebBuss /= overallscale; + addShortBuss /= overallscale; + double limitingBass = 0.39; + double limitingTreb = 0.6; + double limiting = 0.36; + double maxfeedBass = 0.972; + double maxfeedTreb = 0.972; + double maxfeed = 0.975; + double bridgerectifier; + long double inputSampleL; + double lowSampleL = 0.0; + double highSampleL; + double distSampleL; + double minusSampleL; + double plusSampleL; + long double inputSampleR; + double lowSampleR = 0.0; + double highSampleR; + double distSampleR; + double minusSampleR; + double plusSampleR; + double gain = pow(10.0,(A*24.0)/20); + double outlevel = B; + + + 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. + } + + inputSampleL *= gain; + inputSampleR *= gain; + + if (flip) + { + iirSampleAL = (iirSampleAL * 0.9) + (inputSampleL * 0.1); + lowSampleL = iirSampleAL; + iirSampleAR = (iirSampleAR * 0.9) + (inputSampleR * 0.1); + lowSampleR = iirSampleAR; + } + else + { + iirSampleBL = (iirSampleBL * 0.9) + (inputSampleL * 0.1); + lowSampleL = iirSampleBL; + iirSampleBR = (iirSampleBR * 0.9) + (inputSampleR * 0.1); + lowSampleR = iirSampleBR; + } + highSampleL = inputSampleL - lowSampleL; + highSampleR = inputSampleR - lowSampleR; + flip = !flip; + //we now have two bands and the original source + + inP2 = lowSampleL * lowSampleL; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= basslevL; + //second harmonic max +1 + if (basslevL > 0) basslevL -= bassScaleback; + if (basslevL < 0) basslevL += bassScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(lowSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleL > 0.0) distSampleL = bridgerectifier; + else distSampleL = -bridgerectifier; + minusSampleL = lowSampleL - distSampleL; + plusSampleL = lowSampleL + distSampleL; + if (minusSampleL > maxfeedBass) minusSampleL = maxfeedBass; + if (plusSampleL > maxfeedBass) plusSampleL = maxfeedBass; + if (plusSampleL < -maxfeedBass) plusSampleL = -maxfeedBass; + if (minusSampleL < -maxfeedBass) minusSampleL = -maxfeedBass; + if (lowSampleL > distSampleL) basslevL += (minusSampleL*addBassBuss); + if (lowSampleL < -distSampleL) basslevL -= (plusSampleL*addBassBuss); + if (basslevL > 1.0) basslevL = 1.0; + if (basslevL < -1.0) basslevL = -1.0; + bridgerectifier = fabs(lowSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleL > 0.0) lowSampleL = bridgerectifier; + else lowSampleL = -bridgerectifier; + //apply the distortion transform for reals + lowSampleL /= (1.0+fabs(basslevL*limitingBass)); + lowSampleL += chebyshev; + //apply the correction measuresL + + inP2 = lowSampleR * lowSampleR; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= basslevR; + //second harmonic max +1 + if (basslevR > 0) basslevR -= bassScaleback; + if (basslevR < 0) basslevR += bassScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(lowSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleR > 0.0) distSampleR = bridgerectifier; + else distSampleR = -bridgerectifier; + minusSampleR = lowSampleR - distSampleR; + plusSampleR = lowSampleR + distSampleR; + if (minusSampleR > maxfeedBass) minusSampleR = maxfeedBass; + if (plusSampleR > maxfeedBass) plusSampleR = maxfeedBass; + if (plusSampleR < -maxfeedBass) plusSampleR = -maxfeedBass; + if (minusSampleR < -maxfeedBass) minusSampleR = -maxfeedBass; + if (lowSampleR > distSampleR) basslevR += (minusSampleR*addBassBuss); + if (lowSampleR < -distSampleR) basslevR -= (plusSampleR*addBassBuss); + if (basslevR > 1.0) basslevR = 1.0; + if (basslevR < -1.0) basslevR = -1.0; + bridgerectifier = fabs(lowSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleR > 0.0) lowSampleR = bridgerectifier; + else lowSampleR = -bridgerectifier; + //apply the distortion transform for reals + lowSampleR /= (1.0+fabs(basslevR*limitingBass)); + lowSampleR += chebyshev; + //apply the correction measuresR + + inP2 = highSampleL * highSampleL; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= treblevL; + //second harmonic max +1 + if (treblevL > 0) treblevL -= trebleScaleback; + if (treblevL < 0) treblevL += trebleScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(highSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleL > 0.0) distSampleL = bridgerectifier; + else distSampleL = -bridgerectifier; + minusSampleL = highSampleL - distSampleL; + plusSampleL = highSampleL + distSampleL; + if (minusSampleL > maxfeedTreb) minusSampleL = maxfeedTreb; + if (plusSampleL > maxfeedTreb) plusSampleL = maxfeedTreb; + if (plusSampleL < -maxfeedTreb) plusSampleL = -maxfeedTreb; + if (minusSampleL < -maxfeedTreb) minusSampleL = -maxfeedTreb; + if (highSampleL > distSampleL) treblevL += (minusSampleL*addTrebBuss); + if (highSampleL < -distSampleL) treblevL -= (plusSampleL*addTrebBuss); + if (treblevL > 1.0) treblevL = 1.0; + if (treblevL < -1.0) treblevL = -1.0; + bridgerectifier = fabs(highSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleL > 0.0) highSampleL = bridgerectifier; + else highSampleL = -bridgerectifier; + //apply the distortion transform for reals + highSampleL /= (1.0+fabs(treblevL*limitingTreb)); + highSampleL += chebyshev; + //apply the correction measuresL + + inP2 = highSampleR * highSampleR; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= treblevR; + //second harmonic max +1 + if (treblevR > 0) treblevR -= trebleScaleback; + if (treblevR < 0) treblevR += trebleScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(highSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleR > 0.0) distSampleR = bridgerectifier; + else distSampleR = -bridgerectifier; + minusSampleR = highSampleR - distSampleR; + plusSampleR = highSampleR + distSampleR; + if (minusSampleR > maxfeedTreb) minusSampleR = maxfeedTreb; + if (plusSampleR > maxfeedTreb) plusSampleR = maxfeedTreb; + if (plusSampleR < -maxfeedTreb) plusSampleR = -maxfeedTreb; + if (minusSampleR < -maxfeedTreb) minusSampleR = -maxfeedTreb; + if (highSampleR > distSampleR) treblevR += (minusSampleR*addTrebBuss); + if (highSampleR < -distSampleR) treblevR -= (plusSampleR*addTrebBuss); + if (treblevR > 1.0) treblevR = 1.0; + if (treblevR < -1.0) treblevR = -1.0; + bridgerectifier = fabs(highSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleR > 0.0) highSampleR = bridgerectifier; + else highSampleR = -bridgerectifier; + //apply the distortion transform for reals + highSampleR /= (1.0+fabs(treblevR*limitingTreb)); + highSampleR += chebyshev; + //apply the correction measuresR + + inputSampleL = lowSampleL + highSampleL; + inputSampleR = lowSampleR + highSampleR; + + inP2 = inputSampleL * inputSampleL; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= cheblevL; + //third harmonic max -1 + if (cheblevL > 0) cheblevL -= (IIRscaleback); + if (cheblevL < 0) cheblevL += (IIRscaleback); + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0.0) distSampleL = bridgerectifier; + else distSampleL = -bridgerectifier; + minusSampleL = inputSampleL - distSampleL; + plusSampleL = inputSampleL + distSampleL; + if (minusSampleL > maxfeed) minusSampleL = maxfeed; + if (plusSampleL > maxfeed) plusSampleL = maxfeed; + if (plusSampleL < -maxfeed) plusSampleL = -maxfeed; + if (minusSampleL < -maxfeed) minusSampleL = -maxfeed; + if (inputSampleL > distSampleL) cheblevL += (minusSampleL*addShortBuss); + if (inputSampleL < -distSampleL) cheblevL -= (plusSampleL*addShortBuss); + if (cheblevL > 1.0) cheblevL = 1.0; + if (cheblevL < -1.0) cheblevL = -1.0; + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0.0) inputSampleL = bridgerectifier; + else inputSampleL = -bridgerectifier; + //apply the distortion transform for reals + inputSampleL /= (1.0+fabs(cheblevL*limiting)); + inputSampleL += chebyshev; + //apply the correction measuresL + + inP2 = inputSampleR * inputSampleR; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= cheblevR; + //third harmonic max -1 + if (cheblevR > 0) cheblevR -= IIRscaleback; + if (cheblevR < 0) cheblevR += IIRscaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0.0) distSampleR = bridgerectifier; + else distSampleR = -bridgerectifier; + minusSampleR = inputSampleR - distSampleR; + plusSampleR = inputSampleR + distSampleR; + if (minusSampleR > maxfeed) minusSampleR = maxfeed; + if (plusSampleR > maxfeed) plusSampleR = maxfeed; + if (plusSampleR < -maxfeed) plusSampleR = -maxfeed; + if (minusSampleR < -maxfeed) minusSampleR = -maxfeed; + if (inputSampleR > distSampleR) cheblevR += (minusSampleR*addShortBuss); + if (inputSampleR < -distSampleR) cheblevR -= (plusSampleR*addShortBuss); + if (cheblevR > 1.0) cheblevR = 1.0; + if (cheblevR < -1.0) cheblevR = -1.0; + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0.0) inputSampleR = bridgerectifier; + else inputSampleR = -bridgerectifier; + //apply the distortion transform for reals + inputSampleR /= (1.0+fabs(cheblevR*limiting)); + inputSampleR += chebyshev; + //apply the correction measuresR + + if (outlevel < 1.0) { + inputSampleL *= outlevel; + inputSampleR *= outlevel; + } + + if (inputSampleL > 0.95) inputSampleL = 0.95; + if (inputSampleL < -0.95) inputSampleL = -0.95; + if (inputSampleR > 0.95) inputSampleR = 0.95; + if (inputSampleR < -0.95) inputSampleR = -0.95; + //iron bar + + //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 NCSeventeen::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double inP2; + double chebyshev; + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + + double IIRscaleback = 0.0004716; + double bassScaleback = 0.0002364; + double trebleScaleback = 0.0005484; + double addBassBuss = 0.000243; + double addTrebBuss = 0.000407; + double addShortBuss = 0.000326; + IIRscaleback /= overallscale; + bassScaleback /= overallscale; + trebleScaleback /= overallscale; + addBassBuss /= overallscale; + addTrebBuss /= overallscale; + addShortBuss /= overallscale; + double limitingBass = 0.39; + double limitingTreb = 0.6; + double limiting = 0.36; + double maxfeedBass = 0.972; + double maxfeedTreb = 0.972; + double maxfeed = 0.975; + double bridgerectifier; + long double inputSampleL; + double lowSampleL = 0.0; + double highSampleL; + double distSampleL; + double minusSampleL; + double plusSampleL; + long double inputSampleR; + double lowSampleR = 0.0; + double highSampleR; + double distSampleR; + double minusSampleR; + double plusSampleR; + double gain = pow(10.0,(A*24.0)/20); + double outlevel = B; + + 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. + } + + inputSampleL *= gain; + inputSampleR *= gain; + + if (flip) + { + iirSampleAL = (iirSampleAL * 0.9) + (inputSampleL * 0.1); + lowSampleL = iirSampleAL; + iirSampleAR = (iirSampleAR * 0.9) + (inputSampleR * 0.1); + lowSampleR = iirSampleAR; + } + else + { + iirSampleBL = (iirSampleBL * 0.9) + (inputSampleL * 0.1); + lowSampleL = iirSampleBL; + iirSampleBR = (iirSampleBR * 0.9) + (inputSampleR * 0.1); + lowSampleR = iirSampleBR; + } + highSampleL = inputSampleL - lowSampleL; + highSampleR = inputSampleR - lowSampleR; + flip = !flip; + //we now have two bands and the original source + + inP2 = lowSampleL * lowSampleL; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= basslevL; + //second harmonic max +1 + if (basslevL > 0) basslevL -= bassScaleback; + if (basslevL < 0) basslevL += bassScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(lowSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleL > 0.0) distSampleL = bridgerectifier; + else distSampleL = -bridgerectifier; + minusSampleL = lowSampleL - distSampleL; + plusSampleL = lowSampleL + distSampleL; + if (minusSampleL > maxfeedBass) minusSampleL = maxfeedBass; + if (plusSampleL > maxfeedBass) plusSampleL = maxfeedBass; + if (plusSampleL < -maxfeedBass) plusSampleL = -maxfeedBass; + if (minusSampleL < -maxfeedBass) minusSampleL = -maxfeedBass; + if (lowSampleL > distSampleL) basslevL += (minusSampleL*addBassBuss); + if (lowSampleL < -distSampleL) basslevL -= (plusSampleL*addBassBuss); + if (basslevL > 1.0) basslevL = 1.0; + if (basslevL < -1.0) basslevL = -1.0; + bridgerectifier = fabs(lowSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleL > 0.0) lowSampleL = bridgerectifier; + else lowSampleL = -bridgerectifier; + //apply the distortion transform for reals + lowSampleL /= (1.0+fabs(basslevL*limitingBass)); + lowSampleL += chebyshev; + //apply the correction measuresL + + inP2 = lowSampleR * lowSampleR; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= basslevR; + //second harmonic max +1 + if (basslevR > 0) basslevR -= bassScaleback; + if (basslevR < 0) basslevR += bassScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(lowSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleR > 0.0) distSampleR = bridgerectifier; + else distSampleR = -bridgerectifier; + minusSampleR = lowSampleR - distSampleR; + plusSampleR = lowSampleR + distSampleR; + if (minusSampleR > maxfeedBass) minusSampleR = maxfeedBass; + if (plusSampleR > maxfeedBass) plusSampleR = maxfeedBass; + if (plusSampleR < -maxfeedBass) plusSampleR = -maxfeedBass; + if (minusSampleR < -maxfeedBass) minusSampleR = -maxfeedBass; + if (lowSampleR > distSampleR) basslevR += (minusSampleR*addBassBuss); + if (lowSampleR < -distSampleR) basslevR -= (plusSampleR*addBassBuss); + if (basslevR > 1.0) basslevR = 1.0; + if (basslevR < -1.0) basslevR = -1.0; + bridgerectifier = fabs(lowSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (lowSampleR > 0.0) lowSampleR = bridgerectifier; + else lowSampleR = -bridgerectifier; + //apply the distortion transform for reals + lowSampleR /= (1.0+fabs(basslevR*limitingBass)); + lowSampleR += chebyshev; + //apply the correction measuresR + + inP2 = highSampleL * highSampleL; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= treblevL; + //second harmonic max +1 + if (treblevL > 0) treblevL -= trebleScaleback; + if (treblevL < 0) treblevL += trebleScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(highSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleL > 0.0) distSampleL = bridgerectifier; + else distSampleL = -bridgerectifier; + minusSampleL = highSampleL - distSampleL; + plusSampleL = highSampleL + distSampleL; + if (minusSampleL > maxfeedTreb) minusSampleL = maxfeedTreb; + if (plusSampleL > maxfeedTreb) plusSampleL = maxfeedTreb; + if (plusSampleL < -maxfeedTreb) plusSampleL = -maxfeedTreb; + if (minusSampleL < -maxfeedTreb) minusSampleL = -maxfeedTreb; + if (highSampleL > distSampleL) treblevL += (minusSampleL*addTrebBuss); + if (highSampleL < -distSampleL) treblevL -= (plusSampleL*addTrebBuss); + if (treblevL > 1.0) treblevL = 1.0; + if (treblevL < -1.0) treblevL = -1.0; + bridgerectifier = fabs(highSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleL > 0.0) highSampleL = bridgerectifier; + else highSampleL = -bridgerectifier; + //apply the distortion transform for reals + highSampleL /= (1.0+fabs(treblevL*limitingTreb)); + highSampleL += chebyshev; + //apply the correction measuresL + + inP2 = highSampleR * highSampleR; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= treblevR; + //second harmonic max +1 + if (treblevR > 0) treblevR -= trebleScaleback; + if (treblevR < 0) treblevR += trebleScaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(highSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleR > 0.0) distSampleR = bridgerectifier; + else distSampleR = -bridgerectifier; + minusSampleR = highSampleR - distSampleR; + plusSampleR = highSampleR + distSampleR; + if (minusSampleR > maxfeedTreb) minusSampleR = maxfeedTreb; + if (plusSampleR > maxfeedTreb) plusSampleR = maxfeedTreb; + if (plusSampleR < -maxfeedTreb) plusSampleR = -maxfeedTreb; + if (minusSampleR < -maxfeedTreb) minusSampleR = -maxfeedTreb; + if (highSampleR > distSampleR) treblevR += (minusSampleR*addTrebBuss); + if (highSampleR < -distSampleR) treblevR -= (plusSampleR*addTrebBuss); + if (treblevR > 1.0) treblevR = 1.0; + if (treblevR < -1.0) treblevR = -1.0; + bridgerectifier = fabs(highSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (highSampleR > 0.0) highSampleR = bridgerectifier; + else highSampleR = -bridgerectifier; + //apply the distortion transform for reals + highSampleR /= (1.0+fabs(treblevR*limitingTreb)); + highSampleR += chebyshev; + //apply the correction measuresR + + inputSampleL = lowSampleL + highSampleL; + inputSampleR = lowSampleR + highSampleR; + + inP2 = inputSampleL * inputSampleL; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= cheblevL; + //third harmonic max -1 + if (cheblevL > 0) cheblevL -= (IIRscaleback); + if (cheblevL < 0) cheblevL += (IIRscaleback); + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0.0) distSampleL = bridgerectifier; + else distSampleL = -bridgerectifier; + minusSampleL = inputSampleL - distSampleL; + plusSampleL = inputSampleL + distSampleL; + if (minusSampleL > maxfeed) minusSampleL = maxfeed; + if (plusSampleL > maxfeed) plusSampleL = maxfeed; + if (plusSampleL < -maxfeed) plusSampleL = -maxfeed; + if (minusSampleL < -maxfeed) minusSampleL = -maxfeed; + if (inputSampleL > distSampleL) cheblevL += (minusSampleL*addShortBuss); + if (inputSampleL < -distSampleL) cheblevL -= (plusSampleL*addShortBuss); + if (cheblevL > 1.0) cheblevL = 1.0; + if (cheblevL < -1.0) cheblevL = -1.0; + bridgerectifier = fabs(inputSampleL); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleL > 0.0) inputSampleL = bridgerectifier; + else inputSampleL = -bridgerectifier; + //apply the distortion transform for reals + inputSampleL /= (1.0+fabs(cheblevL*limiting)); + inputSampleL += chebyshev; + //apply the correction measuresL + + inP2 = inputSampleR * inputSampleR; + if (inP2 > 1.0) inP2 = 1.0; if (inP2 < -1.0) inP2 = -1.0; + chebyshev = (2 * inP2); + chebyshev *= cheblevR; + //third harmonic max -1 + if (cheblevR > 0) cheblevR -= IIRscaleback; + if (cheblevR < 0) cheblevR += IIRscaleback; + //this is ShortBuss, IIRscaleback is the decay speed. *2 for second harmonic, and so on + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0.0) distSampleR = bridgerectifier; + else distSampleR = -bridgerectifier; + minusSampleR = inputSampleR - distSampleR; + plusSampleR = inputSampleR + distSampleR; + if (minusSampleR > maxfeed) minusSampleR = maxfeed; + if (plusSampleR > maxfeed) plusSampleR = maxfeed; + if (plusSampleR < -maxfeed) plusSampleR = -maxfeed; + if (minusSampleR < -maxfeed) minusSampleR = -maxfeed; + if (inputSampleR > distSampleR) cheblevR += (minusSampleR*addShortBuss); + if (inputSampleR < -distSampleR) cheblevR -= (plusSampleR*addShortBuss); + if (cheblevR > 1.0) cheblevR = 1.0; + if (cheblevR < -1.0) cheblevR = -1.0; + bridgerectifier = fabs(inputSampleR); + if (bridgerectifier > 1.57079633) bridgerectifier = 1.57079633; + //max value for sine function + bridgerectifier = sin(bridgerectifier); + if (inputSampleR > 0.0) inputSampleR = bridgerectifier; + else inputSampleR = -bridgerectifier; + //apply the distortion transform for reals + inputSampleR /= (1.0+fabs(cheblevR*limiting)); + inputSampleR += chebyshev; + //apply the correction measuresR + + if (outlevel < 1.0) { + inputSampleL *= outlevel; + inputSampleR *= outlevel; + } + + if (inputSampleL > 0.95) inputSampleL = 0.95; + if (inputSampleL < -0.95) inputSampleL = -0.95; + if (inputSampleR > 0.95) inputSampleR = 0.95; + if (inputSampleR < -0.95) inputSampleR = -0.95; + //iron bar + + //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++; + } +}
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