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Diffstat (limited to 'plugins/WinVST/Average/AverageProc.cpp')
| -rwxr-xr-x | plugins/WinVST/Average/AverageProc.cpp | 361 |
1 files changed, 361 insertions, 0 deletions
diff --git a/plugins/WinVST/Average/AverageProc.cpp b/plugins/WinVST/Average/AverageProc.cpp new file mode 100755 index 0000000..2b1c355 --- /dev/null +++ b/plugins/WinVST/Average/AverageProc.cpp @@ -0,0 +1,361 @@ +/* ======================================== + * Average - Average.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Average_H +#include "Average.h" +#endif + +void Average::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double correctionSample; + double accumulatorSampleL; + double accumulatorSampleR; + double drySampleL; + double drySampleR; + double inputSampleL; + double inputSampleR; + + double overallscale = (A * 9.0)+1.0; + double wet = B; + double dry = 1.0 - wet; + double gain = overallscale; + + if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} + if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} + if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} + if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} + if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} + if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} + if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} + if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} + if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} + if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + + if (overallscale < 1.0) overallscale = 1.0; + f[0] /= overallscale; + f[1] /= overallscale; + f[2] /= overallscale; + f[3] /= overallscale; + f[4] /= overallscale; + f[5] /= overallscale; + f[6] /= overallscale; + f[7] /= overallscale; + f[8] /= overallscale; + f[9] /= overallscale; + //and now it's neatly scaled, too + + 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. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + 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] = accumulatorSampleL = inputSampleL; + + 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] = accumulatorSampleR = inputSampleR; + //primitive way of doing this: for larger batches of samples, you might + //try using a circular buffer like in a reverb. If you add the new sample + //and subtract the one on the end you can keep a running tally of the samples + //between. Beware of tiny floating-point math errors eventually screwing up + //your system, though! + + accumulatorSampleL *= f[0]; + accumulatorSampleL += (bL[1] * f[1]); + accumulatorSampleL += (bL[2] * f[2]); + accumulatorSampleL += (bL[3] * f[3]); + accumulatorSampleL += (bL[4] * f[4]); + accumulatorSampleL += (bL[5] * f[5]); + accumulatorSampleL += (bL[6] * f[6]); + accumulatorSampleL += (bL[7] * f[7]); + accumulatorSampleL += (bL[8] * f[8]); + accumulatorSampleL += (bL[9] * f[9]); + + accumulatorSampleR *= f[0]; + accumulatorSampleR += (bR[1] * f[1]); + accumulatorSampleR += (bR[2] * f[2]); + accumulatorSampleR += (bR[3] * f[3]); + accumulatorSampleR += (bR[4] * f[4]); + accumulatorSampleR += (bR[5] * f[5]); + accumulatorSampleR += (bR[6] * f[6]); + accumulatorSampleR += (bR[7] * f[7]); + accumulatorSampleR += (bR[8] * f[8]); + accumulatorSampleR += (bR[9] * f[9]); + //we are doing our repetitive calculations on a separate value + + correctionSample = inputSampleL - accumulatorSampleL; + //we're gonna apply the total effect of all these calculations as a single subtract + inputSampleL -= correctionSample; + + correctionSample = inputSampleR - accumulatorSampleR; + inputSampleR -= correctionSample; + //our one math operation on the input data coming in + + if (wet < 1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + //dry/wet control only applies if you're using it. We don't do a multiply by 1.0 + //if it 'won't change anything' but our sample might be at a very different scaling + //in the floating point system. + + + //noise shaping to 32-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 32 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Average::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + double correctionSample; + double accumulatorSampleL; + double accumulatorSampleR; + double drySampleL; + double drySampleR; + double inputSampleL; + double inputSampleR; + + double overallscale = (A * 9.0)+1.0; + double wet = B; + double dry = 1.0 - wet; + double gain = overallscale; + + if (gain > 1.0) {f[0] = 1.0; gain -= 1.0;} else {f[0] = gain; gain = 0.0;} + if (gain > 1.0) {f[1] = 1.0; gain -= 1.0;} else {f[1] = gain; gain = 0.0;} + if (gain > 1.0) {f[2] = 1.0; gain -= 1.0;} else {f[2] = gain; gain = 0.0;} + if (gain > 1.0) {f[3] = 1.0; gain -= 1.0;} else {f[3] = gain; gain = 0.0;} + if (gain > 1.0) {f[4] = 1.0; gain -= 1.0;} else {f[4] = gain; gain = 0.0;} + if (gain > 1.0) {f[5] = 1.0; gain -= 1.0;} else {f[5] = gain; gain = 0.0;} + if (gain > 1.0) {f[6] = 1.0; gain -= 1.0;} else {f[6] = gain; gain = 0.0;} + if (gain > 1.0) {f[7] = 1.0; gain -= 1.0;} else {f[7] = gain; gain = 0.0;} + if (gain > 1.0) {f[8] = 1.0; gain -= 1.0;} else {f[8] = gain; gain = 0.0;} + if (gain > 1.0) {f[9] = 1.0; gain -= 1.0;} else {f[9] = gain; gain = 0.0;} + //there, now we have a neat little moving average with remainders + + if (overallscale < 1.0) overallscale = 1.0; + f[0] /= overallscale; + f[1] /= overallscale; + f[2] /= overallscale; + f[3] /= overallscale; + f[4] /= overallscale; + f[5] /= overallscale; + f[6] /= overallscale; + f[7] /= overallscale; + f[8] /= overallscale; + f[9] /= overallscale; + //and now it's neatly scaled, too + + 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. + } + drySampleL = inputSampleL; + drySampleR = inputSampleR; + + 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] = accumulatorSampleL = inputSampleL; + + 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] = accumulatorSampleR = inputSampleR; + //primitive way of doing this: for larger batches of samples, you might + //try using a circular buffer like in a reverb. If you add the new sample + //and subtract the one on the end you can keep a running tally of the samples + //between. Beware of tiny floating-point math errors eventually screwing up + //your system, though! + + accumulatorSampleL *= f[0]; + accumulatorSampleL += (bL[1] * f[1]); + accumulatorSampleL += (bL[2] * f[2]); + accumulatorSampleL += (bL[3] * f[3]); + accumulatorSampleL += (bL[4] * f[4]); + accumulatorSampleL += (bL[5] * f[5]); + accumulatorSampleL += (bL[6] * f[6]); + accumulatorSampleL += (bL[7] * f[7]); + accumulatorSampleL += (bL[8] * f[8]); + accumulatorSampleL += (bL[9] * f[9]); + + accumulatorSampleR *= f[0]; + accumulatorSampleR += (bR[1] * f[1]); + accumulatorSampleR += (bR[2] * f[2]); + accumulatorSampleR += (bR[3] * f[3]); + accumulatorSampleR += (bR[4] * f[4]); + accumulatorSampleR += (bR[5] * f[5]); + accumulatorSampleR += (bR[6] * f[6]); + accumulatorSampleR += (bR[7] * f[7]); + accumulatorSampleR += (bR[8] * f[8]); + accumulatorSampleR += (bR[9] * f[9]); + //we are doing our repetitive calculations on a separate value + + correctionSample = inputSampleL - accumulatorSampleL; + //we're gonna apply the total effect of all these calculations as a single subtract + inputSampleL -= correctionSample; + + correctionSample = inputSampleR - accumulatorSampleR; + inputSampleR -= correctionSample; + //our one math operation on the input data coming in + + if (wet < 1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * dry); + inputSampleR = (inputSampleR * wet) + (drySampleR * dry); + } + //dry/wet control only applies if you're using it. We don't do a multiply by 1.0 + //if it 'won't change anything' but our sample might be at a very different scaling + //in the floating point system. + + //noise shaping to 64-bit floating point + if (fpFlip) { + fpTemp = inputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLA; + fpTemp = inputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRA; + } + else { + fpTemp = inputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((inputSampleL-fpTemp)*fpNew); + inputSampleL += fpNShapeLB; + fpTemp = inputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((inputSampleR-fpTemp)*fpNew); + inputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 64 bit output + + *out1 = inputSampleL; + *out2 = inputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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