/* ======================================== * 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]; 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. //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 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 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. //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++; } }