/* ======================================== * PurestEcho - PurestEcho.h * Copyright (c) 2016 airwindows, All rights reserved * ======================================== */ #ifndef __PurestEcho_H #include "PurestEcho.h" #endif void PurestEcho::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) { float* in1 = inputs[0]; float* in2 = inputs[1]; float* out1 = outputs[0]; float* out2 = outputs[1]; int loopLimit = (int)(totalsamples * 0.499); //this is a double buffer so we will be splitting it in two double time = pow(A,2) * 0.999; double tap1 = B; double tap2 = C; double tap3 = D; double tap4 = E; double gainTrim = 1.0 / (1.0 + tap1 + tap2 + tap3 + tap4); //this becomes our equal-loudness mechanic. 0.2 to 1.0 gain on all things. double tapsTrim = gainTrim * 0.5; //the taps interpolate and require half that gain: 0.1 to 0.5 on all taps. int position1 = (int)(loopLimit * time * 0.25); int position2 = (int)(loopLimit * time * 0.5); int position3 = (int)(loopLimit * time * 0.75); int position4 = (int)(loopLimit * time); //basic echo information: we're taking four equally spaced echoes and setting their levels as desired. //position4 is what you'd have for 'just set a delay time' double volAfter1 = (loopLimit * time * 0.25) - position1; double volAfter2 = (loopLimit * time * 0.5) - position2; double volAfter3 = (loopLimit * time * 0.75) - position3; double volAfter4 = (loopLimit * time) - position4; //these are 0-1: casting to an (int) truncates fractional numbers towards zero (and is faster than floor() ) //so, when we take the integer number (all above zero) and subtract it from the real value, we get 0-1 double volBefore1 = (1.0 - volAfter1) * tap1; double volBefore2 = (1.0 - volAfter2) * tap2; double volBefore3 = (1.0 - volAfter3) * tap3; double volBefore4 = (1.0 - volAfter4) * tap4; //and if we are including a bit of the previous/next sample to interpolate, then if the sample position is 1.0001 //we'll be leaning most heavily on the 'before' sample which is nearer to us, and the 'after' sample is almost not used. //if the sample position is 1.9999, the 'after' sample is strong and 'before' is almost not used. volAfter1 *= tap1; volAfter2 *= tap2; volAfter3 *= tap3; volAfter4 *= tap4; //and like with volBefore, we also want to scale this 'interpolate' to the loudness of this tap. //We do it here because we can do it only once per audio buffer, not on every sample. This assumes we're //not moving the tap every sample: if so we'd have to do this every sample as well. int oneBefore1 = position1 - 1; int oneBefore2 = position2 - 1; int oneBefore3 = position3 - 1; int oneBefore4 = position4 - 1; if (oneBefore1 < 0) oneBefore1 = 0; if (oneBefore2 < 0) oneBefore2 = 0; if (oneBefore3 < 0) oneBefore3 = 0; if (oneBefore4 < 0) oneBefore4 = 0; int oneAfter1 = position1 + 1; int oneAfter2 = position2 + 1; int oneAfter3 = position3 + 1; int oneAfter4 = position4 + 1; //this is setting up the way we interpolate samples: we're doing an echo-darkening thing //to make it sound better. Pretty much no acoustic delay in human-breathable air will give //you zero attenuation at 22 kilohertz: forget this at your peril ;) double delaysBufferL; double delaysBufferR; long double inputSampleL; long double inputSampleR; 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. } if (gcount < 0 || gcount > loopLimit) gcount = loopLimit; dL[gcount+loopLimit] = dL[gcount] = inputSampleL * tapsTrim; dR[gcount+loopLimit] = dR[gcount] = inputSampleR * tapsTrim; //this is how the double buffer works: //we can look for delay taps without ever having to 'wrap around' within our calculation. //As long as the delay tap is less than our loop limit we can always just add it to where we're //at, and get a valid sample back right away, no matter where we are in the buffer. //The 0.5 is taking into account the interpolation, by padding down the whole buffer. delaysBufferL = (dL[gcount+oneBefore4]*volBefore4); delaysBufferL += (dL[gcount+oneAfter4]*volAfter4); delaysBufferL += (dL[gcount+oneBefore3]*volBefore3); delaysBufferL += (dL[gcount+oneAfter3]*volAfter3); delaysBufferL += (dL[gcount+oneBefore2]*volBefore2); delaysBufferL += (dL[gcount+oneAfter2]*volAfter2); delaysBufferL += (dL[gcount+oneBefore1]*volBefore1); delaysBufferL += (dL[gcount+oneAfter1]*volAfter1); delaysBufferR = (dR[gcount+oneBefore4]*volBefore4); delaysBufferR += (dR[gcount+oneAfter4]*volAfter4); delaysBufferR += (dR[gcount+oneBefore3]*volBefore3); delaysBufferR += (dR[gcount+oneAfter3]*volAfter3); delaysBufferR += (dR[gcount+oneBefore2]*volBefore2); delaysBufferR += (dR[gcount+oneAfter2]*volAfter2); delaysBufferR += (dR[gcount+oneBefore1]*volBefore1); delaysBufferR += (dR[gcount+oneAfter1]*volAfter1); //These are the interpolated samples. We're adding them first, because we know they're smaller //and while the value of delaysBuffer is small we'll add similarly small values to it. Note the order. delaysBufferL += (dL[gcount+position4]*tap4); delaysBufferL += (dL[gcount+position3]*tap3); delaysBufferL += (dL[gcount+position2]*tap2); delaysBufferL += (dL[gcount+position1]*tap1); delaysBufferR += (dR[gcount+position4]*tap4); delaysBufferR += (dR[gcount+position3]*tap3); delaysBufferR += (dR[gcount+position2]*tap2); delaysBufferR += (dR[gcount+position1]*tap1); //These are the primary samples for the echo, and we're adding them last. As before we're starting with the //most delayed echoes, and ending with what we think might be the loudest echo. We're building this delaybuffer //from the faintest noises to the loudest, to avoid adding a bunch of teeny values at the end. //You can of course put the last echo as loudest, but with diminishing echo volumes this is optimal. //This technique is also present in other plugins such as Iron Oxide. inputSampleL = (inputSampleL * gainTrim) + delaysBufferL; inputSampleR = (inputSampleR * gainTrim) + delaysBufferR; //this could be just inputSample += d[gcount+position1]; //for literally a single, full volume echo combined with dry. //What I'm doing is making the echoes more interesting. gcount--; //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 PurestEcho::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) { double* in1 = inputs[0]; double* in2 = inputs[1]; double* out1 = outputs[0]; double* out2 = outputs[1]; int loopLimit = (int)(totalsamples * 0.499); //this is a double buffer so we will be splitting it in two double time = pow(A,2) * 0.999; double tap1 = B; double tap2 = C; double tap3 = D; double tap4 = E; double gainTrim = 1.0 / (1.0 + tap1 + tap2 + tap3 + tap4); //this becomes our equal-loudness mechanic. 0.2 to 1.0 gain on all things. double tapsTrim = gainTrim * 0.5; //the taps interpolate and require half that gain: 0.1 to 0.5 on all taps. int position1 = (int)(loopLimit * time * 0.25); int position2 = (int)(loopLimit * time * 0.5); int position3 = (int)(loopLimit * time * 0.75); int position4 = (int)(loopLimit * time); //basic echo information: we're taking four equally spaced echoes and setting their levels as desired. //position4 is what you'd have for 'just set a delay time' double volAfter1 = (loopLimit * time * 0.25) - position1; double volAfter2 = (loopLimit * time * 0.5) - position2; double volAfter3 = (loopLimit * time * 0.75) - position3; double volAfter4 = (loopLimit * time) - position4; //these are 0-1: casting to an (int) truncates fractional numbers towards zero (and is faster than floor() ) //so, when we take the integer number (all above zero) and subtract it from the real value, we get 0-1 double volBefore1 = (1.0 - volAfter1) * tap1; double volBefore2 = (1.0 - volAfter2) * tap2; double volBefore3 = (1.0 - volAfter3) * tap3; double volBefore4 = (1.0 - volAfter4) * tap4; //and if we are including a bit of the previous/next sample to interpolate, then if the sample position is 1.0001 //we'll be leaning most heavily on the 'before' sample which is nearer to us, and the 'after' sample is almost not used. //if the sample position is 1.9999, the 'after' sample is strong and 'before' is almost not used. volAfter1 *= tap1; volAfter2 *= tap2; volAfter3 *= tap3; volAfter4 *= tap4; //and like with volBefore, we also want to scale this 'interpolate' to the loudness of this tap. //We do it here because we can do it only once per audio buffer, not on every sample. This assumes we're //not moving the tap every sample: if so we'd have to do this every sample as well. int oneBefore1 = position1 - 1; int oneBefore2 = position2 - 1; int oneBefore3 = position3 - 1; int oneBefore4 = position4 - 1; if (oneBefore1 < 0) oneBefore1 = 0; if (oneBefore2 < 0) oneBefore2 = 0; if (oneBefore3 < 0) oneBefore3 = 0; if (oneBefore4 < 0) oneBefore4 = 0; int oneAfter1 = position1 + 1; int oneAfter2 = position2 + 1; int oneAfter3 = position3 + 1; int oneAfter4 = position4 + 1; //this is setting up the way we interpolate samples: we're doing an echo-darkening thing //to make it sound better. Pretty much no acoustic delay in human-breathable air will give //you zero attenuation at 22 kilohertz: forget this at your peril ;) double delaysBufferL; double delaysBufferR; long double inputSampleL; long double inputSampleR; 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. } if (gcount < 0 || gcount > loopLimit) gcount = loopLimit; dL[gcount+loopLimit] = dL[gcount] = inputSampleL * tapsTrim; dR[gcount+loopLimit] = dR[gcount] = inputSampleR * tapsTrim; //this is how the double buffer works: //we can look for delay taps without ever having to 'wrap around' within our calculation. //As long as the delay tap is less than our loop limit we can always just add it to where we're //at, and get a valid sample back right away, no matter where we are in the buffer. //The 0.5 is taking into account the interpolation, by padding down the whole buffer. delaysBufferL = (dL[gcount+oneBefore4]*volBefore4); delaysBufferL += (dL[gcount+oneAfter4]*volAfter4); delaysBufferL += (dL[gcount+oneBefore3]*volBefore3); delaysBufferL += (dL[gcount+oneAfter3]*volAfter3); delaysBufferL += (dL[gcount+oneBefore2]*volBefore2); delaysBufferL += (dL[gcount+oneAfter2]*volAfter2); delaysBufferL += (dL[gcount+oneBefore1]*volBefore1); delaysBufferL += (dL[gcount+oneAfter1]*volAfter1); delaysBufferR = (dR[gcount+oneBefore4]*volBefore4); delaysBufferR += (dR[gcount+oneAfter4]*volAfter4); delaysBufferR += (dR[gcount+oneBefore3]*volBefore3); delaysBufferR += (dR[gcount+oneAfter3]*volAfter3); delaysBufferR += (dR[gcount+oneBefore2]*volBefore2); delaysBufferR += (dR[gcount+oneAfter2]*volAfter2); delaysBufferR += (dR[gcount+oneBefore1]*volBefore1); delaysBufferR += (dR[gcount+oneAfter1]*volAfter1); //These are the interpolated samples. We're adding them first, because we know they're smaller //and while the value of delaysBuffer is small we'll add similarly small values to it. Note the order. delaysBufferL += (dL[gcount+position4]*tap4); delaysBufferL += (dL[gcount+position3]*tap3); delaysBufferL += (dL[gcount+position2]*tap2); delaysBufferL += (dL[gcount+position1]*tap1); delaysBufferR += (dR[gcount+position4]*tap4); delaysBufferR += (dR[gcount+position3]*tap3); delaysBufferR += (dR[gcount+position2]*tap2); delaysBufferR += (dR[gcount+position1]*tap1); //These are the primary samples for the echo, and we're adding them last. As before we're starting with the //most delayed echoes, and ending with what we think might be the loudest echo. We're building this delaybuffer //from the faintest noises to the loudest, to avoid adding a bunch of teeny values at the end. //You can of course put the last echo as loudest, but with diminishing echo volumes this is optimal. //This technique is also present in other plugins such as Iron Oxide. inputSampleL = (inputSampleL * gainTrim) + delaysBufferL; inputSampleR = (inputSampleR * gainTrim) + delaysBufferR; //this could be just inputSample += d[gcount+position1]; //for literally a single, full volume echo combined with dry. //What I'm doing is making the echoes more interesting. gcount--; //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++; } }