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