mirror of
https://github.com/airwindows/airwindows.git
synced 2026-05-15 22:01:19 -06:00
406 lines
No EOL
12 KiB
C++
Executable file
406 lines
No EOL
12 KiB
C++
Executable file
/* ========================================
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* Thunder - Thunder.h
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* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
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* ======================================== */
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#ifndef __Thunder_H
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#include "Thunder.h"
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#endif
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void Thunder::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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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double thunder = A * 0.4;
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double threshold = 1.0 - (thunder * 2.0);
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if (threshold < 0.01) threshold = 0.01;
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double muMakeupGain = 1.0 / threshold;
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double release = pow((1.28-thunder),5)*32768.0;
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release /= overallscale;
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double fastest = sqrt(release);
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double EQ = ((0.0275 / getSampleRate())*32000.0);
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double dcblock = EQ / 300.0;
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double basstrim = (0.01/EQ)+1.0;
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//FF parameters also ride off Speed
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double outputGain = B;
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double coefficient;
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double inputSense;
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double resultL;
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double resultR;
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double resultM;
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double resultML;
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double resultMR;
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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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inputSampleL = inputSampleL * muMakeupGain;
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inputSampleR = inputSampleR * muMakeupGain;
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if (gateL < fabs(inputSampleL)) gateL = inputSampleL;
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else gateL -= dcblock;
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if (gateR < fabs(inputSampleR)) gateR = inputSampleR;
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else gateR -= dcblock;
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//setting up gated DC blocking to control the tendency for rumble and offset
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//begin three FathomFive stages
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iirSampleAL += (inputSampleL * EQ * thunder);
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iirSampleAL -= (iirSampleAL * iirSampleAL * iirSampleAL * EQ);
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if (iirSampleAL > gateL) iirSampleAL -= dcblock;
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if (iirSampleAL < -gateL) iirSampleAL += dcblock;
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resultL = iirSampleAL*basstrim;
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iirSampleBL = (iirSampleBL * (1 - EQ)) + (resultL * EQ);
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resultL = iirSampleBL;
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iirSampleAR += (inputSampleR * EQ * thunder);
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iirSampleAR -= (iirSampleAR * iirSampleAR * iirSampleAR * EQ);
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if (iirSampleAR > gateR) iirSampleAR -= dcblock;
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if (iirSampleAR < -gateR) iirSampleAR += dcblock;
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resultR = iirSampleAR*basstrim;
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iirSampleBR = (iirSampleBR * (1 - EQ)) + (resultR * EQ);
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resultR = iirSampleBR;
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iirSampleAM += ((inputSampleL + inputSampleR) * EQ * thunder);
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iirSampleAM -= (iirSampleAM * iirSampleAM * iirSampleAM * EQ);
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resultM = iirSampleAM*basstrim;
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iirSampleBM = (iirSampleBM * (1 - EQ)) + (resultM * EQ);
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resultM = iirSampleBM;
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iirSampleCM = (iirSampleCM * (1 - EQ)) + (resultM * EQ);
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resultM = fabs(iirSampleCM);
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resultML = fabs(resultL);
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resultMR = fabs(resultR);
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if (resultM > resultML) resultML = resultM;
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if (resultM > resultMR) resultMR = resultM;
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//trying to restrict the buzziness
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if (resultML > 1.0) resultML = 1.0;
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if (resultMR > 1.0) resultMR = 1.0;
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//now we have result L, R and M the trigger modulator which must be 0-1
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//begin compressor section
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inputSampleL -= (iirSampleBL * thunder);
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inputSampleR -= (iirSampleBR * thunder);
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//highpass the comp section by sneaking out what will be the reinforcement
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inputSense = fabs(inputSampleL);
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if (fabs(inputSampleR) > inputSense)
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inputSense = fabs(inputSampleR);
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//we will take the greater of either channel and just use that, then apply the result
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//to both stereo channels.
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if (flip)
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{
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if (inputSense > threshold)
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{
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muVary = threshold / inputSense;
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muAttack = sqrt(fabs(muSpeedA));
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muCoefficientA = muCoefficientA * (muAttack-1.0);
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if (muVary < threshold)
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{
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muCoefficientA = muCoefficientA + threshold;
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}
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else
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{
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muCoefficientA = muCoefficientA + muVary;
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}
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muCoefficientA = muCoefficientA / muAttack;
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}
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else
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{
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muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0);
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muCoefficientA = muCoefficientA + 1.0;
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muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA);
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}
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muNewSpeed = muSpeedA * (muSpeedA-1);
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muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
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muSpeedA = muNewSpeed / muSpeedA;
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}
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else
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{
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if (inputSense > threshold)
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{
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muVary = threshold / inputSense;
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muAttack = sqrt(fabs(muSpeedB));
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muCoefficientB = muCoefficientB * (muAttack-1);
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if (muVary < threshold)
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{
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muCoefficientB = muCoefficientB + threshold;
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}
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else
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{
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muCoefficientB = muCoefficientB + muVary;
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}
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muCoefficientB = muCoefficientB / muAttack;
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}
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else
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{
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muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0);
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muCoefficientB = muCoefficientB + 1.0;
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muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB);
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}
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muNewSpeed = muSpeedB * (muSpeedB-1);
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muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
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muSpeedB = muNewSpeed / muSpeedB;
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}
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//got coefficients, adjusted speeds
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if (flip)
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{
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coefficient = pow(muCoefficientA,2);
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inputSampleL *= coefficient;
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inputSampleR *= coefficient;
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}
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else
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{
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coefficient = pow(muCoefficientB,2);
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inputSampleL *= coefficient;
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inputSampleR *= coefficient;
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}
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//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
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//applied gain correction to control output level- tends to constrain sound rather than inflate it
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inputSampleL += (resultL * resultM);
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inputSampleR += (resultR * resultM);
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//combine the two by adding the summed channnel of lows
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if (outputGain != 1.0) {
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inputSampleL *= outputGain;
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inputSampleR *= outputGain;
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}
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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 Thunder::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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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= getSampleRate();
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double thunder = A * 0.4;
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double threshold = 1.0 - (thunder * 2.0);
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if (threshold < 0.01) threshold = 0.01;
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double muMakeupGain = 1.0 / threshold;
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double release = pow((1.28-thunder),5)*32768.0;
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release /= overallscale;
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double fastest = sqrt(release);
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double EQ = ((0.0275 / getSampleRate())*32000.0);
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double dcblock = EQ / 300.0;
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double basstrim = (0.01/EQ)+1.0;
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//FF parameters also ride off Speed
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double outputGain = B;
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double coefficient;
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double inputSense;
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double resultL;
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double resultR;
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double resultM;
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double resultML;
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double resultMR;
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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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inputSampleL = inputSampleL * muMakeupGain;
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inputSampleR = inputSampleR * muMakeupGain;
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if (gateL < fabs(inputSampleL)) gateL = inputSampleL;
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else gateL -= dcblock;
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if (gateR < fabs(inputSampleR)) gateR = inputSampleR;
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else gateR -= dcblock;
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//setting up gated DC blocking to control the tendency for rumble and offset
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//begin three FathomFive stages
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iirSampleAL += (inputSampleL * EQ * thunder);
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iirSampleAL -= (iirSampleAL * iirSampleAL * iirSampleAL * EQ);
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if (iirSampleAL > gateL) iirSampleAL -= dcblock;
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if (iirSampleAL < -gateL) iirSampleAL += dcblock;
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resultL = iirSampleAL*basstrim;
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iirSampleBL = (iirSampleBL * (1 - EQ)) + (resultL * EQ);
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resultL = iirSampleBL;
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iirSampleAR += (inputSampleR * EQ * thunder);
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iirSampleAR -= (iirSampleAR * iirSampleAR * iirSampleAR * EQ);
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if (iirSampleAR > gateR) iirSampleAR -= dcblock;
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if (iirSampleAR < -gateR) iirSampleAR += dcblock;
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resultR = iirSampleAR*basstrim;
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iirSampleBR = (iirSampleBR * (1 - EQ)) + (resultR * EQ);
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resultR = iirSampleBR;
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iirSampleAM += ((inputSampleL + inputSampleR) * EQ * thunder);
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iirSampleAM -= (iirSampleAM * iirSampleAM * iirSampleAM * EQ);
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resultM = iirSampleAM*basstrim;
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iirSampleBM = (iirSampleBM * (1 - EQ)) + (resultM * EQ);
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resultM = iirSampleBM;
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iirSampleCM = (iirSampleCM * (1 - EQ)) + (resultM * EQ);
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resultM = fabs(iirSampleCM);
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resultML = fabs(resultL);
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resultMR = fabs(resultR);
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if (resultM > resultML) resultML = resultM;
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if (resultM > resultMR) resultMR = resultM;
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//trying to restrict the buzziness
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if (resultML > 1.0) resultML = 1.0;
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if (resultMR > 1.0) resultMR = 1.0;
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//now we have result L, R and M the trigger modulator which must be 0-1
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//begin compressor section
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inputSampleL -= (iirSampleBL * thunder);
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inputSampleR -= (iirSampleBR * thunder);
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//highpass the comp section by sneaking out what will be the reinforcement
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inputSense = fabs(inputSampleL);
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if (fabs(inputSampleR) > inputSense)
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inputSense = fabs(inputSampleR);
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//we will take the greater of either channel and just use that, then apply the result
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//to both stereo channels.
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if (flip)
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{
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if (inputSense > threshold)
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{
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muVary = threshold / inputSense;
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muAttack = sqrt(fabs(muSpeedA));
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muCoefficientA = muCoefficientA * (muAttack-1.0);
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if (muVary < threshold)
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{
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muCoefficientA = muCoefficientA + threshold;
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}
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else
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{
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muCoefficientA = muCoefficientA + muVary;
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}
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muCoefficientA = muCoefficientA / muAttack;
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}
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else
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{
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muCoefficientA = muCoefficientA * ((muSpeedA * muSpeedA)-1.0);
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muCoefficientA = muCoefficientA + 1.0;
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muCoefficientA = muCoefficientA / (muSpeedA * muSpeedA);
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}
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muNewSpeed = muSpeedA * (muSpeedA-1);
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muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
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muSpeedA = muNewSpeed / muSpeedA;
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}
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else
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{
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if (inputSense > threshold)
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{
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muVary = threshold / inputSense;
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muAttack = sqrt(fabs(muSpeedB));
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muCoefficientB = muCoefficientB * (muAttack-1);
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if (muVary < threshold)
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{
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muCoefficientB = muCoefficientB + threshold;
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}
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else
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{
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muCoefficientB = muCoefficientB + muVary;
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}
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muCoefficientB = muCoefficientB / muAttack;
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}
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else
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{
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muCoefficientB = muCoefficientB * ((muSpeedB * muSpeedB)-1.0);
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muCoefficientB = muCoefficientB + 1.0;
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muCoefficientB = muCoefficientB / (muSpeedB * muSpeedB);
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}
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muNewSpeed = muSpeedB * (muSpeedB-1);
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muNewSpeed = muNewSpeed + fabs(inputSense*release)+fastest;
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muSpeedB = muNewSpeed / muSpeedB;
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}
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//got coefficients, adjusted speeds
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if (flip)
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{
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coefficient = pow(muCoefficientA,2);
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inputSampleL *= coefficient;
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inputSampleR *= coefficient;
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}
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else
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{
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coefficient = pow(muCoefficientB,2);
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inputSampleL *= coefficient;
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inputSampleR *= coefficient;
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}
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//applied compression with vari-vari-µ-µ-µ-µ-µ-µ-is-the-kitten-song o/~
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//applied gain correction to control output level- tends to constrain sound rather than inflate it
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inputSampleL += (resultL * resultM);
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inputSampleR += (resultR * resultM);
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//combine the two by adding the summed channnel of lows
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if (outputGain != 1.0) {
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inputSampleL *= outputGain;
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inputSampleR *= outputGain;
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}
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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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} |