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188 lines
7.1 KiB
C++
Executable file
188 lines
7.1 KiB
C++
Executable file
/* ========================================
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* Exciter - Exciter.h
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* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
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* ======================================== */
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#ifndef __Exciter_H
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#include "Exciter.h"
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#endif
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void Exciter::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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biquad[0] = ((A*7000.0)+8000.0)/getSampleRate();
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biquad[1] = A+B+0.7071;
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//tighter resonance as frequency and boost increases
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double boost = pow(B,2.0)*16.0;
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double K = tan(M_PI * biquad[0]);
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double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
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biquad[2] = K / 0.7071 * norm;
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biquad[4] = -biquad[2];
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biquad[5] = 2.0 * (K * K - 1.0) * norm;
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biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
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//bandpass to focus the intensity of the effect
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while (--sampleFrames >= 0)
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{
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double inputSampleL = *in1;
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double 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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double outSampleL = (inputSampleL * biquad[2]) + biquad[7];
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biquad[7] = (inputSampleL * biquad[3]) - (outSampleL * biquad[5]) + biquad[8];
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biquad[8] = (inputSampleL * biquad[4]) - (outSampleL * biquad[6]);
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double outSampleR = (inputSampleR * biquad[2]) + biquad[9];
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biquad[9] = (inputSampleR * biquad[3]) - (outSampleR * biquad[5]) + biquad[10];
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biquad[10] = (inputSampleR * biquad[4]) - (outSampleR * biquad[6]);
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//so the audio we're working with is going to be a bandpassed signal: only high mids
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outSampleL *= boost;
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if (outSampleL > M_PI) outSampleL = M_PI;
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if (outSampleL < -M_PI) outSampleL = -M_PI;
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outSampleL -= sin(outSampleL);
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outSampleR *= boost;
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if (outSampleR > M_PI) outSampleR = M_PI;
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if (outSampleR < -M_PI) outSampleR = -M_PI;
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outSampleR -= sin(outSampleR);
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//so we're clipping to generate artifacts, but subtracting them, meaning that
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//our deviations from the unclipped waveform are now negative in form.
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outSampleL *= boost;
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if (outSampleL > M_PI) outSampleL = M_PI;
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if (outSampleL < -M_PI) outSampleL = -M_PI;
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outSampleL = sin(outSampleL);
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outSampleR *= boost;
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if (outSampleR > M_PI) outSampleR = M_PI;
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if (outSampleR < -M_PI) outSampleR = -M_PI;
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outSampleR = sin(outSampleR);
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//now we're clipping the result, to make the peaks less intense
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inputSampleL -= outSampleL;
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inputSampleR -= outSampleR;
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//and we apply only those deviations from distorted high mids, by subtracting them from
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//the original full bandwidth audio. And that's how analog aural exciters basically work.
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//The real ones used a 4049 chip sometimes to produce the soft saturation we're getting with sin()
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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//and we'll do the harshest of hardclips to cope with how intense the new peaks can get,
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//without in any way dialing back the ruthless brightness the exciter can create.
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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 Exciter::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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biquad[0] = ((A*7000.0)+8000.0)/getSampleRate();
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biquad[1] = A+B+0.7071;
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//tighter resonance as frequency and boost increases
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double boost = pow(B,2.0)*16.0;
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double K = tan(M_PI * biquad[0]);
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double norm = 1.0 / (1.0 + K / biquad[1] + K * K);
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biquad[2] = K / 0.7071 * norm;
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biquad[4] = -biquad[2];
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biquad[5] = 2.0 * (K * K - 1.0) * norm;
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biquad[6] = (1.0 - K / biquad[1] + K * K) * norm;
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//bandpass to focus the intensity of the effect
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while (--sampleFrames >= 0)
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{
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double inputSampleL = *in1;
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double 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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double outSampleL = (inputSampleL * biquad[2]) + biquad[7];
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biquad[7] = (inputSampleL * biquad[3]) - (outSampleL * biquad[5]) + biquad[8];
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biquad[8] = (inputSampleL * biquad[4]) - (outSampleL * biquad[6]);
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double outSampleR = (inputSampleR * biquad[2]) + biquad[9];
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biquad[9] = (inputSampleR * biquad[3]) - (outSampleR * biquad[5]) + biquad[10];
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biquad[10] = (inputSampleR * biquad[4]) - (outSampleR * biquad[6]);
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//so the audio we're working with is going to be a bandpassed signal: only high mids
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outSampleL *= boost;
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if (outSampleL > M_PI) outSampleL = M_PI;
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if (outSampleL < -M_PI) outSampleL = -M_PI;
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outSampleL -= sin(outSampleL);
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outSampleR *= boost;
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if (outSampleR > M_PI) outSampleR = M_PI;
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if (outSampleR < -M_PI) outSampleR = -M_PI;
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outSampleR -= sin(outSampleR);
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//so we're clipping to generate artifacts, but subtracting them, meaning that
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//our deviations from the unclipped waveform are now negative in form.
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outSampleL *= boost;
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if (outSampleL > M_PI) outSampleL = M_PI;
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if (outSampleL < -M_PI) outSampleL = -M_PI;
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outSampleL = sin(outSampleL);
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outSampleR *= boost;
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if (outSampleR > M_PI) outSampleR = M_PI;
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if (outSampleR < -M_PI) outSampleR = -M_PI;
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outSampleR = sin(outSampleR);
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//now we're clipping the result, to make the peaks less intense
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inputSampleL -= outSampleL;
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inputSampleR -= outSampleR;
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//and we apply only those deviations from distorted high mids, by subtracting them from
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//the original full bandwidth audio. And that's how analog aural exciters basically work.
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//The real ones used a 4049 chip sometimes to produce the soft saturation we're getting with sin()
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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//and we'll do the harshest of hardclips to cope with how intense the new peaks can get,
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//without in any way dialing back the ruthless brightness the exciter can create.
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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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}
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