mirror of
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214 lines
10 KiB
C++
Executable file
214 lines
10 KiB
C++
Executable file
/* ========================================
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* VoiceTrick - VoiceTrick.h
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* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
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* ======================================== */
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#ifndef __VoiceTrick_H
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#include "VoiceTrick.h"
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#endif
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void VoiceTrick::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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lowpassChase = pow(A,2);
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//should not scale with sample rate, because values reaching 1 are important
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//to its ability to bypass when set to max
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double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0);
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lastLowpass = lowpassChase;
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double invLowpass;
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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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lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount;
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//setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control,
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//but if I say it's the lowpass out of Capacitor it should literally be that in every behavior.
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double inputSample = (inputSampleL + inputSampleR) * 0.5;
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//this is now our mono audio
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count++; if (count > 5) count = 0; switch (count)
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{
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case 0:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
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iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
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break;
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case 1:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
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iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
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break;
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case 2:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
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iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
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break;
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case 3:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
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iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
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break;
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case 4:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
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iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
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break;
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case 5:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
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iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
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break;
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}
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//Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively
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//steepens the filter after minimizing artifacts.
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inputSampleL = -inputSample;
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inputSampleR = inputSample;
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//and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone.
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//The purpose of all this is to allow for recording of lead vocals without use of headphones:
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//or at least sealed headphones. You should be able to use this to record vocals with either
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//open-back headphones, or literally speakers in the room so long as the mic is exactly
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//equidistant from each speaker/headphone side.
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//You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor
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//only reverb and echo for vibe. Direct sound is the singer's direct sound.
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//The filtering is because, if you use open-back headphones and move your head, highs will
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//bleed through first like a through-zero flange coming out of cancellation (which it is).
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//Therefore, you can filter off highs until the bleed isn't annoying.
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//Or just run with it, it shouldn't be that loud.
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//Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :)
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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 VoiceTrick::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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lowpassChase = pow(A,2);
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//should not scale with sample rate, because values reaching 1 are important
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//to its ability to bypass when set to max
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double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0);
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lastLowpass = lowpassChase;
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double invLowpass;
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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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lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount;
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//setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control,
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//but if I say it's the lowpass out of Capacitor it should literally be that in every behavior.
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double inputSample = (inputSampleL + inputSampleR) * 0.5;
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//this is now our mono audio
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count++; if (count > 5) count = 0; switch (count)
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{
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case 0:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
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iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
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break;
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case 1:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
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iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
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break;
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case 2:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
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iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
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break;
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case 3:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
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iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD;
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break;
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case 4:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB;
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iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE;
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break;
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case 5:
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iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA;
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iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC;
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iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF;
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break;
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}
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//Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively
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//steepens the filter after minimizing artifacts.
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inputSampleL = -inputSample;
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inputSampleR = inputSample;
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//and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone.
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//The purpose of all this is to allow for recording of lead vocals without use of headphones:
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//or at least sealed headphones. You should be able to use this to record vocals with either
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//open-back headphones, or literally speakers in the room so long as the mic is exactly
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//equidistant from each speaker/headphone side.
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//You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor
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//only reverb and echo for vibe. Direct sound is the singer's direct sound.
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//The filtering is because, if you use open-back headphones and move your head, highs will
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//bleed through first like a through-zero flange coming out of cancellation (which it is).
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//Therefore, you can filter off highs until the bleed isn't annoying.
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//Or just run with it, it shouldn't be that loud.
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//Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :)
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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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