/* ======================================== * VoiceTrick - VoiceTrick.h * Copyright (c) 2016 airwindows, All rights reserved * ======================================== */ #ifndef __VoiceTrick_H #include "VoiceTrick.h" #endif void VoiceTrick::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) { float* in1 = inputs[0]; float* in2 = inputs[1]; float* out1 = outputs[0]; float* out2 = outputs[1]; lowpassChase = pow(A,2); //should not scale with sample rate, because values reaching 1 are important //to its ability to bypass when set to max double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0); lastLowpass = lowpassChase; double invLowpass; while (--sampleFrames >= 0) { long double inputSampleL = *in1; long double inputSampleR = *in2; if (fabs(inputSampleL)<1.18e-37) inputSampleL = fpd * 1.18e-37; if (fabs(inputSampleR)<1.18e-37) inputSampleR = fpd * 1.18e-37; lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount; //setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control, //but if I say it's the lowpass out of Capacitor it should literally be that in every behavior. long double inputSample = (inputSampleL + inputSampleR) * 0.5; //this is now our mono audio count++; if (count > 5) count = 0; switch (count) { case 0: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB; iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD; break; case 1: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC; iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE; break; case 2: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB; iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF; break; case 3: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC; iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD; break; case 4: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB; iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE; break; case 5: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC; iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF; break; } //Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively //steepens the filter after minimizing artifacts. inputSampleL = -inputSample; inputSampleR = inputSample; //and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone. //The purpose of all this is to allow for recording of lead vocals without use of headphones: //or at least sealed headphones. You should be able to use this to record vocals with either //open-back headphones, or literally speakers in the room so long as the mic is exactly //equidistant from each speaker/headphone side. //You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor //only reverb and echo for vibe. Direct sound is the singer's direct sound. //The filtering is because, if you use open-back headphones and move your head, highs will //bleed through first like a through-zero flange coming out of cancellation (which it is). //Therefore, you can filter off highs until the bleed isn't annoying. //Or just run with it, it shouldn't be that loud. //Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :) //begin 32 bit stereo floating point dither int expon; frexpf((float)inputSampleL, &expon); fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62)); frexpf((float)inputSampleR, &expon); fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62)); //end 32 bit stereo floating point dither *out1 = inputSampleL; *out2 = inputSampleR; *in1++; *in2++; *out1++; *out2++; } } void VoiceTrick::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) { double* in1 = inputs[0]; double* in2 = inputs[1]; double* out1 = outputs[0]; double* out2 = outputs[1]; lowpassChase = pow(A,2); //should not scale with sample rate, because values reaching 1 are important //to its ability to bypass when set to max double lowpassSpeed = 300 / (fabs( lastLowpass - lowpassChase)+1.0); lastLowpass = lowpassChase; double invLowpass; while (--sampleFrames >= 0) { long double inputSampleL = *in1; long double inputSampleR = *in2; if (fabs(inputSampleL)<1.18e-43) inputSampleL = fpd * 1.18e-43; if (fabs(inputSampleR)<1.18e-43) inputSampleR = fpd * 1.18e-43; lowpassAmount = (((lowpassAmount*lowpassSpeed)+lowpassChase)/(lowpassSpeed + 1.0)); invLowpass = 1.0 - lowpassAmount; //setting chase functionality of Capacitor Lowpass. I could just use this value directly from the control, //but if I say it's the lowpass out of Capacitor it should literally be that in every behavior. long double inputSample = (inputSampleL + inputSampleR) * 0.5; //this is now our mono audio count++; if (count > 5) count = 0; switch (count) { case 0: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB; iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD; break; case 1: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC; iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE; break; case 2: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB; iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF; break; case 3: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC; iirLowpassD = (iirLowpassD * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassD; break; case 4: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassB = (iirLowpassB * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassB; iirLowpassE = (iirLowpassE * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassE; break; case 5: iirLowpassA = (iirLowpassA * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassA; iirLowpassC = (iirLowpassC * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassC; iirLowpassF = (iirLowpassF * invLowpass) + (inputSample * lowpassAmount); inputSample = iirLowpassF; break; } //Highpass Filter chunk. This is three poles of IIR highpass, with a 'gearbox' that progressively //steepens the filter after minimizing artifacts. inputSampleL = -inputSample; inputSampleR = inputSample; //and now the output is mono, maybe filtered, and phase flipped to cancel at the microphone. //The purpose of all this is to allow for recording of lead vocals without use of headphones: //or at least sealed headphones. You should be able to use this to record vocals with either //open-back headphones, or literally speakers in the room so long as the mic is exactly //equidistant from each speaker/headphone side. //You'll probably want to not use voice monitoring: just mute the track being recorded, or monitor //only reverb and echo for vibe. Direct sound is the singer's direct sound. //The filtering is because, if you use open-back headphones and move your head, highs will //bleed through first like a through-zero flange coming out of cancellation (which it is). //Therefore, you can filter off highs until the bleed isn't annoying. //Or just run with it, it shouldn't be that loud. //Thanks to Peter Gabriel for many great examples of hit vocals recorded just like this :) //begin 64 bit stereo floating point dither int expon; frexp((double)inputSampleL, &expon); fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; inputSampleL += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62)); frexp((double)inputSampleR, &expon); fpd ^= fpd << 13; fpd ^= fpd >> 17; fpd ^= fpd << 5; inputSampleR += ((double(fpd)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62)); //end 64 bit stereo floating point dither *out1 = inputSampleL; *out2 = inputSampleR; *in1++; *in2++; *out1++; *out2++; } }