diff options
Diffstat (limited to 'plugins/WinVST/VoiceTrick/VoiceTrickProc.cpp')
| -rwxr-xr-x | plugins/WinVST/VoiceTrick/VoiceTrickProc.cpp | 214 |
1 files changed, 214 insertions, 0 deletions
diff --git a/plugins/WinVST/VoiceTrick/VoiceTrickProc.cpp b/plugins/WinVST/VoiceTrick/VoiceTrickProc.cpp new file mode 100755 index 0000000..b57b695 --- /dev/null +++ b/plugins/WinVST/VoiceTrick/VoiceTrickProc.cpp @@ -0,0 +1,214 @@ +/* ======================================== + * 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++; + } +} |