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Diffstat (limited to 'plugins/WinVST/Coils/CoilsProc.cpp')
| -rwxr-xr-x | plugins/WinVST/Coils/CoilsProc.cpp | 174 |
1 files changed, 174 insertions, 0 deletions
diff --git a/plugins/WinVST/Coils/CoilsProc.cpp b/plugins/WinVST/Coils/CoilsProc.cpp new file mode 100755 index 0000000..cc0456a --- /dev/null +++ b/plugins/WinVST/Coils/CoilsProc.cpp @@ -0,0 +1,174 @@ +/* ======================================== + * Coils - Coils.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Coils_H +#include "Coils.h" +#endif + +void Coils::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + //[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist + //[1] is resonance, 0.7071 is Butterworth. Also can't be zero + double boost = 1.0-pow(A,2); + if (boost < 0.001) boost = 0.001; //there's a divide, we can't have this be zero + figureL[0] = figureR[0] = 600.0/getSampleRate(); //fixed frequency, 600hz + figureL[1] = figureR[1] = 0.023; //resonance + double offset = B; + double wet = C; + double K = tan(M_PI * figureR[0]); + double norm = 1.0 / (1.0 + K / figureR[1] + K * K); + figureL[2] = figureR[2] = K / figureR[1] * norm; + figureL[4] = figureR[4] = -figureR[2]; + figureL[5] = figureR[5] = 2.0 * (K * K - 1.0) * norm; + figureL[6] = figureR[6] = (1.0 - K / figureR[1] + K * K) * norm; + + 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; + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + //long double tempSample = (inputSample * figure[2]) + figure[7]; + //figure[7] = -(tempSample * figure[5]) + figure[8]; + //figure[8] = (inputSample * figure[4]) - (tempSample * figure[6]); + //inputSample = tempSample + sin(drySample-tempSample); + //or + //inputSample = tempSample + ((sin(((drySample-tempSample)/boost)+offset)-offset)*boost); + // + //given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of + //the full frequencies but distorts like a real transformer. Purest case, and since + //we are not using a high Q we can remove the extra sin/asin on the biquad. + + + long double tempSample = (inputSampleL * figureL[2]) + figureL[7]; + figureL[7] = -(tempSample * figureL[5]) + figureL[8]; + figureL[8] = (inputSampleL * figureL[4]) - (tempSample * figureL[6]); + inputSampleL = tempSample + ((sin(((drySampleL-tempSample)/boost)+offset)-offset)*boost); + //given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of + //the full frequencies but distorts like a real transformer. Since + //we are not using a high Q we can remove the extra sin/asin on the biquad. + + tempSample = (inputSampleR * figureR[2]) + figureR[7]; + figureR[7] = -(tempSample * figureR[5]) + figureR[8]; + figureR[8] = (inputSampleR * figureR[4]) - (tempSample * figureR[6]); + inputSampleR = tempSample + ((sin(((drySampleR-tempSample)/boost)+offset)-offset)*boost); + //given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of + //the full frequencies but distorts like a real transformer. Since + //we are not using a high Q we can remove the extra sin/asin on the biquad. + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet)); + inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet)); + } + + //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 Coils::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + //[0] is frequency: 0.000001 to 0.499999 is near-zero to near-Nyquist + //[1] is resonance, 0.7071 is Butterworth. Also can't be zero + double boost = 1.0-pow(A,2); + if (boost < 0.001) boost = 0.001; //there's a divide, we can't have this be zero + figureL[0] = figureR[0] = 600.0/getSampleRate(); //fixed frequency, 600hz + figureL[1] = figureR[1] = 0.023; //resonance + double offset = B; + double wet = C; + double K = tan(M_PI * figureR[0]); + double norm = 1.0 / (1.0 + K / figureR[1] + K * K); + figureL[2] = figureR[2] = K / figureR[1] * norm; + figureL[4] = figureR[4] = -figureR[2]; + figureL[5] = figureR[5] = 2.0 * (K * K - 1.0) * norm; + figureL[6] = figureR[6] = (1.0 - K / figureR[1] + K * K) * norm; + + 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; + long double drySampleL = inputSampleL; + long double drySampleR = inputSampleR; + + //long double tempSample = (inputSample * figure[2]) + figure[7]; + //figure[7] = -(tempSample * figure[5]) + figure[8]; + //figure[8] = (inputSample * figure[4]) - (tempSample * figure[6]); + //inputSample = tempSample + sin(drySample-tempSample); + //or + //inputSample = tempSample + ((sin(((drySample-tempSample)/boost)+offset)-offset)*boost); + // + //given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of + //the full frequencies but distorts like a real transformer. Purest case, and since + //we are not using a high Q we can remove the extra sin/asin on the biquad. + + + long double tempSample = (inputSampleL * figureL[2]) + figureL[7]; + figureL[7] = -(tempSample * figureL[5]) + figureL[8]; + figureL[8] = (inputSampleL * figureL[4]) - (tempSample * figureL[6]); + inputSampleL = tempSample + ((sin(((drySampleL-tempSample)/boost)+offset)-offset)*boost); + //given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of + //the full frequencies but distorts like a real transformer. Since + //we are not using a high Q we can remove the extra sin/asin on the biquad. + + tempSample = (inputSampleR * figureR[2]) + figureR[7]; + figureR[7] = -(tempSample * figureR[5]) + figureR[8]; + figureR[8] = (inputSampleR * figureR[4]) - (tempSample * figureR[6]); + inputSampleR = tempSample + ((sin(((drySampleR-tempSample)/boost)+offset)-offset)*boost); + //given a bandlimited inputSample, freq 600hz and Q of 0.023, this restores a lot of + //the full frequencies but distorts like a real transformer. Since + //we are not using a high Q we can remove the extra sin/asin on the biquad. + + if (wet !=1.0) { + inputSampleL = (inputSampleL * wet) + (drySampleL * (1.0-wet)); + inputSampleR = (inputSampleR * wet) + (drySampleR * (1.0-wet)); + } + + //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++; + } +} |