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Diffstat (limited to 'plugins/MacVST/Highpass/source/HighpassProc.cpp')
| -rwxr-xr-x | plugins/MacVST/Highpass/source/HighpassProc.cpp | 298 |
1 files changed, 298 insertions, 0 deletions
diff --git a/plugins/MacVST/Highpass/source/HighpassProc.cpp b/plugins/MacVST/Highpass/source/HighpassProc.cpp new file mode 100755 index 0000000..b1e7de9 --- /dev/null +++ b/plugins/MacVST/Highpass/source/HighpassProc.cpp @@ -0,0 +1,298 @@ +/* ======================================== + * Highpass - Highpass.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Highpass_H +#include "Highpass.h" +#endif + +void Highpass::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + double iirAmount = pow(A,3)/overallscale; + double tight = (B*2.0)-1.0; + double wet = C; + double dry = 1.0 - wet; + double offset; + double inputSampleL; + double inputSampleR; + double outputSampleL; + double outputSampleR; + float fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + iirAmount += (iirAmount * tight * tight); + if (tight > 0) tight /= 1.5; + else tight /= 3.0; + //we are setting it up so that to either extreme we can get an audible sound, + //but sort of scaled so small adjustments don't shift the cutoff frequency yet. + if (iirAmount <= 0.0) iirAmount = 0.0; + if (iirAmount > 1.0) iirAmount = 1.0; + //handle the change in cutoff frequency + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + outputSampleL = inputSampleL; + outputSampleR = inputSampleR; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) + { + iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + outputSampleL = outputSampleL - iirSampleAL; + } + else + { + iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + outputSampleL = outputSampleL - iirSampleBL; + } + + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) + { + iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + outputSampleR = outputSampleR - iirSampleAR; + } + else + { + iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + outputSampleR = outputSampleR - iirSampleBR; + } + + + + if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry); + if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry); + + //noise shaping to 32-bit floating point + if (fpFlip) { + fpTemp = outputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((outputSampleL-fpTemp)*fpNew); + outputSampleL += fpNShapeLA; + + fpTemp = outputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((outputSampleR-fpTemp)*fpNew); + outputSampleR += fpNShapeRA; + } + else { + fpTemp = outputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((outputSampleL-fpTemp)*fpNew); + outputSampleL += fpNShapeLB; + + fpTemp = outputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((outputSampleR-fpTemp)*fpNew); + outputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 32 bit output + + *out1 = outputSampleL; + *out2 = outputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +} + +void Highpass::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double overallscale = 1.0; + overallscale /= 44100.0; + overallscale *= getSampleRate(); + double iirAmount = pow(A,3)/overallscale; + double tight = (B*2.0)-1.0; + double wet = C; + double dry = 1.0 - wet; + double offset; + double inputSampleL; + double inputSampleR; + double outputSampleL; + double outputSampleR; + double fpTemp; + long double fpOld = 0.618033988749894848204586; //golden ratio! + long double fpNew = 1.0 - fpOld; + + iirAmount += (iirAmount * tight * tight); + if (tight > 0) tight /= 1.5; + else tight /= 3.0; + //we are setting it up so that to either extreme we can get an audible sound, + //but sort of scaled so small adjustments don't shift the cutoff frequency yet. + if (iirAmount <= 0.0) iirAmount = 0.0; + if (iirAmount > 1.0) iirAmount = 1.0; + //handle the change in cutoff frequency + + while (--sampleFrames >= 0) + { + inputSampleL = *in1; + inputSampleR = *in2; + if (inputSampleL<1.2e-38 && -inputSampleL<1.2e-38) { + static int noisesource = 0; + //this declares a variable before anything else is compiled. It won't keep assigning + //it to 0 for every sample, it's as if the declaration doesn't exist in this context, + //but it lets me add this denormalization fix in a single place rather than updating + //it in three different locations. The variable isn't thread-safe but this is only + //a random seed and we can share it with whatever. + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleL = applyresidue; + } + if (inputSampleR<1.2e-38 && -inputSampleR<1.2e-38) { + static int noisesource = 0; + noisesource = noisesource % 1700021; noisesource++; + int residue = noisesource * noisesource; + residue = residue % 170003; residue *= residue; + residue = residue % 17011; residue *= residue; + residue = residue % 1709; residue *= residue; + residue = residue % 173; residue *= residue; + residue = residue % 17; + double applyresidue = residue; + applyresidue *= 0.00000001; + applyresidue *= 0.00000001; + inputSampleR = applyresidue; + //this denormalization routine produces a white noise at -300 dB which the noise + //shaping will interact with to produce a bipolar output, but the noise is actually + //all positive. That should stop any variables from going denormal, and the routine + //only kicks in if digital black is input. As a final touch, if you save to 24-bit + //the silence will return to being digital black again. + } + outputSampleL = inputSampleL; + outputSampleR = inputSampleR; + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleL)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleL))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) + { + iirSampleAL = (iirSampleAL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + outputSampleL = outputSampleL - iirSampleAL; + } + else + { + iirSampleBL = (iirSampleBL * (1 - (offset * iirAmount))) + (inputSampleL * (offset * iirAmount)); + outputSampleL = outputSampleL - iirSampleBL; + } + + + if (tight > 0) offset = (1 - tight) + (fabs(inputSampleR)*tight); + else offset = (1 + tight) + ((1-fabs(inputSampleR))*tight); + if (offset < 0) offset = 0; + if (offset > 1) offset = 1; + if (fpFlip) + { + iirSampleAR = (iirSampleAR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + outputSampleR = outputSampleR - iirSampleAR; + } + else + { + iirSampleBR = (iirSampleBR * (1 - (offset * iirAmount))) + (inputSampleR * (offset * iirAmount)); + outputSampleR = outputSampleR - iirSampleBR; + } + + + + if (wet < 1.0) outputSampleL = (outputSampleL * wet) + (inputSampleL * dry); + if (wet < 1.0) outputSampleR = (outputSampleR * wet) + (inputSampleR * dry); + + //noise shaping to 32-bit floating point + if (fpFlip) { + fpTemp = outputSampleL; + fpNShapeLA = (fpNShapeLA*fpOld)+((outputSampleL-fpTemp)*fpNew); + outputSampleL += fpNShapeLA; + + fpTemp = outputSampleR; + fpNShapeRA = (fpNShapeRA*fpOld)+((outputSampleR-fpTemp)*fpNew); + outputSampleR += fpNShapeRA; + } + else { + fpTemp = outputSampleL; + fpNShapeLB = (fpNShapeLB*fpOld)+((outputSampleL-fpTemp)*fpNew); + outputSampleL += fpNShapeLB; + + fpTemp = outputSampleR; + fpNShapeRB = (fpNShapeRB*fpOld)+((outputSampleR-fpTemp)*fpNew); + outputSampleR += fpNShapeRB; + } + fpFlip = !fpFlip; + //end noise shaping on 32 bit output + + *out1 = outputSampleL; + *out2 = outputSampleR; + + *in1++; + *in2++; + *out1++; + *out2++; + } +}
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