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Diffstat (limited to 'plugins/LinuxVST/src/BiquadOneHalf/BiquadOneHalfProc.cpp')
| -rwxr-xr-x | plugins/LinuxVST/src/BiquadOneHalf/BiquadOneHalfProc.cpp | 313 |
1 files changed, 313 insertions, 0 deletions
diff --git a/plugins/LinuxVST/src/BiquadOneHalf/BiquadOneHalfProc.cpp b/plugins/LinuxVST/src/BiquadOneHalf/BiquadOneHalfProc.cpp new file mode 100755 index 0000000..e126dcf --- /dev/null +++ b/plugins/LinuxVST/src/BiquadOneHalf/BiquadOneHalfProc.cpp @@ -0,0 +1,313 @@ +/* ======================================== + * BiquadOneHalf - BiquadOneHalf.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __BiquadOneHalf_H +#include "BiquadOneHalf.h" +#endif + +void BiquadOneHalf::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(); + + int type = ceil((A*3.999)+0.00001); + + biquadAL[0] = ((B*B*B*0.9999)+0.0001)*0.499; + if (biquadAL[0] < 0.0001) biquadAL[0] = 0.0001; + + biquadAL[1] = (C*C*C*29.99)+0.01; + if (biquadAL[1] < 0.0001) biquadAL[1] = 0.0001; + + double wet = (D*2.0)-1.0; + + //biquad contains these values: + //[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 + //[2] is a0 but you need distinct ones for additional biquad instances so it's here + //[3] is a1 but you need distinct ones for additional biquad instances so it's here + //[4] is a2 but you need distinct ones for additional biquad instances so it's here + //[5] is b1 but you need distinct ones for additional biquad instances so it's here + //[6] is b2 but you need distinct ones for additional biquad instances so it's here + //[7] is stored delayed sample (freq and res are stored so you can move them sample by sample) + //[8] is stored delayed sample (you have to include the coefficient making code if you do that) + + //to build a dedicated filter, rename 'biquad' to whatever the new filter is, then + //put this code either within the sample buffer (for smoothly modulating freq or res) + //or in this 'read the controls' area (for letting you change freq and res with controls) + //or in 'reset' if the freq and res are absolutely fixed (use GetSampleRate to define freq) + + if (type == 1) { //lowpass + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = K * K * norm; + biquadAL[3] = 2.0 * biquadAL[2]; + biquadAL[4] = biquadAL[2]; + biquadAL[5] = 2.0 * (K * K - 1.0) * norm; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + + if (type == 2) { //highpass + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = norm; + biquadAL[3] = -2.0 * biquadAL[2]; + biquadAL[4] = biquadAL[2]; + biquadAL[5] = 2.0 * (K * K - 1.0) * norm; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + + if (type == 3) { //bandpass + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = K / biquadAL[1] * norm; + biquadAL[3] = 0.0; //bandpass can simplify the biquad kernel: leave out this multiply + biquadAL[4] = -biquadAL[2]; + biquadAL[5] = 2.0 * (K * K - 1.0) * norm; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + + if (type == 4) { //notch + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = (1.0 + K * K) * norm; + biquadAL[3] = 2.0 * (K * K - 1) * norm; + biquadAL[4] = biquadAL[2]; + biquadAL[5] = biquadAL[3]; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + for (int x = 0; x < 7; x++) {biquadAR[x] = biquadBL[x] = biquadBR[x] = biquadAL[x];} + + 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; + + inputSampleL = sin(inputSampleL); + inputSampleR = sin(inputSampleR); + //encode Console5: good cleanness + + long double tempSampleL; + long double tempSampleR; + + if (flip) + { + tempSampleL = (inputSampleL * biquadAL[2]) + biquadAL[7]; + biquadAL[7] = (inputSampleL * biquadAL[3]) - (tempSampleL * biquadAL[5]) + biquadAL[8]; + biquadAL[8] = (inputSampleL * biquadAL[4]) - (tempSampleL * biquadAL[6]); + inputSampleL = tempSampleL; + tempSampleR = (inputSampleR * biquadAR[2]) + biquadAR[7]; + biquadAR[7] = (inputSampleR * biquadAR[3]) - (tempSampleR * biquadAR[5]) + biquadAR[8]; + biquadAR[8] = (inputSampleR * biquadAR[4]) - (tempSampleR * biquadAR[6]); + inputSampleR = tempSampleR; + } + else + { + tempSampleL = (inputSampleL * biquadBL[2]) + biquadBL[7]; + biquadBL[7] = (inputSampleL * biquadBL[3]) - (tempSampleL * biquadBL[5]) + biquadBL[8]; + biquadBL[8] = (inputSampleL * biquadBL[4]) - (tempSampleL * biquadBL[6]); + inputSampleL = tempSampleL; + tempSampleR = (inputSampleR * biquadBR[2]) + biquadBR[7]; + biquadBR[7] = (inputSampleR * biquadBR[3]) - (tempSampleR * biquadBR[5]) + biquadBR[8]; + biquadBR[8] = (inputSampleR * biquadBR[4]) - (tempSampleR * biquadBR[6]); + inputSampleR = tempSampleR; + } + flip = !flip; + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + //without this, you can get a NaN condition where it spits out DC offset at full blast! + inputSampleL = asin(inputSampleL); + inputSampleR = asin(inputSampleR); + //amplitude aspect + + if (wet < 1.0) { + inputSampleL = (inputSampleL*wet) + (drySampleL*(1.0-fabs(wet))); + inputSampleR = (inputSampleR*wet) + (drySampleR*(1.0-fabs(wet))); + //inv/dry/wet lets us turn LP into HP and band into notch + } + + //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 BiquadOneHalf::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(); + + int type = ceil((A*3.999)+0.00001); + + biquadAL[0] = ((B*B*B*0.9999)+0.0001)*0.499; + if (biquadAL[0] < 0.0001) biquadAL[0] = 0.0001; + + biquadAL[1] = (C*C*C*29.99)+0.01; + if (biquadAL[1] < 0.0001) biquadAL[1] = 0.0001; + + double wet = (D*2.0)-1.0; + + //biquad contains these values: + //[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 + //[2] is a0 but you need distinct ones for additional biquad instances so it's here + //[3] is a1 but you need distinct ones for additional biquad instances so it's here + //[4] is a2 but you need distinct ones for additional biquad instances so it's here + //[5] is b1 but you need distinct ones for additional biquad instances so it's here + //[6] is b2 but you need distinct ones for additional biquad instances so it's here + //[7] is stored delayed sample (freq and res are stored so you can move them sample by sample) + //[8] is stored delayed sample (you have to include the coefficient making code if you do that) + + //to build a dedicated filter, rename 'biquad' to whatever the new filter is, then + //put this code either within the sample buffer (for smoothly modulating freq or res) + //or in this 'read the controls' area (for letting you change freq and res with controls) + //or in 'reset' if the freq and res are absolutely fixed (use GetSampleRate to define freq) + + if (type == 1) { //lowpass + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = K * K * norm; + biquadAL[3] = 2.0 * biquadAL[2]; + biquadAL[4] = biquadAL[2]; + biquadAL[5] = 2.0 * (K * K - 1.0) * norm; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + + if (type == 2) { //highpass + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = norm; + biquadAL[3] = -2.0 * biquadAL[2]; + biquadAL[4] = biquadAL[2]; + biquadAL[5] = 2.0 * (K * K - 1.0) * norm; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + + if (type == 3) { //bandpass + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = K / biquadAL[1] * norm; + biquadAL[3] = 0.0; //bandpass can simplify the biquad kernel: leave out this multiply + biquadAL[4] = -biquadAL[2]; + biquadAL[5] = 2.0 * (K * K - 1.0) * norm; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + + if (type == 4) { //notch + double K = tan(M_PI * biquadAL[0]); + double norm = 1.0 / (1.0 + K / biquadAL[1] + K * K); + biquadAL[2] = (1.0 + K * K) * norm; + biquadAL[3] = 2.0 * (K * K - 1) * norm; + biquadAL[4] = biquadAL[2]; + biquadAL[5] = biquadAL[3]; + biquadAL[6] = (1.0 - K / biquadAL[1] + K * K) * norm; + } + for (int x = 0; x < 7; x++) {biquadAR[x] = biquadBL[x] = biquadBR[x] = biquadAL[x];} + + 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; + + + inputSampleL = sin(inputSampleL); + inputSampleR = sin(inputSampleR); + //encode Console5: good cleanness + + long double tempSampleL; + long double tempSampleR; + + if (flip) + { + tempSampleL = (inputSampleL * biquadAL[2]) + biquadAL[7]; + biquadAL[7] = (inputSampleL * biquadAL[3]) - (tempSampleL * biquadAL[5]) + biquadAL[8]; + biquadAL[8] = (inputSampleL * biquadAL[4]) - (tempSampleL * biquadAL[6]); + inputSampleL = tempSampleL; + tempSampleR = (inputSampleR * biquadAR[2]) + biquadAR[7]; + biquadAR[7] = (inputSampleR * biquadAR[3]) - (tempSampleR * biquadAR[5]) + biquadAR[8]; + biquadAR[8] = (inputSampleR * biquadAR[4]) - (tempSampleR * biquadAR[6]); + inputSampleR = tempSampleR; + } + else + { + tempSampleL = (inputSampleL * biquadBL[2]) + biquadBL[7]; + biquadBL[7] = (inputSampleL * biquadBL[3]) - (tempSampleL * biquadBL[5]) + biquadBL[8]; + biquadBL[8] = (inputSampleL * biquadBL[4]) - (tempSampleL * biquadBL[6]); + inputSampleL = tempSampleL; + tempSampleR = (inputSampleR * biquadBR[2]) + biquadBR[7]; + biquadBR[7] = (inputSampleR * biquadBR[3]) - (tempSampleR * biquadBR[5]) + biquadBR[8]; + biquadBR[8] = (inputSampleR * biquadBR[4]) - (tempSampleR * biquadBR[6]); + inputSampleR = tempSampleR; + } + flip = !flip; + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + if (inputSampleR > 1.0) inputSampleR = 1.0; + if (inputSampleR < -1.0) inputSampleR = -1.0; + //without this, you can get a NaN condition where it spits out DC offset at full blast! + inputSampleL = asin(inputSampleL); + inputSampleR = asin(inputSampleR); + //amplitude aspect + + if (wet < 1.0) { + inputSampleL = (inputSampleL*wet) + (drySampleL*(1.0-fabs(wet))); + inputSampleR = (inputSampleR*wet) + (drySampleR*(1.0-fabs(wet))); + //inv/dry/wet lets us turn LP into HP and band into notch + } + + //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++; + } +} |