diff options
Diffstat (limited to 'plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp')
| -rwxr-xr-x | plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp | 282 |
1 files changed, 282 insertions, 0 deletions
diff --git a/plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp b/plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp new file mode 100755 index 0000000..123d328 --- /dev/null +++ b/plugins/LinuxVST/src/Baxandall/BaxandallProc.cpp @@ -0,0 +1,282 @@ +/* ======================================== + * Baxandall - Baxandall.h + * Copyright (c) 2016 airwindows, All rights reserved + * ======================================== */ + +#ifndef __Baxandall_H +#include "Baxandall.h" +#endif + +void Baxandall::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames) +{ + float* in1 = inputs[0]; + float* in2 = inputs[1]; + float* out1 = outputs[0]; + float* out2 = outputs[1]; + + double trebleGain = pow(10.0,((A*30.0)-15.0)/20.0); + double trebleFreq = (4410.0*trebleGain)/getSampleRate(); + if (trebleFreq > 0.45) trebleFreq = 0.45; + trebleAL[0] = trebleBL[0] = trebleAR[0] = trebleBR[0] = trebleFreq; + double bassGain = pow(10.0,((B*30.0)-15.0)/20.0); + double bassFreq = pow(10.0,-((B*30.0)-15.0)/20.0); + bassFreq = (4410.0*bassFreq)/getSampleRate(); + if (bassFreq > 0.45) bassFreq = 0.45; + bassAL[0] = bassBL[0] = bassAR[0] = bassBR[0] = bassFreq; + trebleAL[1] = trebleBL[1] = trebleAR[1] = trebleBR[1] = 0.4; + bassAL[1] = bassBL[1] = bassAR[1] = bassBR[1] = 0.2; + double output = pow(10.0,((C*30.0)-15.0)/20.0); + + double K = tan(M_PI * trebleAL[0]); + double norm = 1.0 / (1.0 + K / trebleAL[1] + K * K); + trebleBL[2] = trebleAL[2] = trebleBR[2] = trebleAR[2] = K * K * norm; + trebleBL[3] = trebleAL[3] = trebleBR[3] = trebleAR[3] = 2.0 * trebleAL[2]; + trebleBL[4] = trebleAL[4] = trebleBR[4] = trebleAR[4] = trebleAL[2]; + trebleBL[5] = trebleAL[5] = trebleBR[5] = trebleAR[5] = 2.0 * (K * K - 1.0) * norm; + trebleBL[6] = trebleAL[6] = trebleBR[6] = trebleAR[6] = (1.0 - K / trebleAL[1] + K * K) * norm; + + K = tan(M_PI * bassAL[0]); + norm = 1.0 / (1.0 + K / bassAL[1] + K * K); + bassBL[2] = bassAL[2] = bassBR[2] = bassAR[2] = K * K * norm; + bassBL[3] = bassAL[3] = bassBR[3] = bassAR[3] = 2.0 * bassAL[2]; + bassBL[4] = bassAL[4] = bassBR[4] = bassAR[4] = bassAL[2]; + bassBL[5] = bassAL[5] = bassBR[5] = bassAR[5] = 2.0 * (K * K - 1.0) * norm; + bassBL[6] = bassAL[6] = bassBR[6] = bassAR[6] = (1.0 - K / bassAL[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; + + if (output != 1.0) { + inputSampleL *= output; + inputSampleR *= output; + }//gain trim in front of plugin, in case Console stage clips + + inputSampleL = sin(inputSampleL); + inputSampleR = sin(inputSampleR); + //encode Console5: good cleanness + + long double trebleSampleL; + long double bassSampleL; + long double trebleSampleR; + long double bassSampleR; + + if (flip) + { + trebleSampleL = (inputSampleL * trebleAL[2]) + trebleAL[7]; + trebleAL[7] = (inputSampleL * trebleAL[3]) - (trebleSampleL * trebleAL[5]) + trebleAL[8]; + trebleAL[8] = (inputSampleL * trebleAL[4]) - (trebleSampleL * trebleAL[6]); + trebleSampleL = inputSampleL - trebleSampleL; + + bassSampleL = (inputSampleL * bassAL[2]) + bassAL[7]; + bassAL[7] = (inputSampleL * bassAL[3]) - (bassSampleL * bassAL[5]) + bassAL[8]; + bassAL[8] = (inputSampleL * bassAL[4]) - (bassSampleL * bassAL[6]); + + trebleSampleR = (inputSampleR * trebleAR[2]) + trebleAR[7]; + trebleAR[7] = (inputSampleR * trebleAR[3]) - (trebleSampleR * trebleAR[5]) + trebleAR[8]; + trebleAR[8] = (inputSampleR * trebleAR[4]) - (trebleSampleR * trebleAR[6]); + trebleSampleR = inputSampleR - trebleSampleR; + + bassSampleR = (inputSampleR * bassAR[2]) + bassAR[7]; + bassAR[7] = (inputSampleR * bassAR[3]) - (bassSampleR * bassAR[5]) + bassAR[8]; + bassAR[8] = (inputSampleR * bassAR[4]) - (bassSampleR * bassAR[6]); + } + else + { + trebleSampleL = (inputSampleL * trebleBL[2]) + trebleBL[7]; + trebleBL[7] = (inputSampleL * trebleBL[3]) - (trebleSampleL * trebleBL[5]) + trebleBL[8]; + trebleBL[8] = (inputSampleL * trebleBL[4]) - (trebleSampleL * trebleBL[6]); + trebleSampleL = inputSampleL - trebleSampleL; + + bassSampleL = (inputSampleL * bassBL[2]) + bassBL[7]; + bassBL[7] = (inputSampleL * bassBL[3]) - (bassSampleL * bassBL[5]) + bassBL[8]; + bassBL[8] = (inputSampleL * bassBL[4]) - (bassSampleL * bassBL[6]); + + trebleSampleR = (inputSampleR * trebleBR[2]) + trebleBR[7]; + trebleBR[7] = (inputSampleR * trebleBR[3]) - (trebleSampleR * trebleBR[5]) + trebleBR[8]; + trebleBR[8] = (inputSampleR * trebleBR[4]) - (trebleSampleR * trebleBR[6]); + trebleSampleR = inputSampleR - trebleSampleR; + + bassSampleR = (inputSampleR * bassBR[2]) + bassBR[7]; + bassBR[7] = (inputSampleR * bassBR[3]) - (bassSampleR * bassBR[5]) + bassBR[8]; + bassBR[8] = (inputSampleR * bassBR[4]) - (bassSampleR * bassBR[6]); + } + flip = !flip; + + trebleSampleL *= trebleGain; + bassSampleL *= bassGain; + inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad + trebleSampleR *= trebleGain; + bassSampleR *= bassGain; + inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + //without this, you can get a NaN condition where it spits out DC offset at full blast! + inputSampleL = asin(inputSampleL); + //amplitude aspect + + 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! + inputSampleR = asin(inputSampleR); + //amplitude aspect + + //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 Baxandall::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames) +{ + double* in1 = inputs[0]; + double* in2 = inputs[1]; + double* out1 = outputs[0]; + double* out2 = outputs[1]; + + double trebleGain = pow(10.0,((A*30.0)-15.0)/20.0); + double trebleFreq = (4410.0*trebleGain)/getSampleRate(); + if (trebleFreq > 0.45) trebleFreq = 0.45; + trebleAL[0] = trebleBL[0] = trebleAR[0] = trebleBR[0] = trebleFreq; + double bassGain = pow(10.0,((B*30.0)-15.0)/20.0); + double bassFreq = pow(10.0,-((B*30.0)-15.0)/20.0); + bassFreq = (4410.0*bassFreq)/getSampleRate(); + if (bassFreq > 0.45) bassFreq = 0.45; + bassAL[0] = bassBL[0] = bassAR[0] = bassBR[0] = bassFreq; + trebleAL[1] = trebleBL[1] = trebleAR[1] = trebleBR[1] = 0.4; + bassAL[1] = bassBL[1] = bassAR[1] = bassBR[1] = 0.2; + double output = pow(10.0,((C*30.0)-15.0)/20.0); + + double K = tan(M_PI * trebleAL[0]); + double norm = 1.0 / (1.0 + K / trebleAL[1] + K * K); + trebleBL[2] = trebleAL[2] = trebleBR[2] = trebleAR[2] = K * K * norm; + trebleBL[3] = trebleAL[3] = trebleBR[3] = trebleAR[3] = 2.0 * trebleAL[2]; + trebleBL[4] = trebleAL[4] = trebleBR[4] = trebleAR[4] = trebleAL[2]; + trebleBL[5] = trebleAL[5] = trebleBR[5] = trebleAR[5] = 2.0 * (K * K - 1.0) * norm; + trebleBL[6] = trebleAL[6] = trebleBR[6] = trebleAR[6] = (1.0 - K / trebleAL[1] + K * K) * norm; + + K = tan(M_PI * bassAL[0]); + norm = 1.0 / (1.0 + K / bassAL[1] + K * K); + bassBL[2] = bassAL[2] = bassBR[2] = bassAR[2] = K * K * norm; + bassBL[3] = bassAL[3] = bassBR[3] = bassAR[3] = 2.0 * bassAL[2]; + bassBL[4] = bassAL[4] = bassBR[4] = bassAR[4] = bassAL[2]; + bassBL[5] = bassAL[5] = bassBR[5] = bassAR[5] = 2.0 * (K * K - 1.0) * norm; + bassBL[6] = bassAL[6] = bassBR[6] = bassAR[6] = (1.0 - K / bassAL[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; + + if (output != 1.0) { + inputSampleL *= output; + inputSampleR *= output; + }//gain trim in front of plugin, in case Console stage clips + + inputSampleL = sin(inputSampleL); + inputSampleR = sin(inputSampleR); + //encode Console5: good cleanness + + long double trebleSampleL; + long double bassSampleL; + long double trebleSampleR; + long double bassSampleR; + + if (flip) + { + trebleSampleL = (inputSampleL * trebleAL[2]) + trebleAL[7]; + trebleAL[7] = (inputSampleL * trebleAL[3]) - (trebleSampleL * trebleAL[5]) + trebleAL[8]; + trebleAL[8] = (inputSampleL * trebleAL[4]) - (trebleSampleL * trebleAL[6]); + trebleSampleL = inputSampleL - trebleSampleL; + + bassSampleL = (inputSampleL * bassAL[2]) + bassAL[7]; + bassAL[7] = (inputSampleL * bassAL[3]) - (bassSampleL * bassAL[5]) + bassAL[8]; + bassAL[8] = (inputSampleL * bassAL[4]) - (bassSampleL * bassAL[6]); + + trebleSampleR = (inputSampleR * trebleAR[2]) + trebleAR[7]; + trebleAR[7] = (inputSampleR * trebleAR[3]) - (trebleSampleR * trebleAR[5]) + trebleAR[8]; + trebleAR[8] = (inputSampleR * trebleAR[4]) - (trebleSampleR * trebleAR[6]); + trebleSampleR = inputSampleR - trebleSampleR; + + bassSampleR = (inputSampleR * bassAR[2]) + bassAR[7]; + bassAR[7] = (inputSampleR * bassAR[3]) - (bassSampleR * bassAR[5]) + bassAR[8]; + bassAR[8] = (inputSampleR * bassAR[4]) - (bassSampleR * bassAR[6]); + } + else + { + trebleSampleL = (inputSampleL * trebleBL[2]) + trebleBL[7]; + trebleBL[7] = (inputSampleL * trebleBL[3]) - (trebleSampleL * trebleBL[5]) + trebleBL[8]; + trebleBL[8] = (inputSampleL * trebleBL[4]) - (trebleSampleL * trebleBL[6]); + trebleSampleL = inputSampleL - trebleSampleL; + + bassSampleL = (inputSampleL * bassBL[2]) + bassBL[7]; + bassBL[7] = (inputSampleL * bassBL[3]) - (bassSampleL * bassBL[5]) + bassBL[8]; + bassBL[8] = (inputSampleL * bassBL[4]) - (bassSampleL * bassBL[6]); + + trebleSampleR = (inputSampleR * trebleBR[2]) + trebleBR[7]; + trebleBR[7] = (inputSampleR * trebleBR[3]) - (trebleSampleR * trebleBR[5]) + trebleBR[8]; + trebleBR[8] = (inputSampleR * trebleBR[4]) - (trebleSampleR * trebleBR[6]); + trebleSampleR = inputSampleR - trebleSampleR; + + bassSampleR = (inputSampleR * bassBR[2]) + bassBR[7]; + bassBR[7] = (inputSampleR * bassBR[3]) - (bassSampleR * bassBR[5]) + bassBR[8]; + bassBR[8] = (inputSampleR * bassBR[4]) - (bassSampleR * bassBR[6]); + } + flip = !flip; + + trebleSampleL *= trebleGain; + bassSampleL *= bassGain; + inputSampleL = bassSampleL + trebleSampleL; //interleaved biquad + trebleSampleR *= trebleGain; + bassSampleR *= bassGain; + inputSampleR = bassSampleR + trebleSampleR; //interleaved biquad + + if (inputSampleL > 1.0) inputSampleL = 1.0; + if (inputSampleL < -1.0) inputSampleL = -1.0; + //without this, you can get a NaN condition where it spits out DC offset at full blast! + inputSampleL = asin(inputSampleL); + //amplitude aspect + + 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! + inputSampleR = asin(inputSampleR); + //amplitude aspect + + //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++; + } +} |