mirror of
https://github.com/airwindows/airwindows.git
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168 lines
8.1 KiB
C++
Executable file
168 lines
8.1 KiB
C++
Executable file
/* ========================================
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* Console7Channel - Console7Channel.h
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* Copyright (c) 2016 airwindows, Airwindows uses the MIT license
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* ======================================== */
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#ifndef __Console7Channel_H
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#include "Console7Channel.h"
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#endif
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void Console7Channel::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
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{
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float* in1 = inputs[0];
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float* in2 = inputs[1];
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float* out1 = outputs[0];
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float* out2 = outputs[1];
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double inputgain = A*1.272019649514069;
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//which is, in fact, the square root of 1.618033988749894848204586...
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//this happens to give us a boost factor where the track continues to get louder even
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//as it saturates and loses a bit of peak energy. Console7Channel channels go to 12! (.272,etc)
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//Neutral gain through the whole system with a full scale sine ia 0.772 on the gain knob
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if (gainchase != inputgain) chasespeed *= 2.0;
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if (chasespeed > sampleFrames) chasespeed = sampleFrames;
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if (gainchase < 0.0) gainchase = inputgain;
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biquadA[0] = 20000.0 / getSampleRate();
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biquadA[1] = 1.618033988749894848204586;
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double K = tan(M_PI * biquadA[0]); //lowpass
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double norm = 1.0 / (1.0 + K / biquadA[1] + K * K);
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biquadA[2] = K * K * norm;
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biquadA[3] = 2.0 * biquadA[2];
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biquadA[4] = biquadA[2];
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biquadA[5] = 2.0 * (K * K - 1.0) * norm;
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biquadA[6] = (1.0 - K / biquadA[1] + K * K) * norm;
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while (--sampleFrames >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
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if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
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double outSampleL = biquadA[2]*inputSampleL+biquadA[3]*biquadA[7]+biquadA[4]*biquadA[8]-biquadA[5]*biquadA[9]-biquadA[6]*biquadA[10];
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biquadA[8] = biquadA[7]; biquadA[7] = inputSampleL; inputSampleL = outSampleL; biquadA[10] = biquadA[9]; biquadA[9] = inputSampleL; //DF1 left
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double outSampleR = biquadA[2]*inputSampleR+biquadA[3]*biquadA[11]+biquadA[4]*biquadA[12]-biquadA[5]*biquadA[13]-biquadA[6]*biquadA[14];
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biquadA[12] = biquadA[11]; biquadA[11] = inputSampleR; inputSampleR = outSampleR; biquadA[14] = biquadA[13]; biquadA[13] = inputSampleR; //DF1 right
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chasespeed *= 0.9999; chasespeed -= 0.01; if (chasespeed < 64.0) chasespeed = 64.0;
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//we have our chase speed compensated for recent fader activity
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gainchase = (((gainchase*chasespeed)+inputgain)/(chasespeed+1.0));
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//gainchase is chasing the target, as a simple multiply gain factor
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if (1.0 != gainchase) {inputSampleL *= pow(gainchase,3); inputSampleR *= pow(gainchase,3);}
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//this trim control cuts back extra hard because we will amplify after the distortion
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//that will shift the distortion/antidistortion curve, in order to make faded settings
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//slightly 'expanded' and fall back in the soundstage, subtly
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if (inputSampleL > 1.097) inputSampleL = 1.097;
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if (inputSampleL < -1.097) inputSampleL = -1.097;
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inputSampleL = ((sin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.8)+(sin(inputSampleL)*0.2);
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if (inputSampleR > 1.097) inputSampleR = 1.097;
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if (inputSampleR < -1.097) inputSampleR = -1.097;
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inputSampleR = ((sin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.8)+(sin(inputSampleR)*0.2);
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//this is a version of Spiral blended 80/20 with regular Density ConsoleChannel.
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//It's blending between two different harmonics in the overtones of the algorithm
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if (1.0 != gainchase && 0.0 != gainchase) {inputSampleL /= gainchase; inputSampleR /= gainchase;}
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//we re-amplify after the distortion relative to how much we cut back previously.
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//begin 32 bit stereo floating point dither
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int expon; frexpf((float)inputSampleL, &expon);
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fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
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inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
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frexpf((float)inputSampleR, &expon);
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fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
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inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 5.5e-36l * pow(2,expon+62));
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//end 32 bit stereo floating point dither
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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*in1++;
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*in2++;
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*out1++;
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*out2++;
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}
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}
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void Console7Channel::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
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{
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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double inputgain = A*1.272019649514069;
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//which is, in fact, the square root of 1.618033988749894848204586...
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//this happens to give us a boost factor where the track continues to get louder even
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//as it saturates and loses a bit of peak energy. Console7Channel channels go to 12! (.272,etc)
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//Neutral gain through the whole system with a full scale sine ia 0.772 on the gain knob
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if (gainchase != inputgain) chasespeed *= 2.0;
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if (chasespeed > sampleFrames) chasespeed = sampleFrames;
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if (gainchase < 0.0) gainchase = inputgain;
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biquadA[0] = 20000.0 / getSampleRate();
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biquadA[1] = 1.618033988749894848204586;
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double K = tan(M_PI * biquadA[0]); //lowpass
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double norm = 1.0 / (1.0 + K / biquadA[1] + K * K);
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biquadA[2] = K * K * norm;
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biquadA[3] = 2.0 * biquadA[2];
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biquadA[4] = biquadA[2];
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biquadA[5] = 2.0 * (K * K - 1.0) * norm;
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biquadA[6] = (1.0 - K / biquadA[1] + K * K) * norm;
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while (--sampleFrames >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
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if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
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double outSampleL = biquadA[2]*inputSampleL+biquadA[3]*biquadA[7]+biquadA[4]*biquadA[8]-biquadA[5]*biquadA[9]-biquadA[6]*biquadA[10];
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biquadA[8] = biquadA[7]; biquadA[7] = inputSampleL; inputSampleL = outSampleL; biquadA[10] = biquadA[9]; biquadA[9] = inputSampleL; //DF1 left
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double outSampleR = biquadA[2]*inputSampleR+biquadA[3]*biquadA[11]+biquadA[4]*biquadA[12]-biquadA[5]*biquadA[13]-biquadA[6]*biquadA[14];
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biquadA[12] = biquadA[11]; biquadA[11] = inputSampleR; inputSampleR = outSampleR; biquadA[14] = biquadA[13]; biquadA[13] = inputSampleR; //DF1 right
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chasespeed *= 0.9999; chasespeed -= 0.01; if (chasespeed < 64.0) chasespeed = 64.0;
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//we have our chase speed compensated for recent fader activity
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gainchase = (((gainchase*chasespeed)+inputgain)/(chasespeed+1.0));
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//gainchase is chasing the target, as a simple multiply gain factor
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if (1.0 != gainchase) {inputSampleL *= pow(gainchase,3); inputSampleR *= pow(gainchase,3);}
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//this trim control cuts back extra hard because we will amplify after the distortion
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//that will shift the distortion/antidistortion curve, in order to make faded settings
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//slightly 'expanded' and fall back in the soundstage, subtly
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if (inputSampleL > 1.097) inputSampleL = 1.097;
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if (inputSampleL < -1.097) inputSampleL = -1.097;
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inputSampleL = ((sin(inputSampleL*fabs(inputSampleL))/((fabs(inputSampleL) == 0.0) ?1:fabs(inputSampleL)))*0.8)+(sin(inputSampleL)*0.2);
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if (inputSampleR > 1.097) inputSampleR = 1.097;
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if (inputSampleR < -1.097) inputSampleR = -1.097;
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inputSampleR = ((sin(inputSampleR*fabs(inputSampleR))/((fabs(inputSampleR) == 0.0) ?1:fabs(inputSampleR)))*0.8)+(sin(inputSampleR)*0.2);
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//this is a version of Spiral blended 80/20 with regular Density ConsoleChannel.
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//It's blending between two different harmonics in the overtones of the algorithm
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if (1.0 != gainchase && 0.0 != gainchase) {inputSampleL /= gainchase; inputSampleR /= gainchase;}
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//we re-amplify after the distortion relative to how much we cut back previously.
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//begin 64 bit stereo floating point dither
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//int expon; frexp((double)inputSampleL, &expon);
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fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
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//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//frexp((double)inputSampleR, &expon);
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fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
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//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//end 64 bit stereo floating point dither
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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*in1++;
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*in2++;
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*out1++;
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*out2++;
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}
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}
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