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-rwxr-xr-xplugins/MacVST/Recurve/source/RecurveProc.cpp118
1 files changed, 118 insertions, 0 deletions
diff --git a/plugins/MacVST/Recurve/source/RecurveProc.cpp b/plugins/MacVST/Recurve/source/RecurveProc.cpp
new file mode 100755
index 0000000..8a0b85d
--- /dev/null
+++ b/plugins/MacVST/Recurve/source/RecurveProc.cpp
@@ -0,0 +1,118 @@
+/* ========================================
+ * Recurve - Recurve.h
+ * Copyright (c) 2016 airwindows, All rights reserved
+ * ======================================== */
+
+#ifndef __Recurve_H
+#include "Recurve.h"
+#endif
+
+void Recurve::processReplacing(float **inputs, float **outputs, VstInt32 sampleFrames)
+{
+ float* in1 = inputs[0];
+ float* in2 = inputs[1];
+ float* out1 = outputs[0];
+ float* out2 = outputs[1];
+
+ 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;
+
+ inputSampleL *= 0.5;
+ inputSampleR *= 0.5;
+
+ if (gain < 0.0078125) gain = 0.0078125; if (gain > 2.0) gain = 2.0;
+ //gain of 1,0 gives you a super-clean one, gain of 2 is obviously compressing
+ //smaller number is maximum clamping, if too small it'll take a while to bounce back
+ inputSampleL *= gain; inputSampleR *= gain;
+
+ gain += sin((fabs(inputSampleL*4)>1)?4:fabs(inputSampleL*4))*pow(inputSampleL,4);
+ gain += sin((fabs(inputSampleR*4)>1)?4:fabs(inputSampleR*4))*pow(inputSampleR,4);
+ //4.71239 radians sined will turn to -1 which is the maximum gain reduction speed
+
+ inputSampleL *= 2.0;
+ inputSampleR *= 2.0;
+
+ if (inputSampleL > 0.9999) inputSampleL = 0.9999;
+ if (inputSampleR > 0.9999) inputSampleR = 0.9999;
+ if (inputSampleL < -0.9999) inputSampleL = -0.9999;
+ if (inputSampleR < -0.9999) inputSampleR = -0.9999;
+ //iron bar so people can play with this as a loudenator
+
+ //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 Recurve::processDoubleReplacing(double **inputs, double **outputs, VstInt32 sampleFrames)
+{
+ double* in1 = inputs[0];
+ double* in2 = inputs[1];
+ double* out1 = outputs[0];
+ double* out2 = outputs[1];
+
+ 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;
+
+ inputSampleL *= 0.5;
+ inputSampleR *= 0.5;
+
+ if (gain < 0.0078125) gain = 0.0078125; if (gain > 2.0) gain = 2.0;
+ //gain of 1,0 gives you a super-clean one, gain of 2 is obviously compressing
+ //smaller number is maximum clamping, if too small it'll take a while to bounce back
+ inputSampleL *= gain; inputSampleR *= gain;
+
+ gain += sin((fabs(inputSampleL*4)>1)?4:fabs(inputSampleL*4))*pow(inputSampleL,4);
+ gain += sin((fabs(inputSampleR*4)>1)?4:fabs(inputSampleR*4))*pow(inputSampleR,4);
+ //4.71239 radians sined will turn to -1 which is the maximum gain reduction speed
+
+ inputSampleL *= 2.0;
+ inputSampleR *= 2.0;
+
+ if (inputSampleL > 0.9999) inputSampleL = 0.9999;
+ if (inputSampleR > 0.9999) inputSampleR = 0.9999;
+ if (inputSampleL < -0.9999) inputSampleL = -0.9999;
+ if (inputSampleR < -0.9999) inputSampleR = -0.9999;
+ //iron bar so people can play with this as a loudenator
+
+ //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++;
+ }
+}