1 files changed, 104 insertions, 0 deletions

diff --git a/core/multimedia/opieplayer/modplug/snd_flt.cpp b/core/multimedia/opieplayer/modplug/snd_flt.cpp
new file mode 100644
index 0000000..5ac241e
--- a/dev/null
+++ b/core/multimedia/opieplayer/modplug/snd_flt.cpp
@@ -0,0 +1,104 @@
+/*
+ * This program is  free software; you can redistribute it  and modify it
+ * under the terms of the GNU  General Public License as published by the
+ * Free Software Foundation; either version 2  of the license or (at your
+ * option) any later version.
+ *
+ * Authors: Olivier Lapicque <olivierl@jps.net>
+*/
+
+#include "stdafx.h"
+#include "sndfile.h"
+
+// AWE32: cutoff = reg[0-255] * 31.25 + 100 -> [100Hz-8060Hz]
+// EMU10K1 docs: cutoff = reg[0-127]*62+100
+        #define FILTER_PRECISION8192
+
+#ifndef NO_FILTER
+
+#ifdef WIN32
+#define _ASM_MATH
+#endif
+
+#ifdef _ASM_MATH
+
+// pow(a,b) returns a^^b -> 2^^(b.log2(a))
+static float pow(float a, float b)
+{
+        long tmpint;
+        float result;
+        _asm {
+                fld b                   // Load b
+                fld a                   // Load a
+                fyl2x                   // ST(0) = b.log2(a)
+                fist tmpint             // Store integer exponent
+                fisub tmpint    // ST(0) = -1 <= (b*log2(a)) <= 1
+                f2xm1                   // ST(0) = 2^(x)-1
+                fild tmpint             // load integer exponent
+                fld1                    // Load 1
+                fscale                  // ST(0) = 2^ST(1)
+                fstp ST(1)              // Remove the integer from the stack
+                fmul ST(1), ST(0)// multiply with fractional part
+                faddp ST(1), ST(0)// add integer_part
+                fstp result             // Store the result
+        }
+        return result;
+}
+
+
+#else
+
+#include <math.h>
+
+#endif // _ASM_MATH
+
+
+DWORD CSoundFile::CutOffToFrequency(UINT nCutOff, int flt_modifier) const
+//-----------------------------------------------------------------------
+{
+        float Fc;
+
+        if (m_dwSongFlags & SONG_EXFILTERRANGE)
+                Fc = 110.0f * pow(2.0f, 0.25f + ((float)(nCutOff*(flt_modifier+256)))/(21.0f*512.0f));
+        else
+                Fc = 110.0f * pow(2.0f, 0.25f + ((float)(nCutOff*(flt_modifier+256)))/(24.0f*512.0f));
+        LONG freq = (LONG)Fc;
+        if (freq < 120) return 120;
+        if (freq > 10000) return 10000;
+        if (freq*2 > (LONG)gdwMixingFreq) freq = gdwMixingFreq>>1;
+        return (DWORD)freq;
+}
+
+
+// Simple 2-poles resonant filter
+void CSoundFile::SetupChannelFilter(MODCHANNEL *pChn, BOOL bReset, int flt_modifier) const
+//----------------------------------------------------------------------------------------
+{
+        float fc = (float)CutOffToFrequency(pChn->nCutOff, flt_modifier);
+        float fs = (float)gdwMixingFreq;
+        float fg, fb0, fb1;
+
+        fc *= (float)(2.0*3.14159265358/fs);
+        float dmpfac = pow(10.0f, -((24.0f / 128.0f)*(float)pChn->nResonance) / 20.0f);
+        float d = (1.0f-2.0f*dmpfac)* fc;
+        if (d>2.0) d = 2.0;
+        d = (2.0f*dmpfac - d)/fc;
+        float e = pow(1.0f/fc,2.0);
+
+        fg=1/(1+d+e);
+        fb0=(d+e+e)/(1+d+e);
+        fb1=-e/(1+d+e);
+
+        pChn->nFilter_A0 = (int)(fg * FILTER_PRECISION);
+        pChn->nFilter_B0 = (int)(fb0 * FILTER_PRECISION);
+        pChn->nFilter_B1 = (int)(fb1 * FILTER_PRECISION);
+        
+        if (bReset)
+        {
+                pChn->nFilter_Y1 = pChn->nFilter_Y2 = 0;
+                pChn->nFilter_Y3 = pChn->nFilter_Y4 = 0;
+        }
+        pChn->dwFlags |= CHN_FILTER;
+}
+
+#endif // NO_FILTER