summaryrefslogtreecommitdiff
path: root/gst/modplug/libmodplug/snd_flt.cpp
blob: 00506d4af0c46d049bec394c96a4075dd9c2e8e4 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
/*
 * This source code is public domain.
 *
 * Authors: Olivier Lapicque <olivierl@jps.net>
*/

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#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_PRECISION	8192

#ifndef NO_FILTER

#ifdef MSC_VER
#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);

	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