annotate libaf/filter.c @ 25170:b0e5772d519e

Fix multiple inclusion guards, identifiers starting with __ are reserved for the system.
author diego
date Thu, 29 Nov 2007 14:23:24 +0000
parents a54a25221b79
children 07abe94a9cc4
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
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1 /*=============================================================================
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2 //
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3 // This software has been released under the terms of the GNU General Public
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4 // license. See http://www.gnu.org/copyleft/gpl.html for details.
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5 //
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6 // Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
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7 //
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8 //=============================================================================
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9 */
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10
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11 /* Design and implementation of different types of digital filters
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12
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13 */
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14 #include <string.h>
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15 #include <math.h>
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16 #include "dsp.h"
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17
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18 /******************************************************************************
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19 * FIR filter implementations
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20 ******************************************************************************/
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21
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22 /* C implementation of FIR filter y=w*x
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23
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24 n number of filter taps, where mod(n,4)==0
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25 w filter taps
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26 x input signal must be a circular buffer which is indexed backwards
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27 */
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28 inline _ftype_t af_filter_fir(register unsigned int n, _ftype_t* w, _ftype_t* x)
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29 {
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30 register _ftype_t y; // Output
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31 y = 0.0;
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32 do{
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33 n--;
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34 y+=w[n]*x[n];
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35 }while(n != 0);
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36 return y;
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37 }
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38
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39 /* C implementation of parallel FIR filter y(k)=w(k) * x(k) (where * denotes convolution)
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40
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41 n number of filter taps, where mod(n,4)==0
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42 d number of filters
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43 xi current index in xq
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44 w filter taps k by n big
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45 x input signal must be a circular buffers which are indexed backwards
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46 y output buffer
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47 s output buffer stride
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48 */
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49 _ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s)
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50 {
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51 register _ftype_t* xt = *x + xi;
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52 register _ftype_t* wt = *w;
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53 register int nt = 2*n;
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54 while(d-- > 0){
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55 *y = af_filter_fir(n,wt,xt);
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56 wt+=n;
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57 xt+=nt;
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58 y+=s;
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59 }
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60 return y;
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61 }
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62
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63 /* Add new data to circular queue designed to be used with a parallel
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64 FIR filter, with d filters. xq is the circular queue, in pointing
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65 at the new samples, xi current index in xq and n the length of the
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66 filter. xq must be n*2 by k big, s is the index for in.
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67 */
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68 int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s)
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69 {
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70 register _ftype_t* txq = *xq + xi;
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71 register int nt = n*2;
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72
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73 while(d-- >0){
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74 *txq= *(txq+n) = *in;
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75 txq+=nt;
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76 in+=s;
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77 }
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78 return (++xi)&(n-1);
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79 }
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80
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81 /******************************************************************************
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82 * FIR filter design
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83 ******************************************************************************/
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84
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85 /* Design FIR filter using the Window method
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86
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87 n filter length must be odd for HP and BS filters
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88 w buffer for the filter taps (must be n long)
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89 fc cutoff frequencies (1 for LP and HP, 2 for BP and BS)
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90 0 < fc < 1 where 1 <=> Fs/2
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91 flags window and filter type as defined in filter.h
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92 variables are ored together: i.e. LP|HAMMING will give a
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93 low pass filter designed using a hamming window
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94 opt beta constant used only when designing using kaiser windows
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95
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96 returns 0 if OK, -1 if fail
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97 */
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98 int af_filter_design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _ftype_t opt)
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99 {
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100 unsigned int o = n & 1; // Indicator for odd filter length
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101 unsigned int end = ((n + 1) >> 1) - o; // Loop end
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102 unsigned int i; // Loop index
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103
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104 _ftype_t k1 = 2 * M_PI; // 2*pi*fc1
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105 _ftype_t k2 = 0.5 * (_ftype_t)(1 - o);// Constant used if the filter has even length
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106 _ftype_t k3; // 2*pi*fc2 Constant used in BP and BS design
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107 _ftype_t g = 0.0; // Gain
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108 _ftype_t t1,t2,t3; // Temporary variables
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109 _ftype_t fc1,fc2; // Cutoff frequencies
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110
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111 // Sanity check
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112 if(!w || (n == 0)) return -1;
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113
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114 // Get window coefficients
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115 switch(flags & WINDOW_MASK){
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116 case(BOXCAR):
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117 af_window_boxcar(n,w); break;
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118 case(TRIANG):
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119 af_window_triang(n,w); break;
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120 case(HAMMING):
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121 af_window_hamming(n,w); break;
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122 case(HANNING):
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123 af_window_hanning(n,w); break;
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124 case(BLACKMAN):
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125 af_window_blackman(n,w); break;
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126 case(FLATTOP):
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127 af_window_flattop(n,w); break;
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128 case(KAISER):
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129 af_window_kaiser(n,w,opt); break;
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130 default:
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131 return -1;
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132 }
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133
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134 if(flags & (LP | HP)){
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135 fc1=*fc;
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136 // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
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137 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
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138 k1 *= fc1;
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139
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140 if(flags & LP){ // Low pass filter
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141
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142 // If the filter length is odd, there is one point which is exactly
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143 // in the middle. The value at this point is 2*fCutoff*sin(x)/x,
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144 // where x is zero. To make sure nothing strange happens, we set this
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145 // value separately.
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146 if (o){
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147 w[end] = fc1 * w[end] * 2.0;
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148 g=w[end];
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149 }
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150
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151 // Create filter
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152 for (i=0 ; i<end ; i++){
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153 t1 = (_ftype_t)(i+1) - k2;
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154 w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
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155 g += 2*w[end-i-1]; // Total gain in filter
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156 }
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157 }
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158 else{ // High pass filter
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159 if (!o) // High pass filters must have odd length
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160 return -1;
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161 w[end] = 1.0 - (fc1 * w[end] * 2.0);
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162 g= w[end];
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163
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164 // Create filter
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165 for (i=0 ; i<end ; i++){
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166 t1 = (_ftype_t)(i+1);
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167 w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
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168 g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
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169 }
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170 }
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171 }
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172
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173 if(flags & (BP | BS)){
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174 fc1=fc[0];
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175 fc2=fc[1];
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176 // Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2
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177 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
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178 fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25;
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179 k3 = k1 * fc2; // 2*pi*fc2
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180 k1 *= fc1; // 2*pi*fc1
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181
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182 if(flags & BP){ // Band pass
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183 // Calculate center tap
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184 if (o){
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185 g=w[end]*(fc1+fc2);
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186 w[end] = (fc2 - fc1) * w[end] * 2.0;
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187 }
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188
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189 // Create filter
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190 for (i=0 ; i<end ; i++){
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191 t1 = (_ftype_t)(i+1) - k2;
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192 t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
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193 t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
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194 g += w[end-i-1] * (t3 + t2); // Total gain in filter
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195 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
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196 }
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197 }
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198 else{ // Band stop
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199 if (!o) // Band stop filters must have odd length
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200 return -1;
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201 w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
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202 g= w[end];
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203
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204 // Create filter
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205 for (i=0 ; i<end ; i++){
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206 t1 = (_ftype_t)(i+1);
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207 t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
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208 t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
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209 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
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210 g += 2*w[end-i-1]; // Total gain in filter
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211 }
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212 }
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213 }
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214
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215 // Normalize gain
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216 g=1/g;
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217 for (i=0; i<n; i++)
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218 w[i] *= g;
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219
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220 return 0;
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221 }
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222
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223 /* Design polyphase FIR filter from prototype filter
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224
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225 n length of prototype filter
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226 k number of polyphase components
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227 w prototype filter taps
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228 pw Parallel FIR filter
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229 g Filter gain
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230 flags FWD forward indexing
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231 REW reverse indexing
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232 ODD multiply every 2nd filter tap by -1 => HP filter
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233
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234 returns 0 if OK, -1 if fail
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235 */
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236 int af_filter_design_pfir(unsigned int n, unsigned int k, _ftype_t* w, _ftype_t** pw, _ftype_t g, unsigned int flags)
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237 {
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238 int l = (int)n/k; // Length of individual FIR filters
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239 int i; // Counters
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240 int j;
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241 _ftype_t t; // g * w[i]
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242
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243 // Sanity check
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244 if(l<1 || k<1 || !w || !pw)
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245 return -1;
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246
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247 // Do the stuff
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248 if(flags&REW){
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249 for(j=l-1;j>-1;j--){//Columns
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250 for(i=0;i<(int)k;i++){//Rows
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251 t=g * *w++;
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252 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? -1 : 1) : 1);
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253 }
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254 }
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255 }
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256 else{
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257 for(j=0;j<l;j++){//Columns
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258 for(i=0;i<(int)k;i++){//Rows
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259 t=g * *w++;
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260 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? 1 : -1) : 1);
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261 }
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262 }
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263 }
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264 return -1;
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265 }
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266
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267 /******************************************************************************
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268 * IIR filter design
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269 ******************************************************************************/
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270
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271 /* Helper functions for the bilinear transform */
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272
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273 /* Pre-warp the coefficients of a numerator or denominator.
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274 Note that a0 is assumed to be 1, so there is no wrapping
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275 of it.
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276 */
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277 static void af_filter_prewarp(_ftype_t* a, _ftype_t fc, _ftype_t fs)
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278 {
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279 _ftype_t wp;
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280 wp = 2.0 * fs * tan(M_PI * fc / fs);
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281 a[2] = a[2]/(wp * wp);
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282 a[1] = a[1]/wp;
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283 }
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284
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285 /* Transform the numerator and denominator coefficients of s-domain
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286 biquad section into corresponding z-domain coefficients.
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287
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288 The transfer function for z-domain is:
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289
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290 1 + alpha1 * z^(-1) + alpha2 * z^(-2)
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291 H(z) = -------------------------------------
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292 1 + beta1 * z^(-1) + beta2 * z^(-2)
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293
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294 Store the 4 IIR coefficients in array pointed by coef in following
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295 order:
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296 beta1, beta2 (denominator)
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297 alpha1, alpha2 (numerator)
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298
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299 Arguments:
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300 a - s-domain numerator coefficients
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301 b - s-domain denominator coefficients
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302 k - filter gain factor. Initially set to 1 and modified by each
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303 biquad section in such a way, as to make it the
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304 coefficient by which to multiply the overall filter gain
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305 in order to achieve a desired overall filter gain,
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306 specified in initial value of k.
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307 fs - sampling rate (Hz)
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308 coef - array of z-domain coefficients to be filled in.
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309
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310 Return: On return, set coef z-domain coefficients and k to the gain
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311 required to maintain overall gain = 1.0;
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312 */
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313 static void af_filter_bilinear(_ftype_t* a, _ftype_t* b, _ftype_t* k, _ftype_t fs, _ftype_t *coef)
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314 {
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315 _ftype_t ad, bd;
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316
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317 /* alpha (Numerator in s-domain) */
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318 ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0];
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319 /* beta (Denominator in s-domain) */
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320 bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0];
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321
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322 /* Update gain constant for this section */
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323 *k *= ad/bd;
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324
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325 /* Denominator */
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326 *coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */
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327 *coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */
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328
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329 /* Numerator */
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330 *coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */
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331 *coef = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad; /* alpha2 */
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332 }
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333
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334
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335
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336 /* IIR filter design using bilinear transform and prewarp. Transforms
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337 2nd order s domain analog filter into a digital IIR biquad link. To
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338 create a filter fill in a, b, Q and fs and make space for coef and k.
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339
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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340
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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341 Example Butterworth design:
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342
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343 Below are Butterworth polynomials, arranged as a series of 2nd
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344 order sections:
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345
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346 Note: n is filter order.
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347
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348 n Polynomials
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349 -------------------------------------------------------------------
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350 2 s^2 + 1.4142s + 1
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351 4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
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352 6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
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353
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354 For n=4 we have following equation for the filter transfer function:
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355 1 1
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356 T(s) = --------------------------- * ----------------------------
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357 s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1
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358
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359 The filter consists of two 2nd order sections since highest s power
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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360 is 2. Now we can take the coefficients, or the numbers by which s
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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361 is multiplied and plug them into a standard formula to be used by
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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362 bilinear transform.
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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363
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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364 Our standard form for each 2nd order section is:
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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365
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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366 a2 * s^2 + a1 * s + a0
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367 H(s) = ----------------------
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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368 b2 * s^2 + b1 * s + b0
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369
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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370 Note that Butterworth numerator is 1 for all filter sections, which
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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371 means s^2 = 0 and s^1 = 0
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372
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373 Let's convert standard Butterworth polynomials into this form:
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374
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375 0 + 0 + 1 0 + 0 + 1
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376 --------------------------- * --------------------------
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377 1 + ((1/Q) * 0.765367) + 1 1 + ((1/Q) * 1.847759) + 1
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378
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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379 Section 1:
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380 a2 = 0; a1 = 0; a0 = 1;
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381 b2 = 1; b1 = 0.765367; b0 = 1;
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382
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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383 Section 2:
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384 a2 = 0; a1 = 0; a0 = 1;
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385 b2 = 1; b1 = 1.847759; b0 = 1;
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386
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387 Q is filter quality factor or resonance, in the range of 1 to
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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388 1000. The overall filter Q is a product of all 2nd order stages.
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389 For example, the 6th order filter (3 stages, or biquads) with
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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390 individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
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391
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392
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393 Arguments:
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394 a - s-domain numerator coefficients, a[1] is always assumed to be 1.0
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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395 b - s-domain denominator coefficients
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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396 Q - Q value for the filter
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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397 k - filter gain factor. Initially set to 1 and modified by each
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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398 biquad section in such a way, as to make it the
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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399 coefficient by which to multiply the overall filter gain
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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400 in order to achieve a desired overall filter gain,
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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401 specified in initial value of k.
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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402 fs - sampling rate (Hz)
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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403 coef - array of z-domain coefficients to be filled in.
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404
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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405 Note: Upon return from each call, the k argument will be set to a
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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406 value, by which to multiply our actual signal in order for the gain
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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407 to be one. On second call to szxform() we provide k that was
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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408 changed by the previous section. During actual audio filtering
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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409 k can be used for gain compensation.
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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410
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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411 return -1 if fail 0 if success.
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412 */
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413 int af_filter_szxform(_ftype_t* a, _ftype_t* b, _ftype_t Q, _ftype_t fc, _ftype_t fs, _ftype_t *k, _ftype_t *coef)
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414 {
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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415 _ftype_t at[3];
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416 _ftype_t bt[3];
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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417
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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418 if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
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419 return -1;
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420
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421 memcpy(at,a,3*sizeof(_ftype_t));
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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422 memcpy(bt,b,3*sizeof(_ftype_t));
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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423
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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424 bt[1]/=Q;
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425
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426 /* Calculate a and b and overwrite the original values */
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427 af_filter_prewarp(at, fc, fs);
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428 af_filter_prewarp(bt, fc, fs);
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429 /* Execute bilinear transform */
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430 af_filter_bilinear(at, bt, k, fs, coef);
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431
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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432 return 0;
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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433 }
a1578b329cc0 Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
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434