annotate libaf/filter.c @ 35704:ec0cc8013b3a

(Partially) synchronize .gitignore with svn:ignore properties.
author diego
date Sun, 20 Jan 2013 15:41:53 +0000
parents 32725ca88fed
children
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1 /*
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2 * design and implementation of different types of digital filters
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3 *
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4 * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
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5 *
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6 * This file is part of MPlayer.
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7 *
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8 * MPlayer is free software; you can redistribute it and/or modify
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9 * it under the terms of the GNU General Public License as published by
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10 * the Free Software Foundation; either version 2 of the License, or
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11 * (at your option) any later version.
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12 *
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13 * MPlayer is distributed in the hope that it will be useful,
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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16 * GNU General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU General Public License along
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19 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
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20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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21 */
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22
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23 #include <string.h>
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24 #include <math.h>
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25 #include "dsp.h"
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26
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27 /******************************************************************************
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28 * FIR filter implementations
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29 ******************************************************************************/
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30
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31 /* C implementation of FIR filter y=w*x
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32
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33 n number of filter taps, where mod(n,4)==0
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34 w filter taps
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35 x input signal must be a circular buffer which is indexed backwards
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36 */
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37 inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
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38 const FLOAT_TYPE* x)
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39 {
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40 register FLOAT_TYPE y; // Output
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41 y = 0.0;
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42 do{
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43 n--;
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44 y+=w[n]*x[n];
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45 }while(n != 0);
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46 return y;
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47 }
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48
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49 /* C implementation of parallel FIR filter y(k)=w(k) * x(k) (where * denotes convolution)
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50
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51 n number of filter taps, where mod(n,4)==0
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52 d number of filters
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53 xi current index in xq
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54 w filter taps k by n big
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55 x input signal must be a circular buffers which are indexed backwards
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56 y output buffer
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57 s output buffer stride
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58 */
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59 FLOAT_TYPE* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi,
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60 const FLOAT_TYPE** w, const FLOAT_TYPE** x, FLOAT_TYPE* y,
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61 unsigned int s)
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62 {
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63 register const FLOAT_TYPE* xt = *x + xi;
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64 register const FLOAT_TYPE* wt = *w;
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65 register int nt = 2*n;
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66 while(d-- > 0){
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67 *y = af_filter_fir(n,wt,xt);
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68 wt+=n;
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69 xt+=nt;
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70 y+=s;
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71 }
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72 return y;
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73 }
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74
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75 /* Add new data to circular queue designed to be used with a parallel
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76 FIR filter, with d filters. xq is the circular queue, in pointing
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77 at the new samples, xi current index in xq and n the length of the
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78 filter. xq must be n*2 by k big, s is the index for in.
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79 */
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80 int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi,
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81 FLOAT_TYPE** xq, const FLOAT_TYPE* in, unsigned int s)
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82 {
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83 register FLOAT_TYPE* txq = *xq + xi;
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84 register int nt = n*2;
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85
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86 while(d-- >0){
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87 *txq= *(txq+n) = *in;
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88 txq+=nt;
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89 in+=s;
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90 }
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91 return (++xi)&(n-1);
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92 }
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93
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94 /******************************************************************************
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95 * FIR filter design
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96 ******************************************************************************/
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97
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98 /* Design FIR filter using the Window method
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99
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100 n filter length must be odd for HP and BS filters
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101 w buffer for the filter taps (must be n long)
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102 fc cutoff frequencies (1 for LP and HP, 2 for BP and BS)
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103 0 < fc < 1 where 1 <=> Fs/2
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104 flags window and filter type as defined in filter.h
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105 variables are ored together: i.e. LP|HAMMING will give a
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106 low pass filter designed using a hamming window
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107 opt beta constant used only when designing using kaiser windows
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108
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109 returns 0 if OK, -1 if fail
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110 */
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111 int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
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112 unsigned int flags, FLOAT_TYPE opt)
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113 {
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114 unsigned int o = n & 1; // Indicator for odd filter length
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115 unsigned int end = ((n + 1) >> 1) - o; // Loop end
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116 unsigned int i; // Loop index
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117
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118 FLOAT_TYPE k1 = 2 * M_PI; // 2*pi*fc1
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119 FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length
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120 FLOAT_TYPE k3; // 2*pi*fc2 Constant used in BP and BS design
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121 FLOAT_TYPE g = 0.0; // Gain
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122 FLOAT_TYPE t1,t2,t3; // Temporary variables
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123 FLOAT_TYPE fc1,fc2; // Cutoff frequencies
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124
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125 // Sanity check
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126 if(!w || (n == 0)) return -1;
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127
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128 // Get window coefficients
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129 switch(flags & WINDOW_MASK){
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130 case(BOXCAR):
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131 af_window_boxcar(n,w); break;
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132 case(TRIANG):
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133 af_window_triang(n,w); break;
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134 case(HAMMING):
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135 af_window_hamming(n,w); break;
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136 case(HANNING):
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137 af_window_hanning(n,w); break;
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138 case(BLACKMAN):
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139 af_window_blackman(n,w); break;
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140 case(FLATTOP):
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141 af_window_flattop(n,w); break;
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142 case(KAISER):
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143 af_window_kaiser(n,w,opt); break;
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144 default:
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145 return -1;
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146 }
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147
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148 if(flags & (LP | HP)){
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149 fc1=*fc;
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150 // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
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151 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
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152 k1 *= fc1;
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153
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154 if(flags & LP){ // Low pass filter
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155
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156 // If the filter length is odd, there is one point which is exactly
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157 // in the middle. The value at this point is 2*fCutoff*sin(x)/x,
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158 // where x is zero. To make sure nothing strange happens, we set this
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159 // value separately.
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160 if (o){
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161 w[end] = fc1 * w[end] * 2.0;
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162 g=w[end];
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163 }
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164
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165 // Create filter
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166 for (i=0 ; i<end ; i++){
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167 t1 = (FLOAT_TYPE)(i+1) - k2;
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168 w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
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169 g += 2*w[end-i-1]; // Total gain in filter
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170 }
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171 }
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172 else{ // High pass filter
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173 if (!o) // High pass filters must have odd length
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174 return -1;
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175 w[end] = 1.0 - (fc1 * w[end] * 2.0);
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176 g= w[end];
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177
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178 // Create filter
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179 for (i=0 ; i<end ; i++){
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180 t1 = (FLOAT_TYPE)(i+1);
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181 w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
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182 g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
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183 }
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184 }
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185 }
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186
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187 if(flags & (BP | BS)){
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188 fc1=fc[0];
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189 fc2=fc[1];
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190 // Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2
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191 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
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192 fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25;
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193 k3 = k1 * fc2; // 2*pi*fc2
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194 k1 *= fc1; // 2*pi*fc1
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195
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196 if(flags & BP){ // Band pass
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197 // Calculate center tap
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198 if (o){
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199 g=w[end]*(fc1+fc2);
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200 w[end] = (fc2 - fc1) * w[end] * 2.0;
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201 }
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202
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203 // Create filter
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204 for (i=0 ; i<end ; i++){
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205 t1 = (FLOAT_TYPE)(i+1) - k2;
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206 t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
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207 t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
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208 g += w[end-i-1] * (t3 + t2); // Total gain in filter
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209 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
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210 }
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211 }
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212 else{ // Band stop
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213 if (!o) // Band stop filters must have odd length
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214 return -1;
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215 w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
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216 g= w[end];
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217
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218 // Create filter
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219 for (i=0 ; i<end ; i++){
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220 t1 = (FLOAT_TYPE)(i+1);
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221 t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
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222 t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
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223 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
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224 g += 2*w[end-i-1]; // Total gain in filter
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225 }
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226 }
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227 }
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228
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229 // Normalize gain
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230 g=1/g;
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231 for (i=0; i<n; i++)
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232 w[i] *= g;
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233
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234 return 0;
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235 }
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236
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237 /* Design polyphase FIR filter from prototype filter
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238
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239 n length of prototype filter
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240 k number of polyphase components
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241 w prototype filter taps
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242 pw Parallel FIR filter
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243 g Filter gain
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244 flags FWD forward indexing
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245 REW reverse indexing
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246 ODD multiply every 2nd filter tap by -1 => HP filter
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247
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248 returns 0 if OK, -1 if fail
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249 */
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250 int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w,
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251 FLOAT_TYPE** pw, FLOAT_TYPE g, unsigned int flags)
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252 {
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253 int l = (int)n/k; // Length of individual FIR filters
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254 int i; // Counters
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255 int j;
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256 FLOAT_TYPE t; // g * w[i]
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257
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258 // Sanity check
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259 if(l<1 || k<1 || !w || !pw)
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260 return -1;
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261
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262 // Do the stuff
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263 if(flags&REW){
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264 for(j=l-1;j>-1;j--){//Columns
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265 for(i=0;i<(int)k;i++){//Rows
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266 t=g * *w++;
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267 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? -1 : 1) : 1);
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268 }
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269 }
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270 }
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271 else{
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272 for(j=0;j<l;j++){//Columns
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273 for(i=0;i<(int)k;i++){//Rows
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274 t=g * *w++;
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275 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? 1 : -1) : 1);
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276 }
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277 }
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278 }
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279 return -1;
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280 }
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281
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282 /******************************************************************************
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283 * IIR filter design
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284 ******************************************************************************/
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285
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286 /* Helper functions for the bilinear transform */
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287
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288 /* Pre-warp the coefficients of a numerator or denominator.
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289 Note that a0 is assumed to be 1, so there is no wrapping
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290 of it.
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291 */
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292 static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
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293 {
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294 FLOAT_TYPE wp;
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295 wp = 2.0 * fs * tan(M_PI * fc / fs);
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296 a[2] = a[2]/(wp * wp);
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297 a[1] = a[1]/wp;
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298 }
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299
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300 /* Transform the numerator and denominator coefficients of s-domain
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301 biquad section into corresponding z-domain coefficients.
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302
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303 The transfer function for z-domain is:
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304
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305 1 + alpha1 * z^(-1) + alpha2 * z^(-2)
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306 H(z) = -------------------------------------
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307 1 + beta1 * z^(-1) + beta2 * z^(-2)
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308
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309 Store the 4 IIR coefficients in array pointed by coef in following
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310 order:
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311 beta1, beta2 (denominator)
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312 alpha1, alpha2 (numerator)
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313
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314 Arguments:
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315 a - s-domain numerator coefficients
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316 b - s-domain denominator coefficients
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317 k - filter gain factor. Initially set to 1 and modified by each
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318 biquad section in such a way, as to make it the
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319 coefficient by which to multiply the overall filter gain
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320 in order to achieve a desired overall filter gain,
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321 specified in initial value of k.
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322 fs - sampling rate (Hz)
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323 coef - array of z-domain coefficients to be filled in.
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324
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325 Return: On return, set coef z-domain coefficients and k to the gain
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326 required to maintain overall gain = 1.0;
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327 */
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328 static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE* k,
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329 FLOAT_TYPE fs, FLOAT_TYPE *coef)
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330 {
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331 FLOAT_TYPE ad, bd;
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332
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333 /* alpha (Numerator in s-domain) */
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334 ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0];
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335 /* beta (Denominator in s-domain) */
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336 bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0];
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337
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338 /* Update gain constant for this section */
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339 *k *= ad/bd;
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340
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341 /* Denominator */
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342 *coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */
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343 *coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */
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344
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345 /* Numerator */
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346 *coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */
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347 *coef = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad; /* alpha2 */
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348 }
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349
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350
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351
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352 /* IIR filter design using bilinear transform and prewarp. Transforms
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353 2nd order s domain analog filter into a digital IIR biquad link. To
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354 create a filter fill in a, b, Q and fs and make space for coef and k.
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355
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356
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357 Example Butterworth design:
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358
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359 Below are Butterworth polynomials, arranged as a series of 2nd
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360 order sections:
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361
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362 Note: n is filter order.
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363
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364 n Polynomials
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365 -------------------------------------------------------------------
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366 2 s^2 + 1.4142s + 1
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367 4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
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368 6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
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369
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370 For n=4 we have following equation for the filter transfer function:
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371 1 1
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372 T(s) = --------------------------- * ----------------------------
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373 s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1
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374
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375 The filter consists of two 2nd order sections since highest s power
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376 is 2. Now we can take the coefficients, or the numbers by which s
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377 is multiplied and plug them into a standard formula to be used by
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378 bilinear transform.
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379
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380 Our standard form for each 2nd order section is:
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381
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382 a2 * s^2 + a1 * s + a0
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383 H(s) = ----------------------
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384 b2 * s^2 + b1 * s + b0
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385
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386 Note that Butterworth numerator is 1 for all filter sections, which
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387 means s^2 = 0 and s^1 = 0
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388
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389 Let's convert standard Butterworth polynomials into this form:
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390
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391 0 + 0 + 1 0 + 0 + 1
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392 --------------------------- * --------------------------
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393 1 + ((1/Q) * 0.765367) + 1 1 + ((1/Q) * 1.847759) + 1
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394
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395 Section 1:
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396 a2 = 0; a1 = 0; a0 = 1;
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397 b2 = 1; b1 = 0.765367; b0 = 1;
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398
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399 Section 2:
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400 a2 = 0; a1 = 0; a0 = 1;
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401 b2 = 1; b1 = 1.847759; b0 = 1;
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402
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403 Q is filter quality factor or resonance, in the range of 1 to
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404 1000. The overall filter Q is a product of all 2nd order stages.
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405 For example, the 6th order filter (3 stages, or biquads) with
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406 individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
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407
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408
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409 Arguments:
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410 a - s-domain numerator coefficients, a[1] is always assumed to be 1.0
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411 b - s-domain denominator coefficients
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412 Q - Q value for the filter
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413 k - filter gain factor. Initially set to 1 and modified by each
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414 biquad section in such a way, as to make it the
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415 coefficient by which to multiply the overall filter gain
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416 in order to achieve a desired overall filter gain,
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417 specified in initial value of k.
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418 fs - sampling rate (Hz)
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419 coef - array of z-domain coefficients to be filled in.
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420
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421 Note: Upon return from each call, the k argument will be set to a
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422 value, by which to multiply our actual signal in order for the gain
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423 to be one. On second call to szxform() we provide k that was
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424 changed by the previous section. During actual audio filtering
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425 k can be used for gain compensation.
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426
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427 return -1 if fail 0 if success.
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428 */
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429 int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FLOAT_TYPE fc,
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430 FLOAT_TYPE fs, FLOAT_TYPE *k, FLOAT_TYPE *coef)
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431 {
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432 FLOAT_TYPE at[3];
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433 FLOAT_TYPE bt[3];
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434
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435 if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
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436 return -1;
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437
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438 memcpy(at,a,3*sizeof(FLOAT_TYPE));
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439 memcpy(bt,b,3*sizeof(FLOAT_TYPE));
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440
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441 bt[1]/=Q;
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442
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443 /* Calculate a and b and overwrite the original values */
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444 af_filter_prewarp(at, fc, fs);
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445 af_filter_prewarp(bt, fc, fs);
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446 /* Execute bilinear transform */
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447 af_filter_bilinear(at, bt, k, fs, coef);
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448
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449 return 0;
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450 }