annotate libaf/af_equalizer.c @ 26080:90e9ac3ee4c0

joystick.c is only ever compiled on Linux, remove pointless #ifdef around the whole file and dummy functions.
author diego
date Thu, 28 Feb 2008 18:32:17 +0000
parents 9079c9745ff9
children 72d0b1444141
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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 /* Equalizer filter, implementation of a 10 band time domain graphic
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12 equalizer using IIR filters. The IIR filters are implemented using a
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13 Direct Form II approach, but has been modified (b1 == 0 always) to
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14 save computation.
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15 */
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16
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17 #include <stdio.h>
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18 #include <stdlib.h>
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19
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20 #include <inttypes.h>
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21 #include <math.h>
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22
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23 #include "af.h"
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24
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25 #define L 2 // Storage for filter taps
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26 #define KM 10 // Max number of bands
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27
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28 #define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
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29 gives 4dB suppression @ Fc*2 and Fc/2 */
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30
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31 /* Center frequencies for band-pass filters
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32 The different frequency bands are:
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33 nr. center frequency
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34 0 31.25 Hz
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35 1 62.50 Hz
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36 2 125.0 Hz
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37 3 250.0 Hz
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38 4 500.0 Hz
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39 5 1.000 kHz
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40 6 2.000 kHz
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41 7 4.000 kHz
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42 8 8.000 kHz
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43 9 16.00 kHz
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44 */
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45 #define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
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46
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47 // Maximum and minimum gain for the bands
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48 #define G_MAX +12.0
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49 #define G_MIN -12.0
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50
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51 // Data for specific instances of this filter
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52 typedef struct af_equalizer_s
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53 {
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54 float a[KM][L]; // A weights
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55 float b[KM][L]; // B weights
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56 float wq[AF_NCH][KM][L]; // Circular buffer for W data
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57 float g[AF_NCH][KM]; // Gain factor for each channel and band
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58 int K; // Number of used eq bands
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59 int channels; // Number of channels
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60 float gain_factor; // applied at output to avoid clipping
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61 } af_equalizer_t;
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62
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63 // 2nd order Band-pass Filter design
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64 static void bp2(float* a, float* b, float fc, float q){
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65 double th= 2.0 * M_PI * fc;
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66 double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
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67
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68 a[0] = (1.0 + C) * cos(th);
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69 a[1] = -1 * C;
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70
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71 b[0] = (1.0 - C)/2.0;
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72 b[1] = -1.0050;
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73 }
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74
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75 // Initialization and runtime control
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76 static int control(struct af_instance_s* af, int cmd, void* arg)
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77 {
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78 af_equalizer_t* s = (af_equalizer_t*)af->setup;
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79
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80 switch(cmd){
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81 case AF_CONTROL_REINIT:{
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82 int k =0, i =0;
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83 float F[KM] = CF;
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84
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85 s->gain_factor=0.0;
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86
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87 // Sanity check
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88 if(!arg) return AF_ERROR;
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89
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90 af->data->rate = ((af_data_t*)arg)->rate;
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91 af->data->nch = ((af_data_t*)arg)->nch;
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92 af->data->format = AF_FORMAT_FLOAT_NE;
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93 af->data->bps = 4;
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94
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95 // Calculate number of active filters
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96 s->K=KM;
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97 while(F[s->K-1] > (float)af->data->rate/2.2)
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98 s->K--;
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99
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100 if(s->K != KM)
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101 af_msg(AF_MSG_INFO,"[equalizer] Limiting the number of filters to"
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102 " %i due to low sample rate.\n",s->K);
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103
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104 // Generate filter taps
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105 for(k=0;k<s->K;k++)
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106 bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
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107
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108 // Calculate how much this plugin adds to the overall time delay
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109 af->delay = 2 * af->data->nch * af->data->bps;
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110
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111 // Calculate gain factor to prevent clipping at output
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112 for(k=0;k<AF_NCH;k++)
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113 {
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114 for(i=0;i<KM;i++)
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115 {
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116 if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
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117 }
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118 }
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119
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120 s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
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121
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122 if(s->gain_factor > 0.0)
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123 {
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124 s->gain_factor=0.1+(s->gain_factor/12.0);
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125 }else{
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126 s->gain_factor=1;
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127 }
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128
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129 return af_test_output(af,arg);
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130 }
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131 case AF_CONTROL_COMMAND_LINE:{
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132 float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0};
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133 int i,j;
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134 sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
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135 &g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
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136 for(i=0;i<AF_NCH;i++){
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137 for(j=0;j<KM;j++){
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138 ((af_equalizer_t*)af->setup)->g[i][j] =
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139 pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
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140 }
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141 }
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142 return AF_OK;
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143 }
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144 case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
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145 float* gain = ((af_control_ext_t*)arg)->arg;
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146 int ch = ((af_control_ext_t*)arg)->ch;
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147 int k;
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148 if(ch >= AF_NCH || ch < 0)
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149 return AF_ERROR;
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150
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151 for(k = 0 ; k<KM ; k++)
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152 s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
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153
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154 return AF_OK;
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155 }
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156 case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
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157 float* gain = ((af_control_ext_t*)arg)->arg;
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158 int ch = ((af_control_ext_t*)arg)->ch;
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159 int k;
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160 if(ch >= AF_NCH || ch < 0)
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161 return AF_ERROR;
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162
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163 for(k = 0 ; k<KM ; k++)
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164 gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
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165
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166 return AF_OK;
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167 }
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168 }
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169 return AF_UNKNOWN;
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170 }
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171
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172 // Deallocate memory
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173 static void uninit(struct af_instance_s* af)
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174 {
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175 if(af->data)
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176 free(af->data);
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177 if(af->setup)
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178 free(af->setup);
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179 }
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180
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181 // Filter data through filter
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182 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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183 {
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184 af_data_t* c = data; // Current working data
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185 af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup
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186 uint32_t ci = af->data->nch; // Index for channels
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187 uint32_t nch = af->data->nch; // Number of channels
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188
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189 while(ci--){
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190 float* g = s->g[ci]; // Gain factor
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191 float* in = ((float*)c->audio)+ci;
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192 float* out = ((float*)c->audio)+ci;
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193 float* end = in + c->len/4; // Block loop end
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194
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195 while(in < end){
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196 register int k = 0; // Frequency band index
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197 register float yt = *in; // Current input sample
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198 in+=nch;
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199
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200 // Run the filters
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201 for(;k<s->K;k++){
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202 // Pointer to circular buffer wq
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203 register float* wq = s->wq[ci][k];
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204 // Calculate output from AR part of current filter
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205 register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
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206 // Calculate output form MA part of current filter
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207 yt+=(w + wq[1]*s->b[k][1])*g[k];
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208 // Update circular buffer
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209 wq[1] = wq[0];
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210 wq[0] = w;
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211 }
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212 // Calculate output
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213 *out=yt*s->gain_factor;
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214 out+=nch;
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215 }
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216 }
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217 return c;
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218 }
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219
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220 // Allocate memory and set function pointers
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221 static int af_open(af_instance_t* af){
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222 af->control=control;
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223 af->uninit=uninit;
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224 af->play=play;
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225 af->mul=1;
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226 af->data=calloc(1,sizeof(af_data_t));
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227 af->setup=calloc(1,sizeof(af_equalizer_t));
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228 if(af->data == NULL || af->setup == NULL)
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229 return AF_ERROR;
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230 return AF_OK;
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231 }
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232
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233 // Description of this filter
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234 af_info_t af_info_equalizer = {
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235 "Equalizer audio filter",
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236 "equalizer",
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237 "Anders",
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238 "",
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239 AF_FLAGS_NOT_REENTRANT,
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240 af_open
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241 };
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242
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243
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244
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245
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246
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247
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248