annotate libaf/af_equalizer.c @ 28498:d492a439d7d0

Document sws_getIdentityVec().
author stefano
date Wed, 11 Feb 2009 21:34:44 +0000
parents 72d0b1444141
children 8c706ce21c6f
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1 /*
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2 * Equalizer filter, implementation of a 10 band time domain graphic
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3 * equalizer using IIR filters. The IIR filters are implemented using a
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4 * Direct Form II approach, but has been modified (b1 == 0 always) to
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5 * save computation.
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6 *
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7 * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
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8 *
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9 * This file is part of MPlayer.
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10 *
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11 * MPlayer is free software; you can redistribute it and/or modify
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12 * it under the terms of the GNU General Public License as published by
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13 * the Free Software Foundation; either version 2 of the License, or
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14 * (at your option) any later version.
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15 *
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16 * MPlayer is distributed in the hope that it will be useful,
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17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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19 * GNU General Public License for more details.
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20 *
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21 * You should have received a copy of the GNU General Public License along
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22 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
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23 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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24 */
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25
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26 #include <stdio.h>
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27 #include <stdlib.h>
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28
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29 #include <inttypes.h>
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30 #include <math.h>
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31
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32 #include "af.h"
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33
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34 #define L 2 // Storage for filter taps
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35 #define KM 10 // Max number of bands
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36
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37 #define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
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38 gives 4dB suppression @ Fc*2 and Fc/2 */
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39
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40 /* Center frequencies for band-pass filters
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41 The different frequency bands are:
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42 nr. center frequency
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43 0 31.25 Hz
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44 1 62.50 Hz
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45 2 125.0 Hz
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46 3 250.0 Hz
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47 4 500.0 Hz
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48 5 1.000 kHz
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49 6 2.000 kHz
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50 7 4.000 kHz
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51 8 8.000 kHz
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52 9 16.00 kHz
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53 */
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54 #define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
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55
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56 // Maximum and minimum gain for the bands
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57 #define G_MAX +12.0
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58 #define G_MIN -12.0
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59
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60 // Data for specific instances of this filter
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61 typedef struct af_equalizer_s
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62 {
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63 float a[KM][L]; // A weights
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64 float b[KM][L]; // B weights
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65 float wq[AF_NCH][KM][L]; // Circular buffer for W data
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66 float g[AF_NCH][KM]; // Gain factor for each channel and band
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67 int K; // Number of used eq bands
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68 int channels; // Number of channels
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69 float gain_factor; // applied at output to avoid clipping
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70 } af_equalizer_t;
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71
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72 // 2nd order Band-pass Filter design
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73 static void bp2(float* a, float* b, float fc, float q){
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74 double th= 2.0 * M_PI * fc;
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75 double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
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76
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77 a[0] = (1.0 + C) * cos(th);
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78 a[1] = -1 * C;
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79
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80 b[0] = (1.0 - C)/2.0;
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81 b[1] = -1.0050;
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82 }
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83
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84 // Initialization and runtime control
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85 static int control(struct af_instance_s* af, int cmd, void* arg)
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86 {
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87 af_equalizer_t* s = (af_equalizer_t*)af->setup;
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88
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89 switch(cmd){
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90 case AF_CONTROL_REINIT:{
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91 int k =0, i =0;
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92 float F[KM] = CF;
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93
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94 s->gain_factor=0.0;
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95
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96 // Sanity check
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97 if(!arg) return AF_ERROR;
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98
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99 af->data->rate = ((af_data_t*)arg)->rate;
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100 af->data->nch = ((af_data_t*)arg)->nch;
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101 af->data->format = AF_FORMAT_FLOAT_NE;
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102 af->data->bps = 4;
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103
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104 // Calculate number of active filters
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105 s->K=KM;
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106 while(F[s->K-1] > (float)af->data->rate/2.2)
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107 s->K--;
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108
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109 if(s->K != KM)
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110 af_msg(AF_MSG_INFO,"[equalizer] Limiting the number of filters to"
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111 " %i due to low sample rate.\n",s->K);
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112
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113 // Generate filter taps
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114 for(k=0;k<s->K;k++)
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115 bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
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116
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117 // Calculate how much this plugin adds to the overall time delay
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118 af->delay = 2 * af->data->nch * af->data->bps;
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119
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120 // Calculate gain factor to prevent clipping at output
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121 for(k=0;k<AF_NCH;k++)
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122 {
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123 for(i=0;i<KM;i++)
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124 {
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125 if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
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126 }
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127 }
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128
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129 s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
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130
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131 if(s->gain_factor > 0.0)
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132 {
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133 s->gain_factor=0.1+(s->gain_factor/12.0);
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134 }else{
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135 s->gain_factor=1;
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136 }
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137
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138 return af_test_output(af,arg);
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139 }
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140 case AF_CONTROL_COMMAND_LINE:{
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141 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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142 int i,j;
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143 sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
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144 &g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
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145 for(i=0;i<AF_NCH;i++){
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146 for(j=0;j<KM;j++){
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147 ((af_equalizer_t*)af->setup)->g[i][j] =
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148 pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
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149 }
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150 }
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151 return AF_OK;
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152 }
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153 case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
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154 float* gain = ((af_control_ext_t*)arg)->arg;
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155 int ch = ((af_control_ext_t*)arg)->ch;
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156 int k;
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157 if(ch >= AF_NCH || ch < 0)
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158 return AF_ERROR;
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159
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160 for(k = 0 ; k<KM ; k++)
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161 s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
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162
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163 return AF_OK;
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164 }
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165 case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
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166 float* gain = ((af_control_ext_t*)arg)->arg;
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167 int ch = ((af_control_ext_t*)arg)->ch;
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168 int k;
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169 if(ch >= AF_NCH || ch < 0)
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170 return AF_ERROR;
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171
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172 for(k = 0 ; k<KM ; k++)
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173 gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
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174
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175 return AF_OK;
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176 }
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177 }
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178 return AF_UNKNOWN;
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179 }
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180
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181 // Deallocate memory
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182 static void uninit(struct af_instance_s* af)
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183 {
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184 if(af->data)
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185 free(af->data);
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186 if(af->setup)
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187 free(af->setup);
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188 }
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189
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190 // Filter data through filter
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191 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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192 {
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193 af_data_t* c = data; // Current working data
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194 af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup
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195 uint32_t ci = af->data->nch; // Index for channels
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196 uint32_t nch = af->data->nch; // Number of channels
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197
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198 while(ci--){
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199 float* g = s->g[ci]; // Gain factor
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200 float* in = ((float*)c->audio)+ci;
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201 float* out = ((float*)c->audio)+ci;
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202 float* end = in + c->len/4; // Block loop end
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203
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204 while(in < end){
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205 register int k = 0; // Frequency band index
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206 register float yt = *in; // Current input sample
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207 in+=nch;
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208
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209 // Run the filters
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210 for(;k<s->K;k++){
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211 // Pointer to circular buffer wq
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212 register float* wq = s->wq[ci][k];
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213 // Calculate output from AR part of current filter
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214 register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
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215 // Calculate output form MA part of current filter
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216 yt+=(w + wq[1]*s->b[k][1])*g[k];
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217 // Update circular buffer
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218 wq[1] = wq[0];
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219 wq[0] = w;
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220 }
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221 // Calculate output
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222 *out=yt*s->gain_factor;
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223 out+=nch;
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224 }
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225 }
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226 return c;
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227 }
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228
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229 // Allocate memory and set function pointers
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230 static int af_open(af_instance_t* af){
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231 af->control=control;
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232 af->uninit=uninit;
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233 af->play=play;
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234 af->mul=1;
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235 af->data=calloc(1,sizeof(af_data_t));
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236 af->setup=calloc(1,sizeof(af_equalizer_t));
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237 if(af->data == NULL || af->setup == NULL)
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238 return AF_ERROR;
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239 return AF_OK;
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240 }
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241
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242 // Description of this filter
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243 af_info_t af_info_equalizer = {
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244 "Equalizer audio filter",
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245 "equalizer",
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246 "Anders",
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247 "",
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248 AF_FLAGS_NOT_REENTRANT,
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249 af_open
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250 };
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251
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252
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253
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254
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255
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256
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257