Mercurial > libavcodec.hg
annotate resample2.c @ 2097:dd81e5be0c7a libavcodec
make the tables static
author | alex |
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date | Sat, 26 Jun 2004 10:08:49 +0000 |
parents | 76cdbe832239 |
children | 84637c6ca74e |
rev | line source |
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2082
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1 /* |
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2 * audio resampling |
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3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> |
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4 * |
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5 * This library is free software; you can redistribute it and/or |
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6 * modify it under the terms of the GNU Lesser General Public |
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7 * License as published by the Free Software Foundation; either |
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8 * version 2 of the License, or (at your option) any later version. |
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9 * |
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10 * This library is distributed in the hope that it will be useful, |
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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13 * Lesser General Public License for more details. |
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14 * |
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15 * You should have received a copy of the GNU Lesser General Public |
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16 * License along with this library; if not, write to the Free Software |
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17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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18 * |
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19 */ |
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20 |
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21 /** |
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22 * @file resample2.c |
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23 * audio resampling |
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24 * @author Michael Niedermayer <michaelni@gmx.at> |
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25 */ |
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26 |
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27 #include "avcodec.h" |
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28 #include "common.h" |
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29 |
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30 #define PHASE_SHIFT 10 |
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31 #define PHASE_COUNT (1<<PHASE_SHIFT) |
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32 #define PHASE_MASK (PHASE_COUNT-1) |
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33 #define FILTER_SHIFT 15 |
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34 |
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35 typedef struct AVResampleContext{ |
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36 short *filter_bank; |
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37 int filter_length; |
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38 int ideal_dst_incr; |
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39 int dst_incr; |
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40 int index; |
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41 int frac; |
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42 int src_incr; |
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43 int compensation_distance; |
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44 }AVResampleContext; |
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45 |
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46 /** |
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47 * 0th order modified bessel function of the first kind. |
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48 */ |
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49 double bessel(double x){ |
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50 double v=1; |
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51 double t=1; |
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52 int i; |
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53 |
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54 for(i=1; i<50; i++){ |
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55 t *= i; |
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56 v += pow(x*x/4, i)/(t*t); |
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57 } |
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58 return v; |
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59 } |
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60 |
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61 /** |
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62 * builds a polyphase filterbank. |
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63 * @param factor resampling factor |
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64 * @param scale wanted sum of coefficients for each filter |
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65 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16 |
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66 */ |
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67 void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){ |
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68 int ph, i, v; |
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69 double x, y, w, tab[tap_count]; |
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70 const int center= (tap_count-1)/2; |
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71 |
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72 /* if upsampling, only need to interpolate, no filter */ |
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73 if (factor > 1.0) |
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74 factor = 1.0; |
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75 |
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76 for(ph=0;ph<phase_count;ph++) { |
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77 double norm = 0; |
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78 double e= 0; |
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79 for(i=0;i<tap_count;i++) { |
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80 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
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81 if (x == 0) y = 1.0; |
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82 else y = sin(x) / x; |
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83 switch(type){ |
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84 case 0:{ |
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85 const float d= -0.5; //first order derivative = -0.5 |
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86 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
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87 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
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88 else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
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89 break;} |
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90 case 1: |
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91 w = 2.0*x / (factor*tap_count) + M_PI; |
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92 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
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93 break; |
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94 case 2: |
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95 w = 2.0*x / (factor*tap_count*M_PI); |
2083 | 96 y *= bessel(16*sqrt(FFMAX(1-w*w, 0))); |
2082
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97 break; |
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98 } |
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99 |
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100 tab[i] = y; |
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101 norm += y; |
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102 } |
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103 |
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104 /* normalize so that an uniform color remains the same */ |
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105 for(i=0;i<tap_count;i++) { |
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106 v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767); |
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107 filter[ph * tap_count + i] = v; |
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108 e += tab[i] * scale / norm - v; |
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109 } |
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110 } |
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111 } |
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112 |
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113 /** |
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114 * initalizes a audio resampler. |
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115 * note, if either rate is not a integer then simply scale both rates up so they are |
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116 */ |
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117 AVResampleContext *av_resample_init(int out_rate, int in_rate){ |
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118 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); |
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119 double factor= FFMIN(out_rate / (double)in_rate, 1.0); |
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120 |
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121 memset(c, 0, sizeof(AVResampleContext)); |
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122 |
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123 c->filter_length= ceil(16.0/factor); |
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124 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short)); |
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125 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1); |
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126 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 1]= (1<<FILTER_SHIFT)-1; |
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127 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 1; |
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128 |
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129 c->src_incr= out_rate; |
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130 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT; |
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131 c->index= -PHASE_COUNT*((c->filter_length-1)/2); |
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132 |
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133 return c; |
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134 } |
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135 |
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136 void av_resample_close(AVResampleContext *c){ |
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137 av_freep(&c->filter_bank); |
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138 av_freep(&c); |
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139 } |
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140 |
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141 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
2083 | 142 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; |
2082
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143 c->compensation_distance= compensation_distance; |
2083 | 144 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; |
2082
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145 } |
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146 |
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147 /** |
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148 * resamples. |
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149 * @param src an array of unconsumed samples |
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150 * @param consumed the number of samples of src which have been consumed are returned here |
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151 * @param src_size the number of unconsumed samples available |
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152 * @param dst_size the amount of space in samples available in dst |
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153 * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context |
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154 * @return the number of samples written in dst or -1 if an error occured |
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155 */ |
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156 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ |
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157 int dst_index, i; |
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158 int index= c->index; |
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159 int frac= c->frac; |
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160 int dst_incr_frac= c->dst_incr % c->src_incr; |
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161 int dst_incr= c->dst_incr / c->src_incr; |
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162 |
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163 if(c->compensation_distance && c->compensation_distance < dst_size) |
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164 dst_size= c->compensation_distance; |
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165 |
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166 for(dst_index=0; dst_index < dst_size; dst_index++){ |
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167 short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK); |
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168 int sample_index= index >> PHASE_SHIFT; |
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169 int val=0; |
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170 |
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171 if(sample_index < 0){ |
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172 for(i=0; i<c->filter_length; i++) |
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173 val += src[ABS(sample_index + i)] * filter[i]; |
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174 }else if(sample_index + c->filter_length > src_size){ |
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175 break; |
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176 }else{ |
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177 #if 0 |
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178 int64_t v=0; |
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179 int sub_phase= (frac<<12) / c->src_incr; |
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180 for(i=0; i<c->filter_length; i++){ |
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181 int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase; |
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182 v += src[sample_index + i] * coeff; |
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183 } |
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184 val= v>>12; |
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185 #else |
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186 for(i=0; i<c->filter_length; i++){ |
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187 val += src[sample_index + i] * filter[i]; |
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188 } |
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189 #endif |
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190 } |
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191 |
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192 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; |
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193 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; |
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194 |
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195 frac += dst_incr_frac; |
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196 index += dst_incr; |
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197 if(frac >= c->src_incr){ |
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198 frac -= c->src_incr; |
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199 index++; |
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200 } |
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201 } |
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202 if(update_ctx){ |
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203 if(c->compensation_distance){ |
2083 | 204 c->compensation_distance -= dst_index; |
2082
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205 if(!c->compensation_distance) |
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206 c->dst_incr= c->ideal_dst_incr; |
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207 } |
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208 c->frac= frac; |
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209 c->index=0; |
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210 } |
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211 *consumed= index >> PHASE_SHIFT; |
2083 | 212 #if 0 |
213 if(update_ctx && !c->compensation_distance){ | |
214 #undef rand | |
215 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); | |
216 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); | |
217 } | |
218 #endif | |
219 | |
2082
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220 return dst_index; |
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221 } |