Mercurial > libavcodec.hg
annotate resample2.c @ 2203:edf4787f3baf libavcodec
typo
author | michael |
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date | Thu, 02 Sep 2004 18:46:41 +0000 |
parents | 3f52c129d00f |
children | b88e05b9b445 |
rev | line source |
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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 #include "dsputil.h" |
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30 |
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31 #define PHASE_SHIFT 10 |
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32 #define PHASE_COUNT (1<<PHASE_SHIFT) |
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33 #define PHASE_MASK (PHASE_COUNT-1) |
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34 #define FILTER_SHIFT 15 |
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35 |
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36 typedef struct AVResampleContext{ |
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37 short *filter_bank; |
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38 int filter_length; |
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39 int ideal_dst_incr; |
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40 int dst_incr; |
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41 int index; |
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42 int frac; |
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43 int src_incr; |
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44 int compensation_distance; |
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45 }AVResampleContext; |
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46 |
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47 /** |
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48 * 0th order modified bessel function of the first kind. |
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49 */ |
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50 double bessel(double x){ |
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51 double v=1; |
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52 double t=1; |
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53 int i; |
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54 |
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55 for(i=1; i<50; i++){ |
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56 t *= i; |
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57 v += pow(x*x/4, i)/(t*t); |
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58 } |
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59 return v; |
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60 } |
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61 |
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62 /** |
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63 * builds a polyphase filterbank. |
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64 * @param factor resampling factor |
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65 * @param scale wanted sum of coefficients for each filter |
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66 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16 |
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67 */ |
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68 void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){ |
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69 int ph, i, v; |
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70 double x, y, w, tab[tap_count]; |
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71 const int center= (tap_count-1)/2; |
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72 |
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73 /* if upsampling, only need to interpolate, no filter */ |
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74 if (factor > 1.0) |
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75 factor = 1.0; |
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76 |
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77 for(ph=0;ph<phase_count;ph++) { |
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78 double norm = 0; |
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79 double e= 0; |
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80 for(i=0;i<tap_count;i++) { |
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81 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
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82 if (x == 0) y = 1.0; |
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83 else y = sin(x) / x; |
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84 switch(type){ |
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85 case 0:{ |
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86 const float d= -0.5; //first order derivative = -0.5 |
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87 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
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88 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
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89 else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
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90 break;} |
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91 case 1: |
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92 w = 2.0*x / (factor*tap_count) + M_PI; |
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93 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
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94 break; |
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95 case 2: |
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96 w = 2.0*x / (factor*tap_count*M_PI); |
2083 | 97 y *= bessel(16*sqrt(FFMAX(1-w*w, 0))); |
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98 break; |
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99 } |
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100 |
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101 tab[i] = y; |
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102 norm += y; |
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103 } |
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104 |
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105 /* normalize so that an uniform color remains the same */ |
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106 for(i=0;i<tap_count;i++) { |
2171 | 107 v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767); |
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108 filter[ph * tap_count + i] = v; |
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109 e += tab[i] * scale / norm - v; |
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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 /** |
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115 * initalizes a audio resampler. |
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116 * note, if either rate is not a integer then simply scale both rates up so they are |
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117 */ |
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118 AVResampleContext *av_resample_init(int out_rate, int in_rate){ |
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119 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); |
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120 double factor= FFMIN(out_rate / (double)in_rate, 1.0); |
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121 |
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122 memset(c, 0, sizeof(AVResampleContext)); |
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123 |
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124 c->filter_length= ceil(16.0/factor); |
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125 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short)); |
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126 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1); |
2109 | 127 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1; |
128 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 1; | |
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129 |
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130 c->src_incr= out_rate; |
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131 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT; |
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132 c->index= -PHASE_COUNT*((c->filter_length-1)/2); |
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133 |
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134 return c; |
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135 } |
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136 |
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137 void av_resample_close(AVResampleContext *c){ |
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138 av_freep(&c->filter_bank); |
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139 av_freep(&c); |
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140 } |
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141 |
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142 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
2083 | 143 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; |
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144 c->compensation_distance= compensation_distance; |
2083 | 145 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; |
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146 } |
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147 |
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148 /** |
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149 * resamples. |
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150 * @param src an array of unconsumed samples |
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151 * @param consumed the number of samples of src which have been consumed are returned here |
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152 * @param src_size the number of unconsumed samples available |
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153 * @param dst_size the amount of space in samples available in dst |
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154 * @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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155 * @return the number of samples written in dst or -1 if an error occured |
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156 */ |
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157 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ |
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158 int dst_index, i; |
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159 int index= c->index; |
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160 int frac= c->frac; |
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161 int dst_incr_frac= c->dst_incr % c->src_incr; |
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162 int dst_incr= c->dst_incr / c->src_incr; |
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163 |
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164 if(c->compensation_distance && c->compensation_distance < dst_size) |
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165 dst_size= c->compensation_distance; |
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166 |
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167 for(dst_index=0; dst_index < dst_size; dst_index++){ |
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168 short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK); |
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169 int sample_index= index >> PHASE_SHIFT; |
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170 int val=0; |
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171 |
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172 if(sample_index < 0){ |
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173 for(i=0; i<c->filter_length; i++) |
2109 | 174 val += src[ABS(sample_index + i) % src_size] * filter[i]; |
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175 }else if(sample_index + c->filter_length > src_size){ |
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176 break; |
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177 }else{ |
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178 #if 0 |
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179 int64_t v=0; |
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180 int sub_phase= (frac<<12) / c->src_incr; |
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181 for(i=0; i<c->filter_length; i++){ |
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182 int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase; |
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183 v += src[sample_index + i] * coeff; |
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184 } |
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185 val= v>>12; |
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186 #else |
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187 for(i=0; i<c->filter_length; i++){ |
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188 val += src[sample_index + i] * filter[i]; |
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189 } |
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190 #endif |
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191 } |
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192 |
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193 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; |
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194 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; |
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195 |
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196 frac += dst_incr_frac; |
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197 index += dst_incr; |
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198 if(frac >= c->src_incr){ |
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199 frac -= c->src_incr; |
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200 index++; |
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201 } |
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202 } |
2109 | 203 *consumed= FFMAX(index, 0) >> PHASE_SHIFT; |
204 index= FFMIN(index, 0); | |
205 | |
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206 if(update_ctx){ |
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207 if(c->compensation_distance){ |
2083 | 208 c->compensation_distance -= dst_index; |
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209 if(!c->compensation_distance) |
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210 c->dst_incr= c->ideal_dst_incr; |
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211 } |
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212 c->frac= frac; |
2109 | 213 c->index= index; |
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214 } |
2083 | 215 #if 0 |
216 if(update_ctx && !c->compensation_distance){ | |
217 #undef rand | |
218 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); | |
219 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); | |
220 } | |
221 #endif | |
222 | |
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223 return dst_index; |
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224 } |