annotate resample2.c @ 2174:6d614374d907 libavcodec

Get H.264 frame rate from SPS/VUI patch by (Mns Rullgrd <mru at kth dot se>)
author michael
date Mon, 16 Aug 2004 22:48:05 +0000
parents 3f52c129d00f
children b88e05b9b445
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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);
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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++) {
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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);
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127 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
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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){
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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;
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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++)
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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 }
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203 *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
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204 index= FFMIN(index, 0);
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205
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206 if(update_ctx){
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207 if(c->compensation_distance){
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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;
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213 c->index= index;
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214 }
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215 #if 0
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216 if(update_ctx && !c->compensation_distance){
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217 #undef rand
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218 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
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219 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
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220 }
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221 #endif
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222
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223 return dst_index;
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224 }