annotate resample2.c @ 8442:60be1c7d8345 libavcodec

Make h264 parser 50% faster.
author michael
date Tue, 23 Dec 2008 02:34:58 +0000
parents 3f819263176e
children e9d9d946f213
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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 file is part of FFmpeg.
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6 *
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7 * FFmpeg is free software; you can redistribute it and/or
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8 * modify it under the terms of the GNU Lesser General Public
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9 * License as published by the Free Software Foundation; either
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10 * version 2.1 of the License, or (at your option) any later version.
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11 *
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12 * FFmpeg is distributed in the hope that it will be useful,
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 * Lesser General Public License for more details.
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16 *
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17 * You should have received a copy of the GNU Lesser General Public
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18 * License along with FFmpeg; if not, write to the Free Software
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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20 */
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21
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22 /**
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23 * @file resample2.c
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24 * audio resampling
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25 * @author Michael Niedermayer <michaelni@gmx.at>
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26 */
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27
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28 #include "avcodec.h"
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29 #include "dsputil.h"
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30
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31 #ifndef CONFIG_RESAMPLE_HP
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32 #define FILTER_SHIFT 15
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33
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34 #define FELEM int16_t
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35 #define FELEM2 int32_t
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36 #define FELEML int64_t
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37 #define FELEM_MAX INT16_MAX
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38 #define FELEM_MIN INT16_MIN
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39 #define WINDOW_TYPE 9
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40 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
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41 #define FILTER_SHIFT 30
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42
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43 #define FELEM int32_t
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44 #define FELEM2 int64_t
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45 #define FELEML int64_t
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46 #define FELEM_MAX INT32_MAX
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47 #define FELEM_MIN INT32_MIN
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48 #define WINDOW_TYPE 12
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49 #else
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50 #define FILTER_SHIFT 0
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51
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52 #define FELEM double
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53 #define FELEM2 double
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54 #define FELEML double
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55 #define WINDOW_TYPE 24
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56 #endif
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57
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58
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59 typedef struct AVResampleContext{
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60 FELEM *filter_bank;
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61 int filter_length;
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62 int ideal_dst_incr;
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63 int dst_incr;
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64 int index;
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65 int frac;
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66 int src_incr;
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67 int compensation_distance;
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68 int phase_shift;
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69 int phase_mask;
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70 int linear;
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71 }AVResampleContext;
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72
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73 /**
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74 * 0th order modified bessel function of the first kind.
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75 */
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76 static double bessel(double x){
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77 double v=1;
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78 double t=1;
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79 int i;
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80
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81 x= x*x/4;
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82 for(i=1; i<50; i++){
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83 t *= x/(i*i);
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84 v += t;
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85 }
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86 return v;
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87 }
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88
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89 /**
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90 * builds a polyphase filterbank.
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91 * @param factor resampling factor
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92 * @param scale wanted sum of coefficients for each filter
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93 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
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94 */
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95 void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
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96 int ph, i;
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97 double x, y, w, tab[tap_count];
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98 const int center= (tap_count-1)/2;
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99
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100 /* if upsampling, only need to interpolate, no filter */
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101 if (factor > 1.0)
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102 factor = 1.0;
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103
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104 for(ph=0;ph<phase_count;ph++) {
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105 double norm = 0;
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106 for(i=0;i<tap_count;i++) {
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107 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
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108 if (x == 0) y = 1.0;
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109 else y = sin(x) / x;
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110 switch(type){
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111 case 0:{
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112 const float d= -0.5; //first order derivative = -0.5
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113 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
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114 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
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115 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
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116 break;}
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117 case 1:
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118 w = 2.0*x / (factor*tap_count) + M_PI;
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119 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
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120 break;
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121 default:
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122 w = 2.0*x / (factor*tap_count*M_PI);
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123 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
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124 break;
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125 }
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126
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127 tab[i] = y;
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128 norm += y;
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129 }
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130
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131 /* normalize so that an uniform color remains the same */
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132 for(i=0;i<tap_count;i++) {
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133 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
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134 filter[ph * tap_count + i] = tab[i] / norm;
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135 #else
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136 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
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137 #endif
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138 }
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139 }
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140 #if 0
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141 {
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142 #define LEN 1024
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143 int j,k;
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144 double sine[LEN + tap_count];
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145 double filtered[LEN];
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146 double maxff=-2, minff=2, maxsf=-2, minsf=2;
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147 for(i=0; i<LEN; i++){
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148 double ss=0, sf=0, ff=0;
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149 for(j=0; j<LEN+tap_count; j++)
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150 sine[j]= cos(i*j*M_PI/LEN);
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151 for(j=0; j<LEN; j++){
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152 double sum=0;
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153 ph=0;
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154 for(k=0; k<tap_count; k++)
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155 sum += filter[ph * tap_count + k] * sine[k+j];
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156 filtered[j]= sum / (1<<FILTER_SHIFT);
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157 ss+= sine[j + center] * sine[j + center];
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158 ff+= filtered[j] * filtered[j];
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159 sf+= sine[j + center] * filtered[j];
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160 }
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161 ss= sqrt(2*ss/LEN);
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162 ff= sqrt(2*ff/LEN);
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163 sf= 2*sf/LEN;
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164 maxff= FFMAX(maxff, ff);
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165 minff= FFMIN(minff, ff);
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166 maxsf= FFMAX(maxsf, sf);
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167 minsf= FFMIN(minsf, sf);
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168 if(i%11==0){
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169 av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
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170 minff=minsf= 2;
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171 maxff=maxsf= -2;
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172 }
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173 }
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174 }
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175 #endif
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176 }
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177
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178 /**
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179 * Initializes an audio resampler.
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180 * Note, if either rate is not an integer then simply scale both rates up so they are.
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181 */
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182 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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183 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
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184 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
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185 int phase_count= 1<<phase_shift;
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186
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187 c->phase_shift= phase_shift;
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188 c->phase_mask= phase_count-1;
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189 c->linear= linear;
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190
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191 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
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192 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
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193 av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
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194 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
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195 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
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196
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197 c->src_incr= out_rate;
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198 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
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199 c->index= -phase_count*((c->filter_length-1)/2);
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200
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201 return c;
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202 }
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203
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204 void av_resample_close(AVResampleContext *c){
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205 av_freep(&c->filter_bank);
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206 av_freep(&c);
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207 }
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208
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209 /**
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210 * Compensates samplerate/timestamp drift. The compensation is done by changing
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211 * the resampler parameters, so no audible clicks or similar distortions occur
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212 * @param compensation_distance distance in output samples over which the compensation should be performed
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213 * @param sample_delta number of output samples which should be output less
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214 *
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215 * example: av_resample_compensate(c, 10, 500)
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216 * here instead of 510 samples only 500 samples would be output
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217 *
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218 * note, due to rounding the actual compensation might be slightly different,
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219 * especially if the compensation_distance is large and the in_rate used during init is small
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220 */
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221 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
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222 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
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223 c->compensation_distance= compensation_distance;
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224 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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225 }
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226
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227 /**
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228 * resamples.
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229 * @param src an array of unconsumed samples
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230 * @param consumed the number of samples of src which have been consumed are returned here
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231 * @param src_size the number of unconsumed samples available
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232 * @param dst_size the amount of space in samples available in dst
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233 * @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
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234 * @return the number of samples written in dst or -1 if an error occurred
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235 */
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236 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
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237 int dst_index, i;
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238 int index= c->index;
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239 int frac= c->frac;
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240 int dst_incr_frac= c->dst_incr % c->src_incr;
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241 int dst_incr= c->dst_incr / c->src_incr;
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242 int compensation_distance= c->compensation_distance;
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243
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244 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
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245 int64_t index2= ((int64_t)index)<<32;
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246 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
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247 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
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248
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249 for(dst_index=0; dst_index < dst_size; dst_index++){
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250 dst[dst_index] = src[index2>>32];
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251 index2 += incr;
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252 }
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253 frac += dst_index * dst_incr_frac;
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254 index += dst_index * dst_incr;
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255 index += frac / c->src_incr;
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256 frac %= c->src_incr;
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257 }else{
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258 for(dst_index=0; dst_index < dst_size; dst_index++){
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259 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
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260 int sample_index= index >> c->phase_shift;
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261 FELEM2 val=0;
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262
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263 if(sample_index < 0){
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264 for(i=0; i<c->filter_length; i++)
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265 val += src[FFABS(sample_index + i) % src_size] * filter[i];
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266 }else if(sample_index + c->filter_length > src_size){
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267 break;
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268 }else if(c->linear){
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269 FELEM2 v2=0;
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270 for(i=0; i<c->filter_length; i++){
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271 val += src[sample_index + i] * (FELEM2)filter[i];
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272 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
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273 }
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274 val+=(v2-val)*(FELEML)frac / c->src_incr;
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275 }else{
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276 for(i=0; i<c->filter_length; i++){
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277 val += src[sample_index + i] * (FELEM2)filter[i];
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278 }
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279 }
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280
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281 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
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282 dst[dst_index] = av_clip_int16(lrintf(val));
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283 #else
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284 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
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285 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
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286 #endif
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287
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288 frac += dst_incr_frac;
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289 index += dst_incr;
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290 if(frac >= c->src_incr){
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291 frac -= c->src_incr;
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292 index++;
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293 }
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294
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295 if(dst_index + 1 == compensation_distance){
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296 compensation_distance= 0;
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297 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
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298 dst_incr= c->ideal_dst_incr / c->src_incr;
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299 }
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300 }
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301 }
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302 *consumed= FFMAX(index, 0) >> c->phase_shift;
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303 if(index>=0) index &= c->phase_mask;
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304
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305 if(compensation_distance){
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306 compensation_distance -= dst_index;
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307 assert(compensation_distance > 0);
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308 }
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309 if(update_ctx){
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310 c->frac= frac;
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311 c->index= index;
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312 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
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313 c->compensation_distance= compensation_distance;
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314 }
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315 #if 0
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316 if(update_ctx && !c->compensation_distance){
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317 #undef rand
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318 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
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319 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
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320 }
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321 #endif
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322
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323 return dst_index;
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324 }