annotate resample2.c @ 4096:99616a74260a libavcodec

use lambda/lmax instead of qscale/qmax as end check for vbv retry
author michael
date Sat, 28 Oct 2006 11:15:06 +0000
parents 34fdffe98bd0
children a96d905dcbaa
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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 /**
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24 * @file resample2.c
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25 * audio resampling
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26 * @author Michael Niedermayer <michaelni@gmx.at>
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27 */
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28
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29 #include "avcodec.h"
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30 #include "common.h"
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31 #include "dsputil.h"
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32
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33 #if 1
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34 #define FILTER_SHIFT 15
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35
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36 #define FELEM int16_t
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37 #define FELEM2 int32_t
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38 #define FELEM_MAX INT16_MAX
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39 #define FELEM_MIN INT16_MIN
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40 #else
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41 #define FILTER_SHIFT 22
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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 FELEM_MAX INT32_MAX
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46 #define FELEM_MIN INT32_MIN
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47 #endif
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48
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49
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50 typedef struct AVResampleContext{
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51 FELEM *filter_bank;
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52 int filter_length;
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53 int ideal_dst_incr;
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54 int dst_incr;
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55 int index;
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56 int frac;
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57 int src_incr;
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58 int compensation_distance;
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59 int phase_shift;
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60 int phase_mask;
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61 int linear;
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62 }AVResampleContext;
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63
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64 /**
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65 * 0th order modified bessel function of the first kind.
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66 */
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67 static double bessel(double x){
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68 double v=1;
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69 double t=1;
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70 int i;
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71
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72 for(i=1; i<50; i++){
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73 t *= i;
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74 v += pow(x*x/4, i)/(t*t);
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75 }
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76 return v;
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77 }
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78
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79 /**
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80 * builds a polyphase filterbank.
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81 * @param factor resampling factor
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82 * @param scale wanted sum of coefficients for each filter
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83 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
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84 */
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85 void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
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86 int ph, i, v;
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87 double x, y, w, tab[tap_count];
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88 const int center= (tap_count-1)/2;
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89
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90 /* if upsampling, only need to interpolate, no filter */
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91 if (factor > 1.0)
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92 factor = 1.0;
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93
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94 for(ph=0;ph<phase_count;ph++) {
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95 double norm = 0;
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96 double e= 0;
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97 for(i=0;i<tap_count;i++) {
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98 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
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99 if (x == 0) y = 1.0;
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100 else y = sin(x) / x;
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101 switch(type){
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102 case 0:{
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103 const float d= -0.5; //first order derivative = -0.5
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104 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
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105 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
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106 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
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107 break;}
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108 case 1:
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109 w = 2.0*x / (factor*tap_count) + M_PI;
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110 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
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111 break;
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112 case 2:
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113 w = 2.0*x / (factor*tap_count*M_PI);
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114 y *= bessel(16*sqrt(FFMAX(1-w*w, 0)));
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115 break;
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116 }
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117
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118 tab[i] = y;
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119 norm += y;
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120 }
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121
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122 /* normalize so that an uniform color remains the same */
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123 for(i=0;i<tap_count;i++) {
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124 v = clip(lrintf(tab[i] * scale / norm + e), FELEM_MIN, FELEM_MAX);
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125 filter[ph * tap_count + i] = v;
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126 e += tab[i] * scale / norm - v;
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127 }
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128 }
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129 }
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130
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131 /**
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132 * initalizes a audio resampler.
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133 * note, if either rate is not a integer then simply scale both rates up so they are
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134 */
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135 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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136 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
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137 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
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138 int phase_count= 1<<phase_shift;
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139
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140 c->phase_shift= phase_shift;
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141 c->phase_mask= phase_count-1;
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142 c->linear= linear;
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143
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144 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
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145 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
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146 av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, 1);
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147 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
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148 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
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149
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150 c->src_incr= out_rate;
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151 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
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152 c->index= -phase_count*((c->filter_length-1)/2);
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153
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154 return c;
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155 }
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156
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157 void av_resample_close(AVResampleContext *c){
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158 av_freep(&c->filter_bank);
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159 av_freep(&c);
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160 }
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161
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162 /**
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163 * Compensates samplerate/timestamp drift. The compensation is done by changing
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164 * the resampler parameters, so no audible clicks or similar distortions ocur
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165 * @param compensation_distance distance in output samples over which the compensation should be performed
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166 * @param sample_delta number of output samples which should be output less
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167 *
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168 * example: av_resample_compensate(c, 10, 500)
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169 * here instead of 510 samples only 500 samples would be output
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170 *
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171 * note, due to rounding the actual compensation might be slightly different,
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172 * especially if the compensation_distance is large and the in_rate used during init is small
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173 */
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174 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
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175 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
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176 c->compensation_distance= compensation_distance;
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177 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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178 }
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179
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180 /**
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181 * resamples.
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182 * @param src an array of unconsumed samples
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183 * @param consumed the number of samples of src which have been consumed are returned here
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184 * @param src_size the number of unconsumed samples available
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185 * @param dst_size the amount of space in samples available in dst
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186 * @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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187 * @return the number of samples written in dst or -1 if an error occured
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188 */
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189 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
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190 int dst_index, i;
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191 int index= c->index;
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192 int frac= c->frac;
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193 int dst_incr_frac= c->dst_incr % c->src_incr;
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194 int dst_incr= c->dst_incr / c->src_incr;
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195 int compensation_distance= c->compensation_distance;
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196
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197 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
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198 int64_t index2= ((int64_t)index)<<32;
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199 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
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200 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
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201
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202 for(dst_index=0; dst_index < dst_size; dst_index++){
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203 dst[dst_index] = src[index2>>32];
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204 index2 += incr;
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205 }
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206 frac += dst_index * dst_incr_frac;
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207 index += dst_index * dst_incr;
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208 index += frac / c->src_incr;
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209 frac %= c->src_incr;
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210 }else{
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211 for(dst_index=0; dst_index < dst_size; dst_index++){
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212 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
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213 int sample_index= index >> c->phase_shift;
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214 FELEM2 val=0;
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215
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216 if(sample_index < 0){
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217 for(i=0; i<c->filter_length; i++)
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218 val += src[FFABS(sample_index + i) % src_size] * filter[i];
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219 }else if(sample_index + c->filter_length > src_size){
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220 break;
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221 }else if(c->linear){
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222 int64_t v=0;
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223 int sub_phase= (frac<<8) / c->src_incr;
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224 for(i=0; i<c->filter_length; i++){
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225 int64_t coeff= filter[i]*(256 - sub_phase) + filter[i + c->filter_length]*sub_phase;
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226 v += src[sample_index + i] * coeff;
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227 }
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228 val= v>>8;
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229 }else{
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230 for(i=0; i<c->filter_length; i++){
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231 val += src[sample_index + i] * (FELEM2)filter[i];
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232 }
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233 }
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234
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235 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
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236 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
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237
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238 frac += dst_incr_frac;
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239 index += dst_incr;
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240 if(frac >= c->src_incr){
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241 frac -= c->src_incr;
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242 index++;
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243 }
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244
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245 if(dst_index + 1 == compensation_distance){
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246 compensation_distance= 0;
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247 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
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248 dst_incr= c->ideal_dst_incr / c->src_incr;
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249 }
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250 }
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251 }
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252 *consumed= FFMAX(index, 0) >> c->phase_shift;
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253 if(index>=0) index &= c->phase_mask;
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254
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255 if(compensation_distance){
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256 compensation_distance -= dst_index;
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257 assert(compensation_distance > 0);
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258 }
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259 if(update_ctx){
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260 c->frac= frac;
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261 c->index= index;
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262 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
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263 c->compensation_distance= compensation_distance;
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264 }
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265 #if 0
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266 if(update_ctx && !c->compensation_distance){
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267 #undef rand
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268 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
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269 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
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270 }
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271 #endif
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272
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273 return dst_index;
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274 }