Mercurial > libavcodec.hg
annotate resample2.c @ 4418:4cceb7c877af libavcodec
rename CMOV_IS_FAST to HAVE_FAST_CMOV and simplify configure
author | mru |
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date | Sat, 27 Jan 2007 14:10:57 +0000 |
parents | 34fdffe98bd0 |
children | a96d905dcbaa |
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 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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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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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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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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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 */ |
2308 | 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)); |
2308 | 137 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
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138 int phase_count= 1<<phase_shift; |
2967 | 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 |
2346 | 162 /** |
163 * Compensates samplerate/timestamp drift. The compensation is done by changing | |
164 * the resampler parameters, so no audible clicks or similar distortions ocur | |
165 * @param compensation_distance distance in output samples over which the compensation should be performed | |
166 * @param sample_delta number of output samples which should be output less | |
167 * | |
168 * example: av_resample_compensate(c, 10, 500) | |
169 * here instead of 510 samples only 500 samples would be output | |
170 * | |
2967 | 171 * note, due to rounding the actual compensation might be slightly different, |
2346 | 172 * especially if the compensation_distance is large and the in_rate used during init is small |
173 */ | |
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174 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
2083 | 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; |
2083 | 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; |
2277 | 195 int compensation_distance= c->compensation_distance; |
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196 |
2403 | 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); |
2967 | 201 |
2403 | 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; |
2403 | 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; |
2403 | 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; |
2967 | 215 |
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216 if(sample_index < 0){ |
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217 for(i=0; i<c->filter_length; i++) |
4001 | 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; |
2305 | 223 int sub_phase= (frac<<8) / c->src_incr; |
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224 for(i=0; i<c->filter_length; i++){ |
2305 | 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 } |
2305 | 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 } |
2277 | 244 |
245 if(dst_index + 1 == compensation_distance){ | |
246 compensation_distance= 0; | |
247 dst_incr_frac= c->ideal_dst_incr % c->src_incr; | |
248 dst_incr= c->ideal_dst_incr / c->src_incr; | |
249 } | |
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250 } |
2403 | 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; |
2109 | 254 |
2277 | 255 if(compensation_distance){ |
256 compensation_distance -= dst_index; | |
257 assert(compensation_distance > 0); | |
258 } | |
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259 if(update_ctx){ |
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260 c->frac= frac; |
2109 | 261 c->index= index; |
2277 | 262 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; |
263 c->compensation_distance= compensation_distance; | |
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264 } |
2967 | 265 #if 0 |
2083 | 266 if(update_ctx && !c->compensation_distance){ |
267 #undef rand | |
268 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); | |
269 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); | |
270 } | |
271 #endif | |
2967 | 272 |
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273 return dst_index; |
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274 } |