Mercurial > libavcodec.hg
annotate resample2.c @ 8108:5de6db4225d6 libavcodec
Fix test program build: ff_eval was replaced by ff_eval2.
author | diego |
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date | Wed, 05 Nov 2008 13:32:06 +0000 |
parents | 3f819263176e |
children | e9d9d946f213 |
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 */ |
2967 | 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 |
4702 | 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 |
4702 | 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 |
4702 | 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 |
4702 | 49 #else |
50 #define FILTER_SHIFT 0 | |
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 |
4702 | 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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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){ |
4822 | 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++) { |
4702 | 133 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE |
134 filter[ph * tap_count + i] = tab[i] / norm; | |
135 #else | |
136 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); | |
137 #endif | |
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138 } |
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139 } |
4695 | 140 #if 0 |
141 { | |
142 #define LEN 1024 | |
143 int j,k; | |
144 double sine[LEN + tap_count]; | |
145 double filtered[LEN]; | |
146 double maxff=-2, minff=2, maxsf=-2, minsf=2; | |
147 for(i=0; i<LEN; i++){ | |
148 double ss=0, sf=0, ff=0; | |
149 for(j=0; j<LEN+tap_count; j++) | |
150 sine[j]= cos(i*j*M_PI/LEN); | |
151 for(j=0; j<LEN; j++){ | |
152 double sum=0; | |
153 ph=0; | |
154 for(k=0; k<tap_count; k++) | |
155 sum += filter[ph * tap_count + k] * sine[k+j]; | |
156 filtered[j]= sum / (1<<FILTER_SHIFT); | |
157 ss+= sine[j + center] * sine[j + center]; | |
158 ff+= filtered[j] * filtered[j]; | |
159 sf+= sine[j + center] * filtered[j]; | |
160 } | |
161 ss= sqrt(2*ss/LEN); | |
162 ff= sqrt(2*ff/LEN); | |
163 sf= 2*sf/LEN; | |
164 maxff= FFMAX(maxff, ff); | |
165 minff= FFMIN(minff, ff); | |
166 maxsf= FFMAX(maxsf, sf); | |
167 minsf= FFMIN(minsf, sf); | |
168 if(i%11==0){ | |
4702 | 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); |
4695 | 170 minff=minsf= 2; |
171 maxff=maxsf= -2; | |
172 } | |
173 } | |
174 } | |
175 #endif | |
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176 } |
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177 |
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178 /** |
5129 | 179 * Initializes an audio resampler. |
180 * Note, if either rate is not an integer then simply scale both rates up so they are. | |
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181 */ |
2308 | 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)); |
2308 | 184 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
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185 int phase_count= 1<<phase_shift; |
2967 | 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 |
2346 | 209 /** |
210 * Compensates samplerate/timestamp drift. The compensation is done by changing | |
6524 | 211 * the resampler parameters, so no audible clicks or similar distortions occur |
2346 | 212 * @param compensation_distance distance in output samples over which the compensation should be performed |
213 * @param sample_delta number of output samples which should be output less | |
214 * | |
215 * example: av_resample_compensate(c, 10, 500) | |
216 * here instead of 510 samples only 500 samples would be output | |
217 * | |
2967 | 218 * note, due to rounding the actual compensation might be slightly different, |
2346 | 219 * especially if the compensation_distance is large and the in_rate used during init is small |
220 */ | |
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221 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
2083 | 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; |
2083 | 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. |
6524 | 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; |
2277 | 242 int compensation_distance= c->compensation_distance; |
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243 |
2403 | 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); |
2967 | 248 |
2403 | 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; |
2403 | 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; |
2403 | 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; |
2967 | 262 |
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263 if(sample_index < 0){ |
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264 for(i=0; i<c->filter_length; i++) |
4001 | 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 |
4702 | 281 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE |
5523 | 282 dst[dst_index] = av_clip_int16(lrintf(val)); |
4702 | 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; |
4702 | 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 } |
2277 | 294 |
295 if(dst_index + 1 == compensation_distance){ | |
296 compensation_distance= 0; | |
297 dst_incr_frac= c->ideal_dst_incr % c->src_incr; | |
298 dst_incr= c->ideal_dst_incr / c->src_incr; | |
299 } | |
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300 } |
2403 | 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; |
2109 | 304 |
2277 | 305 if(compensation_distance){ |
306 compensation_distance -= dst_index; | |
307 assert(compensation_distance > 0); | |
308 } | |
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309 if(update_ctx){ |
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310 c->frac= frac; |
2109 | 311 c->index= index; |
2277 | 312 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; |
313 c->compensation_distance= compensation_distance; | |
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314 } |
2967 | 315 #if 0 |
2083 | 316 if(update_ctx && !c->compensation_distance){ |
317 #undef rand | |
318 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); | |
319 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); | |
320 } | |
321 #endif | |
2967 | 322 |
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323 return dst_index; |
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324 } |