Mercurial > libavcodec.hg
annotate resample2.c @ 6323:e6da66f378c7 libavcodec
mpegvideo.h has two function declarations with the 'inline' specifier
but no definition for those functions. The C standard requires a
definition to appear in the same translation unit for any function
declared with 'inline'. Most of the files including mpegvideo.h do not
define those functions. Fix this by removing the 'inline' specifiers
from the header.
patch by Uoti Urpala
author | diego |
---|---|
date | Sun, 03 Feb 2008 17:54:30 +0000 |
parents | c2ab2ac31edb |
children | 01647ac078a7 |
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 | |
211 * the resampler parameters, so no audible clicks or similar distortions ocur | |
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 wont 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 occured |
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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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polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters
michael
parents:
diff
changeset
|
324 } |