Libav 0.7.1
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00001 /* 00002 * audio resampling 00003 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> 00004 * 00005 * This file is part of Libav. 00006 * 00007 * Libav is free software; you can redistribute it and/or 00008 * modify it under the terms of the GNU Lesser General Public 00009 * License as published by the Free Software Foundation; either 00010 * version 2.1 of the License, or (at your option) any later version. 00011 * 00012 * Libav is distributed in the hope that it will be useful, 00013 * but WITHOUT ANY WARRANTY; without even the implied warranty of 00014 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 00015 * Lesser General Public License for more details. 00016 * 00017 * You should have received a copy of the GNU Lesser General Public 00018 * License along with Libav; if not, write to the Free Software 00019 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 00020 */ 00021 00028 #include "avcodec.h" 00029 #include "dsputil.h" 00030 00031 #ifndef CONFIG_RESAMPLE_HP 00032 #define FILTER_SHIFT 15 00033 00034 #define FELEM int16_t 00035 #define FELEM2 int32_t 00036 #define FELEML int64_t 00037 #define FELEM_MAX INT16_MAX 00038 #define FELEM_MIN INT16_MIN 00039 #define WINDOW_TYPE 9 00040 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) 00041 #define FILTER_SHIFT 30 00042 00043 #define FELEM int32_t 00044 #define FELEM2 int64_t 00045 #define FELEML int64_t 00046 #define FELEM_MAX INT32_MAX 00047 #define FELEM_MIN INT32_MIN 00048 #define WINDOW_TYPE 12 00049 #else 00050 #define FILTER_SHIFT 0 00051 00052 #define FELEM double 00053 #define FELEM2 double 00054 #define FELEML double 00055 #define WINDOW_TYPE 24 00056 #endif 00057 00058 00059 typedef struct AVResampleContext{ 00060 const AVClass *av_class; 00061 FELEM *filter_bank; 00062 int filter_length; 00063 int ideal_dst_incr; 00064 int dst_incr; 00065 int index; 00066 int frac; 00067 int src_incr; 00068 int compensation_distance; 00069 int phase_shift; 00070 int phase_mask; 00071 int linear; 00072 }AVResampleContext; 00073 00077 static double bessel(double x){ 00078 double v=1; 00079 double lastv=0; 00080 double t=1; 00081 int i; 00082 00083 x= x*x/4; 00084 for(i=1; v != lastv; i++){ 00085 lastv=v; 00086 t *= x/(i*i); 00087 v += t; 00088 } 00089 return v; 00090 } 00091 00099 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ 00100 int ph, i; 00101 double x, y, w; 00102 double *tab = av_malloc(tap_count * sizeof(*tab)); 00103 const int center= (tap_count-1)/2; 00104 00105 if (!tab) 00106 return AVERROR(ENOMEM); 00107 00108 /* if upsampling, only need to interpolate, no filter */ 00109 if (factor > 1.0) 00110 factor = 1.0; 00111 00112 for(ph=0;ph<phase_count;ph++) { 00113 double norm = 0; 00114 for(i=0;i<tap_count;i++) { 00115 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; 00116 if (x == 0) y = 1.0; 00117 else y = sin(x) / x; 00118 switch(type){ 00119 case 0:{ 00120 const float d= -0.5; //first order derivative = -0.5 00121 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); 00122 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); 00123 else y= d*(-4 + 8*x - 5*x*x + x*x*x); 00124 break;} 00125 case 1: 00126 w = 2.0*x / (factor*tap_count) + M_PI; 00127 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); 00128 break; 00129 default: 00130 w = 2.0*x / (factor*tap_count*M_PI); 00131 y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); 00132 break; 00133 } 00134 00135 tab[i] = y; 00136 norm += y; 00137 } 00138 00139 /* normalize so that an uniform color remains the same */ 00140 for(i=0;i<tap_count;i++) { 00141 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 00142 filter[ph * tap_count + i] = tab[i] / norm; 00143 #else 00144 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); 00145 #endif 00146 } 00147 } 00148 #if 0 00149 { 00150 #define LEN 1024 00151 int j,k; 00152 double sine[LEN + tap_count]; 00153 double filtered[LEN]; 00154 double maxff=-2, minff=2, maxsf=-2, minsf=2; 00155 for(i=0; i<LEN; i++){ 00156 double ss=0, sf=0, ff=0; 00157 for(j=0; j<LEN+tap_count; j++) 00158 sine[j]= cos(i*j*M_PI/LEN); 00159 for(j=0; j<LEN; j++){ 00160 double sum=0; 00161 ph=0; 00162 for(k=0; k<tap_count; k++) 00163 sum += filter[ph * tap_count + k] * sine[k+j]; 00164 filtered[j]= sum / (1<<FILTER_SHIFT); 00165 ss+= sine[j + center] * sine[j + center]; 00166 ff+= filtered[j] * filtered[j]; 00167 sf+= sine[j + center] * filtered[j]; 00168 } 00169 ss= sqrt(2*ss/LEN); 00170 ff= sqrt(2*ff/LEN); 00171 sf= 2*sf/LEN; 00172 maxff= FFMAX(maxff, ff); 00173 minff= FFMIN(minff, ff); 00174 maxsf= FFMAX(maxsf, sf); 00175 minsf= FFMIN(minsf, sf); 00176 if(i%11==0){ 00177 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); 00178 minff=minsf= 2; 00179 maxff=maxsf= -2; 00180 } 00181 } 00182 } 00183 #endif 00184 00185 av_free(tab); 00186 return 0; 00187 } 00188 00189 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ 00190 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); 00191 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); 00192 int phase_count= 1<<phase_shift; 00193 00194 if (!c) 00195 return NULL; 00196 00197 c->phase_shift= phase_shift; 00198 c->phase_mask= phase_count-1; 00199 c->linear= linear; 00200 00201 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); 00202 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); 00203 if (!c->filter_bank) 00204 goto error; 00205 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE)) 00206 goto error; 00207 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); 00208 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; 00209 00210 c->src_incr= out_rate; 00211 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; 00212 c->index= -phase_count*((c->filter_length-1)/2); 00213 00214 return c; 00215 error: 00216 av_free(c->filter_bank); 00217 av_free(c); 00218 return NULL; 00219 } 00220 00221 void av_resample_close(AVResampleContext *c){ 00222 av_freep(&c->filter_bank); 00223 av_freep(&c); 00224 } 00225 00226 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ 00227 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; 00228 c->compensation_distance= compensation_distance; 00229 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; 00230 } 00231 00232 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ 00233 int dst_index, i; 00234 int index= c->index; 00235 int frac= c->frac; 00236 int dst_incr_frac= c->dst_incr % c->src_incr; 00237 int dst_incr= c->dst_incr / c->src_incr; 00238 int compensation_distance= c->compensation_distance; 00239 00240 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ 00241 int64_t index2= ((int64_t)index)<<32; 00242 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; 00243 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); 00244 00245 for(dst_index=0; dst_index < dst_size; dst_index++){ 00246 dst[dst_index] = src[index2>>32]; 00247 index2 += incr; 00248 } 00249 frac += dst_index * dst_incr_frac; 00250 index += dst_index * dst_incr; 00251 index += frac / c->src_incr; 00252 frac %= c->src_incr; 00253 }else{ 00254 for(dst_index=0; dst_index < dst_size; dst_index++){ 00255 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); 00256 int sample_index= index >> c->phase_shift; 00257 FELEM2 val=0; 00258 00259 if(sample_index < 0){ 00260 for(i=0; i<c->filter_length; i++) 00261 val += src[FFABS(sample_index + i) % src_size] * filter[i]; 00262 }else if(sample_index + c->filter_length > src_size){ 00263 break; 00264 }else if(c->linear){ 00265 FELEM2 v2=0; 00266 for(i=0; i<c->filter_length; i++){ 00267 val += src[sample_index + i] * (FELEM2)filter[i]; 00268 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; 00269 } 00270 val+=(v2-val)*(FELEML)frac / c->src_incr; 00271 }else{ 00272 for(i=0; i<c->filter_length; i++){ 00273 val += src[sample_index + i] * (FELEM2)filter[i]; 00274 } 00275 } 00276 00277 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 00278 dst[dst_index] = av_clip_int16(lrintf(val)); 00279 #else 00280 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; 00281 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; 00282 #endif 00283 00284 frac += dst_incr_frac; 00285 index += dst_incr; 00286 if(frac >= c->src_incr){ 00287 frac -= c->src_incr; 00288 index++; 00289 } 00290 00291 if(dst_index + 1 == compensation_distance){ 00292 compensation_distance= 0; 00293 dst_incr_frac= c->ideal_dst_incr % c->src_incr; 00294 dst_incr= c->ideal_dst_incr / c->src_incr; 00295 } 00296 } 00297 } 00298 *consumed= FFMAX(index, 0) >> c->phase_shift; 00299 if(index>=0) index &= c->phase_mask; 00300 00301 if(compensation_distance){ 00302 compensation_distance -= dst_index; 00303 assert(compensation_distance > 0); 00304 } 00305 if(update_ctx){ 00306 c->frac= frac; 00307 c->index= index; 00308 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; 00309 c->compensation_distance= compensation_distance; 00310 } 00311 #if 0 00312 if(update_ctx && !c->compensation_distance){ 00313 #undef rand 00314 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); 00315 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); 00316 } 00317 #endif 00318 00319 return dst_index; 00320 }