1/* 2 * audio resampling 3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> 4 * 5 * This file is part of Libav. 6 * 7 * Libav is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Lesser General Public 9 * License as published by the Free Software Foundation; either 10 * version 2.1 of the License, or (at your option) any later version. 11 * 12 * Libav is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Lesser General Public License for more details. 16 * 17 * You should have received a copy of the GNU Lesser General Public 18 * License along with Libav; if not, write to the Free Software 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 20 */ 21 22/** 23 * @file 24 * audio resampling 25 * @author Michael Niedermayer <michaelni@gmx.at> 26 */ 27 28#include "avcodec.h" 29#include "dsputil.h" 30 31#ifndef CONFIG_RESAMPLE_HP 32#define FILTER_SHIFT 15 33 34#define FELEM int16_t 35#define FELEM2 int32_t 36#define FELEML int64_t 37#define FELEM_MAX INT16_MAX 38#define FELEM_MIN INT16_MIN 39#define WINDOW_TYPE 9 40#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) 41#define FILTER_SHIFT 30 42 43#define FELEM int32_t 44#define FELEM2 int64_t 45#define FELEML int64_t 46#define FELEM_MAX INT32_MAX 47#define FELEM_MIN INT32_MIN 48#define WINDOW_TYPE 12 49#else 50#define FILTER_SHIFT 0 51 52#define FELEM double 53#define FELEM2 double 54#define FELEML double 55#define WINDOW_TYPE 24 56#endif 57 58 59typedef struct AVResampleContext{ 60 const AVClass *av_class; 61 FELEM *filter_bank; 62 int filter_length; 63 int ideal_dst_incr; 64 int dst_incr; 65 int index; 66 int frac; 67 int src_incr; 68 int compensation_distance; 69 int phase_shift; 70 int phase_mask; 71 int linear; 72}AVResampleContext; 73 74/** 75 * 0th order modified bessel function of the first kind. 76 */ 77static double bessel(double x){ 78 double v=1; 79 double lastv=0; 80 double t=1; 81 int i; 82 83 x= x*x/4; 84 for(i=1; v != lastv; i++){ 85 lastv=v; 86 t *= x/(i*i); 87 v += t; 88 } 89 return v; 90} 91 92/** 93 * Build a polyphase filterbank. 94 * @param factor resampling factor 95 * @param scale wanted sum of coefficients for each filter 96 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 97 * @return 0 on success, negative on error 98 */ 99static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ 100 int ph, i; 101 double x, y, w; 102 double *tab = av_malloc(tap_count * sizeof(*tab)); 103 const int center= (tap_count-1)/2; 104 105 if (!tab) 106 return AVERROR(ENOMEM); 107 108 /* if upsampling, only need to interpolate, no filter */ 109 if (factor > 1.0) 110 factor = 1.0; 111 112 for(ph=0;ph<phase_count;ph++) { 113 double norm = 0; 114 for(i=0;i<tap_count;i++) { 115 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; 116 if (x == 0) y = 1.0; 117 else y = sin(x) / x; 118 switch(type){ 119 case 0:{ 120 const float d= -0.5; //first order derivative = -0.5 121 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); 122 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); 123 else y= d*(-4 + 8*x - 5*x*x + x*x*x); 124 break;} 125 case 1: 126 w = 2.0*x / (factor*tap_count) + M_PI; 127 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); 128 break; 129 default: 130 w = 2.0*x / (factor*tap_count*M_PI); 131 y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); 132 break; 133 } 134 135 tab[i] = y; 136 norm += y; 137 } 138 139 /* normalize so that an uniform color remains the same */ 140 for(i=0;i<tap_count;i++) { 141#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 142 filter[ph * tap_count + i] = tab[i] / norm; 143#else 144 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); 145#endif 146 } 147 } 148#if 0 149 { 150#define LEN 1024 151 int j,k; 152 double sine[LEN + tap_count]; 153 double filtered[LEN]; 154 double maxff=-2, minff=2, maxsf=-2, minsf=2; 155 for(i=0; i<LEN; i++){ 156 double ss=0, sf=0, ff=0; 157 for(j=0; j<LEN+tap_count; j++) 158 sine[j]= cos(i*j*M_PI/LEN); 159 for(j=0; j<LEN; j++){ 160 double sum=0; 161 ph=0; 162 for(k=0; k<tap_count; k++) 163 sum += filter[ph * tap_count + k] * sine[k+j]; 164 filtered[j]= sum / (1<<FILTER_SHIFT); 165 ss+= sine[j + center] * sine[j + center]; 166 ff+= filtered[j] * filtered[j]; 167 sf+= sine[j + center] * filtered[j]; 168 } 169 ss= sqrt(2*ss/LEN); 170 ff= sqrt(2*ff/LEN); 171 sf= 2*sf/LEN; 172 maxff= FFMAX(maxff, ff); 173 minff= FFMIN(minff, ff); 174 maxsf= FFMAX(maxsf, sf); 175 minsf= FFMIN(minsf, sf); 176 if(i%11==0){ 177 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); 178 minff=minsf= 2; 179 maxff=maxsf= -2; 180 } 181 } 182 } 183#endif 184 185 av_free(tab); 186 return 0; 187} 188 189AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ 190 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); 191 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); 192 int phase_count= 1<<phase_shift; 193 194 if (!c) 195 return NULL; 196 197 c->phase_shift= phase_shift; 198 c->phase_mask= phase_count-1; 199 c->linear= linear; 200 201 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); 202 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); 203 if (!c->filter_bank) 204 goto error; 205 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE)) 206 goto error; 207 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); 208 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; 209 210 c->src_incr= out_rate; 211 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; 212 c->index= -phase_count*((c->filter_length-1)/2); 213 214 return c; 215error: 216 av_free(c->filter_bank); 217 av_free(c); 218 return NULL; 219} 220 221void av_resample_close(AVResampleContext *c){ 222 av_freep(&c->filter_bank); 223 av_freep(&c); 224} 225 226void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ 227// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; 228 c->compensation_distance= compensation_distance; 229 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; 230} 231 232int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ 233 int dst_index, i; 234 int index= c->index; 235 int frac= c->frac; 236 int dst_incr_frac= c->dst_incr % c->src_incr; 237 int dst_incr= c->dst_incr / c->src_incr; 238 int compensation_distance= c->compensation_distance; 239 240 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ 241 int64_t index2= ((int64_t)index)<<32; 242 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; 243 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); 244 245 for(dst_index=0; dst_index < dst_size; dst_index++){ 246 dst[dst_index] = src[index2>>32]; 247 index2 += incr; 248 } 249 frac += dst_index * dst_incr_frac; 250 index += dst_index * dst_incr; 251 index += frac / c->src_incr; 252 frac %= c->src_incr; 253 }else{ 254 for(dst_index=0; dst_index < dst_size; dst_index++){ 255 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); 256 int sample_index= index >> c->phase_shift; 257 FELEM2 val=0; 258 259 if(sample_index < 0){ 260 for(i=0; i<c->filter_length; i++) 261 val += src[FFABS(sample_index + i) % src_size] * filter[i]; 262 }else if(sample_index + c->filter_length > src_size){ 263 break; 264 }else if(c->linear){ 265 FELEM2 v2=0; 266 for(i=0; i<c->filter_length; i++){ 267 val += src[sample_index + i] * (FELEM2)filter[i]; 268 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; 269 } 270 val+=(v2-val)*(FELEML)frac / c->src_incr; 271 }else{ 272 for(i=0; i<c->filter_length; i++){ 273 val += src[sample_index + i] * (FELEM2)filter[i]; 274 } 275 } 276 277#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 278 dst[dst_index] = av_clip_int16(lrintf(val)); 279#else 280 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; 281 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; 282#endif 283 284 frac += dst_incr_frac; 285 index += dst_incr; 286 if(frac >= c->src_incr){ 287 frac -= c->src_incr; 288 index++; 289 } 290 291 if(dst_index + 1 == compensation_distance){ 292 compensation_distance= 0; 293 dst_incr_frac= c->ideal_dst_incr % c->src_incr; 294 dst_incr= c->ideal_dst_incr / c->src_incr; 295 } 296 } 297 } 298 *consumed= FFMAX(index, 0) >> c->phase_shift; 299 if(index>=0) index &= c->phase_mask; 300 301 if(compensation_distance){ 302 compensation_distance -= dst_index; 303 assert(compensation_distance > 0); 304 } 305 if(update_ctx){ 306 c->frac= frac; 307 c->index= index; 308 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; 309 c->compensation_distance= compensation_distance; 310 } 311#if 0 312 if(update_ctx && !c->compensation_distance){ 313#undef rand 314 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); 315av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); 316 } 317#endif 318 319 return dst_index; 320} 321