1/* 2 * audio resampling 3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> 4 * 5 * This file is part of FFmpeg. 6 * 7 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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 libavcodec/resample2.c 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 FELEM *filter_bank; 61 int filter_length; 62 int ideal_dst_incr; 63 int dst_incr; 64 int index; 65 int frac; 66 int src_incr; 67 int compensation_distance; 68 int phase_shift; 69 int phase_mask; 70 int linear; 71}AVResampleContext; 72 73/** 74 * 0th order modified bessel function of the first kind. 75 */ 76static double bessel(double x){ 77 double v=1; 78 double t=1; 79 int i; 80 81 x= x*x/4; 82 for(i=1; i<50; i++){ 83 t *= x/(i*i); 84 v += t; 85 } 86 return v; 87} 88 89/** 90 * builds a polyphase filterbank. 91 * @param factor resampling factor 92 * @param scale wanted sum of coefficients for each filter 93 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 94 */ 95void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ 96 int ph, i; 97 double x, y, w, tab[tap_count]; 98 const int center= (tap_count-1)/2; 99 100 /* if upsampling, only need to interpolate, no filter */ 101 if (factor > 1.0) 102 factor = 1.0; 103 104 for(ph=0;ph<phase_count;ph++) { 105 double norm = 0; 106 for(i=0;i<tap_count;i++) { 107 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; 108 if (x == 0) y = 1.0; 109 else y = sin(x) / x; 110 switch(type){ 111 case 0:{ 112 const float d= -0.5; //first order derivative = -0.5 113 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); 114 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); 115 else y= d*(-4 + 8*x - 5*x*x + x*x*x); 116 break;} 117 case 1: 118 w = 2.0*x / (factor*tap_count) + M_PI; 119 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); 120 break; 121 default: 122 w = 2.0*x / (factor*tap_count*M_PI); 123 y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); 124 break; 125 } 126 127 tab[i] = y; 128 norm += y; 129 } 130 131 /* normalize so that an uniform color remains the same */ 132 for(i=0;i<tap_count;i++) { 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 138 } 139 } 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){ 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); 170 minff=minsf= 2; 171 maxff=maxsf= -2; 172 } 173 } 174 } 175#endif 176} 177 178AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ 179 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); 180 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); 181 int phase_count= 1<<phase_shift; 182 183 c->phase_shift= phase_shift; 184 c->phase_mask= phase_count-1; 185 c->linear= linear; 186 187 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); 188 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); 189 av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE); 190 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); 191 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; 192 193 c->src_incr= out_rate; 194 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; 195 c->index= -phase_count*((c->filter_length-1)/2); 196 197 return c; 198} 199 200void av_resample_close(AVResampleContext *c){ 201 av_freep(&c->filter_bank); 202 av_freep(&c); 203} 204 205void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ 206// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; 207 c->compensation_distance= compensation_distance; 208 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; 209} 210 211int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ 212 int dst_index, i; 213 int index= c->index; 214 int frac= c->frac; 215 int dst_incr_frac= c->dst_incr % c->src_incr; 216 int dst_incr= c->dst_incr / c->src_incr; 217 int compensation_distance= c->compensation_distance; 218 219 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ 220 int64_t index2= ((int64_t)index)<<32; 221 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; 222 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); 223 224 for(dst_index=0; dst_index < dst_size; dst_index++){ 225 dst[dst_index] = src[index2>>32]; 226 index2 += incr; 227 } 228 frac += dst_index * dst_incr_frac; 229 index += dst_index * dst_incr; 230 index += frac / c->src_incr; 231 frac %= c->src_incr; 232 }else{ 233 for(dst_index=0; dst_index < dst_size; dst_index++){ 234 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); 235 int sample_index= index >> c->phase_shift; 236 FELEM2 val=0; 237 238 if(sample_index < 0){ 239 for(i=0; i<c->filter_length; i++) 240 val += src[FFABS(sample_index + i) % src_size] * filter[i]; 241 }else if(sample_index + c->filter_length > src_size){ 242 break; 243 }else if(c->linear){ 244 FELEM2 v2=0; 245 for(i=0; i<c->filter_length; i++){ 246 val += src[sample_index + i] * (FELEM2)filter[i]; 247 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; 248 } 249 val+=(v2-val)*(FELEML)frac / c->src_incr; 250 }else{ 251 for(i=0; i<c->filter_length; i++){ 252 val += src[sample_index + i] * (FELEM2)filter[i]; 253 } 254 } 255 256#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 257 dst[dst_index] = av_clip_int16(lrintf(val)); 258#else 259 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; 260 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; 261#endif 262 263 frac += dst_incr_frac; 264 index += dst_incr; 265 if(frac >= c->src_incr){ 266 frac -= c->src_incr; 267 index++; 268 } 269 270 if(dst_index + 1 == compensation_distance){ 271 compensation_distance= 0; 272 dst_incr_frac= c->ideal_dst_incr % c->src_incr; 273 dst_incr= c->ideal_dst_incr / c->src_incr; 274 } 275 } 276 } 277 *consumed= FFMAX(index, 0) >> c->phase_shift; 278 if(index>=0) index &= c->phase_mask; 279 280 if(compensation_distance){ 281 compensation_distance -= dst_index; 282 assert(compensation_distance > 0); 283 } 284 if(update_ctx){ 285 c->frac= frac; 286 c->index= index; 287 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; 288 c->compensation_distance= compensation_distance; 289 } 290#if 0 291 if(update_ctx && !c->compensation_distance){ 292#undef rand 293 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); 294av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); 295 } 296#endif 297 298 return dst_index; 299} 300