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, see <http://www.gnu.org/licenses/>. 19 */ 20 21/** 22 * @file libavcodec/resample2.c 23 * audio resampling 24 * @author Michael Niedermayer <michaelni@gmx.at> 25 */ 26 27#include "avcodec.h" 28#include "dsputil.h" 29 30#ifndef CONFIG_RESAMPLE_HP 31#define FILTER_SHIFT 15 32 33#define FELEM int16_t 34#define FELEM2 int32_t 35#define FELEML int64_t 36#define FELEM_MAX INT16_MAX 37#define FELEM_MIN INT16_MIN 38#define WINDOW_TYPE 9 39#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) 40#define FILTER_SHIFT 30 41 42#define FELEM int32_t 43#define FELEM2 int64_t 44#define FELEML int64_t 45#define FELEM_MAX INT32_MAX 46#define FELEM_MIN INT32_MIN 47#define WINDOW_TYPE 12 48#else 49#define FILTER_SHIFT 0 50 51#define FELEM double 52#define FELEM2 double 53#define FELEML double 54#define WINDOW_TYPE 24 55#endif 56 57 58typedef struct AVResampleContext{ 59 FELEM *filter_bank; 60 int filter_length; 61 int ideal_dst_incr; 62 int dst_incr; 63 int index; 64 int frac; 65 int src_incr; 66 int compensation_distance; 67 int phase_shift; 68 int phase_mask; 69 int linear; 70}AVResampleContext; 71 72/** 73 * 0th order modified bessel function of the first kind. 74 */ 75static double bessel(double x){ 76 double v=1; 77 double t=1; 78 int i; 79 80 x= x*x/4; 81 for(i=1; i<50; i++){ 82 t *= x/(i*i); 83 v += t; 84 } 85 return v; 86} 87 88/** 89 * builds a polyphase filterbank. 90 * @param factor resampling factor 91 * @param scale wanted sum of coefficients for each filter 92 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 93 */ 94void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ 95 int ph, i; 96 double x, y, w, tab[tap_count]; 97 const int center= (tap_count-1)/2; 98 99 /* if upsampling, only need to interpolate, no filter */ 100 if (factor > 1.0) 101 factor = 1.0; 102 103 for(ph=0;ph<phase_count;ph++) { 104 double norm = 0; 105 for(i=0;i<tap_count;i++) { 106 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; 107 if (x == 0) y = 1.0; 108 else y = sin(x) / x; 109 switch(type){ 110 case 0:{ 111 const float d= -0.5; //first order derivative = -0.5 112 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); 113 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); 114 else y= d*(-4 + 8*x - 5*x*x + x*x*x); 115 break;} 116 case 1: 117 w = 2.0*x / (factor*tap_count) + M_PI; 118 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); 119 break; 120 default: 121 w = 2.0*x / (factor*tap_count*M_PI); 122 y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); 123 break; 124 } 125 126 tab[i] = y; 127 norm += y; 128 } 129 130 /* normalize so that an uniform color remains the same */ 131 for(i=0;i<tap_count;i++) { 132#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 133 filter[ph * tap_count + i] = tab[i] / norm; 134#else 135 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); 136#endif 137 } 138 } 139#if 0 140 { 141#define LEN 1024 142 int j,k; 143 double sine[LEN + tap_count]; 144 double filtered[LEN]; 145 double maxff=-2, minff=2, maxsf=-2, minsf=2; 146 for(i=0; i<LEN; i++){ 147 double ss=0, sf=0, ff=0; 148 for(j=0; j<LEN+tap_count; j++) 149 sine[j]= cos(i*j*M_PI/LEN); 150 for(j=0; j<LEN; j++){ 151 double sum=0; 152 ph=0; 153 for(k=0; k<tap_count; k++) 154 sum += filter[ph * tap_count + k] * sine[k+j]; 155 filtered[j]= sum / (1<<FILTER_SHIFT); 156 ss+= sine[j + center] * sine[j + center]; 157 ff+= filtered[j] * filtered[j]; 158 sf+= sine[j + center] * filtered[j]; 159 } 160 ss= sqrt(2*ss/LEN); 161 ff= sqrt(2*ff/LEN); 162 sf= 2*sf/LEN; 163 maxff= FFMAX(maxff, ff); 164 minff= FFMIN(minff, ff); 165 maxsf= FFMAX(maxsf, sf); 166 minsf= FFMIN(minsf, sf); 167 if(i%11==0){ 168 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); 169 minff=minsf= 2; 170 maxff=maxsf= -2; 171 } 172 } 173 } 174#endif 175} 176 177AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ 178 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); 179 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); 180 int phase_count= 1<<phase_shift; 181 182 c->phase_shift= phase_shift; 183 c->phase_mask= phase_count-1; 184 c->linear= linear; 185 186 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); 187 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); 188 av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE); 189 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); 190 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; 191 192 c->src_incr= out_rate; 193 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; 194 c->index= -phase_count*((c->filter_length-1)/2); 195 196 return c; 197} 198 199void av_resample_close(AVResampleContext *c){ 200 av_freep(&c->filter_bank); 201 av_freep(&c); 202} 203 204void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ 205// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; 206 c->compensation_distance= compensation_distance; 207 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; 208} 209 210int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ 211 int dst_index, i; 212 int index= c->index; 213 int frac= c->frac; 214 int dst_incr_frac= c->dst_incr % c->src_incr; 215 int dst_incr= c->dst_incr / c->src_incr; 216 int compensation_distance= c->compensation_distance; 217 218 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ 219 int64_t index2= ((int64_t)index)<<32; 220 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; 221 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); 222 223 for(dst_index=0; dst_index < dst_size; dst_index++){ 224 dst[dst_index] = src[index2>>32]; 225 index2 += incr; 226 } 227 frac += dst_index * dst_incr_frac; 228 index += dst_index * dst_incr; 229 index += frac / c->src_incr; 230 frac %= c->src_incr; 231 }else{ 232 for(dst_index=0; dst_index < dst_size; dst_index++){ 233 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); 234 int sample_index= index >> c->phase_shift; 235 FELEM2 val=0; 236 237 if(sample_index < 0){ 238 for(i=0; i<c->filter_length; i++) 239 val += src[FFABS(sample_index + i) % src_size] * filter[i]; 240 }else if(sample_index + c->filter_length > src_size){ 241 break; 242 }else if(c->linear){ 243 FELEM2 v2=0; 244 for(i=0; i<c->filter_length; i++){ 245 val += src[sample_index + i] * (FELEM2)filter[i]; 246 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; 247 } 248 val+=(v2-val)*(FELEML)frac / c->src_incr; 249 }else{ 250 for(i=0; i<c->filter_length; i++){ 251 val += src[sample_index + i] * (FELEM2)filter[i]; 252 } 253 } 254 255#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 256 dst[dst_index] = av_clip_int16(lrintf(val)); 257#else 258 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; 259 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; 260#endif 261 262 frac += dst_incr_frac; 263 index += dst_incr; 264 if(frac >= c->src_incr){ 265 frac -= c->src_incr; 266 index++; 267 } 268 269 if(dst_index + 1 == compensation_distance){ 270 compensation_distance= 0; 271 dst_incr_frac= c->ideal_dst_incr % c->src_incr; 272 dst_incr= c->ideal_dst_incr / c->src_incr; 273 } 274 } 275 } 276 *consumed= FFMAX(index, 0) >> c->phase_shift; 277 if(index>=0) index &= c->phase_mask; 278 279 if(compensation_distance){ 280 compensation_distance -= dst_index; 281 assert(compensation_distance > 0); 282 } 283 if(update_ctx){ 284 c->frac= frac; 285 c->index= index; 286 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; 287 c->compensation_distance= compensation_distance; 288 } 289#if 0 290 if(update_ctx && !c->compensation_distance){ 291#undef rand 292 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); 293av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); 294 } 295#endif 296 297 return dst_index; 298} 299