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 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 * builds 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 */ 98static void build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ 99 int ph, i; 100 double x, y, w, tab[tap_count]; 101 const int center= (tap_count-1)/2; 102 103 /* if upsampling, only need to interpolate, no filter */ 104 if (factor > 1.0) 105 factor = 1.0; 106 107 for(ph=0;ph<phase_count;ph++) { 108 double norm = 0; 109 for(i=0;i<tap_count;i++) { 110 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; 111 if (x == 0) y = 1.0; 112 else y = sin(x) / x; 113 switch(type){ 114 case 0:{ 115 const float d= -0.5; //first order derivative = -0.5 116 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); 117 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); 118 else y= d*(-4 + 8*x - 5*x*x + x*x*x); 119 break;} 120 case 1: 121 w = 2.0*x / (factor*tap_count) + M_PI; 122 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); 123 break; 124 default: 125 w = 2.0*x / (factor*tap_count*M_PI); 126 y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); 127 break; 128 } 129 130 tab[i] = y; 131 norm += y; 132 } 133 134 /* normalize so that an uniform color remains the same */ 135 for(i=0;i<tap_count;i++) { 136#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 137 filter[ph * tap_count + i] = tab[i] / norm; 138#else 139 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); 140#endif 141 } 142 } 143#if 0 144 { 145#define LEN 1024 146 int j,k; 147 double sine[LEN + tap_count]; 148 double filtered[LEN]; 149 double maxff=-2, minff=2, maxsf=-2, minsf=2; 150 for(i=0; i<LEN; i++){ 151 double ss=0, sf=0, ff=0; 152 for(j=0; j<LEN+tap_count; j++) 153 sine[j]= cos(i*j*M_PI/LEN); 154 for(j=0; j<LEN; j++){ 155 double sum=0; 156 ph=0; 157 for(k=0; k<tap_count; k++) 158 sum += filter[ph * tap_count + k] * sine[k+j]; 159 filtered[j]= sum / (1<<FILTER_SHIFT); 160 ss+= sine[j + center] * sine[j + center]; 161 ff+= filtered[j] * filtered[j]; 162 sf+= sine[j + center] * filtered[j]; 163 } 164 ss= sqrt(2*ss/LEN); 165 ff= sqrt(2*ff/LEN); 166 sf= 2*sf/LEN; 167 maxff= FFMAX(maxff, ff); 168 minff= FFMIN(minff, ff); 169 maxsf= FFMAX(maxsf, sf); 170 minsf= FFMIN(minsf, sf); 171 if(i%11==0){ 172 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); 173 minff=minsf= 2; 174 maxff=maxsf= -2; 175 } 176 } 177 } 178#endif 179} 180 181AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ 182 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); 183 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); 184 int phase_count= 1<<phase_shift; 185 186 c->phase_shift= phase_shift; 187 c->phase_mask= phase_count-1; 188 c->linear= linear; 189 190 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); 191 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); 192 build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE); 193 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); 194 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; 195 196 c->src_incr= out_rate; 197 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count; 198 c->index= -phase_count*((c->filter_length-1)/2); 199 200 return c; 201} 202 203void av_resample_close(AVResampleContext *c){ 204 av_freep(&c->filter_bank); 205 av_freep(&c); 206} 207 208void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ 209// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; 210 c->compensation_distance= compensation_distance; 211 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; 212} 213 214int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ 215 int dst_index, i; 216 int index= c->index; 217 int frac= c->frac; 218 int dst_incr_frac= c->dst_incr % c->src_incr; 219 int dst_incr= c->dst_incr / c->src_incr; 220 int compensation_distance= c->compensation_distance; 221 222 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ 223 int64_t index2= ((int64_t)index)<<32; 224 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; 225 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); 226 227 for(dst_index=0; dst_index < dst_size; dst_index++){ 228 dst[dst_index] = src[index2>>32]; 229 index2 += incr; 230 } 231 frac += dst_index * dst_incr_frac; 232 index += dst_index * dst_incr; 233 index += frac / c->src_incr; 234 frac %= c->src_incr; 235 }else{ 236 for(dst_index=0; dst_index < dst_size; dst_index++){ 237 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); 238 int sample_index= index >> c->phase_shift; 239 FELEM2 val=0; 240 241 if(sample_index < 0){ 242 for(i=0; i<c->filter_length; i++) 243 val += src[FFABS(sample_index + i) % src_size] * filter[i]; 244 }else if(sample_index + c->filter_length > src_size){ 245 break; 246 }else if(c->linear){ 247 FELEM2 v2=0; 248 for(i=0; i<c->filter_length; i++){ 249 val += src[sample_index + i] * (FELEM2)filter[i]; 250 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; 251 } 252 val+=(v2-val)*(FELEML)frac / c->src_incr; 253 }else{ 254 for(i=0; i<c->filter_length; i++){ 255 val += src[sample_index + i] * (FELEM2)filter[i]; 256 } 257 } 258 259#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE 260 dst[dst_index] = av_clip_int16(lrintf(val)); 261#else 262 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; 263 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; 264#endif 265 266 frac += dst_incr_frac; 267 index += dst_incr; 268 if(frac >= c->src_incr){ 269 frac -= c->src_incr; 270 index++; 271 } 272 273 if(dst_index + 1 == compensation_distance){ 274 compensation_distance= 0; 275 dst_incr_frac= c->ideal_dst_incr % c->src_incr; 276 dst_incr= c->ideal_dst_incr / c->src_incr; 277 } 278 } 279 } 280 *consumed= FFMAX(index, 0) >> c->phase_shift; 281 if(index>=0) index &= c->phase_mask; 282 283 if(compensation_distance){ 284 compensation_distance -= dst_index; 285 assert(compensation_distance > 0); 286 } 287 if(update_ctx){ 288 c->frac= frac; 289 c->index= index; 290 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; 291 c->compensation_distance= compensation_distance; 292 } 293#if 0 294 if(update_ctx && !c->compensation_distance){ 295#undef rand 296 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); 297av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); 298 } 299#endif 300 301 return dst_index; 302} 303