| 1 | /* |
| 2 | * audio resampling |
| 3 | * Copyright (c) 2004-2012 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 "libavutil/avassert.h" |
| 29 | #include "resample.h" |
| 30 | |
| 31 | /** |
| 32 | * 0th order modified bessel function of the first kind. |
| 33 | */ |
| 34 | static double bessel(double x){ |
| 35 | double v=1; |
| 36 | double lastv=0; |
| 37 | double t=1; |
| 38 | int i; |
| 39 | static const double inv[100]={ |
| 40 | 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10), |
| 41 | 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20), |
| 42 | 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30), |
| 43 | 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40), |
| 44 | 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50), |
| 45 | 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60), |
| 46 | 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70), |
| 47 | 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80), |
| 48 | 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90), |
| 49 | 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000) |
| 50 | }; |
| 51 | |
| 52 | x= x*x/4; |
| 53 | for(i=0; v != lastv; i++){ |
| 54 | lastv=v; |
| 55 | t *= x*inv[i]; |
| 56 | v += t; |
| 57 | av_assert2(i<99); |
| 58 | } |
| 59 | return v; |
| 60 | } |
| 61 | |
| 62 | /** |
| 63 | * builds a polyphase filterbank. |
| 64 | * @param factor resampling factor |
| 65 | * @param scale wanted sum of coefficients for each filter |
| 66 | * @param filter_type filter type |
| 67 | * @param kaiser_beta kaiser window beta |
| 68 | * @return 0 on success, negative on error |
| 69 | */ |
| 70 | static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, |
| 71 | int filter_type, int kaiser_beta){ |
| 72 | int ph, i; |
| 73 | double x, y, w; |
| 74 | double *tab = av_malloc_array(tap_count, sizeof(*tab)); |
| 75 | const int center= (tap_count-1)/2; |
| 76 | |
| 77 | if (!tab) |
| 78 | return AVERROR(ENOMEM); |
| 79 | |
| 80 | /* if upsampling, only need to interpolate, no filter */ |
| 81 | if (factor > 1.0) |
| 82 | factor = 1.0; |
| 83 | |
| 84 | for(ph=0;ph<phase_count;ph++) { |
| 85 | double norm = 0; |
| 86 | for(i=0;i<tap_count;i++) { |
| 87 | x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
| 88 | if (x == 0) y = 1.0; |
| 89 | else y = sin(x) / x; |
| 90 | switch(filter_type){ |
| 91 | case SWR_FILTER_TYPE_CUBIC:{ |
| 92 | const float d= -0.5; //first order derivative = -0.5 |
| 93 | x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
| 94 | if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
| 95 | else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
| 96 | break;} |
| 97 | case SWR_FILTER_TYPE_BLACKMAN_NUTTALL: |
| 98 | w = 2.0*x / (factor*tap_count) + M_PI; |
| 99 | y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
| 100 | break; |
| 101 | case SWR_FILTER_TYPE_KAISER: |
| 102 | w = 2.0*x / (factor*tap_count*M_PI); |
| 103 | y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); |
| 104 | break; |
| 105 | default: |
| 106 | av_assert0(0); |
| 107 | } |
| 108 | |
| 109 | tab[i] = y; |
| 110 | norm += y; |
| 111 | } |
| 112 | |
| 113 | /* normalize so that an uniform color remains the same */ |
| 114 | switch(c->format){ |
| 115 | case AV_SAMPLE_FMT_S16P: |
| 116 | for(i=0;i<tap_count;i++) |
| 117 | ((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX); |
| 118 | break; |
| 119 | case AV_SAMPLE_FMT_S32P: |
| 120 | for(i=0;i<tap_count;i++) |
| 121 | ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm)); |
| 122 | break; |
| 123 | case AV_SAMPLE_FMT_FLTP: |
| 124 | for(i=0;i<tap_count;i++) |
| 125 | ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm; |
| 126 | break; |
| 127 | case AV_SAMPLE_FMT_DBLP: |
| 128 | for(i=0;i<tap_count;i++) |
| 129 | ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm; |
| 130 | break; |
| 131 | } |
| 132 | } |
| 133 | #if 0 |
| 134 | { |
| 135 | #define LEN 1024 |
| 136 | int j,k; |
| 137 | double sine[LEN + tap_count]; |
| 138 | double filtered[LEN]; |
| 139 | double maxff=-2, minff=2, maxsf=-2, minsf=2; |
| 140 | for(i=0; i<LEN; i++){ |
| 141 | double ss=0, sf=0, ff=0; |
| 142 | for(j=0; j<LEN+tap_count; j++) |
| 143 | sine[j]= cos(i*j*M_PI/LEN); |
| 144 | for(j=0; j<LEN; j++){ |
| 145 | double sum=0; |
| 146 | ph=0; |
| 147 | for(k=0; k<tap_count; k++) |
| 148 | sum += filter[ph * tap_count + k] * sine[k+j]; |
| 149 | filtered[j]= sum / (1<<FILTER_SHIFT); |
| 150 | ss+= sine[j + center] * sine[j + center]; |
| 151 | ff+= filtered[j] * filtered[j]; |
| 152 | sf+= sine[j + center] * filtered[j]; |
| 153 | } |
| 154 | ss= sqrt(2*ss/LEN); |
| 155 | ff= sqrt(2*ff/LEN); |
| 156 | sf= 2*sf/LEN; |
| 157 | maxff= FFMAX(maxff, ff); |
| 158 | minff= FFMIN(minff, ff); |
| 159 | maxsf= FFMAX(maxsf, sf); |
| 160 | minsf= FFMIN(minsf, sf); |
| 161 | if(i%11==0){ |
| 162 | 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); |
| 163 | minff=minsf= 2; |
| 164 | maxff=maxsf= -2; |
| 165 | } |
| 166 | } |
| 167 | } |
| 168 | #endif |
| 169 | |
| 170 | av_free(tab); |
| 171 | return 0; |
| 172 | } |
| 173 | |
| 174 | static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, |
| 175 | double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, |
| 176 | double precision, int cheby) |
| 177 | { |
| 178 | double cutoff = cutoff0? cutoff0 : 0.97; |
| 179 | double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
| 180 | int phase_count= 1<<phase_shift; |
| 181 | |
| 182 | if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor |
| 183 | || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format |
| 184 | || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { |
| 185 | c = av_mallocz(sizeof(*c)); |
| 186 | if (!c) |
| 187 | return NULL; |
| 188 | |
| 189 | c->format= format; |
| 190 | |
| 191 | c->felem_size= av_get_bytes_per_sample(c->format); |
| 192 | |
| 193 | switch(c->format){ |
| 194 | case AV_SAMPLE_FMT_S16P: |
| 195 | c->filter_shift = 15; |
| 196 | break; |
| 197 | case AV_SAMPLE_FMT_S32P: |
| 198 | c->filter_shift = 30; |
| 199 | break; |
| 200 | case AV_SAMPLE_FMT_FLTP: |
| 201 | case AV_SAMPLE_FMT_DBLP: |
| 202 | c->filter_shift = 0; |
| 203 | break; |
| 204 | default: |
| 205 | av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n"); |
| 206 | av_assert0(0); |
| 207 | } |
| 208 | |
| 209 | if (filter_size/factor > INT32_MAX/256) { |
| 210 | av_log(NULL, AV_LOG_ERROR, "Filter length too large\n"); |
| 211 | goto error; |
| 212 | } |
| 213 | |
| 214 | c->phase_shift = phase_shift; |
| 215 | c->phase_mask = phase_count - 1; |
| 216 | c->linear = linear; |
| 217 | c->factor = factor; |
| 218 | c->filter_length = FFMAX((int)ceil(filter_size/factor), 1); |
| 219 | c->filter_alloc = FFALIGN(c->filter_length, 8); |
| 220 | c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size); |
| 221 | c->filter_type = filter_type; |
| 222 | c->kaiser_beta = kaiser_beta; |
| 223 | if (!c->filter_bank) |
| 224 | goto error; |
| 225 | if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta)) |
| 226 | goto error; |
| 227 | memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size); |
| 228 | memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size); |
| 229 | } |
| 230 | |
| 231 | c->compensation_distance= 0; |
| 232 | if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2)) |
| 233 | goto error; |
| 234 | c->ideal_dst_incr = c->dst_incr; |
| 235 | c->dst_incr_div = c->dst_incr / c->src_incr; |
| 236 | c->dst_incr_mod = c->dst_incr % c->src_incr; |
| 237 | |
| 238 | c->index= -phase_count*((c->filter_length-1)/2); |
| 239 | c->frac= 0; |
| 240 | |
| 241 | swri_resample_dsp_init(c); |
| 242 | |
| 243 | return c; |
| 244 | error: |
| 245 | av_freep(&c->filter_bank); |
| 246 | av_free(c); |
| 247 | return NULL; |
| 248 | } |
| 249 | |
| 250 | static void resample_free(ResampleContext **c){ |
| 251 | if(!*c) |
| 252 | return; |
| 253 | av_freep(&(*c)->filter_bank); |
| 254 | av_freep(c); |
| 255 | } |
| 256 | |
| 257 | static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ |
| 258 | c->compensation_distance= compensation_distance; |
| 259 | if (compensation_distance) |
| 260 | c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; |
| 261 | else |
| 262 | c->dst_incr = c->ideal_dst_incr; |
| 263 | |
| 264 | c->dst_incr_div = c->dst_incr / c->src_incr; |
| 265 | c->dst_incr_mod = c->dst_incr % c->src_incr; |
| 266 | |
| 267 | return 0; |
| 268 | } |
| 269 | |
| 270 | static int swri_resample(ResampleContext *c, |
| 271 | uint8_t *dst, const uint8_t *src, int *consumed, |
| 272 | int src_size, int dst_size, int update_ctx) |
| 273 | { |
| 274 | if (c->filter_length == 1 && c->phase_shift == 0) { |
| 275 | int index= c->index; |
| 276 | int frac= c->frac; |
| 277 | int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index; |
| 278 | int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; |
| 279 | int new_size = (src_size * (int64_t)c->src_incr - frac + c->dst_incr - 1) / c->dst_incr; |
| 280 | |
| 281 | dst_size= FFMIN(dst_size, new_size); |
| 282 | c->dsp.resample_one(dst, src, dst_size, index2, incr); |
| 283 | |
| 284 | index += dst_size * c->dst_incr_div; |
| 285 | index += (frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr; |
| 286 | av_assert2(index >= 0); |
| 287 | *consumed= index; |
| 288 | if (update_ctx) { |
| 289 | c->frac = (frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr; |
| 290 | c->index = 0; |
| 291 | } |
| 292 | } else { |
| 293 | int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift; |
| 294 | int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac; |
| 295 | int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr; |
| 296 | |
| 297 | dst_size = FFMIN(dst_size, delta_n); |
| 298 | if (dst_size > 0) { |
| 299 | *consumed = c->dsp.resample(c, dst, src, dst_size, update_ctx); |
| 300 | } else { |
| 301 | *consumed = 0; |
| 302 | } |
| 303 | } |
| 304 | |
| 305 | return dst_size; |
| 306 | } |
| 307 | |
| 308 | static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ |
| 309 | int i, ret= -1; |
| 310 | int av_unused mm_flags = av_get_cpu_flags(); |
| 311 | int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 && |
| 312 | (mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2; |
| 313 | int64_t max_src_size = (INT64_MAX >> (c->phase_shift+1)) / c->src_incr; |
| 314 | |
| 315 | if (c->compensation_distance) |
| 316 | dst_size = FFMIN(dst_size, c->compensation_distance); |
| 317 | src_size = FFMIN(src_size, max_src_size); |
| 318 | |
| 319 | for(i=0; i<dst->ch_count; i++){ |
| 320 | ret= swri_resample(c, dst->ch[i], src->ch[i], |
| 321 | consumed, src_size, dst_size, i+1==dst->ch_count); |
| 322 | } |
| 323 | if(need_emms) |
| 324 | emms_c(); |
| 325 | |
| 326 | if (c->compensation_distance) { |
| 327 | c->compensation_distance -= ret; |
| 328 | if (!c->compensation_distance) { |
| 329 | c->dst_incr = c->ideal_dst_incr; |
| 330 | c->dst_incr_div = c->dst_incr / c->src_incr; |
| 331 | c->dst_incr_mod = c->dst_incr % c->src_incr; |
| 332 | } |
| 333 | } |
| 334 | |
| 335 | return ret; |
| 336 | } |
| 337 | |
| 338 | static int64_t get_delay(struct SwrContext *s, int64_t base){ |
| 339 | ResampleContext *c = s->resample; |
| 340 | int64_t num = s->in_buffer_count - (c->filter_length-1)/2; |
| 341 | num <<= c->phase_shift; |
| 342 | num -= c->index; |
| 343 | num *= c->src_incr; |
| 344 | num -= c->frac; |
| 345 | return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); |
| 346 | } |
| 347 | |
| 348 | static int resample_flush(struct SwrContext *s) { |
| 349 | AudioData *a= &s->in_buffer; |
| 350 | int i, j, ret; |
| 351 | if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0) |
| 352 | return ret; |
| 353 | av_assert0(a->planar); |
| 354 | for(i=0; i<a->ch_count; i++){ |
| 355 | for(j=0; j<s->in_buffer_count; j++){ |
| 356 | memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps, |
| 357 | a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps); |
| 358 | } |
| 359 | } |
| 360 | s->in_buffer_count += (s->in_buffer_count+1)/2; |
| 361 | return 0; |
| 362 | } |
| 363 | |
| 364 | // in fact the whole handle multiple ridiculously small buffers might need more thinking... |
| 365 | static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src, |
| 366 | int in_count, int *out_idx, int *out_sz) |
| 367 | { |
| 368 | int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res; |
| 369 | |
| 370 | if (c->index >= 0) |
| 371 | return 0; |
| 372 | |
| 373 | if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0) |
| 374 | return res; |
| 375 | |
| 376 | // copy |
| 377 | for (n = *out_sz; n < num; n++) { |
| 378 | for (ch = 0; ch < src->ch_count; ch++) { |
| 379 | memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size), |
| 380 | src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size); |
| 381 | } |
| 382 | } |
| 383 | |
| 384 | // if not enough data is in, return and wait for more |
| 385 | if (num < c->filter_length + 1) { |
| 386 | *out_sz = num; |
| 387 | *out_idx = c->filter_length; |
| 388 | return INT_MAX; |
| 389 | } |
| 390 | |
| 391 | // else invert |
| 392 | for (n = 1; n <= c->filter_length; n++) { |
| 393 | for (ch = 0; ch < src->ch_count; ch++) { |
| 394 | memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size), |
| 395 | dst->ch[ch] + ((c->filter_length + n) * c->felem_size), |
| 396 | c->felem_size); |
| 397 | } |
| 398 | } |
| 399 | |
| 400 | res = num - *out_sz; |
| 401 | *out_idx = c->filter_length + (c->index >> c->phase_shift); |
| 402 | *out_sz = FFMAX(*out_sz + c->filter_length, |
| 403 | 1 + c->filter_length * 2) - *out_idx; |
| 404 | c->index &= c->phase_mask; |
| 405 | |
| 406 | return FFMAX(res, 0); |
| 407 | } |
| 408 | |
| 409 | struct Resampler const swri_resampler={ |
| 410 | resample_init, |
| 411 | resample_free, |
| 412 | multiple_resample, |
| 413 | resample_flush, |
| 414 | set_compensation, |
| 415 | get_delay, |
| 416 | invert_initial_buffer, |
| 417 | }; |