ffmpeg/libswresample/resample.c

   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 };