libavresample/resample.c

   1 /*
   2  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
   3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
   4  *
   5  * This file is part of Libav.
   6  *
   7  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 #include "libavutil/common.h"
  23 #include "libavutil/libm.h"
  24 #include "libavutil/log.h"
  25 #include "internal.h"
  26 #include "audio_data.h"
  27
  28 struct ResampleContext {
  29     AVAudioResampleContext *avr;
  30     AudioData *buffer;
  31     uint8_t *filter_bank;
  32     int filter_length;
  33     int ideal_dst_incr;
  34     int dst_incr;
  35     int index;
  36     int frac;
  37     int src_incr;
  38     int compensation_distance;
  39     int phase_shift;
  40     int phase_mask;
  41     int linear;
  42     enum AVResampleFilterType filter_type;
  43     int kaiser_beta;
  44     double factor;
  45     void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
  46     void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
  47                          int dst_index, const void *src0, int src_size,
  48                          int index, int frac);
  49 };
  50
  51
  52 /* double template */
  53 #define CONFIG_RESAMPLE_DBL
  54 #include "resample_template.c"
  55 #undef CONFIG_RESAMPLE_DBL
  56
  57 /* float template */
  58 #define CONFIG_RESAMPLE_FLT
  59 #include "resample_template.c"
  60 #undef CONFIG_RESAMPLE_FLT
  61
  62 /* s32 template */
  63 #define CONFIG_RESAMPLE_S32
  64 #include "resample_template.c"
  65 #undef CONFIG_RESAMPLE_S32
  66
  67 /* s16 template */
  68 #include "resample_template.c"
  69
  70
  71 /* 0th order modified bessel function of the first kind. */
  72 static double bessel(double x)
  73 {
  74     double v     = 1;
  75     double lastv = 0;
  76     double t     = 1;
  77     int i;
  78
  79     x = x * x / 4;
  80     for (i = 1; v != lastv; i++) {
  81         lastv = v;
  82         t    *= x / (i * i);
  83         v    += t;
  84     }
  85     return v;
  86 }
  87
  88 /* Build a polyphase filterbank. */
  89 static int build_filter(ResampleContext *c)
  90 {
  91     int ph, i;
  92     double x, y, w, factor;
  93     double *tab;
  94     int tap_count    = c->filter_length;
  95     int phase_count  = 1 << c->phase_shift;
  96     const int center = (tap_count - 1) / 2;
  97
  98     tab = av_malloc(tap_count * sizeof(*tab));
  99     if (!tab)
 100         return AVERROR(ENOMEM);
 101
 102     /* if upsampling, only need to interpolate, no filter */
 103     factor = FFMIN(c->factor, 1.0);
 104
 105     for (ph = 0; ph < phase_count; ph++) {
 106         double norm = 0;
 107         for (i = 0; i < tap_count; i++) {
 108             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
 109             if (x == 0) y = 1.0;
 110             else        y = sin(x) / x;
 111             switch (c->filter_type) {
 112             case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
 113                 const float d = -0.5; //first order derivative = -0.5
 114                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
 115                 if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * (                -x*x + x*x*x);
 116                 else         y =                           d * (-4 + 8 * x - 5 * x*x + x*x*x);
 117                 break;
 118             }
 119             case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
 120                 w  = 2.0 * x / (factor * tap_count) + M_PI;
 121                 y *= 0.3635819 - 0.4891775 * cos(    w) +
 122                                  0.1365995 * cos(2 * w) -
 123                                  0.0106411 * cos(3 * w);
 124                 break;
 125             case AV_RESAMPLE_FILTER_TYPE_KAISER:
 126                 w  = 2.0 * x / (factor * tap_count * M_PI);
 127                 y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
 128                 break;
 129             }
 130
 131             tab[i] = y;
 132             norm  += y;
 133         }
 134         /* normalize so that an uniform color remains the same */
 135         for (i = 0; i < tap_count; i++)
 136             tab[i] = tab[i] / norm;
 137
 138         c->set_filter(c->filter_bank, tab, ph, tap_count);
 139     }
 140
 141     av_free(tab);
 142     return 0;
 143 }
 144
 145 ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
 146 {
 147     ResampleContext *c;
 148     int out_rate    = avr->out_sample_rate;
 149     int in_rate     = avr->in_sample_rate;
 150     double factor   = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
 151     int phase_count = 1 << avr->phase_shift;
 152     int felem_size;
 153
 154     if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
 155         avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
 156         avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
 157         avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
 158         av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
 159                "resampling: %s\n",
 160                av_get_sample_fmt_name(avr->internal_sample_fmt));
 161         return NULL;
 162     }
 163     c = av_mallocz(sizeof(*c));
 164     if (!c)
 165         return NULL;
 166
 167     c->avr           = avr;
 168     c->phase_shift   = avr->phase_shift;
 169     c->phase_mask    = phase_count - 1;
 170     c->linear        = avr->linear_interp;
 171     c->factor        = factor;
 172     c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
 173     c->filter_type   = avr->filter_type;
 174     c->kaiser_beta   = avr->kaiser_beta;
 175
 176     switch (avr->internal_sample_fmt) {
 177     case AV_SAMPLE_FMT_DBLP:
 178         c->resample_one  = resample_one_dbl;
 179         c->set_filter    = set_filter_dbl;
 180         break;
 181     case AV_SAMPLE_FMT_FLTP:
 182         c->resample_one  = resample_one_flt;
 183         c->set_filter    = set_filter_flt;
 184         break;
 185     case AV_SAMPLE_FMT_S32P:
 186         c->resample_one  = resample_one_s32;
 187         c->set_filter    = set_filter_s32;
 188         break;
 189     case AV_SAMPLE_FMT_S16P:
 190         c->resample_one  = resample_one_s16;
 191         c->set_filter    = set_filter_s16;
 192         break;
 193     }
 194
 195     felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
 196     c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
 197     if (!c->filter_bank)
 198         goto error;
 199
 200     if (build_filter(c) < 0)
 201         goto error;
 202
 203     memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
 204            c->filter_bank, (c->filter_length - 1) * felem_size);
 205     memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
 206            &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
 207
 208     c->compensation_distance = 0;
 209     if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
 210                    in_rate * (int64_t)phase_count, INT32_MAX / 2))
 211         goto error;
 212     c->ideal_dst_incr = c->dst_incr;
 213
 214     c->index = -phase_count * ((c->filter_length - 1) / 2);
 215     c->frac  = 0;
 216
 217     /* allocate internal buffer */
 218     c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,
 219                                     avr->internal_sample_fmt,
 220                                     "resample buffer");
 221     if (!c->buffer)
 222         goto error;
 223
 224     av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
 225            av_get_sample_fmt_name(avr->internal_sample_fmt),
 226            avr->in_sample_rate, avr->out_sample_rate);
 227
 228     return c;
 229
 230 error:
 231     ff_audio_data_free(&c->buffer);
 232     av_free(c->filter_bank);
 233     av_free(c);
 234     return NULL;
 235 }
 236
 237 void ff_audio_resample_free(ResampleContext **c)
 238 {
 239     if (!*c)
 240         return;
 241     ff_audio_data_free(&(*c)->buffer);
 242     av_free((*c)->filter_bank);
 243     av_freep(c);
 244 }
 245
 246 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
 247                                 int compensation_distance)
 248 {
 249     ResampleContext *c;
 250     AudioData *fifo_buf = NULL;
 251     int ret = 0;
 252
 253     if (compensation_distance < 0)
 254         return AVERROR(EINVAL);
 255     if (!compensation_distance && sample_delta)
 256         return AVERROR(EINVAL);
 257
 258     /* if resampling was not enabled previously, re-initialize the
 259        AVAudioResampleContext and force resampling */
 260     if (!avr->resample_needed) {
 261         int fifo_samples;
 262         double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };
 263
 264         /* buffer any remaining samples in the output FIFO before closing */
 265         fifo_samples = av_audio_fifo_size(avr->out_fifo);
 266         if (fifo_samples > 0) {
 267             fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
 268                                            avr->out_sample_fmt, NULL);
 269             if (!fifo_buf)
 270                 return AVERROR(EINVAL);
 271             ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
 272                                                fifo_samples);
 273             if (ret < 0)
 274                 goto reinit_fail;
 275         }
 276         /* save the channel mixing matrix */
 277         ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
 278         if (ret < 0)
 279             goto reinit_fail;
 280
 281         /* close the AVAudioResampleContext */
 282         avresample_close(avr);
 283
 284         avr->force_resampling = 1;
 285
 286         /* restore the channel mixing matrix */
 287         ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
 288         if (ret < 0)
 289             goto reinit_fail;
 290
 291         /* re-open the AVAudioResampleContext */
 292         ret = avresample_open(avr);
 293         if (ret < 0)
 294             goto reinit_fail;
 295
 296         /* restore buffered samples to the output FIFO */
 297         if (fifo_samples > 0) {
 298             ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
 299                                             fifo_samples);
 300             if (ret < 0)
 301                 goto reinit_fail;
 302             ff_audio_data_free(&fifo_buf);
 303         }
 304     }
 305     c = avr->resample;
 306     c->compensation_distance = compensation_distance;
 307     if (compensation_distance) {
 308         c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
 309                       (int64_t)sample_delta / compensation_distance;
 310     } else {
 311         c->dst_incr = c->ideal_dst_incr;
 312     }
 313     return 0;
 314
 315 reinit_fail:
 316     ff_audio_data_free(&fifo_buf);
 317     return ret;
 318 }
 319
 320 static int resample(ResampleContext *c, void *dst, const void *src,
 321                     int *consumed, int src_size, int dst_size, int update_ctx)
 322 {
 323     int dst_index;
 324     int index         = c->index;
 325     int frac          = c->frac;
 326     int dst_incr_frac = c->dst_incr % c->src_incr;
 327     int dst_incr      = c->dst_incr / c->src_incr;
 328     int compensation_distance = c->compensation_distance;
 329
 330     if (!dst != !src)
 331         return AVERROR(EINVAL);
 332
 333     if (compensation_distance == 0 && c->filter_length == 1 &&
 334         c->phase_shift == 0) {
 335         int64_t index2 = ((int64_t)index) << 32;
 336         int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;
 337         dst_size       = FFMIN(dst_size,
 338                                (src_size-1-index) * (int64_t)c->src_incr /
 339                                c->dst_incr);
 340
 341         if (dst) {
 342             for(dst_index = 0; dst_index < dst_size; dst_index++) {
 343                 c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
 344                 index2 += incr;
 345             }
 346         } else {
 347             dst_index = dst_size;
 348         }
 349         index += dst_index * dst_incr;
 350         index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
 351         frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
 352     } else {
 353         for (dst_index = 0; dst_index < dst_size; dst_index++) {
 354             int sample_index = index >> c->phase_shift;
 355
 356             if (sample_index + c->filter_length > src_size ||
 357                 -sample_index >= src_size)
 358                 break;
 359
 360             if (dst)
 361                 c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
 362
 363             frac  += dst_incr_frac;
 364             index += dst_incr;
 365             if (frac >= c->src_incr) {
 366                 frac -= c->src_incr;
 367                 index++;
 368             }
 369             if (dst_index + 1 == compensation_distance) {
 370                 compensation_distance = 0;
 371                 dst_incr_frac = c->ideal_dst_incr % c->src_incr;
 372                 dst_incr      = c->ideal_dst_incr / c->src_incr;
 373             }
 374         }
 375     }
 376     if (consumed)
 377         *consumed = FFMAX(index, 0) >> c->phase_shift;
 378
 379     if (update_ctx) {
 380         if (index >= 0)
 381             index &= c->phase_mask;
 382
 383         if (compensation_distance) {
 384             compensation_distance -= dst_index;
 385             if (compensation_distance <= 0)
 386                 return AVERROR_BUG;
 387         }
 388         c->frac     = frac;
 389         c->index    = index;
 390         c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
 391         c->compensation_distance = compensation_distance;
 392     }
 393
 394     return dst_index;
 395 }
 396
 397 int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
 398                       int *consumed)
 399 {
 400     int ch, in_samples, in_leftover, out_samples = 0;
 401     int ret = AVERROR(EINVAL);
 402
 403     in_samples  = src ? src->nb_samples : 0;
 404     in_leftover = c->buffer->nb_samples;
 405
 406     /* add input samples to the internal buffer */
 407     if (src) {
 408         ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
 409         if (ret < 0)
 410             return ret;
 411     } else if (!in_leftover) {
 412         /* no remaining samples to flush */
 413         return 0;
 414     } else {
 415         /* TODO: pad buffer to flush completely */
 416     }
 417
 418     /* calculate output size and reallocate output buffer if needed */
 419     /* TODO: try to calculate this without the dummy resample() run */
 420     if (!dst->read_only && dst->allow_realloc) {
 421         out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
 422                                INT_MAX, 0);
 423         ret = ff_audio_data_realloc(dst, out_samples);
 424         if (ret < 0) {
 425             av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
 426             return ret;
 427         }
 428     }
 429
 430     /* resample each channel plane */
 431     for (ch = 0; ch < c->buffer->channels; ch++) {
 432         out_samples = resample(c, (void *)dst->data[ch],
 433                                (const void *)c->buffer->data[ch], consumed,
 434                                c->buffer->nb_samples, dst->allocated_samples,
 435                                ch + 1 == c->buffer->channels);
 436     }
 437     if (out_samples < 0) {
 438         av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
 439         return out_samples;
 440     }
 441
 442     /* drain consumed samples from the internal buffer */
 443     ff_audio_data_drain(c->buffer, *consumed);
 444
 445     av_dlog(c->avr, "resampled %d in + %d leftover to %d out + %d leftover\n",
 446             in_samples, in_leftover, out_samples, c->buffer->nb_samples);
 447
 448     dst->nb_samples = out_samples;
 449     return 0;
 450 }
 451
 452 int avresample_get_delay(AVAudioResampleContext *avr)
 453 {
 454     if (!avr->resample_needed || !avr->resample)
 455         return 0;
 456
 457     return avr->resample->buffer->nb_samples;
 458 }