git.sesse.net Git - vlc/blob - modules/audio_filter/resampler/bandlimited.c

   1 /*****************************************************************************
   2  * bandlimited.c : band-limited interpolation resampler
   3  *****************************************************************************
   4  * Copyright (C) 2002, 2006 the VideoLAN team
   5  * $Id$
   6  *
   7  * Authors: Gildas Bazin <gbazin@netcourrier.com>
   8  *
   9  * This program is free software; you can redistribute it and/or modify
  10  * it under the terms of the GNU General Public License as published by
  11  * the Free Software Foundation; either version 2 of the License, or
  12  * (at your option) any later version.
  13  *
  14  * This program is distributed in the hope that it will be useful,
  15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  17  * GNU General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU General Public License
  20  * along with this program; if not, write to the Free Software
  21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  22  *****************************************************************************/
  23
  24 /*****************************************************************************
  25  * Preamble:
  26  *
  27  * This implementation of the band-limited interpolationis based on the
  28  * following paper:
  29  * http://ccrma-www.stanford.edu/~jos/resample/resample.html
  30  *
  31  * It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
  32  * filter is 13 samples.
  33  *
  34  *****************************************************************************/
  35 #include <string.h>
  36
  37 #include <vlc/vlc.h>
  38 #include <vlc_aout.h>
  39
  40 #include "bandlimited.h"
  41
  42 /*****************************************************************************
  43  * Local prototypes
  44  *****************************************************************************/
  45 static int  Create    ( vlc_object_t * );
  46 static void Close     ( vlc_object_t * );
  47 static void DoWork    ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
  48                         aout_buffer_t * );
  49
  50 static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing,
  51                            float *f_in, float *f_out, uint32_t ui_remainder,
  52                            uint32_t ui_output_rate, int16_t Inc,
  53                            int i_nb_channels );
  54
  55 static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing,
  56                            float *f_in, float *f_out, uint32_t ui_remainder,
  57                            uint32_t ui_output_rate, uint32_t ui_input_rate,
  58                            int16_t Inc, int i_nb_channels );
  59
  60 /*****************************************************************************
  61  * Local structures
  62  *****************************************************************************/
  63 struct aout_filter_sys_t
  64 {
  65     int32_t *p_buf;                        /* this filter introduces a delay */
  66     int i_buf_size;
  67
  68     int i_old_rate;
  69     double d_old_factor;
  70     int i_old_wing;
  71
  72     unsigned int i_remainder;                /* remainder of previous sample */
  73
  74     audio_date_t end_date;
  75 };
  76
  77 /*****************************************************************************
  78  * Module descriptor
  79  *****************************************************************************/
  80 vlc_module_begin();
  81     set_category( CAT_AUDIO );
  82     set_subcategory( SUBCAT_AUDIO_MISC );
  83     set_description( _("Audio filter for band-limited interpolation resampling") );
  84     set_capability( "audio filter", 20 );
  85     set_callbacks( Create, Close );
  86 vlc_module_end();
  87
  88 /*****************************************************************************
  89  * Create: allocate linear resampler
  90  *****************************************************************************/
  91 static int Create( vlc_object_t *p_this )
  92 {
  93     aout_filter_t * p_filter = (aout_filter_t *)p_this;
  94     double d_factor;
  95     int i_filter_wing;
  96
  97     if ( p_filter->input.i_rate == p_filter->output.i_rate
  98           || p_filter->input.i_format != p_filter->output.i_format
  99           || p_filter->input.i_physical_channels
 100               != p_filter->output.i_physical_channels
 101           || p_filter->input.i_original_channels
 102               != p_filter->output.i_original_channels
 103           || p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
 104     {
 105         return VLC_EGENERIC;
 106     }
 107
 108 #if !defined( __APPLE__ )
 109     if( !config_GetInt( p_this, "hq-resampling" ) )
 110     {
 111         return VLC_EGENERIC;
 112     }
 113 #endif
 114
 115     /* Allocate the memory needed to store the module's structure */
 116     p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
 117     if( p_filter->p_sys == NULL )
 118     {
 119         msg_Err( p_filter, "out of memory" );
 120         return VLC_ENOMEM;
 121     }
 122
 123     /* Calculate worst case for the length of the filter wing */
 124     d_factor = (double)p_filter->output.i_rate
 125                         / p_filter->input.i_rate / AOUT_MAX_INPUT_RATE;
 126     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 127                       * __MAX(1.0, 1.0/d_factor) + 10;
 128     p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
 129         sizeof(int32_t) * 2 * i_filter_wing;
 130
 131     /* Allocate enough memory to buffer previous samples */
 132     p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
 133     if( p_filter->p_sys->p_buf == NULL )
 134     {
 135         msg_Err( p_filter, "out of memory" );
 136         return VLC_ENOMEM;
 137     }
 138
 139     p_filter->p_sys->i_old_wing = 0;
 140     p_filter->pf_do_work = DoWork;
 141
 142     /* We don't want a new buffer to be created because we're not sure we'll
 143      * actually need to resample anything. */
 144     p_filter->b_in_place = VLC_TRUE;
 145
 146     return VLC_SUCCESS;
 147 }
 148
 149 /*****************************************************************************
 150  * Close: free our resources
 151  *****************************************************************************/
 152 static void Close( vlc_object_t * p_this )
 153 {
 154     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 155     free( p_filter->p_sys->p_buf );
 156     free( p_filter->p_sys );
 157 }
 158
 159 /*****************************************************************************
 160  * DoWork: convert a buffer
 161  *****************************************************************************/
 162 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 163                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 164 {
 165     float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
 166
 167     int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
 168     int i_in_nb = p_in_buf->i_nb_samples;
 169     int i_in, i_out = 0;
 170     double d_factor, d_scale_factor, d_old_scale_factor;
 171     int i_filter_wing;
 172 #if 0
 173     int i;
 174 #endif
 175
 176     /* Check if we really need to run the resampler */
 177     if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
 178     {
 179         if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
 180             p_filter->p_sys->i_old_wing &&
 181             p_in_buf->i_size >=
 182               p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
 183               p_filter->input.i_bytes_per_frame )
 184         {
 185             /* output the whole thing with the samples from last time */
 186             memmove( ((float *)(p_in_buf->p_buffer)) +
 187                      i_nb_channels * p_filter->p_sys->i_old_wing,
 188                      p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
 189             memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
 190                     i_nb_channels * p_filter->p_sys->i_old_wing,
 191                     p_filter->p_sys->i_old_wing *
 192                     p_filter->input.i_bytes_per_frame );
 193
 194             p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
 195                 p_filter->p_sys->i_old_wing;
 196
 197             p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
 198             p_out_buf->end_date =
 199                 aout_DateIncrement( &p_filter->p_sys->end_date,
 200                                     p_out_buf->i_nb_samples );
 201
 202             p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 203                 p_filter->input.i_bytes_per_frame;
 204         }
 205         p_filter->b_continuity = VLC_FALSE;
 206         p_filter->p_sys->i_old_wing = 0;
 207         return;
 208     }
 209
 210     if( !p_filter->b_continuity )
 211     {
 212         /* Continuity in sound samples has been broken, we'd better reset
 213          * everything. */
 214         p_filter->b_continuity = VLC_TRUE;
 215         p_filter->p_sys->i_remainder = 0;
 216         aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
 217         aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
 218         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 219         p_filter->p_sys->d_old_factor = 1;
 220         p_filter->p_sys->i_old_wing   = 0;
 221     }
 222
 223 #if 0
 224     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 225              p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
 226              p_filter->p_sys->i_old_wing, i_in_nb );
 227 #endif
 228
 229     /* Prepare the source buffer */
 230     i_in_nb += (p_filter->p_sys->i_old_wing * 2);
 231 #ifdef HAVE_ALLOCA
 232     p_in = p_in_orig = (float *)alloca( i_in_nb *
 233                                         p_filter->input.i_bytes_per_frame );
 234 #else
 235     p_in = p_in_orig = (float *)malloc( i_in_nb *
 236                                         p_filter->input.i_bytes_per_frame );
 237 #endif
 238     if( p_in == NULL )
 239     {
 240         return;
 241     }
 242
 243     /* Copy all our samples in p_in */
 244     if( p_filter->p_sys->i_old_wing )
 245     {
 246         p_aout->p_libvlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
 247                                   p_filter->p_sys->i_old_wing * 2 *
 248                                   p_filter->input.i_bytes_per_frame );
 249     }
 250     p_aout->p_libvlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
 251                               i_nb_channels, p_in_buf->p_buffer,
 252                               p_in_buf->i_nb_samples *
 253                               p_filter->input.i_bytes_per_frame );
 254
 255     /* Make sure the output buffer is reset */
 256     memset( p_out, 0, p_out_buf->i_size );
 257
 258     /* Calculate the new length of the filter wing */
 259     d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
 260     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 261
 262     /* Account for increased filter gain when using factors less than 1 */
 263     d_old_scale_factor = SMALL_FILTER_SCALE *
 264         p_filter->p_sys->d_old_factor + 0.5;
 265     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 266
 267     /* Apply the old rate until we have enough samples for the new one */
 268     i_in = p_filter->p_sys->i_old_wing;
 269     p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
 270     for( ; i_in < i_filter_wing &&
 271            (i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
 272     {
 273         if( p_filter->p_sys->d_old_factor == 1 )
 274         {
 275             /* Just copy the samples */
 276             memcpy( p_out, p_in,
 277                     p_filter->input.i_bytes_per_frame );
 278             p_in += i_nb_channels;
 279             p_out += i_nb_channels;
 280             i_out++;
 281             continue;
 282         }
 283
 284         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 285         {
 286
 287             if( p_filter->p_sys->d_old_factor >= 1 )
 288             {
 289                 /* FilterFloatUP() is faster if we can use it */
 290
 291                 /* Perform left-wing inner product */
 292                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 293                                SMALL_FILTER_NWING, p_in, p_out,
 294                                p_filter->p_sys->i_remainder,
 295                                p_filter->output.i_rate,
 296                                -1, i_nb_channels );
 297                 /* Perform right-wing inner product */
 298                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 299                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 300                                p_filter->output.i_rate -
 301                                p_filter->p_sys->i_remainder,
 302                                p_filter->output.i_rate,
 303                                1, i_nb_channels );
 304
 305 #if 0
 306                 /* Normalize for unity filter gain */
 307                 for( i = 0; i < i_nb_channels; i++ )
 308                 {
 309                     *(p_out+i) *= d_old_scale_factor;
 310                 }
 311 #endif
 312
 313                 /* Sanity check */
 314                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 315                     <= (unsigned int)i_out+1 )
 316                 {
 317                     p_out += i_nb_channels;
 318                     i_out++;
 319                     p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 320                     break;
 321                 }
 322             }
 323             else
 324             {
 325                 /* Perform left-wing inner product */
 326                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 327                                SMALL_FILTER_NWING, p_in, p_out,
 328                                p_filter->p_sys->i_remainder,
 329                                p_filter->output.i_rate, p_filter->input.i_rate,
 330                                -1, i_nb_channels );
 331                 /* Perform right-wing inner product */
 332                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 333                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 334                                p_filter->output.i_rate -
 335                                p_filter->p_sys->i_remainder,
 336                                p_filter->output.i_rate, p_filter->input.i_rate,
 337                                1, i_nb_channels );
 338             }
 339
 340             p_out += i_nb_channels;
 341             i_out++;
 342
 343             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 344         }
 345
 346         p_in += i_nb_channels;
 347         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 348     }
 349
 350     /* Apply the new rate for the rest of the samples */
 351     if( i_in < i_in_nb - i_filter_wing )
 352     {
 353         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 354         p_filter->p_sys->d_old_factor = d_factor;
 355         p_filter->p_sys->i_old_wing   = i_filter_wing;
 356     }
 357     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 358     {
 359         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 360         {
 361
 362             if( d_factor >= 1 )
 363             {
 364                 /* FilterFloatUP() is faster if we can use it */
 365
 366                 /* Perform left-wing inner product */
 367                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 368                                SMALL_FILTER_NWING, p_in, p_out,
 369                                p_filter->p_sys->i_remainder,
 370                                p_filter->output.i_rate,
 371                                -1, i_nb_channels );
 372
 373                 /* Perform right-wing inner product */
 374                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 375                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 376                                p_filter->output.i_rate -
 377                                p_filter->p_sys->i_remainder,
 378                                p_filter->output.i_rate,
 379                                1, i_nb_channels );
 380
 381 #if 0
 382                 /* Normalize for unity filter gain */
 383                 for( i = 0; i < i_nb_channels; i++ )
 384                 {
 385                     *(p_out+i) *= d_old_scale_factor;
 386                 }
 387 #endif
 388                 /* Sanity check */
 389                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 390                     <= (unsigned int)i_out+1 )
 391                 {
 392                     p_out += i_nb_channels;
 393                     i_out++;
 394                     p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 395                     break;
 396                 }
 397             }
 398             else
 399             {
 400                 /* Perform left-wing inner product */
 401                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 402                                SMALL_FILTER_NWING, p_in, p_out,
 403                                p_filter->p_sys->i_remainder,
 404                                p_filter->output.i_rate, p_filter->input.i_rate,
 405                                -1, i_nb_channels );
 406                 /* Perform right-wing inner product */
 407                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 408                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 409                                p_filter->output.i_rate -
 410                                p_filter->p_sys->i_remainder,
 411                                p_filter->output.i_rate, p_filter->input.i_rate,
 412                                1, i_nb_channels );
 413             }
 414
 415             p_out += i_nb_channels;
 416             i_out++;
 417
 418             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 419         }
 420
 421         p_in += i_nb_channels;
 422         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 423     }
 424
 425     /* Buffer i_filter_wing * 2 samples for next time */
 426     if( p_filter->p_sys->i_old_wing )
 427     {
 428         memcpy( p_filter->p_sys->p_buf,
 429                 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
 430                 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
 431                 p_filter->input.i_bytes_per_frame );
 432     }
 433
 434 #if 0
 435     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
 436              i_out * p_filter->input.i_bytes_per_frame );
 437 #endif
 438
 439     /* Free the temp buffer */
 440 #ifndef HAVE_ALLOCA
 441     free( p_in_orig );
 442 #endif
 443
 444     /* Finalize aout buffer */
 445     p_out_buf->i_nb_samples = i_out;
 446     p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
 447     p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
 448                                               p_out_buf->i_nb_samples );
 449
 450     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 451         i_nb_channels * sizeof(int32_t);
 452
 453 }
 454
 455 void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 456                     float *p_out, uint32_t ui_remainder,
 457                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 458 {
 459     float *Hp, *Hdp, *End;
 460     float t, temp;
 461     uint32_t ui_linear_remainder;
 462     int i;
 463
 464     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 465     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 466
 467     End = &Imp[Nwing];
 468
 469     ui_linear_remainder = (ui_remainder<<Nhc) -
 470                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 471
 472     if (Inc == 1)               /* If doing right wing...              */
 473     {                           /* ...drop extra coeff, so when Ph is  */
 474         End--;                  /*    0.5, we don't do too many mult's */
 475         if (ui_remainder == 0)  /* If the phase is zero...           */
 476         {                       /* ...then we've already skipped the */
 477             Hp += Npc;          /*    first sample, so we must also  */
 478             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 479         }
 480     }
 481
 482     while (Hp < End) {
 483         t = *Hp;                /* Get filter coeff */
 484                                 /* t is now interp'd filter coeff */
 485         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 486         for( i = 0; i < i_nb_channels; i++ )
 487         {
 488             temp = t;
 489             temp *= *(p_in+i);  /* Mult coeff by input sample */
 490             *(p_out+i) += temp; /* The filter output */
 491         }
 492         Hdp += Npc;             /* Filter coeff differences step */
 493         Hp += Npc;              /* Filter coeff step */
 494         p_in += (Inc * i_nb_channels); /* Input signal step */
 495     }
 496 }
 497
 498 void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 499                     float *p_out, uint32_t ui_remainder,
 500                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 501                     int16_t Inc, int i_nb_channels )
 502 {
 503     float *Hp, *Hdp, *End;
 504     float t, temp;
 505     uint32_t ui_linear_remainder;
 506     int i, ui_counter = 0;
 507
 508     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 509     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 510
 511     End = &Imp[Nwing];
 512
 513     if (Inc == 1)               /* If doing right wing...              */
 514     {                           /* ...drop extra coeff, so when Ph is  */
 515         End--;                  /*    0.5, we don't do too many mult's */
 516         if (ui_remainder == 0)  /* If the phase is zero...           */
 517         {                       /* ...then we've already skipped the */
 518             Hp = Imp +          /* first sample, so we must also  */
 519                   (ui_output_rate << Nhc) / ui_input_rate;
 520             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 521                   (ui_output_rate << Nhc) / ui_input_rate;
 522             ui_counter++;
 523         }
 524     }
 525
 526     while (Hp < End) {
 527         t = *Hp;                /* Get filter coeff */
 528                                 /* t is now interp'd filter coeff */
 529         ui_linear_remainder =
 530           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 531           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 532           ui_input_rate * ui_input_rate;
 533         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 534         for( i = 0; i < i_nb_channels; i++ )
 535         {
 536             temp = t;
 537             temp *= *(p_in+i);  /* Mult coeff by input sample */
 538             *(p_out+i) += temp; /* The filter output */
 539         }
 540
 541         ui_counter++;
 542
 543         /* Filter coeff step */
 544         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 545                     / ui_input_rate;
 546         /* Filter coeff differences step */
 547         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 548                      / ui_input_rate;
 549
 550         p_in += (Inc * i_nb_channels); /* Input signal step */
 551     }
 552 }