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