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

   1 /*****************************************************************************
   2  * bandlimited.c : bandlimited interpolation resampler
   3  *****************************************************************************
   4  * Copyright (C) 2002 VideoLAN
   5  * $Id: bandlimited.c,v 1.5 2003/03/05 22:37:05 gbazin Exp $
   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., 59 Temple Place - Suite 330, Boston, MA  02111, USA.
  22  *****************************************************************************/
  23
  24 /*****************************************************************************
  25  * Preamble:
  26  *
  27  * This implementation of the bandlimited 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 <stdlib.h>                                      /* malloc(), free() */
  36 #include <string.h>
  37
  38 #include <vlc/vlc.h>
  39 #include "audio_output.h"
  40 #include "aout_internal.h"
  41 #include "bandlimited.h"
  42
  43 /*****************************************************************************
  44  * Local prototypes
  45  *****************************************************************************/
  46 static int  Create    ( vlc_object_t * );
  47 static void Close     ( vlc_object_t * );
  48 static void DoWork    ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
  49                         aout_buffer_t * );
  50
  51 static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing,
  52                            float *f_in, float *f_out, uint32_t ui_remainder,
  53                            uint32_t ui_output_rate, int16_t Inc,
  54                            int i_nb_channels );
  55
  56 static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing,
  57                            float *f_in, float *f_out, uint32_t ui_remainder,
  58                            uint32_t ui_output_rate, uint32_t ui_input_rate,
  59                            int16_t Inc, int i_nb_channels );
  60
  61 /*****************************************************************************
  62  * Local structures
  63  *****************************************************************************/
  64 struct aout_filter_sys_t
  65 {
  66     int32_t *p_buf;                        /* this filter introduces a delay */
  67     int i_buf_size;
  68
  69     int i_old_rate;
  70     double d_old_factor;
  71     int i_old_wing;
  72
  73     unsigned int i_remainder;                /* remainder of previous sample */
  74
  75     audio_date_t end_date;
  76 };
  77
  78 /*****************************************************************************
  79  * Module descriptor
  80  *****************************************************************************/
  81 vlc_module_begin();
  82     set_description( _("audio filter for bandlimited 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     /* Allocate the memory needed to store the module's structure */
 108     p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
 109     if( p_filter->p_sys == NULL )
 110     {
 111         msg_Err( p_filter, "out of memory" );
 112         return VLC_ENOMEM;
 113     }
 114
 115     /* Calculate worst case for the length of the filter wing */
 116     d_factor = (double)p_filter->output.i_rate
 117                         / p_filter->input.i_rate;
 118     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 119                       * __MAX(1.0, 1.0/d_factor) + 10;
 120     p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
 121         sizeof(int32_t) * 2 * i_filter_wing;
 122
 123     /* Allocate enough memory to buffer previous samples */
 124     p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
 125     if( p_filter->p_sys->p_buf == NULL )
 126     {
 127         msg_Err( p_filter, "out of memory" );
 128         return VLC_ENOMEM;
 129     }
 130
 131     p_filter->p_sys->i_old_wing = 0;
 132     p_filter->pf_do_work = DoWork;
 133
 134     /* We don't want a new buffer to be created because we're not sure we'll
 135      * actually need to resample anything. */
 136     p_filter->b_in_place = VLC_TRUE;
 137
 138     return VLC_SUCCESS;
 139 }
 140
 141 /*****************************************************************************
 142  * Close: free our resources
 143  *****************************************************************************/
 144 static void Close( vlc_object_t * p_this )
 145 {
 146     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 147     free( p_filter->p_sys->p_buf );
 148     free( p_filter->p_sys );
 149 }
 150
 151 /*****************************************************************************
 152  * DoWork: convert a buffer
 153  *****************************************************************************/
 154 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 155                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 156 {
 157     float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
 158
 159     int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
 160     int i_in_nb = p_in_buf->i_nb_samples;
 161     int i_in, i_out = 0;
 162     double d_factor, d_scale_factor, d_old_scale_factor;
 163     int i_filter_wing;
 164 #if 0
 165     int i;
 166 #endif
 167
 168     /* Check if we really need to run the resampler */
 169     if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
 170     {
 171         if( //p_filter->b_continuity && /* What difference does it make ? :) */
 172             p_filter->p_sys->i_old_wing &&
 173             p_in_buf->i_size >=
 174               p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
 175               p_filter->input.i_bytes_per_frame )
 176         {
 177             /* output the whole thing with the samples from last time */
 178             memmove( ((float *)(p_in_buf->p_buffer)) +
 179                      i_nb_channels * p_filter->p_sys->i_old_wing,
 180                      p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
 181             memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
 182                     i_nb_channels * p_filter->p_sys->i_old_wing,
 183                     p_filter->p_sys->i_old_wing *
 184                     p_filter->input.i_bytes_per_frame );
 185
 186             p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
 187                 p_filter->p_sys->i_old_wing;
 188
 189             p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
 190             p_out_buf->end_date =
 191                 aout_DateIncrement( &p_filter->p_sys->end_date,
 192                                     p_out_buf->i_nb_samples );
 193
 194             p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 195                 p_filter->input.i_bytes_per_frame;
 196         }
 197         p_filter->b_continuity = VLC_FALSE;
 198         p_filter->p_sys->i_old_wing = 0;
 199         return;
 200     }
 201
 202     if( !p_filter->b_continuity )
 203     {
 204         /* Continuity in sound samples has been broken, we'd better reset
 205          * everything. */
 206         p_filter->b_continuity = VLC_TRUE;
 207         p_filter->p_sys->i_remainder = 0;
 208         aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
 209         aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
 210         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 211         p_filter->p_sys->d_old_factor = 1;
 212         p_filter->p_sys->i_old_wing   = 0;
 213     }
 214
 215 #if 0
 216     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 217              p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
 218              p_filter->p_sys->i_old_wing, i_in_nb );
 219 #endif
 220
 221     /* Prepare the source buffer */
 222     i_in_nb += (p_filter->p_sys->i_old_wing * 2);
 223 #ifdef HAVE_ALLOCA
 224     p_in = p_in_orig = (float *)alloca( i_in_nb *
 225                                         p_filter->input.i_bytes_per_frame );
 226 #else
 227     p_in = p_in_orig = (float *)malloc( i_in_nb *
 228                                         p_filter->input.i_bytes_per_frame );
 229 #endif
 230     if( p_in == NULL )
 231     {
 232         return;
 233     }
 234
 235     /* Copy all our samples in p_in */
 236     if( p_filter->p_sys->i_old_wing )
 237     {
 238         p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
 239                                   p_filter->p_sys->i_old_wing * 2 *
 240                                   p_filter->input.i_bytes_per_frame );
 241     }
 242     p_aout->p_vlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
 243                               i_nb_channels, p_in_buf->p_buffer,
 244                               p_in_buf->i_nb_samples *
 245                               p_filter->input.i_bytes_per_frame );
 246
 247     /* Make sure the output buffer is reset */
 248     memset( p_out, 0, p_out_buf->i_size );
 249
 250     /* Calculate the new length of the filter wing */
 251     d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
 252     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 253
 254     /* Account for increased filter gain when using factors less than 1 */
 255     d_old_scale_factor = SMALL_FILTER_SCALE *
 256         p_filter->p_sys->d_old_factor + 0.5;
 257     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 258
 259     /* Apply the old rate until we have enough samples for the new one */
 260     i_in = p_filter->p_sys->i_old_wing;
 261     p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
 262     for( ; i_in < i_filter_wing &&
 263            (i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
 264     {
 265         if( p_filter->p_sys->d_old_factor == 1 )
 266         {
 267             /* Just copy the samples */
 268             memcpy( p_out, p_in,
 269                     p_filter->input.i_bytes_per_frame );
 270             p_in += i_nb_channels;
 271             p_out += i_nb_channels;
 272             i_out++;
 273             continue;
 274         }
 275
 276         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 277         {
 278
 279             if( p_filter->p_sys->d_old_factor >= 1 )
 280             {
 281                 /* FilterFloatUP() is faster if we can use it */
 282
 283                 /* Perform left-wing inner product */
 284                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 285                                SMALL_FILTER_NWING, p_in, p_out,
 286                                p_filter->p_sys->i_remainder,
 287                                p_filter->output.i_rate,
 288                                -1, i_nb_channels );
 289                 /* Perform right-wing inner product */
 290                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 291                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 292                                p_filter->output.i_rate -
 293                                p_filter->p_sys->i_remainder,
 294                                p_filter->output.i_rate,
 295                                1, i_nb_channels );
 296
 297 #if 0
 298                 /* Normalize for unity filter gain */
 299                 for( i = 0; i < i_nb_channels; i++ )
 300                 {
 301                     *(p_out+i) *= d_old_scale_factor;
 302                 }
 303 #endif
 304
 305                 /* Sanity check */
 306                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 307                     <= (unsigned int)i_out+1 )
 308                 {
 309                     p_out += i_nb_channels;
 310                     i_out++;
 311                     p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 312                     break;
 313                 }
 314             }
 315             else
 316             {
 317                 /* Perform left-wing inner product */
 318                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 319                                SMALL_FILTER_NWING, p_in, p_out,
 320                                p_filter->p_sys->i_remainder,
 321                                p_filter->output.i_rate, p_filter->input.i_rate,
 322                                -1, i_nb_channels );
 323                 /* Perform right-wing inner product */
 324                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 325                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 326                                p_filter->output.i_rate -
 327                                p_filter->p_sys->i_remainder,
 328                                p_filter->output.i_rate, p_filter->input.i_rate,
 329                                1, i_nb_channels );
 330             }
 331
 332             p_out += i_nb_channels;
 333             i_out++;
 334
 335             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 336         }
 337
 338         p_in += i_nb_channels;
 339         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 340     }
 341
 342     /* Apply the new rate for the rest of the samples */
 343     if( i_in < i_in_nb - i_filter_wing )
 344     {
 345         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 346         p_filter->p_sys->d_old_factor = d_factor;
 347         p_filter->p_sys->i_old_wing   = i_filter_wing;
 348     }
 349     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 350     {
 351         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 352         {
 353
 354             if( d_factor >= 1 )
 355             {
 356                 /* FilterFloatUP() is faster if we can use it */
 357
 358                 /* Perform left-wing inner product */
 359                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 360                                SMALL_FILTER_NWING, p_in, p_out,
 361                                p_filter->p_sys->i_remainder,
 362                                p_filter->output.i_rate,
 363                                -1, i_nb_channels );
 364
 365                 /* Perform right-wing inner product */
 366                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 367                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 368                                p_filter->output.i_rate -
 369                                p_filter->p_sys->i_remainder,
 370                                p_filter->output.i_rate,
 371                                1, i_nb_channels );
 372
 373 #if 0
 374                 /* Normalize for unity filter gain */
 375                 for( i = 0; i < i_nb_channels; i++ )
 376                 {
 377                     *(p_out+i) *= d_old_scale_factor;
 378                 }
 379 #endif
 380                 /* Sanity check */
 381                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 382                     <= (unsigned int)i_out+1 )
 383                 {
 384                     p_out += i_nb_channels;
 385                     i_out++;
 386                     p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 387                     break;
 388                 }
 389             }
 390             else
 391             {
 392                 /* Perform left-wing inner product */
 393                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 394                                SMALL_FILTER_NWING, p_in, p_out,
 395                                p_filter->p_sys->i_remainder,
 396                                p_filter->output.i_rate, p_filter->input.i_rate,
 397                                -1, i_nb_channels );
 398                 /* Perform right-wing inner product */
 399                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 400                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 401                                p_filter->output.i_rate -
 402                                p_filter->p_sys->i_remainder,
 403                                p_filter->output.i_rate, p_filter->input.i_rate,
 404                                1, i_nb_channels );
 405             }
 406
 407             p_out += i_nb_channels;
 408             i_out++;
 409
 410             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 411         }
 412
 413         p_in += i_nb_channels;
 414         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 415     }
 416
 417     /* Buffer i_filter_wing * 2 samples for next time */
 418     if( p_filter->p_sys->i_old_wing )
 419     {
 420         memcpy( p_filter->p_sys->p_buf,
 421                 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
 422                 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
 423                 p_filter->input.i_bytes_per_frame );
 424     }
 425
 426 #if 0
 427     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
 428              i_out * p_filter->input.i_bytes_per_frame );
 429 #endif
 430
 431     /* Free the temp buffer */
 432 #ifndef HAVE_ALLOCA
 433     free( p_in_orig );
 434 #endif
 435
 436     /* Finalize aout buffer */
 437     p_out_buf->i_nb_samples = i_out;
 438     p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
 439     p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
 440                                               p_out_buf->i_nb_samples );
 441
 442     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 443         i_nb_channels * sizeof(int32_t);
 444
 445 }
 446
 447 void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 448                     float *p_out, uint32_t ui_remainder,
 449                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 450 {
 451     float *Hp, *Hdp, *End;
 452     float t, temp;
 453     uint32_t ui_linear_remainder;
 454     int i;
 455
 456     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 457     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 458
 459     End = &Imp[Nwing];
 460
 461     ui_linear_remainder = (ui_remainder<<Nhc) -
 462                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 463
 464     if (Inc == 1)               /* If doing right wing...              */
 465     {                           /* ...drop extra coeff, so when Ph is  */
 466         End--;                  /*    0.5, we don't do too many mult's */
 467         if (ui_remainder == 0)  /* If the phase is zero...           */
 468         {                       /* ...then we've already skipped the */
 469             Hp += Npc;          /*    first sample, so we must also  */
 470             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 471         }
 472     }
 473
 474     while (Hp < End) {
 475         t = *Hp;                /* Get filter coeff */
 476                                 /* t is now interp'd filter coeff */
 477         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 478         for( i = 0; i < i_nb_channels; i++ )
 479         {
 480             temp = t;
 481             temp *= *(p_in+i);  /* Mult coeff by input sample */
 482             *(p_out+i) += temp; /* The filter output */
 483         }
 484         Hdp += Npc;             /* Filter coeff differences step */
 485         Hp += Npc;              /* Filter coeff step */
 486         p_in += (Inc * i_nb_channels); /* Input signal step */
 487     }
 488 }
 489
 490 void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 491                     float *p_out, uint32_t ui_remainder,
 492                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 493                     int16_t Inc, int i_nb_channels )
 494 {
 495     float *Hp, *Hdp, *End;
 496     float t, temp;
 497     uint32_t ui_linear_remainder;
 498     int i, ui_counter = 0;
 499
 500     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 501     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 502
 503     End = &Imp[Nwing];
 504
 505     if (Inc == 1)               /* If doing right wing...              */
 506     {                           /* ...drop extra coeff, so when Ph is  */
 507         End--;                  /*    0.5, we don't do too many mult's */
 508         if (ui_remainder == 0)  /* If the phase is zero...           */
 509         {                       /* ...then we've already skipped the */
 510             Hp = Imp +          /* first sample, so we must also  */
 511                   (ui_output_rate << Nhc) / ui_input_rate;
 512             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 513                   (ui_output_rate << Nhc) / ui_input_rate;
 514             ui_counter++;
 515         }
 516     }
 517
 518     while (Hp < End) {
 519         t = *Hp;                /* Get filter coeff */
 520                                 /* t is now interp'd filter coeff */
 521         ui_linear_remainder =
 522           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 523           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 524           ui_input_rate * ui_input_rate;
 525         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 526         for( i = 0; i < i_nb_channels; i++ )
 527         {
 528             temp = t;
 529             temp *= *(p_in+i);  /* Mult coeff by input sample */
 530             *(p_out+i) += temp; /* The filter output */
 531         }
 532
 533         ui_counter++;
 534
 535         /* Filter coeff step */
 536         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 537                     / ui_input_rate;
 538         /* Filter coeff differences step */
 539         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 540                      / ui_input_rate;
 541
 542         p_in += (Inc * i_nb_channels); /* Input signal step */
 543     }
 544 }