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 1.3 2003/03/04 22:08:33 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->pf_do_work = DoWork;
 132
 133     /* We don't want a new buffer to be created because we're not sure we'll
 134      * actually need to resample anything. */
 135     p_filter->b_in_place = VLC_TRUE;
 136
 137     return VLC_SUCCESS;
 138 }
 139
 140 /*****************************************************************************
 141  * Close: free our resources
 142  *****************************************************************************/
 143 static void Close( vlc_object_t * p_this )
 144 {
 145     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 146     free( p_filter->p_sys->p_buf );
 147     free( p_filter->p_sys );
 148 }
 149
 150 /*****************************************************************************
 151  * DoWork: convert a buffer
 152  *****************************************************************************/
 153 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 154                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 155 {
 156     float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
 157
 158     int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
 159     int i_in_nb = p_in_buf->i_nb_samples;
 160     int i_in, i_out = 0;
 161     double d_factor, d_scale_factor, d_old_scale_factor;
 162     int i_filter_wing;
 163 #if 0
 164     int i;
 165 #endif
 166
 167     /* Check if we really need to run the resampler */
 168     if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
 169     {
 170         if( p_filter->b_continuity &&
 171             p_in_buf->i_size >=
 172               p_in_buf->i_nb_bytes + sizeof(float) * i_nb_channels )
 173         {
 174             if( p_filter->p_sys->i_old_wing )
 175             {
 176                 /* output the whole thing with the samples from last time */
 177                 memmove( ((float *)(p_in_buf->p_buffer)) +
 178                          i_nb_channels * p_filter->p_sys->i_old_wing,
 179                          p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
 180                 memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
 181                         i_nb_channels * p_filter->p_sys->i_old_wing,
 182                         p_filter->p_sys->i_old_wing *
 183                         p_filter->input.i_bytes_per_frame );
 184
 185                 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
 186                     p_filter->p_sys->i_old_wing;
 187
 188                 p_out_buf->end_date =
 189                     aout_DateIncrement( &p_filter->p_sys->end_date,
 190                                         p_out_buf->i_nb_samples );
 191
 192                 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 193                     p_filter->input.i_bytes_per_frame;
 194             }
 195         }
 196         p_filter->b_continuity = VLC_FALSE;
 197         return;
 198     }
 199
 200     if( !p_filter->b_continuity )
 201     {
 202         /* Continuity in sound samples has been broken, we'd better reset
 203          * everything. */
 204         p_filter->b_continuity = VLC_TRUE;
 205         p_filter->p_sys->i_remainder = 0;
 206         aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
 207         aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
 208         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 209         p_filter->p_sys->d_old_factor = 1;
 210         p_filter->p_sys->i_old_wing   = 0;
 211     }
 212
 213 #if 0
 214     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 215              p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
 216              p_filter->p_sys->i_old_wing, i_in_nb );
 217 #endif
 218
 219     /* Prepare the source buffer */
 220     i_in_nb += (p_filter->p_sys->i_old_wing * 2);
 221 #ifdef HAVE_ALLOCA
 222     p_in = p_in_orig = (float *)alloca( i_in_nb *
 223                                         p_filter->input.i_bytes_per_frame );
 224 #else
 225     p_in = p_in_orig = (float *)malloc( i_in_nb *
 226                                         p_filter->input.i_bytes_per_frame );
 227 #endif
 228     if( p_in == NULL )
 229     {
 230         return;
 231     }
 232
 233     /* Copy all our samples in p_in */
 234     if( p_filter->p_sys->i_old_wing )
 235     {
 236         p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
 237                                   p_filter->p_sys->i_old_wing * 2 *
 238                                   p_filter->input.i_bytes_per_frame );
 239     }
 240     p_aout->p_vlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
 241                               i_nb_channels, p_in_buf->p_buffer,
 242                               p_in_buf->i_nb_samples *
 243                               p_filter->input.i_bytes_per_frame );
 244
 245     /* Make sure the output buffer is reset */
 246     memset( p_out, 0, p_out_buf->i_size );
 247
 248     /* Calculate the new length of the filter wing */
 249     d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
 250     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 251
 252     /* Account for increased filter gain when using factors less than 1 */
 253     d_old_scale_factor = SMALL_FILTER_SCALE *
 254         p_filter->p_sys->d_old_factor + 0.5;
 255     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 256
 257     /* Apply the old rate until we have enough samples for the new one */
 258     /* TODO: Check we have enough samples */
 259     i_in = p_filter->p_sys->i_old_wing;
 260     p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
 261     for( ; i_in < i_filter_wing; i_in++ )
 262     {
 263         if( p_filter->p_sys->d_old_factor == 1 )
 264         {
 265             /* Just copy the samples */
 266             memcpy( p_out_buf->p_buffer, p_in,
 267                     p_filter->input.i_bytes_per_frame );
 268             p_in += i_nb_channels;
 269             p_out += i_nb_channels;
 270             i_out++;
 271             continue;
 272         }
 273
 274         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 275         {
 276
 277             if( p_filter->p_sys->d_old_factor >= 1 )
 278             {
 279                 /* FilterFloatUP() is faster if we can use it */
 280
 281                 /* Perform left-wing inner product */
 282                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 283                                SMALL_FILTER_NWING, p_in, p_out,
 284                                p_filter->p_sys->i_remainder,
 285                                p_filter->output.i_rate,
 286                                -1, i_nb_channels );
 287                 /* Perform right-wing inner product */
 288                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 289                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 290                                p_filter->output.i_rate -
 291                                p_filter->p_sys->i_remainder,
 292                                p_filter->output.i_rate,
 293                                1, i_nb_channels );
 294
 295 #if 0
 296                 /* Normalize for unity filter gain */
 297                 for( i = 0; i < i_nb_channels; i++ )
 298                 {
 299                     *(p_out+i) *= d_old_scale_factor;
 300                 }
 301 #endif
 302             }
 303             else
 304             {
 305                 /* Perform left-wing inner product */
 306                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 307                                SMALL_FILTER_NWING, p_in, p_out,
 308                                p_filter->p_sys->i_remainder,
 309                                p_filter->output.i_rate, p_filter->input.i_rate,
 310                                -1, i_nb_channels );
 311                 /* Perform right-wing inner product */
 312                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 313                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 314                                p_filter->output.i_rate -
 315                                p_filter->p_sys->i_remainder,
 316                                p_filter->output.i_rate, p_filter->input.i_rate,
 317                                1, i_nb_channels );
 318             }
 319
 320             p_out += i_nb_channels;
 321             i_out++;
 322
 323             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 324         }
 325
 326         p_in += i_nb_channels;
 327         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 328     }
 329
 330     /* Apply the new rate for the rest of the samples */
 331     /* TODO: Check we have enough future samples for the new rate */
 332     if( i_in < i_in_nb - i_filter_wing )
 333     {
 334         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 335         p_filter->p_sys->d_old_factor = d_factor;
 336         p_filter->p_sys->i_old_wing   = i_filter_wing;
 337     }
 338     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 339     {
 340         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 341         {
 342
 343             if( d_factor >= 1 )
 344             {
 345                 /* FilterFloatUP() is faster if we can use it */
 346
 347                 /* Perform left-wing inner product */
 348                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 349                                SMALL_FILTER_NWING, p_in, p_out,
 350                                p_filter->p_sys->i_remainder,
 351                                p_filter->output.i_rate,
 352                                -1, i_nb_channels );
 353
 354                 /* Perform right-wing inner product */
 355                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 356                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 357                                p_filter->output.i_rate -
 358                                p_filter->p_sys->i_remainder,
 359                                p_filter->output.i_rate,
 360                                1, i_nb_channels );
 361
 362 #if 0
 363                 /* Normalize for unity filter gain */
 364                 for( i = 0; i < i_nb_channels; i++ )
 365                 {
 366                     *(p_out+i) *= d_old_scale_factor;
 367                 }
 368 #endif
 369             }
 370             else
 371             {
 372                 /* Perform left-wing inner product */
 373                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 374                                SMALL_FILTER_NWING, p_in, p_out,
 375                                p_filter->p_sys->i_remainder,
 376                                p_filter->output.i_rate, p_filter->input.i_rate,
 377                                -1, i_nb_channels );
 378                 /* Perform right-wing inner product */
 379                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 380                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 381                                p_filter->output.i_rate -
 382                                p_filter->p_sys->i_remainder,
 383                                p_filter->output.i_rate, p_filter->input.i_rate,
 384                                1, i_nb_channels );
 385             }
 386
 387             p_out += i_nb_channels;
 388             i_out++;
 389
 390             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 391         }
 392
 393         p_in += i_nb_channels;
 394         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 395     }
 396
 397     /* Buffer i_filter_wing * 2 samples for next time */
 398     if( p_filter->p_sys->i_old_wing )
 399     {
 400         memcpy( p_filter->p_sys->p_buf,
 401                 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
 402                 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
 403                 p_filter->input.i_bytes_per_frame );
 404     }
 405
 406 #if 0
 407     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
 408              i_out * p_filter->input.i_bytes_per_frame );
 409 #endif
 410
 411     /* Free the temp buffer */
 412 #ifndef HAVE_ALLOCA
 413     free( p_in_orig );
 414 #endif
 415
 416     /* Finalize aout buffer */
 417     p_out_buf->i_nb_samples = i_out;
 418     p_out_buf->start_date = p_in_buf->start_date;
 419
 420     p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
 421                                               p_out_buf->i_nb_samples );
 422
 423     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 424         i_nb_channels * sizeof(int32_t);
 425
 426 }
 427
 428 void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 429                     float *p_out, uint32_t ui_remainder,
 430                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 431 {
 432     float *Hp, *Hdp, *End;
 433     float t, temp;
 434     uint32_t ui_linear_remainder;
 435     int i;
 436
 437     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 438     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 439
 440     End = &Imp[Nwing];
 441
 442     ui_linear_remainder = (ui_remainder<<Nhc) -
 443                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 444
 445     if (Inc == 1)               /* If doing right wing...              */
 446     {                           /* ...drop extra coeff, so when Ph is  */
 447         End--;                  /*    0.5, we don't do too many mult's */
 448         if (ui_remainder == 0)  /* If the phase is zero...           */
 449         {                       /* ...then we've already skipped the */
 450             Hp += Npc;          /*    first sample, so we must also  */
 451             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 452         }
 453     }
 454
 455     while (Hp < End) {
 456         t = *Hp;                /* Get filter coeff */
 457                                 /* t is now interp'd filter coeff */
 458         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 459         for( i = 0; i < i_nb_channels; i++ )
 460         {
 461             temp = t;
 462             temp *= *(p_in+i);  /* Mult coeff by input sample */
 463             *(p_out+i) += temp; /* The filter output */
 464         }
 465         Hdp += Npc;             /* Filter coeff differences step */
 466         Hp += Npc;              /* Filter coeff step */
 467         p_in += (Inc * i_nb_channels); /* Input signal step */
 468     }
 469 }
 470
 471 void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 472                     float *p_out, uint32_t ui_remainder,
 473                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 474                     int16_t Inc, int i_nb_channels )
 475 {
 476     float *Hp, *Hdp, *End;
 477     float t, temp;
 478     uint32_t ui_linear_remainder;
 479     int i, ui_counter = 0;
 480
 481     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 482     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 483
 484     End = &Imp[Nwing];
 485
 486     if (Inc == 1)               /* If doing right wing...              */
 487     {                           /* ...drop extra coeff, so when Ph is  */
 488         End--;                  /*    0.5, we don't do too many mult's */
 489         if (ui_remainder == 0)  /* If the phase is zero...           */
 490         {                       /* ...then we've already skipped the */
 491             Hp = Imp +          /* first sample, so we must also  */
 492                   (ui_output_rate << Nhc) / ui_input_rate;
 493             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 494                   (ui_output_rate << Nhc) / ui_input_rate;
 495             ui_counter++;
 496         }
 497     }
 498
 499     while (Hp < End) {
 500         t = *Hp;                /* Get filter coeff */
 501                                 /* t is now interp'd filter coeff */
 502         ui_linear_remainder =
 503           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 504           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 505           ui_input_rate * ui_input_rate;
 506         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 507         for( i = 0; i < i_nb_channels; i++ )
 508         {
 509             temp = t;
 510             temp *= *(p_in+i);  /* Mult coeff by input sample */
 511             *(p_out+i) += temp; /* The filter output */
 512         }
 513
 514         ui_counter++;
 515
 516         /* Filter coeff step */
 517         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 518                     / ui_input_rate;
 519         /* Filter coeff differences step */
 520         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 521                      / ui_input_rate;
 522
 523         p_in += (Inc * i_nb_channels); /* Input signal step */
 524     }
 525 }