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 VideoLAN
   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., 59 Temple Place - Suite 330, Boston, MA  02111, 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 <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_category( CAT_AUDIO );
  83     set_subcategory( SUBCAT_AUDIO_MISC );
  84     set_description( _("audio filter for band-limited interpolation resampling") );
  85     set_capability( "audio filter", 20 );
  86     set_callbacks( Create, Close );
  87 vlc_module_end();
  88
  89 /*****************************************************************************
  90  * Create: allocate linear resampler
  91  *****************************************************************************/
  92 static int Create( vlc_object_t *p_this )
  93 {
  94     aout_filter_t * p_filter = (aout_filter_t *)p_this;
  95     double d_factor;
  96     int i_filter_wing;
  97
  98     if ( p_filter->input.i_rate == p_filter->output.i_rate
  99           || p_filter->input.i_format != p_filter->output.i_format
 100           || p_filter->input.i_physical_channels
 101               != p_filter->output.i_physical_channels
 102           || p_filter->input.i_original_channels
 103               != p_filter->output.i_original_channels
 104           || p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
 105     {
 106         return VLC_EGENERIC;
 107     }
 108
 109 #if !defined( SYS_DARWIN )
 110     if( !config_GetInt( p_this, "hq-resampling" ) )
 111     {
 112         return VLC_EGENERIC;
 113     }
 114 #endif
 115
 116     /* Allocate the memory needed to store the module's structure */
 117     p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
 118     if( p_filter->p_sys == NULL )
 119     {
 120         msg_Err( p_filter, "out of memory" );
 121         return VLC_ENOMEM;
 122     }
 123
 124     /* Calculate worst case for the length of the filter wing */
 125     d_factor = (double)p_filter->output.i_rate
 126                         / p_filter->input.i_rate;
 127     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 128                       * __MAX(1.0, 1.0/d_factor) + 10;
 129     p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
 130         sizeof(int32_t) * 2 * i_filter_wing;
 131
 132     /* Allocate enough memory to buffer previous samples */
 133     p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
 134     if( p_filter->p_sys->p_buf == NULL )
 135     {
 136         msg_Err( p_filter, "out of memory" );
 137         return VLC_ENOMEM;
 138     }
 139
 140     p_filter->p_sys->i_old_wing = 0;
 141     p_filter->pf_do_work = DoWork;
 142
 143     /* We don't want a new buffer to be created because we're not sure we'll
 144      * actually need to resample anything. */
 145     p_filter->b_in_place = VLC_TRUE;
 146
 147     return VLC_SUCCESS;
 148 }
 149
 150 /*****************************************************************************
 151  * Close: free our resources
 152  *****************************************************************************/
 153 static void Close( vlc_object_t * p_this )
 154 {
 155     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 156     free( p_filter->p_sys->p_buf );
 157     free( p_filter->p_sys );
 158 }
 159
 160 /*****************************************************************************
 161  * DoWork: convert a buffer
 162  *****************************************************************************/
 163 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 164                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 165 {
 166     float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
 167
 168     int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
 169     int i_in_nb = p_in_buf->i_nb_samples;
 170     int i_in, i_out = 0;
 171     double d_factor, d_scale_factor, d_old_scale_factor;
 172     int i_filter_wing;
 173 #if 0
 174     int i;
 175 #endif
 176
 177     /* Check if we really need to run the resampler */
 178     if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
 179     {
 180         if( //p_filter->b_continuity && /* What difference does it make ? :) */
 181             p_filter->p_sys->i_old_wing &&
 182             p_in_buf->i_size >=
 183               p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
 184               p_filter->input.i_bytes_per_frame )
 185         {
 186             /* output the whole thing with the samples from last time */
 187             memmove( ((float *)(p_in_buf->p_buffer)) +
 188                      i_nb_channels * p_filter->p_sys->i_old_wing,
 189                      p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
 190             memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
 191                     i_nb_channels * p_filter->p_sys->i_old_wing,
 192                     p_filter->p_sys->i_old_wing *
 193                     p_filter->input.i_bytes_per_frame );
 194
 195             p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
 196                 p_filter->p_sys->i_old_wing;
 197
 198             p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
 199             p_out_buf->end_date =
 200                 aout_DateIncrement( &p_filter->p_sys->end_date,
 201                                     p_out_buf->i_nb_samples );
 202
 203             p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 204                 p_filter->input.i_bytes_per_frame;
 205         }
 206         p_filter->b_continuity = VLC_FALSE;
 207         p_filter->p_sys->i_old_wing = 0;
 208         return;
 209     }
 210
 211     if( !p_filter->b_continuity )
 212     {
 213         /* Continuity in sound samples has been broken, we'd better reset
 214          * everything. */
 215         p_filter->b_continuity = VLC_TRUE;
 216         p_filter->p_sys->i_remainder = 0;
 217         aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
 218         aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
 219         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 220         p_filter->p_sys->d_old_factor = 1;
 221         p_filter->p_sys->i_old_wing   = 0;
 222     }
 223
 224 #if 0
 225     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 226              p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
 227              p_filter->p_sys->i_old_wing, i_in_nb );
 228 #endif
 229
 230     /* Prepare the source buffer */
 231     i_in_nb += (p_filter->p_sys->i_old_wing * 2);
 232 #ifdef HAVE_ALLOCA
 233     p_in = p_in_orig = (float *)alloca( i_in_nb *
 234                                         p_filter->input.i_bytes_per_frame );
 235 #else
 236     p_in = p_in_orig = (float *)malloc( i_in_nb *
 237                                         p_filter->input.i_bytes_per_frame );
 238 #endif
 239     if( p_in == NULL )
 240     {
 241         return;
 242     }
 243
 244     /* Copy all our samples in p_in */
 245     if( p_filter->p_sys->i_old_wing )
 246     {
 247         p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
 248                                   p_filter->p_sys->i_old_wing * 2 *
 249                                   p_filter->input.i_bytes_per_frame );
 250     }
 251     p_aout->p_vlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
 252                               i_nb_channels, p_in_buf->p_buffer,
 253                               p_in_buf->i_nb_samples *
 254                               p_filter->input.i_bytes_per_frame );
 255
 256     /* Make sure the output buffer is reset */
 257     memset( p_out, 0, p_out_buf->i_size );
 258
 259     /* Calculate the new length of the filter wing */
 260     d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
 261     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 262
 263     /* Account for increased filter gain when using factors less than 1 */
 264     d_old_scale_factor = SMALL_FILTER_SCALE *
 265         p_filter->p_sys->d_old_factor + 0.5;
 266     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 267
 268     /* Apply the old rate until we have enough samples for the new one */
 269     i_in = p_filter->p_sys->i_old_wing;
 270     p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
 271     for( ; i_in < i_filter_wing &&
 272            (i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
 273     {
 274         if( p_filter->p_sys->d_old_factor == 1 )
 275         {
 276             /* Just copy the samples */
 277             memcpy( p_out, p_in,
 278                     p_filter->input.i_bytes_per_frame );
 279             p_in += i_nb_channels;
 280             p_out += i_nb_channels;
 281             i_out++;
 282             continue;
 283         }
 284
 285         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 286         {
 287
 288             if( p_filter->p_sys->d_old_factor >= 1 )
 289             {
 290                 /* FilterFloatUP() is faster if we can use it */
 291
 292                 /* Perform left-wing inner product */
 293                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 294                                SMALL_FILTER_NWING, p_in, p_out,
 295                                p_filter->p_sys->i_remainder,
 296                                p_filter->output.i_rate,
 297                                -1, i_nb_channels );
 298                 /* Perform right-wing inner product */
 299                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 300                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 301                                p_filter->output.i_rate -
 302                                p_filter->p_sys->i_remainder,
 303                                p_filter->output.i_rate,
 304                                1, i_nb_channels );
 305
 306 #if 0
 307                 /* Normalize for unity filter gain */
 308                 for( i = 0; i < i_nb_channels; i++ )
 309                 {
 310                     *(p_out+i) *= d_old_scale_factor;
 311                 }
 312 #endif
 313
 314                 /* Sanity check */
 315                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 316                     <= (unsigned int)i_out+1 )
 317                 {
 318                     p_out += i_nb_channels;
 319                     i_out++;
 320                     p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 321                     break;
 322                 }
 323             }
 324             else
 325             {
 326                 /* Perform left-wing inner product */
 327                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 328                                SMALL_FILTER_NWING, p_in, p_out,
 329                                p_filter->p_sys->i_remainder,
 330                                p_filter->output.i_rate, p_filter->input.i_rate,
 331                                -1, i_nb_channels );
 332                 /* Perform right-wing inner product */
 333                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 334                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 335                                p_filter->output.i_rate -
 336                                p_filter->p_sys->i_remainder,
 337                                p_filter->output.i_rate, p_filter->input.i_rate,
 338                                1, i_nb_channels );
 339             }
 340
 341             p_out += i_nb_channels;
 342             i_out++;
 343
 344             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 345         }
 346
 347         p_in += i_nb_channels;
 348         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 349     }
 350
 351     /* Apply the new rate for the rest of the samples */
 352     if( i_in < i_in_nb - i_filter_wing )
 353     {
 354         p_filter->p_sys->i_old_rate   = p_filter->input.i_rate;
 355         p_filter->p_sys->d_old_factor = d_factor;
 356         p_filter->p_sys->i_old_wing   = i_filter_wing;
 357     }
 358     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 359     {
 360         while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
 361         {
 362
 363             if( d_factor >= 1 )
 364             {
 365                 /* FilterFloatUP() is faster if we can use it */
 366
 367                 /* Perform left-wing inner product */
 368                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 369                                SMALL_FILTER_NWING, p_in, p_out,
 370                                p_filter->p_sys->i_remainder,
 371                                p_filter->output.i_rate,
 372                                -1, i_nb_channels );
 373
 374                 /* Perform right-wing inner product */
 375                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 376                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 377                                p_filter->output.i_rate -
 378                                p_filter->p_sys->i_remainder,
 379                                p_filter->output.i_rate,
 380                                1, i_nb_channels );
 381
 382 #if 0
 383                 /* Normalize for unity filter gain */
 384                 for( i = 0; i < i_nb_channels; i++ )
 385                 {
 386                     *(p_out+i) *= d_old_scale_factor;
 387                 }
 388 #endif
 389                 /* Sanity check */
 390                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 391                     <= (unsigned int)i_out+1 )
 392                 {
 393                     p_out += i_nb_channels;
 394                     i_out++;
 395                     p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 396                     break;
 397                 }
 398             }
 399             else
 400             {
 401                 /* Perform left-wing inner product */
 402                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 403                                SMALL_FILTER_NWING, p_in, p_out,
 404                                p_filter->p_sys->i_remainder,
 405                                p_filter->output.i_rate, p_filter->input.i_rate,
 406                                -1, i_nb_channels );
 407                 /* Perform right-wing inner product */
 408                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 409                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 410                                p_filter->output.i_rate -
 411                                p_filter->p_sys->i_remainder,
 412                                p_filter->output.i_rate, p_filter->input.i_rate,
 413                                1, i_nb_channels );
 414             }
 415
 416             p_out += i_nb_channels;
 417             i_out++;
 418
 419             p_filter->p_sys->i_remainder += p_filter->input.i_rate;
 420         }
 421
 422         p_in += i_nb_channels;
 423         p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
 424     }
 425
 426     /* Buffer i_filter_wing * 2 samples for next time */
 427     if( p_filter->p_sys->i_old_wing )
 428     {
 429         memcpy( p_filter->p_sys->p_buf,
 430                 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
 431                 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
 432                 p_filter->input.i_bytes_per_frame );
 433     }
 434
 435 #if 0
 436     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
 437              i_out * p_filter->input.i_bytes_per_frame );
 438 #endif
 439
 440     /* Free the temp buffer */
 441 #ifndef HAVE_ALLOCA
 442     free( p_in_orig );
 443 #endif
 444
 445     /* Finalize aout buffer */
 446     p_out_buf->i_nb_samples = i_out;
 447     p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
 448     p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
 449                                               p_out_buf->i_nb_samples );
 450
 451     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 452         i_nb_channels * sizeof(int32_t);
 453
 454 }
 455
 456 void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 457                     float *p_out, uint32_t ui_remainder,
 458                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 459 {
 460     float *Hp, *Hdp, *End;
 461     float t, temp;
 462     uint32_t ui_linear_remainder;
 463     int i;
 464
 465     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 466     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 467
 468     End = &Imp[Nwing];
 469
 470     ui_linear_remainder = (ui_remainder<<Nhc) -
 471                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 472
 473     if (Inc == 1)               /* If doing right wing...              */
 474     {                           /* ...drop extra coeff, so when Ph is  */
 475         End--;                  /*    0.5, we don't do too many mult's */
 476         if (ui_remainder == 0)  /* If the phase is zero...           */
 477         {                       /* ...then we've already skipped the */
 478             Hp += Npc;          /*    first sample, so we must also  */
 479             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 480         }
 481     }
 482
 483     while (Hp < End) {
 484         t = *Hp;                /* Get filter coeff */
 485                                 /* t is now interp'd filter coeff */
 486         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 487         for( i = 0; i < i_nb_channels; i++ )
 488         {
 489             temp = t;
 490             temp *= *(p_in+i);  /* Mult coeff by input sample */
 491             *(p_out+i) += temp; /* The filter output */
 492         }
 493         Hdp += Npc;             /* Filter coeff differences step */
 494         Hp += Npc;              /* Filter coeff step */
 495         p_in += (Inc * i_nb_channels); /* Input signal step */
 496     }
 497 }
 498
 499 void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
 500                     float *p_out, uint32_t ui_remainder,
 501                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 502                     int16_t Inc, int i_nb_channels )
 503 {
 504     float *Hp, *Hdp, *End;
 505     float t, temp;
 506     uint32_t ui_linear_remainder;
 507     int i, ui_counter = 0;
 508
 509     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 510     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 511
 512     End = &Imp[Nwing];
 513
 514     if (Inc == 1)               /* If doing right wing...              */
 515     {                           /* ...drop extra coeff, so when Ph is  */
 516         End--;                  /*    0.5, we don't do too many mult's */
 517         if (ui_remainder == 0)  /* If the phase is zero...           */
 518         {                       /* ...then we've already skipped the */
 519             Hp = Imp +          /* first sample, so we must also  */
 520                   (ui_output_rate << Nhc) / ui_input_rate;
 521             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 522                   (ui_output_rate << Nhc) / ui_input_rate;
 523             ui_counter++;
 524         }
 525     }
 526
 527     while (Hp < End) {
 528         t = *Hp;                /* Get filter coeff */
 529                                 /* t is now interp'd filter coeff */
 530         ui_linear_remainder =
 531           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 532           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 533           ui_input_rate * ui_input_rate;
 534         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 535         for( i = 0; i < i_nb_channels; i++ )
 536         {
 537             temp = t;
 538             temp *= *(p_in+i);  /* Mult coeff by input sample */
 539             *(p_out+i) += temp; /* The filter output */
 540         }
 541
 542         ui_counter++;
 543
 544         /* Filter coeff step */
 545         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 546                     / ui_input_rate;
 547         /* Filter coeff differences step */
 548         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 549                      / ui_input_rate;
 550
 551         p_in += (Inc * i_nb_channels); /* Input signal step */
 552     }
 553 }