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