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 #include <vlc_filter.h>
  44 #include <vlc_block.h>
  45
  46 #include "bandlimited.h"
  47
  48 /*****************************************************************************
  49  * Local prototypes
  50  *****************************************************************************/
  51 static int  Create    ( vlc_object_t * );
  52 static void Close     ( vlc_object_t * );
  53 static void DoWork    ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
  54                         aout_buffer_t * );
  55
  56 /* audio filter2 */
  57 static int  OpenFilter ( vlc_object_t * );
  58 static void CloseFilter( vlc_object_t * );
  59 static block_t *Resample( filter_t *, block_t * );
  60
  61
  62 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
  63                            float *f_in, float *f_out, uint32_t ui_remainder,
  64                            uint32_t ui_output_rate, int16_t Inc,
  65                            int i_nb_channels );
  66
  67 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
  68                            float *f_in, float *f_out, uint32_t ui_remainder,
  69                            uint32_t ui_output_rate, uint32_t ui_input_rate,
  70                            int16_t Inc, int i_nb_channels );
  71
  72 /*****************************************************************************
  73  * Local structures
  74  *****************************************************************************/
  75 struct filter_sys_t
  76 {
  77     int32_t *p_buf;                        /* this filter introduces a delay */
  78     int i_buf_size;
  79
  80     int i_old_rate;
  81     double d_old_factor;
  82     int i_old_wing;
  83
  84     unsigned int i_remainder;                /* remainder of previous sample */
  85
  86     audio_date_t end_date;
  87
  88     bool b_first;
  89     bool b_filter2;
  90 };
  91
  92 /*****************************************************************************
  93  * Module descriptor
  94  *****************************************************************************/
  95 vlc_module_begin ()
  96     set_category( CAT_AUDIO )
  97     set_subcategory( SUBCAT_AUDIO_MISC )
  98     set_description( N_("Audio filter for band-limited interpolation resampling") )
  99     set_capability( "audio filter", 20 )
 100     set_callbacks( Create, Close )
 101
 102     add_submodule ()
 103     set_description( _("Audio filter for band-limited interpolation resampling") )
 104     set_capability( "audio filter2", 20 )
 105     set_callbacks( OpenFilter, CloseFilter )
 106 vlc_module_end ()
 107
 108 /*****************************************************************************
 109  * Create: allocate linear resampler
 110  *****************************************************************************/
 111 static int Create( vlc_object_t *p_this )
 112 {
 113     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 114     struct filter_sys_t * p_sys;
 115     double d_factor;
 116     int i_filter_wing;
 117
 118     if ( p_filter->input.i_rate == p_filter->output.i_rate
 119           || p_filter->input.i_format != p_filter->output.i_format
 120           || p_filter->input.i_physical_channels
 121               != p_filter->output.i_physical_channels
 122           || p_filter->input.i_original_channels
 123               != p_filter->output.i_original_channels
 124           || p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
 125     {
 126         return VLC_EGENERIC;
 127     }
 128
 129 #if !defined( __APPLE__ )
 130     if( !config_GetInt( p_this, "hq-resampling" ) )
 131     {
 132         return VLC_EGENERIC;
 133     }
 134 #endif
 135
 136     /* Allocate the memory needed to store the module's structure */
 137     p_sys = malloc( sizeof(filter_sys_t) );
 138     if( p_sys == NULL )
 139         return VLC_ENOMEM;
 140     p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
 141
 142     /* Calculate worst case for the length of the filter wing */
 143     d_factor = (double)p_filter->output.i_rate
 144                         / p_filter->input.i_rate / AOUT_MAX_INPUT_RATE;
 145     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 146                       * __MAX(1.0, 1.0/d_factor) + 10;
 147     p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
 148         sizeof(int32_t) * 2 * i_filter_wing;
 149
 150     /* Allocate enough memory to buffer previous samples */
 151     p_sys->p_buf = malloc( p_sys->i_buf_size );
 152     if( p_sys->p_buf == NULL )
 153     {
 154         free( p_sys );
 155         return VLC_ENOMEM;
 156     }
 157
 158     p_sys->i_old_wing = 0;
 159     p_sys->b_filter2 = false;           /* It seams to be a good valuefor this module */
 160     p_filter->pf_do_work = DoWork;
 161
 162     /* We don't want a new buffer to be created because we're not sure we'll
 163      * actually need to resample anything. */
 164     p_filter->b_in_place = true;
 165
 166     return VLC_SUCCESS;
 167 }
 168
 169 /*****************************************************************************
 170  * Close: free our resources
 171  *****************************************************************************/
 172 static void Close( vlc_object_t * p_this )
 173 {
 174     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 175     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 176     free( p_sys->p_buf );
 177     free( p_sys );
 178 }
 179
 180 /*****************************************************************************
 181  * DoWork: convert a buffer
 182  *****************************************************************************/
 183 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 184                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 185 {
 186     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 187     float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
 188
 189     int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
 190     int i_in_nb = p_in_buf->i_nb_samples;
 191     int i_in, i_out = 0;
 192     unsigned int i_out_rate;
 193     double d_factor, d_scale_factor, d_old_scale_factor;
 194     int i_filter_wing;
 195
 196     if( p_sys->b_filter2 )
 197         i_out_rate = p_filter->output.i_rate;
 198     else
 199         i_out_rate = p_aout->mixer.mixer.i_rate;
 200
 201     /* Check if we really need to run the resampler */
 202     if( i_out_rate == p_filter->input.i_rate )
 203     {
 204         if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
 205             p_sys->i_old_wing &&
 206             p_in_buf->i_size >=
 207               p_in_buf->i_nb_bytes + p_sys->i_old_wing *
 208               p_filter->input.i_bytes_per_frame )
 209         {
 210             /* output the whole thing with the samples from last time */
 211             memmove( ((float *)(p_in_buf->p_buffer)) +
 212                      i_nb_channels * p_sys->i_old_wing,
 213                      p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
 214             memcpy( p_in_buf->p_buffer, p_sys->p_buf +
 215                     i_nb_channels * p_sys->i_old_wing,
 216                     p_sys->i_old_wing *
 217                     p_filter->input.i_bytes_per_frame );
 218
 219             p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
 220                 p_sys->i_old_wing;
 221
 222             p_out_buf->start_date = aout_DateGet( &p_sys->end_date );
 223             p_out_buf->end_date =
 224                 aout_DateIncrement( &p_sys->end_date,
 225                                     p_out_buf->i_nb_samples );
 226
 227             p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 228                 p_filter->input.i_bytes_per_frame;
 229         }
 230         p_filter->b_continuity = false;
 231         p_sys->i_old_wing = 0;
 232         return;
 233     }
 234
 235     if( !p_filter->b_continuity )
 236     {
 237         /* Continuity in sound samples has been broken, we'd better reset
 238          * everything. */
 239         p_filter->b_continuity = true;
 240         p_sys->i_remainder = 0;
 241         aout_DateInit( &p_sys->end_date, i_out_rate );
 242         aout_DateSet( &p_sys->end_date, p_in_buf->start_date );
 243         p_sys->i_old_rate   = p_filter->input.i_rate;
 244         p_sys->d_old_factor = 1;
 245         p_sys->i_old_wing   = 0;
 246     }
 247
 248 #if 0
 249     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 250              p_sys->i_old_rate, p_sys->d_old_factor,
 251              p_sys->i_old_wing, i_in_nb );
 252 #endif
 253
 254     /* Prepare the source buffer */
 255     i_in_nb += (p_sys->i_old_wing * 2);
 256 #ifdef HAVE_ALLOCA
 257     p_in = p_in_orig = (float *)alloca( i_in_nb *
 258                                         p_filter->input.i_bytes_per_frame );
 259 #else
 260     p_in = p_in_orig = (float *)malloc( i_in_nb *
 261                                         p_filter->input.i_bytes_per_frame );
 262 #endif
 263     if( p_in == NULL )
 264     {
 265         return;
 266     }
 267
 268     /* Copy all our samples in p_in */
 269     if( p_sys->i_old_wing )
 270     {
 271         vlc_memcpy( p_in, p_sys->p_buf,
 272                     p_sys->i_old_wing * 2 *
 273                       p_filter->input.i_bytes_per_frame );
 274     }
 275     vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
 276                 p_in_buf->p_buffer,
 277                 p_in_buf->i_nb_samples * p_filter->input.i_bytes_per_frame );
 278
 279     /* Make sure the output buffer is reset */
 280     memset( p_out, 0, p_out_buf->i_size );
 281
 282     /* Calculate the new length of the filter wing */
 283     d_factor = (double)i_out_rate / p_filter->input.i_rate;
 284     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 285
 286     /* Account for increased filter gain when using factors less than 1 */
 287     d_old_scale_factor = SMALL_FILTER_SCALE *
 288         p_sys->d_old_factor + 0.5;
 289     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 290
 291     /* Apply the old rate until we have enough samples for the new one */
 292     i_in = p_sys->i_old_wing;
 293     p_in += p_sys->i_old_wing * i_nb_channels;
 294     for( ; i_in < i_filter_wing &&
 295            (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
 296     {
 297         if( p_sys->d_old_factor == 1 )
 298         {
 299             /* Just copy the samples */
 300             memcpy( p_out, p_in,
 301                     p_filter->input.i_bytes_per_frame );
 302             p_in += i_nb_channels;
 303             p_out += i_nb_channels;
 304             i_out++;
 305             continue;
 306         }
 307
 308         while( p_sys->i_remainder < p_filter->output.i_rate )
 309         {
 310
 311             if( p_sys->d_old_factor >= 1 )
 312             {
 313                 /* FilterFloatUP() is faster if we can use it */
 314
 315                 /* Perform left-wing inner product */
 316                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 317                                SMALL_FILTER_NWING, p_in, p_out,
 318                                p_sys->i_remainder,
 319                                p_filter->output.i_rate,
 320                                -1, i_nb_channels );
 321                 /* Perform right-wing inner product */
 322                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 323                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 324                                p_filter->output.i_rate -
 325                                p_sys->i_remainder,
 326                                p_filter->output.i_rate,
 327                                1, i_nb_channels );
 328
 329 #if 0
 330                 /* Normalize for unity filter gain */
 331                 for( i = 0; i < i_nb_channels; i++ )
 332                 {
 333                     *(p_out+i) *= d_old_scale_factor;
 334                 }
 335 #endif
 336
 337                 /* Sanity check */
 338                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 339                     <= (unsigned int)i_out+1 )
 340                 {
 341                     p_out += i_nb_channels;
 342                     i_out++;
 343                     p_sys->i_remainder += p_filter->input.i_rate;
 344                     break;
 345                 }
 346             }
 347             else
 348             {
 349                 /* Perform left-wing inner product */
 350                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 351                                SMALL_FILTER_NWING, p_in, p_out,
 352                                p_sys->i_remainder,
 353                                p_filter->output.i_rate, p_filter->input.i_rate,
 354                                -1, i_nb_channels );
 355                 /* Perform right-wing inner product */
 356                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 357                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 358                                p_filter->output.i_rate -
 359                                p_sys->i_remainder,
 360                                p_filter->output.i_rate, p_filter->input.i_rate,
 361                                1, i_nb_channels );
 362             }
 363
 364             p_out += i_nb_channels;
 365             i_out++;
 366
 367             p_sys->i_remainder += p_filter->input.i_rate;
 368         }
 369
 370         p_in += i_nb_channels;
 371         p_sys->i_remainder -= p_filter->output.i_rate;
 372     }
 373
 374     /* Apply the new rate for the rest of the samples */
 375     if( i_in < i_in_nb - i_filter_wing )
 376     {
 377         p_sys->i_old_rate   = p_filter->input.i_rate;
 378         p_sys->d_old_factor = d_factor;
 379         p_sys->i_old_wing   = i_filter_wing;
 380     }
 381     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 382     {
 383         while( p_sys->i_remainder < p_filter->output.i_rate )
 384         {
 385
 386             if( d_factor >= 1 )
 387             {
 388                 /* FilterFloatUP() is faster if we can use it */
 389
 390                 /* Perform left-wing inner product */
 391                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 392                                SMALL_FILTER_NWING, p_in, p_out,
 393                                p_sys->i_remainder,
 394                                p_filter->output.i_rate,
 395                                -1, i_nb_channels );
 396
 397                 /* Perform right-wing inner product */
 398                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 399                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 400                                p_filter->output.i_rate -
 401                                p_sys->i_remainder,
 402                                p_filter->output.i_rate,
 403                                1, i_nb_channels );
 404
 405 #if 0
 406                 /* Normalize for unity filter gain */
 407                 for( int i = 0; i < i_nb_channels; i++ )
 408                 {
 409                     *(p_out+i) *= d_old_scale_factor;
 410                 }
 411 #endif
 412                 /* Sanity check */
 413                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 414                     <= (unsigned int)i_out+1 )
 415                 {
 416                     p_out += i_nb_channels;
 417                     i_out++;
 418                     p_sys->i_remainder += p_filter->input.i_rate;
 419                     break;
 420                 }
 421             }
 422             else
 423             {
 424                 /* Perform left-wing inner product */
 425                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 426                                SMALL_FILTER_NWING, p_in, p_out,
 427                                p_sys->i_remainder,
 428                                p_filter->output.i_rate, p_filter->input.i_rate,
 429                                -1, i_nb_channels );
 430                 /* Perform right-wing inner product */
 431                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 432                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 433                                p_filter->output.i_rate -
 434                                p_sys->i_remainder,
 435                                p_filter->output.i_rate, p_filter->input.i_rate,
 436                                1, i_nb_channels );
 437             }
 438
 439             p_out += i_nb_channels;
 440             i_out++;
 441
 442             p_sys->i_remainder += p_filter->input.i_rate;
 443         }
 444
 445         p_in += i_nb_channels;
 446         p_sys->i_remainder -= p_filter->output.i_rate;
 447     }
 448
 449     /* Buffer i_filter_wing * 2 samples for next time */
 450     if( p_sys->i_old_wing )
 451     {
 452         memcpy( p_sys->p_buf,
 453                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 454                 i_nb_channels, (2 * p_sys->i_old_wing) *
 455                 p_filter->input.i_bytes_per_frame );
 456     }
 457
 458 #if 0
 459     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
 460              i_out * p_filter->input.i_bytes_per_frame );
 461 #endif
 462
 463     /* Free the temp buffer */
 464 #ifndef HAVE_ALLOCA
 465     free( p_in_orig );
 466 #endif
 467
 468     /* Finalize aout buffer */
 469     p_out_buf->i_nb_samples = i_out;
 470     p_out_buf->start_date = aout_DateGet( &p_sys->end_date );
 471     p_out_buf->end_date = aout_DateIncrement( &p_sys->end_date,
 472                                               p_out_buf->i_nb_samples );
 473
 474     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 475         i_nb_channels * sizeof(int32_t);
 476
 477 }
 478
 479 /*****************************************************************************
 480  * OpenFilter:
 481  *****************************************************************************/
 482 static int OpenFilter( vlc_object_t *p_this )
 483 {
 484     filter_t *p_filter = (filter_t *)p_this;
 485     filter_sys_t *p_sys;
 486     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 487     double d_factor;
 488     int i_filter_wing;
 489
 490     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 491         p_filter->fmt_in.i_codec != VLC_FOURCC('f','l','3','2') )
 492     {
 493         return VLC_EGENERIC;
 494     }
 495
 496 #if !defined( SYS_DARWIN )
 497     if( !config_GetInt( p_this, "hq-resampling" ) )
 498     {
 499         return VLC_EGENERIC;
 500     }
 501 #endif
 502
 503     /* Allocate the memory needed to store the module's structure */
 504     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 505     if( p_sys == NULL )
 506         return VLC_ENOMEM;
 507
 508     /* Calculate worst case for the length of the filter wing */
 509     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 510     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 511                       * __MAX(1.0, 1.0/d_factor) + 10;
 512     p_filter->p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
 513         sizeof(int32_t) * 2 * i_filter_wing;
 514
 515     /* Allocate enough memory to buffer previous samples */
 516     p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
 517     if( p_filter->p_sys->p_buf == NULL )
 518     {
 519         free( p_sys );
 520         return VLC_ENOMEM;
 521     }
 522
 523     p_filter->p_sys->i_old_wing = 0;
 524     p_sys->b_first = true;
 525     p_sys->b_filter2 = true;
 526     p_filter->pf_audio_filter = Resample;
 527
 528     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 529              (char *)&p_filter->fmt_in.i_codec,
 530              p_filter->fmt_in.audio.i_rate,
 531              p_filter->fmt_in.audio.i_channels,
 532              (char *)&p_filter->fmt_out.i_codec,
 533              p_filter->fmt_out.audio.i_rate,
 534              p_filter->fmt_out.audio.i_channels);
 535
 536     p_filter->fmt_out = p_filter->fmt_in;
 537     p_filter->fmt_out.audio.i_rate = i_out_rate;
 538
 539     return 0;
 540 }
 541
 542 /*****************************************************************************
 543  * CloseFilter : deallocate data structures
 544  *****************************************************************************/
 545 static void CloseFilter( vlc_object_t *p_this )
 546 {
 547     filter_t *p_filter = (filter_t *)p_this;
 548     free( p_filter->p_sys->p_buf );
 549     free( p_filter->p_sys );
 550 }
 551
 552 /*****************************************************************************
 553  * Resample
 554  *****************************************************************************/
 555 static block_t *Resample( filter_t *p_filter, block_t *p_block )
 556 {
 557     aout_filter_t aout_filter;
 558     aout_buffer_t in_buf, out_buf;
 559     block_t *p_out;
 560     int i_out_size;
 561     int i_bytes_per_frame;
 562
 563     if( !p_block || !p_block->i_samples )
 564     {
 565         if( p_block )
 566             block_Release( p_block );
 567         return NULL;
 568     }
 569
 570     i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 571                   p_filter->fmt_out.audio.i_bitspersample / 8;
 572
 573     i_out_size = i_bytes_per_frame * ( 1 + (p_block->i_samples *
 574         p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
 575
 576     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 577     if( !p_out )
 578     {
 579         msg_Warn( p_filter, "can't get output buffer" );
 580         block_Release( p_block );
 581         return NULL;
 582     }
 583
 584     p_out->i_samples = i_out_size / i_bytes_per_frame;
 585     p_out->i_dts = p_block->i_dts;
 586     p_out->i_pts = p_block->i_pts;
 587     p_out->i_length = p_block->i_length;
 588
 589     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 590     aout_filter.input = p_filter->fmt_in.audio;
 591     aout_filter.input.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
 592                   p_filter->fmt_in.audio.i_bitspersample / 8;
 593     aout_filter.output = p_filter->fmt_out.audio;
 594     aout_filter.output.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 595                   p_filter->fmt_out.audio.i_bitspersample / 8;
 596     aout_filter.b_continuity = !p_filter->p_sys->b_first;
 597     p_filter->p_sys->b_first = false;
 598
 599     in_buf.p_buffer = p_block->p_buffer;
 600     in_buf.i_nb_bytes = in_buf.i_size = p_block->i_buffer;
 601     in_buf.i_nb_samples = p_block->i_samples;
 602     out_buf.p_buffer = p_out->p_buffer;
 603     out_buf.i_nb_bytes = out_buf.i_size = p_out->i_buffer;
 604     out_buf.i_nb_samples = p_out->i_samples;
 605
 606     DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
 607
 608     block_Release( p_block );
 609
 610     p_out->i_buffer = out_buf.i_nb_bytes;
 611     p_out->i_samples = out_buf.i_nb_samples;
 612
 613     return p_out;
 614 }
 615
 616 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 617                     float *p_out, uint32_t ui_remainder,
 618                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 619 {
 620     const float *Hp, *Hdp, *End;
 621     float t, temp;
 622     uint32_t ui_linear_remainder;
 623     int i;
 624
 625     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 626     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 627
 628     End = &Imp[Nwing];
 629
 630     ui_linear_remainder = (ui_remainder<<Nhc) -
 631                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 632
 633     if (Inc == 1)               /* If doing right wing...              */
 634     {                           /* ...drop extra coeff, so when Ph is  */
 635         End--;                  /*    0.5, we don't do too many mult's */
 636         if (ui_remainder == 0)  /* If the phase is zero...           */
 637         {                       /* ...then we've already skipped the */
 638             Hp += Npc;          /*    first sample, so we must also  */
 639             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 640         }
 641     }
 642
 643     while (Hp < End) {
 644         t = *Hp;                /* Get filter coeff */
 645                                 /* t is now interp'd filter coeff */
 646         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 647         for( i = 0; i < i_nb_channels; i++ )
 648         {
 649             temp = t;
 650             temp *= *(p_in+i);  /* Mult coeff by input sample */
 651             *(p_out+i) += temp; /* The filter output */
 652         }
 653         Hdp += Npc;             /* Filter coeff differences step */
 654         Hp += Npc;              /* Filter coeff step */
 655         p_in += (Inc * i_nb_channels); /* Input signal step */
 656     }
 657 }
 658
 659 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 660                     float *p_out, uint32_t ui_remainder,
 661                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 662                     int16_t Inc, int i_nb_channels )
 663 {
 664     const float *Hp, *Hdp, *End;
 665     float t, temp;
 666     uint32_t ui_linear_remainder;
 667     int i, ui_counter = 0;
 668
 669     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 670     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 671
 672     End = &Imp[Nwing];
 673
 674     if (Inc == 1)               /* If doing right wing...              */
 675     {                           /* ...drop extra coeff, so when Ph is  */
 676         End--;                  /*    0.5, we don't do too many mult's */
 677         if (ui_remainder == 0)  /* If the phase is zero...           */
 678         {                       /* ...then we've already skipped the */
 679             Hp = Imp +          /* first sample, so we must also  */
 680                   (ui_output_rate << Nhc) / ui_input_rate;
 681             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 682                   (ui_output_rate << Nhc) / ui_input_rate;
 683             ui_counter++;
 684         }
 685     }
 686
 687     while (Hp < End) {
 688         t = *Hp;                /* Get filter coeff */
 689                                 /* t is now interp'd filter coeff */
 690         ui_linear_remainder =
 691           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 692           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 693           ui_input_rate * ui_input_rate;
 694         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 695         for( i = 0; i < i_nb_channels; i++ )
 696         {
 697             temp = t;
 698             temp *= *(p_in+i);  /* Mult coeff by input sample */
 699             *(p_out+i) += temp; /* The filter output */
 700         }
 701
 702         ui_counter++;
 703
 704         /* Filter coeff step */
 705         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 706                     / ui_input_rate;
 707         /* Filter coeff differences step */
 708         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 709                      / ui_input_rate;
 710
 711         p_in += (Inc * i_nb_channels); /* Input signal step */
 712     }
 713 }