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( N_("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_CODEC_FL32 )
 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_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
 257     float p_in_orig[i_in_nb * p_filter->input.i_bytes_per_frame / 4],
 258          *p_in = p_in_orig;
 259
 260     /* Copy all our samples in p_in */
 261     if( p_sys->i_old_wing )
 262     {
 263         vlc_memcpy( p_in, p_sys->p_buf,
 264                     p_sys->i_old_wing * 2 *
 265                       p_filter->input.i_bytes_per_frame );
 266     }
 267     vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
 268                 p_in_buf->p_buffer,
 269                 p_in_buf->i_nb_samples * p_filter->input.i_bytes_per_frame );
 270
 271     /* Make sure the output buffer is reset */
 272     memset( p_out, 0, p_out_buf->i_size );
 273
 274     /* Calculate the new length of the filter wing */
 275     d_factor = (double)i_out_rate / p_filter->input.i_rate;
 276     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 277
 278     /* Account for increased filter gain when using factors less than 1 */
 279     d_old_scale_factor = SMALL_FILTER_SCALE *
 280         p_sys->d_old_factor + 0.5;
 281     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 282
 283     /* Apply the old rate until we have enough samples for the new one */
 284     i_in = p_sys->i_old_wing;
 285     p_in += p_sys->i_old_wing * i_nb_channels;
 286     for( ; i_in < i_filter_wing &&
 287            (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
 288     {
 289         if( p_sys->d_old_factor == 1 )
 290         {
 291             /* Just copy the samples */
 292             memcpy( p_out, p_in,
 293                     p_filter->input.i_bytes_per_frame );
 294             p_in += i_nb_channels;
 295             p_out += i_nb_channels;
 296             i_out++;
 297             continue;
 298         }
 299
 300         while( p_sys->i_remainder < p_filter->output.i_rate )
 301         {
 302
 303             if( p_sys->d_old_factor >= 1 )
 304             {
 305                 /* FilterFloatUP() is faster if we can use it */
 306
 307                 /* Perform left-wing inner product */
 308                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 309                                SMALL_FILTER_NWING, p_in, p_out,
 310                                p_sys->i_remainder,
 311                                p_filter->output.i_rate,
 312                                -1, i_nb_channels );
 313                 /* Perform right-wing inner product */
 314                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 315                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 316                                p_filter->output.i_rate -
 317                                p_sys->i_remainder,
 318                                p_filter->output.i_rate,
 319                                1, i_nb_channels );
 320
 321 #if 0
 322                 /* Normalize for unity filter gain */
 323                 for( i = 0; i < i_nb_channels; i++ )
 324                 {
 325                     *(p_out+i) *= d_old_scale_factor;
 326                 }
 327 #endif
 328
 329                 /* Sanity check */
 330                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 331                     <= (unsigned int)i_out+1 )
 332                 {
 333                     p_out += i_nb_channels;
 334                     i_out++;
 335                     p_sys->i_remainder += p_filter->input.i_rate;
 336                     break;
 337                 }
 338             }
 339             else
 340             {
 341                 /* Perform left-wing inner product */
 342                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 343                                SMALL_FILTER_NWING, p_in, p_out,
 344                                p_sys->i_remainder,
 345                                p_filter->output.i_rate, p_filter->input.i_rate,
 346                                -1, i_nb_channels );
 347                 /* Perform right-wing inner product */
 348                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 349                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 350                                p_filter->output.i_rate -
 351                                p_sys->i_remainder,
 352                                p_filter->output.i_rate, p_filter->input.i_rate,
 353                                1, i_nb_channels );
 354             }
 355
 356             p_out += i_nb_channels;
 357             i_out++;
 358
 359             p_sys->i_remainder += p_filter->input.i_rate;
 360         }
 361
 362         p_in += i_nb_channels;
 363         p_sys->i_remainder -= p_filter->output.i_rate;
 364     }
 365
 366     /* Apply the new rate for the rest of the samples */
 367     if( i_in < i_in_nb - i_filter_wing )
 368     {
 369         p_sys->i_old_rate   = p_filter->input.i_rate;
 370         p_sys->d_old_factor = d_factor;
 371         p_sys->i_old_wing   = i_filter_wing;
 372     }
 373     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 374     {
 375         while( p_sys->i_remainder < p_filter->output.i_rate )
 376         {
 377
 378             if( d_factor >= 1 )
 379             {
 380                 /* FilterFloatUP() is faster if we can use it */
 381
 382                 /* Perform left-wing inner product */
 383                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 384                                SMALL_FILTER_NWING, p_in, p_out,
 385                                p_sys->i_remainder,
 386                                p_filter->output.i_rate,
 387                                -1, i_nb_channels );
 388
 389                 /* Perform right-wing inner product */
 390                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 391                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 392                                p_filter->output.i_rate -
 393                                p_sys->i_remainder,
 394                                p_filter->output.i_rate,
 395                                1, i_nb_channels );
 396
 397 #if 0
 398                 /* Normalize for unity filter gain */
 399                 for( int i = 0; i < i_nb_channels; i++ )
 400                 {
 401                     *(p_out+i) *= d_old_scale_factor;
 402                 }
 403 #endif
 404                 /* Sanity check */
 405                 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
 406                     <= (unsigned int)i_out+1 )
 407                 {
 408                     p_out += i_nb_channels;
 409                     i_out++;
 410                     p_sys->i_remainder += p_filter->input.i_rate;
 411                     break;
 412                 }
 413             }
 414             else
 415             {
 416                 /* Perform left-wing inner product */
 417                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 418                                SMALL_FILTER_NWING, p_in, p_out,
 419                                p_sys->i_remainder,
 420                                p_filter->output.i_rate, p_filter->input.i_rate,
 421                                -1, i_nb_channels );
 422                 /* Perform right-wing inner product */
 423                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 424                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 425                                p_filter->output.i_rate -
 426                                p_sys->i_remainder,
 427                                p_filter->output.i_rate, p_filter->input.i_rate,
 428                                1, i_nb_channels );
 429             }
 430
 431             p_out += i_nb_channels;
 432             i_out++;
 433
 434             p_sys->i_remainder += p_filter->input.i_rate;
 435         }
 436
 437         p_in += i_nb_channels;
 438         p_sys->i_remainder -= p_filter->output.i_rate;
 439     }
 440
 441     /* Buffer i_filter_wing * 2 samples for next time */
 442     if( p_sys->i_old_wing )
 443     {
 444         memcpy( p_sys->p_buf,
 445                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 446                 i_nb_channels, (2 * p_sys->i_old_wing) *
 447                 p_filter->input.i_bytes_per_frame );
 448     }
 449
 450 #if 0
 451     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
 452              i_out * p_filter->input.i_bytes_per_frame );
 453 #endif
 454
 455     /* Finalize aout buffer */
 456     p_out_buf->i_nb_samples = i_out;
 457     p_out_buf->start_date = aout_DateGet( &p_sys->end_date );
 458     p_out_buf->end_date = aout_DateIncrement( &p_sys->end_date,
 459                                               p_out_buf->i_nb_samples );
 460
 461     p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
 462         i_nb_channels * sizeof(int32_t);
 463
 464 }
 465
 466 /*****************************************************************************
 467  * OpenFilter:
 468  *****************************************************************************/
 469 static int OpenFilter( vlc_object_t *p_this )
 470 {
 471     filter_t *p_filter = (filter_t *)p_this;
 472     filter_sys_t *p_sys;
 473     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 474     double d_factor;
 475     int i_filter_wing;
 476
 477     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 478         p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
 479     {
 480         return VLC_EGENERIC;
 481     }
 482
 483 #if !defined( SYS_DARWIN )
 484     if( !config_GetInt( p_this, "hq-resampling" ) )
 485     {
 486         return VLC_EGENERIC;
 487     }
 488 #endif
 489
 490     /* Allocate the memory needed to store the module's structure */
 491     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 492     if( p_sys == NULL )
 493         return VLC_ENOMEM;
 494
 495     /* Calculate worst case for the length of the filter wing */
 496     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 497     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 498                       * __MAX(1.0, 1.0/d_factor) + 10;
 499     p_filter->p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
 500         sizeof(int32_t) * 2 * i_filter_wing;
 501
 502     /* Allocate enough memory to buffer previous samples */
 503     p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
 504     if( p_filter->p_sys->p_buf == NULL )
 505     {
 506         free( p_sys );
 507         return VLC_ENOMEM;
 508     }
 509
 510     p_filter->p_sys->i_old_wing = 0;
 511     p_sys->b_first = true;
 512     p_sys->b_filter2 = true;
 513     p_filter->pf_audio_filter = Resample;
 514
 515     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 516              (char *)&p_filter->fmt_in.i_codec,
 517              p_filter->fmt_in.audio.i_rate,
 518              p_filter->fmt_in.audio.i_channels,
 519              (char *)&p_filter->fmt_out.i_codec,
 520              p_filter->fmt_out.audio.i_rate,
 521              p_filter->fmt_out.audio.i_channels);
 522
 523     p_filter->fmt_out = p_filter->fmt_in;
 524     p_filter->fmt_out.audio.i_rate = i_out_rate;
 525
 526     return 0;
 527 }
 528
 529 /*****************************************************************************
 530  * CloseFilter : deallocate data structures
 531  *****************************************************************************/
 532 static void CloseFilter( vlc_object_t *p_this )
 533 {
 534     filter_t *p_filter = (filter_t *)p_this;
 535     free( p_filter->p_sys->p_buf );
 536     free( p_filter->p_sys );
 537 }
 538
 539 /*****************************************************************************
 540  * Resample
 541  *****************************************************************************/
 542 static block_t *Resample( filter_t *p_filter, block_t *p_block )
 543 {
 544     aout_filter_t aout_filter;
 545     aout_buffer_t in_buf, out_buf;
 546     block_t *p_out;
 547     int i_out_size;
 548     int i_bytes_per_frame;
 549
 550     if( !p_block || !p_block->i_samples )
 551     {
 552         if( p_block )
 553             block_Release( p_block );
 554         return NULL;
 555     }
 556
 557     i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 558                   p_filter->fmt_out.audio.i_bitspersample / 8;
 559
 560     i_out_size = i_bytes_per_frame * ( 1 + ( p_block->i_samples *
 561                                              p_filter->fmt_out.audio.i_rate /
 562                                              p_filter->fmt_in.audio.i_rate) ) +
 563                  p_filter->p_sys->i_buf_size;
 564
 565     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 566     if( !p_out )
 567     {
 568         msg_Warn( p_filter, "can't get output buffer" );
 569         block_Release( p_block );
 570         return NULL;
 571     }
 572
 573     p_out->i_samples = i_out_size / i_bytes_per_frame;
 574     p_out->i_dts = p_block->i_dts;
 575     p_out->i_pts = p_block->i_pts;
 576     p_out->i_length = p_block->i_length;
 577
 578     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 579     aout_filter.input = p_filter->fmt_in.audio;
 580     aout_filter.input.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
 581                   p_filter->fmt_in.audio.i_bitspersample / 8;
 582     aout_filter.output = p_filter->fmt_out.audio;
 583     aout_filter.output.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 584                   p_filter->fmt_out.audio.i_bitspersample / 8;
 585     aout_filter.b_continuity = !p_filter->p_sys->b_first;
 586     p_filter->p_sys->b_first = false;
 587
 588     in_buf.p_buffer = p_block->p_buffer;
 589     in_buf.i_nb_bytes = in_buf.i_size = p_block->i_buffer;
 590     in_buf.i_nb_samples = p_block->i_samples;
 591     out_buf.p_buffer = p_out->p_buffer;
 592     out_buf.i_nb_bytes = out_buf.i_size = p_out->i_buffer;
 593     out_buf.i_nb_samples = p_out->i_samples;
 594
 595     DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
 596
 597     block_Release( p_block );
 598
 599     p_out->i_buffer = out_buf.i_nb_bytes;
 600     p_out->i_samples = out_buf.i_nb_samples;
 601
 602     return p_out;
 603 }
 604
 605 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 606                     float *p_out, uint32_t ui_remainder,
 607                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 608 {
 609     const float *Hp, *Hdp, *End;
 610     float t, temp;
 611     uint32_t ui_linear_remainder;
 612     int i;
 613
 614     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 615     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 616
 617     End = &Imp[Nwing];
 618
 619     ui_linear_remainder = (ui_remainder<<Nhc) -
 620                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 621
 622     if (Inc == 1)               /* If doing right wing...              */
 623     {                           /* ...drop extra coeff, so when Ph is  */
 624         End--;                  /*    0.5, we don't do too many mult's */
 625         if (ui_remainder == 0)  /* If the phase is zero...           */
 626         {                       /* ...then we've already skipped the */
 627             Hp += Npc;          /*    first sample, so we must also  */
 628             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 629         }
 630     }
 631
 632     while (Hp < End) {
 633         t = *Hp;                /* Get filter coeff */
 634                                 /* t is now interp'd filter coeff */
 635         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 636         for( i = 0; i < i_nb_channels; i++ )
 637         {
 638             temp = t;
 639             temp *= *(p_in+i);  /* Mult coeff by input sample */
 640             *(p_out+i) += temp; /* The filter output */
 641         }
 642         Hdp += Npc;             /* Filter coeff differences step */
 643         Hp += Npc;              /* Filter coeff step */
 644         p_in += (Inc * i_nb_channels); /* Input signal step */
 645     }
 646 }
 647
 648 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 649                     float *p_out, uint32_t ui_remainder,
 650                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 651                     int16_t Inc, int i_nb_channels )
 652 {
 653     const float *Hp, *Hdp, *End;
 654     float t, temp;
 655     uint32_t ui_linear_remainder;
 656     int i, ui_counter = 0;
 657
 658     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 659     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 660
 661     End = &Imp[Nwing];
 662
 663     if (Inc == 1)               /* If doing right wing...              */
 664     {                           /* ...drop extra coeff, so when Ph is  */
 665         End--;                  /*    0.5, we don't do too many mult's */
 666         if (ui_remainder == 0)  /* If the phase is zero...           */
 667         {                       /* ...then we've already skipped the */
 668             Hp = Imp +          /* first sample, so we must also  */
 669                   (ui_output_rate << Nhc) / ui_input_rate;
 670             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 671                   (ui_output_rate << Nhc) / ui_input_rate;
 672             ui_counter++;
 673         }
 674     }
 675
 676     while (Hp < End) {
 677         t = *Hp;                /* Get filter coeff */
 678                                 /* t is now interp'd filter coeff */
 679         ui_linear_remainder =
 680           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 681           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 682           ui_input_rate * ui_input_rate;
 683         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 684         for( i = 0; i < i_nb_channels; i++ )
 685         {
 686             temp = t;
 687             temp *= *(p_in+i);  /* Mult coeff by input sample */
 688             *(p_out+i) += temp; /* The filter output */
 689         }
 690
 691         ui_counter++;
 692
 693         /* Filter coeff step */
 694         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 695                     / ui_input_rate;
 696         /* Filter coeff differences step */
 697         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 698                      / ui_input_rate;
 699
 700         p_in += (Inc * i_nb_channels); /* Input signal step */
 701     }
 702 }