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     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->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
 119           || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
 120           || p_filter->fmt_in.audio.i_physical_channels
 121               != p_filter->fmt_out.audio.i_physical_channels
 122           || p_filter->fmt_in.audio.i_original_channels
 123               != p_filter->fmt_out.audio.i_original_channels
 124           || p_filter->fmt_in.audio.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->fmt_out.audio.i_rate
 144                         / p_filter->fmt_in.audio.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->fmt_in.audio ) *
 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->fmt_in.audio );
 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->fmt_out.audio.i_rate;
 198     else
 199         i_out_rate = p_aout->mixer_format.i_rate;
 200
 201     /* Check if we really need to run the resampler */
 202     if( i_out_rate == p_filter->fmt_in.audio.i_rate )
 203     {
 204 #if 0   /* FIXME: needs audio filter2 to use block_Realloc */
 205         if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
 206             p_sys->i_old_wing )
 207         {
 208             /* output the whole thing with the samples from last time */
 209             p_in_buf = block_Realloc( p_in_buf,
 210                 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
 211                 p_in_buf->i_buffer );
 212             if( !p_in_buf )
 213                 abort();
 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->fmt_in.audio.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->i_pts = date_Get( &p_sys->end_date );
 223             p_out_buf->i_length =
 224                 date_Increment( &p_sys->end_date,
 225                                 p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 226
 227             p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 228                 p_filter->fmt_in.audio.i_bytes_per_frame;
 229         }
 230 #endif
 231         p_filter->b_continuity = false;
 232         p_sys->i_old_wing = 0;
 233         return;
 234     }
 235
 236     if( !p_filter->b_continuity )
 237     {
 238         /* Continuity in sound samples has been broken, we'd better reset
 239          * everything. */
 240         p_filter->b_continuity = true;
 241         p_sys->i_remainder = 0;
 242         date_Init( &p_sys->end_date, i_out_rate, 1 );
 243         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 244         p_sys->i_old_rate   = p_filter->fmt_in.audio.i_rate;
 245         p_sys->d_old_factor = 1;
 246         p_sys->i_old_wing   = 0;
 247     }
 248
 249 #if 0
 250     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 251              p_sys->i_old_rate, p_sys->d_old_factor,
 252              p_sys->i_old_wing, i_in_nb );
 253 #endif
 254
 255     /* Prepare the source buffer */
 256     i_in_nb += (p_sys->i_old_wing * 2);
 257
 258     float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
 259          *p_in = p_in_orig;
 260
 261     /* Copy all our samples in p_in */
 262     if( p_sys->i_old_wing )
 263     {
 264         vlc_memcpy( p_in, p_sys->p_buf,
 265                     p_sys->i_old_wing * 2 *
 266                       p_filter->fmt_in.audio.i_bytes_per_frame );
 267     }
 268     vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
 269                 p_in_buf->p_buffer,
 270                 p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
 271
 272     /* Make sure the output buffer is reset */
 273     memset( p_out, 0, p_out_buf->i_buffer );
 274
 275     /* Calculate the new length of the filter wing */
 276     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 277     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 278
 279     /* Account for increased filter gain when using factors less than 1 */
 280     d_old_scale_factor = SMALL_FILTER_SCALE *
 281         p_sys->d_old_factor + 0.5;
 282     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 283
 284     /* Apply the old rate until we have enough samples for the new one */
 285     i_in = p_sys->i_old_wing;
 286     p_in += p_sys->i_old_wing * i_nb_channels;
 287     for( ; i_in < i_filter_wing &&
 288            (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
 289     {
 290         if( p_sys->d_old_factor == 1 )
 291         {
 292             /* Just copy the samples */
 293             memcpy( p_out, p_in,
 294                     p_filter->fmt_in.audio.i_bytes_per_frame );
 295             p_in += i_nb_channels;
 296             p_out += i_nb_channels;
 297             i_out++;
 298             continue;
 299         }
 300
 301         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 302         {
 303
 304             if( p_sys->d_old_factor >= 1 )
 305             {
 306                 /* FilterFloatUP() is faster if we can use it */
 307
 308                 /* Perform left-wing inner product */
 309                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 310                                SMALL_FILTER_NWING, p_in, p_out,
 311                                p_sys->i_remainder,
 312                                p_filter->fmt_out.audio.i_rate,
 313                                -1, i_nb_channels );
 314                 /* Perform right-wing inner product */
 315                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 316                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 317                                p_filter->fmt_out.audio.i_rate -
 318                                p_sys->i_remainder,
 319                                p_filter->fmt_out.audio.i_rate,
 320                                1, i_nb_channels );
 321
 322 #if 0
 323                 /* Normalize for unity filter gain */
 324                 for( i = 0; i < i_nb_channels; i++ )
 325                 {
 326                     *(p_out+i) *= d_old_scale_factor;
 327                 }
 328 #endif
 329
 330                 /* Sanity check */
 331                 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
 332                     <= (unsigned int)i_out+1 )
 333                 {
 334                     p_out += i_nb_channels;
 335                     i_out++;
 336                     p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 337                     break;
 338                 }
 339             }
 340             else
 341             {
 342                 /* Perform left-wing inner product */
 343                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 344                                SMALL_FILTER_NWING, p_in, p_out,
 345                                p_sys->i_remainder,
 346                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 347                                -1, i_nb_channels );
 348                 /* Perform right-wing inner product */
 349                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 350                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 351                                p_filter->fmt_out.audio.i_rate -
 352                                p_sys->i_remainder,
 353                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 354                                1, i_nb_channels );
 355             }
 356
 357             p_out += i_nb_channels;
 358             i_out++;
 359
 360             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 361         }
 362
 363         p_in += i_nb_channels;
 364         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 365     }
 366
 367     /* Apply the new rate for the rest of the samples */
 368     if( i_in < i_in_nb - i_filter_wing )
 369     {
 370         p_sys->i_old_rate   = p_filter->fmt_in.audio.i_rate;
 371         p_sys->d_old_factor = d_factor;
 372         p_sys->i_old_wing   = i_filter_wing;
 373     }
 374     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 375     {
 376         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 377         {
 378
 379             if( d_factor >= 1 )
 380             {
 381                 /* FilterFloatUP() is faster if we can use it */
 382
 383                 /* Perform left-wing inner product */
 384                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 385                                SMALL_FILTER_NWING, p_in, p_out,
 386                                p_sys->i_remainder,
 387                                p_filter->fmt_out.audio.i_rate,
 388                                -1, i_nb_channels );
 389
 390                 /* Perform right-wing inner product */
 391                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 392                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 393                                p_filter->fmt_out.audio.i_rate -
 394                                p_sys->i_remainder,
 395                                p_filter->fmt_out.audio.i_rate,
 396                                1, i_nb_channels );
 397
 398 #if 0
 399                 /* Normalize for unity filter gain */
 400                 for( int i = 0; i < i_nb_channels; i++ )
 401                 {
 402                     *(p_out+i) *= d_old_scale_factor;
 403                 }
 404 #endif
 405                 /* Sanity check */
 406                 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
 407                     <= (unsigned int)i_out+1 )
 408                 {
 409                     p_out += i_nb_channels;
 410                     i_out++;
 411                     p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 412                     break;
 413                 }
 414             }
 415             else
 416             {
 417                 /* Perform left-wing inner product */
 418                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 419                                SMALL_FILTER_NWING, p_in, p_out,
 420                                p_sys->i_remainder,
 421                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 422                                -1, i_nb_channels );
 423                 /* Perform right-wing inner product */
 424                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 425                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 426                                p_filter->fmt_out.audio.i_rate -
 427                                p_sys->i_remainder,
 428                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 429                                1, i_nb_channels );
 430             }
 431
 432             p_out += i_nb_channels;
 433             i_out++;
 434
 435             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 436         }
 437
 438         p_in += i_nb_channels;
 439         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 440     }
 441
 442     /* Buffer i_filter_wing * 2 samples for next time */
 443     if( p_sys->i_old_wing )
 444     {
 445         memcpy( p_sys->p_buf,
 446                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 447                 i_nb_channels, (2 * p_sys->i_old_wing) *
 448                 p_filter->fmt_in.audio.i_bytes_per_frame );
 449     }
 450
 451 #if 0
 452     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 453              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 454 #endif
 455
 456     /* Finalize aout buffer */
 457     p_out_buf->i_nb_samples = i_out;
 458     p_out_buf->i_pts = date_Get( &p_sys->end_date );
 459     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 460                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 461
 462     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 463         i_nb_channels * sizeof(int32_t);
 464
 465 }
 466
 467 /*****************************************************************************
 468  * OpenFilter:
 469  *****************************************************************************/
 470 static int OpenFilter( vlc_object_t *p_this )
 471 {
 472     filter_t *p_filter = (filter_t *)p_this;
 473     filter_sys_t *p_sys;
 474     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 475     double d_factor;
 476     int i_filter_wing;
 477
 478     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 479         p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
 480     {
 481         return VLC_EGENERIC;
 482     }
 483
 484 #if !defined( SYS_DARWIN )
 485     if( !config_GetInt( p_this, "hq-resampling" ) )
 486     {
 487         return VLC_EGENERIC;
 488     }
 489 #endif
 490
 491     /* Allocate the memory needed to store the module's structure */
 492     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 493     if( p_sys == NULL )
 494         return VLC_ENOMEM;
 495
 496     /* Calculate worst case for the length of the filter wing */
 497     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 498     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 499                       * __MAX(1.0, 1.0/d_factor) + 10;
 500     p_filter->p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
 501         sizeof(int32_t) * 2 * i_filter_wing;
 502
 503     /* Allocate enough memory to buffer previous samples */
 504     p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
 505     if( p_filter->p_sys->p_buf == NULL )
 506     {
 507         free( p_sys );
 508         return VLC_ENOMEM;
 509     }
 510
 511     p_filter->p_sys->i_old_wing = 0;
 512     p_sys->b_first = true;
 513     p_sys->b_filter2 = true;
 514     p_filter->pf_audio_filter = Resample;
 515
 516     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 517              (char *)&p_filter->fmt_in.i_codec,
 518              p_filter->fmt_in.audio.i_rate,
 519              p_filter->fmt_in.audio.i_channels,
 520              (char *)&p_filter->fmt_out.i_codec,
 521              p_filter->fmt_out.audio.i_rate,
 522              p_filter->fmt_out.audio.i_channels);
 523
 524     p_filter->fmt_out = p_filter->fmt_in;
 525     p_filter->fmt_out.audio.i_rate = i_out_rate;
 526
 527     return 0;
 528 }
 529
 530 /*****************************************************************************
 531  * CloseFilter : deallocate data structures
 532  *****************************************************************************/
 533 static void CloseFilter( vlc_object_t *p_this )
 534 {
 535     filter_t *p_filter = (filter_t *)p_this;
 536     free( p_filter->p_sys->p_buf );
 537     free( p_filter->p_sys );
 538 }
 539
 540 /*****************************************************************************
 541  * Resample
 542  *****************************************************************************/
 543 static block_t *Resample( filter_t *p_filter, block_t *p_block )
 544 {
 545     aout_filter_t aout_filter;
 546     aout_buffer_t in_buf, out_buf;
 547     block_t *p_out;
 548     int i_out_size;
 549     int i_bytes_per_frame;
 550
 551     if( !p_block || !p_block->i_nb_samples )
 552     {
 553         if( p_block )
 554             block_Release( p_block );
 555         return NULL;
 556     }
 557
 558     i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 559                   p_filter->fmt_out.audio.i_bitspersample / 8;
 560
 561     i_out_size = i_bytes_per_frame * ( 1 + ( p_block->i_nb_samples *
 562                                              p_filter->fmt_out.audio.i_rate /
 563                                              p_filter->fmt_in.audio.i_rate) ) +
 564                  p_filter->p_sys->i_buf_size;
 565
 566     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 567     if( !p_out )
 568     {
 569         msg_Warn( p_filter, "can't get output buffer" );
 570         block_Release( p_block );
 571         return NULL;
 572     }
 573
 574     p_out->i_nb_samples = i_out_size / i_bytes_per_frame;
 575     p_out->i_dts = p_block->i_dts;
 576     p_out->i_pts = p_block->i_pts;
 577     p_out->i_length = p_block->i_length;
 578
 579     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 580     aout_filter.fmt_in.audio = p_filter->fmt_in.audio;
 581     aout_filter.fmt_in.audio.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
 582                   p_filter->fmt_in.audio.i_bitspersample / 8;
 583     aout_filter.fmt_out.audio = p_filter->fmt_out.audio;
 584     aout_filter.fmt_out.audio.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 585                   p_filter->fmt_out.audio.i_bitspersample / 8;
 586     aout_filter.b_continuity = !p_filter->p_sys->b_first;
 587     p_filter->p_sys->b_first = false;
 588
 589     in_buf.p_buffer = p_block->p_buffer;
 590     in_buf.i_buffer = p_block->i_buffer;
 591     in_buf.i_nb_samples = p_block->i_nb_samples;
 592     out_buf.p_buffer = p_out->p_buffer;
 593     out_buf.i_buffer = p_out->i_buffer;
 594     out_buf.i_nb_samples = p_out->i_nb_samples;
 595
 596     DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
 597
 598     block_Release( p_block );
 599
 600     p_out->i_buffer = out_buf.i_buffer;
 601     p_out->i_nb_samples = out_buf.i_nb_samples;
 602
 603     return p_out;
 604 }
 605
 606 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 607                     float *p_out, uint32_t ui_remainder,
 608                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 609 {
 610     const float *Hp, *Hdp, *End;
 611     float t, temp;
 612     uint32_t ui_linear_remainder;
 613     int i;
 614
 615     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 616     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 617
 618     End = &Imp[Nwing];
 619
 620     ui_linear_remainder = (ui_remainder<<Nhc) -
 621                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 622
 623     if (Inc == 1)               /* If doing right wing...              */
 624     {                           /* ...drop extra coeff, so when Ph is  */
 625         End--;                  /*    0.5, we don't do too many mult's */
 626         if (ui_remainder == 0)  /* If the phase is zero...           */
 627         {                       /* ...then we've already skipped the */
 628             Hp += Npc;          /*    first sample, so we must also  */
 629             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 630         }
 631     }
 632
 633     while (Hp < End) {
 634         t = *Hp;                /* Get filter coeff */
 635                                 /* t is now interp'd filter coeff */
 636         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 637         for( i = 0; i < i_nb_channels; i++ )
 638         {
 639             temp = t;
 640             temp *= *(p_in+i);  /* Mult coeff by input sample */
 641             *(p_out+i) += temp; /* The filter output */
 642         }
 643         Hdp += Npc;             /* Filter coeff differences step */
 644         Hp += Npc;              /* Filter coeff step */
 645         p_in += (Inc * i_nb_channels); /* Input signal step */
 646     }
 647 }
 648
 649 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 650                     float *p_out, uint32_t ui_remainder,
 651                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 652                     int16_t Inc, int i_nb_channels )
 653 {
 654     const float *Hp, *Hdp, *End;
 655     float t, temp;
 656     uint32_t ui_linear_remainder;
 657     int i, ui_counter = 0;
 658
 659     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 660     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 661
 662     End = &Imp[Nwing];
 663
 664     if (Inc == 1)               /* If doing right wing...              */
 665     {                           /* ...drop extra coeff, so when Ph is  */
 666         End--;                  /*    0.5, we don't do too many mult's */
 667         if (ui_remainder == 0)  /* If the phase is zero...           */
 668         {                       /* ...then we've already skipped the */
 669             Hp = Imp +          /* first sample, so we must also  */
 670                   (ui_output_rate << Nhc) / ui_input_rate;
 671             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 672                   (ui_output_rate << Nhc) / ui_input_rate;
 673             ui_counter++;
 674         }
 675     }
 676
 677     while (Hp < End) {
 678         t = *Hp;                /* Get filter coeff */
 679                                 /* t is now interp'd filter coeff */
 680         ui_linear_remainder =
 681           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 682           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 683           ui_input_rate * ui_input_rate;
 684         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 685         for( i = 0; i < i_nb_channels; i++ )
 686         {
 687             temp = t;
 688             temp *= *(p_in+i);  /* Mult coeff by input sample */
 689             *(p_out+i) += temp; /* The filter output */
 690         }
 691
 692         ui_counter++;
 693
 694         /* Filter coeff step */
 695         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 696                     / ui_input_rate;
 697         /* Filter coeff differences step */
 698         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 699                      / ui_input_rate;
 700
 701         p_in += (Inc * i_nb_channels); /* Input signal step */
 702     }
 703 }