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
  52 /* audio filter2 */
  53 static int  OpenFilter ( vlc_object_t * );
  54 static void CloseFilter( vlc_object_t * );
  55 static block_t *Resample( filter_t *, block_t * );
  56
  57
  58 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
  59                            float *f_in, float *f_out, uint32_t ui_remainder,
  60                            uint32_t ui_output_rate, int16_t Inc,
  61                            int i_nb_channels );
  62
  63 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
  64                            float *f_in, float *f_out, uint32_t ui_remainder,
  65                            uint32_t ui_output_rate, uint32_t ui_input_rate,
  66                            int16_t Inc, int i_nb_channels );
  67
  68 /*****************************************************************************
  69  * Local structures
  70  *****************************************************************************/
  71 struct filter_sys_t
  72 {
  73     int32_t *p_buf;                        /* this filter introduces a delay */
  74     int i_buf_size;
  75
  76     double d_old_factor;
  77     int i_old_wing;
  78
  79     unsigned int i_remainder;                /* remainder of previous sample */
  80     bool b_first;
  81
  82     date_t end_date;
  83 };
  84
  85 /*****************************************************************************
  86  * Module descriptor
  87  *****************************************************************************/
  88 vlc_module_begin ()
  89     set_category( CAT_AUDIO )
  90     set_subcategory( SUBCAT_AUDIO_MISC )
  91     set_description( N_("Audio filter for band-limited interpolation resampling") )
  92     set_capability( "audio filter2", 20 )
  93     set_callbacks( OpenFilter, CloseFilter )
  94 vlc_module_end ()
  95
  96 /*****************************************************************************
  97  * Resample: convert a buffer
  98  *****************************************************************************/
  99 static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
 100 {
 101     if( !p_in_buf || !p_in_buf->i_nb_samples )
 102     {
 103         if( p_in_buf )
 104             block_Release( p_in_buf );
 105         return NULL;
 106     }
 107
 108     filter_sys_t *p_sys = p_filter->p_sys;
 109     unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
 110     int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 111
 112     /* Check if we really need to run the resampler */
 113     if( i_out_rate == p_filter->fmt_in.audio.i_rate )
 114     {
 115         if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
 116             p_sys->i_old_wing )
 117         {
 118             /* output the whole thing with the samples from last time */
 119             p_in_buf = block_Realloc( p_in_buf,
 120                 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
 121                 p_in_buf->i_buffer );
 122             if( !p_in_buf )
 123                 return NULL;
 124             memcpy( p_in_buf->p_buffer, p_sys->p_buf +
 125                     i_nb_channels * p_sys->i_old_wing,
 126                     p_sys->i_old_wing *
 127                     p_filter->fmt_in.audio.i_bytes_per_frame );
 128
 129             p_in_buf->i_nb_samples += p_sys->i_old_wing;
 130
 131             p_in_buf->i_pts = date_Get( &p_sys->end_date );
 132             p_in_buf->i_length =
 133                 date_Increment( &p_sys->end_date,
 134                                 p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
 135         }
 136         p_sys->i_old_wing = 0;
 137         return p_in_buf;
 138     }
 139
 140     unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 141                                  p_filter->fmt_out.audio.i_bitspersample / 8;
 142     size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
 143               p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
 144             + p_filter->p_sys->i_buf_size;
 145     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
 146     if( !p_out_buf )
 147         return NULL;
 148     float *p_out = (float *)p_out_buf->p_buffer;
 149
 150     if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
 151     {
 152         /* Continuity in sound samples has been broken, we'd better reset
 153          * everything. */
 154         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 155         p_sys->i_remainder = 0;
 156         date_Init( &p_sys->end_date, i_out_rate, 1 );
 157         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 158         p_sys->d_old_factor = 1;
 159         p_sys->i_old_wing   = 0;
 160         p_sys->b_first = false;
 161     }
 162
 163     int i_in_nb = p_in_buf->i_nb_samples;
 164     int i_in, i_out = 0;
 165     double d_factor, d_scale_factor, d_old_scale_factor;
 166     int i_filter_wing;
 167
 168 #if 0
 169     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 170              p_sys->i_old_rate, p_sys->d_old_factor,
 171              p_sys->i_old_wing, i_in_nb );
 172 #endif
 173
 174     /* Prepare the source buffer */
 175     i_in_nb += (p_sys->i_old_wing * 2);
 176
 177     float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
 178          *p_in = p_in_orig;
 179
 180     /* Copy all our samples in p_in */
 181     if( p_sys->i_old_wing )
 182     {
 183         vlc_memcpy( p_in, p_sys->p_buf,
 184                     p_sys->i_old_wing * 2 *
 185                       p_filter->fmt_in.audio.i_bytes_per_frame );
 186     }
 187     /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
 188     vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
 189                 p_in_buf->p_buffer,
 190                 p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
 191     block_Release( p_in_buf );
 192
 193     /* Make sure the output buffer is reset */
 194     memset( p_out, 0, p_out_buf->i_buffer );
 195
 196     /* Calculate the new length of the filter wing */
 197     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 198     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 199
 200     /* Account for increased filter gain when using factors less than 1 */
 201     d_old_scale_factor = SMALL_FILTER_SCALE *
 202         p_sys->d_old_factor + 0.5;
 203     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 204
 205     /* Apply the old rate until we have enough samples for the new one */
 206     i_in = p_sys->i_old_wing;
 207     p_in += p_sys->i_old_wing * i_nb_channels;
 208     for( ; i_in < i_filter_wing &&
 209            (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
 210     {
 211         if( p_sys->d_old_factor == 1 )
 212         {
 213             /* Just copy the samples */
 214             memcpy( p_out, p_in,
 215                     p_filter->fmt_in.audio.i_bytes_per_frame );
 216             p_in += i_nb_channels;
 217             p_out += i_nb_channels;
 218             i_out++;
 219             continue;
 220         }
 221
 222         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 223         {
 224
 225             if( p_sys->d_old_factor >= 1 )
 226             {
 227                 /* FilterFloatUP() is faster if we can use it */
 228
 229                 /* Perform left-wing inner product */
 230                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 231                                SMALL_FILTER_NWING, p_in, p_out,
 232                                p_sys->i_remainder,
 233                                p_filter->fmt_out.audio.i_rate,
 234                                -1, i_nb_channels );
 235                 /* Perform right-wing inner product */
 236                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 237                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 238                                p_filter->fmt_out.audio.i_rate -
 239                                p_sys->i_remainder,
 240                                p_filter->fmt_out.audio.i_rate,
 241                                1, i_nb_channels );
 242
 243 #if 0
 244                 /* Normalize for unity filter gain */
 245                 for( i = 0; i < i_nb_channels; i++ )
 246                 {
 247                     *(p_out+i) *= d_old_scale_factor;
 248                 }
 249 #endif
 250
 251                 /* Sanity check */
 252                 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
 253                     <= (unsigned int)i_out+1 )
 254                 {
 255                     p_out += i_nb_channels;
 256                     i_out++;
 257                     p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 258                     break;
 259                 }
 260             }
 261             else
 262             {
 263                 /* Perform left-wing inner product */
 264                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 265                                SMALL_FILTER_NWING, p_in, p_out,
 266                                p_sys->i_remainder,
 267                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 268                                -1, i_nb_channels );
 269                 /* Perform right-wing inner product */
 270                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 271                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 272                                p_filter->fmt_out.audio.i_rate -
 273                                p_sys->i_remainder,
 274                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 275                                1, i_nb_channels );
 276             }
 277
 278             p_out += i_nb_channels;
 279             i_out++;
 280
 281             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 282         }
 283
 284         p_in += i_nb_channels;
 285         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 286     }
 287
 288     /* Apply the new rate for the rest of the samples */
 289     if( i_in < i_in_nb - i_filter_wing )
 290     {
 291         p_sys->d_old_factor = d_factor;
 292         p_sys->i_old_wing   = i_filter_wing;
 293     }
 294     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 295     {
 296         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 297         {
 298
 299             if( d_factor >= 1 )
 300             {
 301                 /* FilterFloatUP() is faster if we can use it */
 302
 303                 /* Perform left-wing inner product */
 304                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 305                                SMALL_FILTER_NWING, p_in, p_out,
 306                                p_sys->i_remainder,
 307                                p_filter->fmt_out.audio.i_rate,
 308                                -1, i_nb_channels );
 309
 310                 /* Perform right-wing inner product */
 311                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 312                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 313                                p_filter->fmt_out.audio.i_rate -
 314                                p_sys->i_remainder,
 315                                p_filter->fmt_out.audio.i_rate,
 316                                1, i_nb_channels );
 317
 318 #if 0
 319                 /* Normalize for unity filter gain */
 320                 for( int i = 0; i < i_nb_channels; i++ )
 321                 {
 322                     *(p_out+i) *= d_old_scale_factor;
 323                 }
 324 #endif
 325                 /* Sanity check */
 326                 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
 327                     <= (unsigned int)i_out+1 )
 328                 {
 329                     p_out += i_nb_channels;
 330                     i_out++;
 331                     p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 332                     break;
 333                 }
 334             }
 335             else
 336             {
 337                 /* Perform left-wing inner product */
 338                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 339                                SMALL_FILTER_NWING, p_in, p_out,
 340                                p_sys->i_remainder,
 341                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 342                                -1, i_nb_channels );
 343                 /* Perform right-wing inner product */
 344                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 345                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 346                                p_filter->fmt_out.audio.i_rate -
 347                                p_sys->i_remainder,
 348                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 349                                1, i_nb_channels );
 350             }
 351
 352             p_out += i_nb_channels;
 353             i_out++;
 354
 355             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 356         }
 357
 358         p_in += i_nb_channels;
 359         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 360     }
 361
 362     /* Buffer i_filter_wing * 2 samples for next time */
 363     if( p_sys->i_old_wing )
 364     {
 365         memcpy( p_sys->p_buf,
 366                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 367                 i_nb_channels, (2 * p_sys->i_old_wing) *
 368                 p_filter->fmt_in.audio.i_bytes_per_frame );
 369     }
 370
 371 #if 0
 372     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 373              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 374 #endif
 375
 376     /* Finalize aout buffer */
 377     p_out_buf->i_nb_samples = i_out;
 378     p_out_buf->i_pts = date_Get( &p_sys->end_date );
 379     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 380                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 381
 382     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 383         i_nb_channels * sizeof(int32_t);
 384     return p_out_buf;
 385 }
 386
 387 /*****************************************************************************
 388  * OpenFilter:
 389  *****************************************************************************/
 390 static int OpenFilter( vlc_object_t *p_this )
 391 {
 392     filter_t *p_filter = (filter_t *)p_this;
 393     filter_sys_t *p_sys;
 394     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 395     double d_factor;
 396     int i_filter_wing;
 397
 398     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 399         p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
 400     {
 401         return VLC_EGENERIC;
 402     }
 403
 404 #if !defined( SYS_DARWIN )
 405     if( !config_GetInt( p_this, "hq-resampling" ) )
 406     {
 407         return VLC_EGENERIC;
 408     }
 409 #endif
 410
 411     /* Allocate the memory needed to store the module's structure */
 412     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 413     if( p_sys == NULL )
 414         return VLC_ENOMEM;
 415
 416     /* Calculate worst case for the length of the filter wing */
 417     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 418     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 419                       * __MAX(1.0, 1.0/d_factor) + 10;
 420     p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
 421         sizeof(int32_t) * 2 * i_filter_wing;
 422
 423     /* Allocate enough memory to buffer previous samples */
 424     p_sys->p_buf = malloc( p_sys->i_buf_size );
 425     if( p_sys->p_buf == NULL )
 426     {
 427         free( p_sys );
 428         return VLC_ENOMEM;
 429     }
 430
 431     p_sys->i_old_wing = 0;
 432     p_sys->b_first = true;
 433     p_filter->pf_audio_filter = Resample;
 434
 435     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 436              (char *)&p_filter->fmt_in.i_codec,
 437              p_filter->fmt_in.audio.i_rate,
 438              p_filter->fmt_in.audio.i_channels,
 439              (char *)&p_filter->fmt_out.i_codec,
 440              p_filter->fmt_out.audio.i_rate,
 441              p_filter->fmt_out.audio.i_channels);
 442
 443     p_filter->fmt_out = p_filter->fmt_in;
 444     p_filter->fmt_out.audio.i_rate = i_out_rate;
 445
 446     return 0;
 447 }
 448
 449 /*****************************************************************************
 450  * CloseFilter : deallocate data structures
 451  *****************************************************************************/
 452 static void CloseFilter( vlc_object_t *p_this )
 453 {
 454     filter_t *p_filter = (filter_t *)p_this;
 455     free( p_filter->p_sys->p_buf );
 456     free( p_filter->p_sys );
 457 }
 458
 459 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 460                     float *p_out, uint32_t ui_remainder,
 461                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 462 {
 463     const float *Hp, *Hdp, *End;
 464     float t, temp;
 465     uint32_t ui_linear_remainder;
 466     int i;
 467
 468     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 469     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 470
 471     End = &Imp[Nwing];
 472
 473     ui_linear_remainder = (ui_remainder<<Nhc) -
 474                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 475
 476     if (Inc == 1)               /* If doing right wing...              */
 477     {                           /* ...drop extra coeff, so when Ph is  */
 478         End--;                  /*    0.5, we don't do too many mult's */
 479         if (ui_remainder == 0)  /* If the phase is zero...           */
 480         {                       /* ...then we've already skipped the */
 481             Hp += Npc;          /*    first sample, so we must also  */
 482             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 483         }
 484     }
 485
 486     while (Hp < End) {
 487         t = *Hp;                /* Get filter coeff */
 488                                 /* t is now interp'd filter coeff */
 489         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 490         for( i = 0; i < i_nb_channels; i++ )
 491         {
 492             temp = t;
 493             temp *= *(p_in+i);  /* Mult coeff by input sample */
 494             *(p_out+i) += temp; /* The filter output */
 495         }
 496         Hdp += Npc;             /* Filter coeff differences step */
 497         Hp += Npc;              /* Filter coeff step */
 498         p_in += (Inc * i_nb_channels); /* Input signal step */
 499     }
 500 }
 501
 502 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 503                     float *p_out, uint32_t ui_remainder,
 504                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 505                     int16_t Inc, int i_nb_channels )
 506 {
 507     const float *Hp, *Hdp, *End;
 508     float t, temp;
 509     uint32_t ui_linear_remainder;
 510     int i, ui_counter = 0;
 511
 512     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 513     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 514
 515     End = &Imp[Nwing];
 516
 517     if (Inc == 1)               /* If doing right wing...              */
 518     {                           /* ...drop extra coeff, so when Ph is  */
 519         End--;                  /*    0.5, we don't do too many mult's */
 520         if (ui_remainder == 0)  /* If the phase is zero...           */
 521         {                       /* ...then we've already skipped the */
 522             Hp = Imp +          /* first sample, so we must also  */
 523                   (ui_output_rate << Nhc) / ui_input_rate;
 524             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 525                   (ui_output_rate << Nhc) / ui_input_rate;
 526             ui_counter++;
 527         }
 528     }
 529
 530     while (Hp < End) {
 531         t = *Hp;                /* Get filter coeff */
 532                                 /* t is now interp'd filter coeff */
 533         ui_linear_remainder =
 534           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 535           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 536           ui_input_rate * ui_input_rate;
 537         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 538         for( i = 0; i < i_nb_channels; i++ )
 539         {
 540             temp = t;
 541             temp *= *(p_in+i);  /* Mult coeff by input sample */
 542             *(p_out+i) += temp; /* The filter output */
 543         }
 544
 545         ui_counter++;
 546
 547         /* Filter coeff step */
 548         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 549                     / ui_input_rate;
 550         /* Filter coeff differences step */
 551         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 552                      / ui_input_rate;
 553
 554         p_in += (Inc * i_nb_channels); /* Input signal step */
 555     }
 556 }