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 <assert.h>
  47
  48 #include "bandlimited.h"
  49
  50 /*****************************************************************************
  51  * Local prototypes
  52  *****************************************************************************/
  53
  54 /* audio filter */
  55 static int  OpenFilter ( vlc_object_t * );
  56 static void CloseFilter( vlc_object_t * );
  57 static block_t *Resample( filter_t *, block_t * );
  58
  59 static void ResampleFloat( filter_t *p_filter,
  60                            block_t **pp_out_buf,  size_t *pi_out,
  61                            float **pp_in,
  62                            int i_in, int i_in_end,
  63                            double d_factor, bool b_factor_old,
  64                            int i_nb_channels, int i_bytes_per_frame );
  65
  66 /*****************************************************************************
  67  * Local structures
  68  *****************************************************************************/
  69 struct filter_sys_t
  70 {
  71     int32_t *p_buf;                        /* this filter introduces a delay */
  72     size_t i_buf_size;
  73
  74     double d_old_factor;
  75     int i_old_wing;
  76
  77     unsigned int i_remainder;                /* remainder of previous sample */
  78     bool b_first;
  79
  80     date_t end_date;
  81 };
  82
  83 /*****************************************************************************
  84  * Module descriptor
  85  *****************************************************************************/
  86 vlc_module_begin ()
  87     set_category( CAT_AUDIO )
  88     set_subcategory( SUBCAT_AUDIO_MISC )
  89     set_description( N_("Audio filter for band-limited interpolation resampling") )
  90     set_capability( "audio filter", 20 )
  91     set_callbacks( OpenFilter, CloseFilter )
  92 vlc_module_end ()
  93
  94 /*****************************************************************************
  95  * Resample: convert a buffer
  96  *****************************************************************************/
  97 static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
  98 {
  99     if( !p_in_buf || !p_in_buf->i_nb_samples )
 100     {
 101         if( p_in_buf )
 102             block_Release( p_in_buf );
 103         return NULL;
 104     }
 105
 106     filter_sys_t *p_sys = p_filter->p_sys;
 107     unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
 108     int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 109
 110     /* Check if we really need to run the resampler */
 111     if( i_out_rate == p_filter->fmt_in.audio.i_rate )
 112     {
 113         if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
 114             p_sys->i_old_wing )
 115         {
 116             /* output the whole thing with the samples from last time */
 117             p_in_buf = block_Realloc( p_in_buf,
 118                 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
 119                 p_in_buf->i_buffer );
 120             if( !p_in_buf )
 121                 return NULL;
 122             memcpy( p_in_buf->p_buffer, p_sys->p_buf +
 123                     i_nb_channels * p_sys->i_old_wing,
 124                     p_sys->i_old_wing *
 125                     p_filter->fmt_in.audio.i_bytes_per_frame );
 126
 127             p_in_buf->i_nb_samples += p_sys->i_old_wing;
 128
 129             p_in_buf->i_pts = date_Get( &p_sys->end_date );
 130             p_in_buf->i_length =
 131                 date_Increment( &p_sys->end_date,
 132                                 p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
 133         }
 134         p_sys->i_old_wing = 0;
 135         p_sys->b_first = true;
 136         return p_in_buf;
 137     }
 138
 139     unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 140                                  p_filter->fmt_out.audio.i_bitspersample / 8;
 141     size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
 142               p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
 143             + p_filter->p_sys->i_buf_size;
 144     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
 145     if( !p_out_buf )
 146     {
 147         block_Release( p_in_buf );
 148         return NULL;
 149     }
 150
 151     if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
 152     {
 153         /* Continuity in sound samples has been broken, we'd better reset
 154          * everything. */
 155         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 156         p_sys->i_remainder = 0;
 157         date_Init( &p_sys->end_date, i_out_rate, 1 );
 158         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 159         p_sys->d_old_factor = 1;
 160         p_sys->i_old_wing   = 0;
 161         p_sys->b_first = false;
 162     }
 163
 164     size_t i_in_nb = p_in_buf->i_nb_samples;
 165     size_t i_in, i_out = 0;
 166     double d_factor, d_scale_factor, d_old_scale_factor;
 167     size_t i_filter_wing;
 168
 169 #if 0
 170     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 171              p_sys->i_old_rate, p_sys->d_old_factor,
 172              p_sys->i_old_wing, i_in_nb );
 173 #endif
 174
 175     /* Same format in and out... */
 176     assert( p_filter->fmt_in.audio.i_bytes_per_frame == i_bytes_per_frame );
 177
 178     /* Prepare the source buffer */
 179     if( p_sys->i_old_wing )
 180     {   /* Copy all our samples in p_in_buf */
 181         /* Normally, there should be enough room for the old wing in the
 182          * buffer head room. Otherwise, we need to copy memory anyway. */
 183         p_in_buf = block_Realloc( p_in_buf,
 184                                   p_sys->i_old_wing * 2 * i_bytes_per_frame,
 185                                   p_in_buf->i_buffer );
 186         if( unlikely(p_in_buf == NULL) )
 187             return NULL;
 188         memcpy( p_in_buf->p_buffer, p_sys->p_buf,
 189                 p_sys->i_old_wing * 2 * i_bytes_per_frame );
 190     }
 191     i_in_nb += (p_sys->i_old_wing * 2);
 192     float *p_in = (float *)p_in_buf->p_buffer;
 193     const float *p_in_orig = p_in;
 194
 195     /* Make sure the output buffer is reset */
 196     memset( p_out_buf->p_buffer, 0, p_out_buf->i_buffer );
 197
 198     /* Calculate the new length of the filter wing */
 199     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 200     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 201
 202     /* Account for increased filter gain when using factors less than 1 */
 203     d_old_scale_factor = SMALL_FILTER_SCALE *
 204         p_sys->d_old_factor + 0.5;
 205     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 206
 207     /* Apply the old rate until we have enough samples for the new one */
 208     i_in = p_sys->i_old_wing;
 209     p_in += p_sys->i_old_wing * i_nb_channels;
 210
 211     size_t i_old_in_end = 0;
 212     if( p_sys->i_old_wing <= i_in_nb )
 213         i_old_in_end = __MIN( i_filter_wing, i_in_nb - p_sys->i_old_wing );
 214
 215     ResampleFloat( p_filter,
 216                    &p_out_buf, &i_out, &p_in,
 217                    i_in, i_old_in_end,
 218                    p_sys->d_old_factor, true,
 219                    i_nb_channels, i_bytes_per_frame );
 220     i_in = __MAX( i_in, i_old_in_end );
 221
 222     /* Apply the new rate for the rest of the samples */
 223     if( i_in < i_in_nb - i_filter_wing )
 224     {
 225         p_sys->d_old_factor = d_factor;
 226         p_sys->i_old_wing   = i_filter_wing;
 227     }
 228     if( p_out_buf )
 229     {
 230         ResampleFloat( p_filter,
 231                        &p_out_buf, &i_out, &p_in,
 232                        i_in, i_in_nb - i_filter_wing,
 233                        d_factor, false,
 234                        i_nb_channels, i_bytes_per_frame );
 235
 236         /* Finalize aout buffer */
 237         p_out_buf->i_nb_samples = i_out;
 238         p_out_buf->i_dts =
 239         p_out_buf->i_pts = date_Get( &p_sys->end_date );
 240         p_out_buf->i_length = date_Increment( &p_sys->end_date,
 241                                       p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 242
 243         p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 244             i_nb_channels * sizeof(int32_t);
 245     }
 246
 247     /* Buffer i_filter_wing * 2 samples for next time */
 248     if( p_sys->i_old_wing )
 249     {
 250         size_t newsize = p_sys->i_old_wing * 2 * i_bytes_per_frame;
 251         if( newsize > p_sys->i_buf_size )
 252         {
 253             free( p_sys->p_buf );
 254             p_sys->p_buf = malloc( newsize );
 255             if( p_sys->p_buf != NULL )
 256                 p_sys->i_buf_size = newsize;
 257             else
 258             {
 259                 p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
 260                 block_Release( p_in_buf );
 261                 return p_out_buf;
 262             }
 263         }
 264         memcpy( p_sys->p_buf,
 265                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 266                 i_nb_channels, (2 * p_sys->i_old_wing) *
 267                 p_filter->fmt_in.audio.i_bytes_per_frame );
 268     }
 269
 270 #if 0
 271     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 272              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 273 #endif
 274
 275     block_Release( p_in_buf );
 276     return p_out_buf;
 277 }
 278
 279 /*****************************************************************************
 280  * OpenFilter:
 281  *****************************************************************************/
 282 static int OpenFilter( vlc_object_t *p_this )
 283 {
 284     filter_t *p_filter = (filter_t *)p_this;
 285     filter_sys_t *p_sys;
 286     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 287
 288     if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
 289       || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
 290       || p_filter->fmt_in.audio.i_physical_channels
 291               != p_filter->fmt_out.audio.i_physical_channels
 292       || p_filter->fmt_in.audio.i_original_channels
 293               != p_filter->fmt_out.audio.i_original_channels
 294       || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
 295     {
 296         return VLC_EGENERIC;
 297     }
 298
 299 #if !defined( SYS_DARWIN )
 300     if( !var_InheritBool( p_this, "hq-resampling" ) )
 301     {
 302         return VLC_EGENERIC;
 303     }
 304 #endif
 305
 306     /* Allocate the memory needed to store the module's structure */
 307     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 308     if( p_sys == NULL )
 309         return VLC_ENOMEM;
 310
 311     p_sys->p_buf = NULL;
 312     p_sys->i_buf_size = 0;
 313
 314     p_sys->i_old_wing = 0;
 315     p_sys->b_first = true;
 316     p_filter->pf_audio_filter = Resample;
 317
 318     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 319              (char *)&p_filter->fmt_in.i_codec,
 320              p_filter->fmt_in.audio.i_rate,
 321              p_filter->fmt_in.audio.i_channels,
 322              (char *)&p_filter->fmt_out.i_codec,
 323              p_filter->fmt_out.audio.i_rate,
 324              p_filter->fmt_out.audio.i_channels);
 325
 326     p_filter->fmt_out = p_filter->fmt_in;
 327     p_filter->fmt_out.audio.i_rate = i_out_rate;
 328
 329     return 0;
 330 }
 331
 332 /*****************************************************************************
 333  * CloseFilter : deallocate data structures
 334  *****************************************************************************/
 335 static void CloseFilter( vlc_object_t *p_this )
 336 {
 337     filter_t *p_filter = (filter_t *)p_this;
 338     free( p_filter->p_sys->p_buf );
 339     free( p_filter->p_sys );
 340 }
 341
 342 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 343                             float *p_out, uint32_t ui_remainder,
 344                             uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 345 {
 346     const float *Hp, *Hdp, *End;
 347     float t, temp;
 348     uint32_t ui_linear_remainder;
 349     int i;
 350
 351     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 352     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 353
 354     End = &Imp[Nwing];
 355
 356     ui_linear_remainder = (ui_remainder<<Nhc) -
 357                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 358
 359     if (Inc == 1)               /* If doing right wing...              */
 360     {                           /* ...drop extra coeff, so when Ph is  */
 361         End--;                  /*    0.5, we don't do too many mult's */
 362         if (ui_remainder == 0)  /* If the phase is zero...           */
 363         {                       /* ...then we've already skipped the */
 364             Hp += Npc;          /*    first sample, so we must also  */
 365             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 366         }
 367     }
 368
 369     while (Hp < End) {
 370         t = *Hp;                /* Get filter coeff */
 371                                 /* t is now interp'd filter coeff */
 372         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 373         for( i = 0; i < i_nb_channels; i++ )
 374         {
 375             temp = t;
 376             temp *= *(p_in+i);  /* Mult coeff by input sample */
 377             *(p_out+i) += temp; /* The filter output */
 378         }
 379         Hdp += Npc;             /* Filter coeff differences step */
 380         Hp += Npc;              /* Filter coeff step */
 381         p_in += (Inc * i_nb_channels); /* Input signal step */
 382     }
 383 }
 384
 385 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 386                            float *p_out, uint32_t ui_remainder,
 387                            uint32_t ui_output_rate, uint32_t ui_input_rate,
 388                            int16_t Inc, int i_nb_channels )
 389 {
 390     const float *Hp, *Hdp, *End;
 391     float t, temp;
 392     uint32_t ui_linear_remainder;
 393     int i, ui_counter = 0;
 394
 395     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 396     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 397
 398     End = &Imp[Nwing];
 399
 400     if (Inc == 1)               /* If doing right wing...              */
 401     {                           /* ...drop extra coeff, so when Ph is  */
 402         End--;                  /*    0.5, we don't do too many mult's */
 403         if (ui_remainder == 0)  /* If the phase is zero...           */
 404         {                       /* ...then we've already skipped the */
 405             Hp = Imp +          /* first sample, so we must also  */
 406                   (ui_output_rate << Nhc) / ui_input_rate;
 407             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 408                   (ui_output_rate << Nhc) / ui_input_rate;
 409             ui_counter++;
 410         }
 411     }
 412
 413     while (Hp < End) {
 414         t = *Hp;                /* Get filter coeff */
 415                                 /* t is now interp'd filter coeff */
 416         ui_linear_remainder =
 417           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 418           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 419           ui_input_rate * ui_input_rate;
 420         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 421         for( i = 0; i < i_nb_channels; i++ )
 422         {
 423             temp = t;
 424             temp *= *(p_in+i);  /* Mult coeff by input sample */
 425             *(p_out+i) += temp; /* The filter output */
 426         }
 427
 428         ui_counter++;
 429
 430         /* Filter coeff step */
 431         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 432                     / ui_input_rate;
 433         /* Filter coeff differences step */
 434         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 435                      / ui_input_rate;
 436
 437         p_in += (Inc * i_nb_channels); /* Input signal step */
 438     }
 439 }
 440
 441 static int ReallocBuffer( block_t **pp_out_buf,
 442                           float **pp_out, size_t i_out,
 443                           int i_nb_channels, int i_bytes_per_frame )
 444 {
 445     if( i_out < (*pp_out_buf)->i_buffer/i_bytes_per_frame )
 446         return VLC_SUCCESS;
 447
 448     /* It may happen when the wing size changes */
 449     const unsigned i_extra_frame = 256;
 450     *pp_out_buf = block_Realloc( *pp_out_buf, 0,
 451                                  (*pp_out_buf)->i_buffer +
 452                                     i_extra_frame * i_bytes_per_frame );
 453     if( !*pp_out_buf )
 454         return VLC_EGENERIC;
 455
 456     *pp_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
 457     memset( *pp_out, 0, i_extra_frame * i_bytes_per_frame );
 458     return VLC_SUCCESS;
 459 }
 460
 461 static void ResampleFloat( filter_t *p_filter,
 462                            block_t **pp_out_buf,  size_t *pi_out,
 463                            float **pp_in,
 464                            int i_in, int i_in_end,
 465                            double d_factor, bool b_factor_old,
 466                            int i_nb_channels, int i_bytes_per_frame )
 467 {
 468     filter_sys_t *p_sys = p_filter->p_sys;
 469
 470     float *p_in = *pp_in;
 471     size_t i_out = *pi_out;
 472     float *p_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
 473
 474     for( ; i_in < i_in_end; i_in++ )
 475     {
 476         if( b_factor_old && d_factor == 1 )
 477         {
 478             if( ReallocBuffer( pp_out_buf, &p_out,
 479                                i_out, i_nb_channels, i_bytes_per_frame ) )
 480                 return;
 481             /* Just copy the samples */
 482             memcpy( p_out, p_in, i_bytes_per_frame );
 483             p_in += i_nb_channels;
 484             p_out += i_nb_channels;
 485             i_out++;
 486             continue;
 487         }
 488
 489         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 490         {
 491             if( ReallocBuffer( pp_out_buf, &p_out,
 492                                i_out, i_nb_channels, i_bytes_per_frame ) )
 493                 return;
 494
 495             if( d_factor >= 1 )
 496             {
 497                 /* FilterFloatUP() is faster if we can use it */
 498
 499                 /* Perform left-wing inner product */
 500                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 501                                SMALL_FILTER_NWING, p_in, p_out,
 502                                p_sys->i_remainder,
 503                                p_filter->fmt_out.audio.i_rate,
 504                                -1, i_nb_channels );
 505                 /* Perform right-wing inner product */
 506                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 507                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 508                                p_filter->fmt_out.audio.i_rate -
 509                                p_sys->i_remainder,
 510                                p_filter->fmt_out.audio.i_rate,
 511                                1, i_nb_channels );
 512
 513 #if 0
 514                 /* Normalize for unity filter gain */
 515                 for( i = 0; i < i_nb_channels; i++ )
 516                 {
 517                     *(p_out+i) *= d_old_scale_factor;
 518                 }
 519 #endif
 520             }
 521             else
 522             {
 523                 /* Perform left-wing inner product */
 524                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 525                                SMALL_FILTER_NWING, p_in, p_out,
 526                                p_sys->i_remainder,
 527                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 528                                -1, i_nb_channels );
 529                 /* Perform right-wing inner product */
 530                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 531                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 532                                p_filter->fmt_out.audio.i_rate -
 533                                p_sys->i_remainder,
 534                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 535                                1, i_nb_channels );
 536             }
 537
 538             p_out += i_nb_channels;
 539             i_out++;
 540
 541             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 542         }
 543
 544         p_in += i_nb_channels;
 545         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 546     }
 547
 548     *pp_in  = p_in;
 549     *pi_out = i_out;
 550 }
 551
 552