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