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     {
 151         block_Release( p_in_buf );
 152         return NULL;
 153     }
 154     float *p_out = (float *)p_out_buf->p_buffer;
 155
 156     if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
 157     {
 158         /* Continuity in sound samples has been broken, we'd better reset
 159          * everything. */
 160         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 161         p_sys->i_remainder = 0;
 162         date_Init( &p_sys->end_date, i_out_rate, 1 );
 163         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 164         p_sys->d_old_factor = 1;
 165         p_sys->i_old_wing   = 0;
 166         p_sys->b_first = false;
 167     }
 168
 169     size_t i_in_nb = p_in_buf->i_nb_samples;
 170     size_t i_in, i_out = 0;
 171     double d_factor, d_scale_factor, d_old_scale_factor;
 172     size_t i_filter_wing;
 173
 174 #if 0
 175     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 176              p_sys->i_old_rate, p_sys->d_old_factor,
 177              p_sys->i_old_wing, i_in_nb );
 178 #endif
 179
 180     /* Same format in and out... */
 181     assert( p_filter->fmt_in.audio.i_bytes_per_frame == i_bytes_per_frame );
 182
 183     /* Prepare the source buffer */
 184     if( p_sys->i_old_wing )
 185     {   /* Copy all our samples in p_in_buf */
 186         /* Normally, there should be enough room for the old wing in the
 187          * buffer head room. Otherwise, we need to copy memory anyway. */
 188         p_in_buf = block_Realloc( p_in_buf,
 189                                   p_sys->i_old_wing * 2 * i_bytes_per_frame,
 190                                   p_in_buf->i_buffer );
 191         if( unlikely(p_in_buf == NULL) )
 192             return NULL;
 193         memcpy( p_in_buf->p_buffer, p_sys->p_buf,
 194                 p_sys->i_old_wing * 2 * i_bytes_per_frame );
 195     }
 196     i_in_nb += (p_sys->i_old_wing * 2);
 197     float *p_in = (float *)p_in_buf->p_buffer;
 198     const float *p_in_orig = p_in;
 199
 200     /* Make sure the output buffer is reset */
 201     memset( p_out, 0, p_out_buf->i_buffer );
 202
 203     /* Calculate the new length of the filter wing */
 204     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 205     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 206
 207     /* Account for increased filter gain when using factors less than 1 */
 208     d_old_scale_factor = SMALL_FILTER_SCALE *
 209         p_sys->d_old_factor + 0.5;
 210     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 211
 212     /* Apply the old rate until we have enough samples for the new one */
 213     i_in = p_sys->i_old_wing;
 214     p_in += p_sys->i_old_wing * i_nb_channels;
 215     for( ; i_in < i_filter_wing &&
 216            (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
 217     {
 218         if( p_sys->d_old_factor == 1 )
 219         {
 220             /* Just copy the samples */
 221             memcpy( p_out, p_in, i_bytes_per_frame );
 222             p_in += i_nb_channels;
 223             p_out += i_nb_channels;
 224             i_out++;
 225             continue;
 226         }
 227
 228         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 229         {
 230             if( p_out_buf->i_buffer/i_bytes_per_frame <= i_out )
 231                 break;
 232
 233             if( p_sys->d_old_factor >= 1 )
 234             {
 235                 /* FilterFloatUP() is faster if we can use it */
 236
 237                 /* Perform left-wing inner product */
 238                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 239                                SMALL_FILTER_NWING, p_in, p_out,
 240                                p_sys->i_remainder,
 241                                p_filter->fmt_out.audio.i_rate,
 242                                -1, i_nb_channels );
 243                 /* Perform right-wing inner product */
 244                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 245                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 246                                p_filter->fmt_out.audio.i_rate -
 247                                p_sys->i_remainder,
 248                                p_filter->fmt_out.audio.i_rate,
 249                                1, i_nb_channels );
 250
 251 #if 0
 252                 /* Normalize for unity filter gain */
 253                 for( i = 0; i < i_nb_channels; i++ )
 254                 {
 255                     *(p_out+i) *= d_old_scale_factor;
 256                 }
 257 #endif
 258             }
 259             else
 260             {
 261                 /* Perform left-wing inner product */
 262                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 263                                SMALL_FILTER_NWING, p_in, p_out,
 264                                p_sys->i_remainder,
 265                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 266                                -1, i_nb_channels );
 267                 /* Perform right-wing inner product */
 268                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 269                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 270                                p_filter->fmt_out.audio.i_rate -
 271                                p_sys->i_remainder,
 272                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 273                                1, i_nb_channels );
 274             }
 275
 276             p_out += i_nb_channels;
 277             i_out++;
 278
 279             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 280         }
 281
 282         p_in += i_nb_channels;
 283         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 284     }
 285
 286     /* Apply the new rate for the rest of the samples */
 287     if( i_in < i_in_nb - i_filter_wing )
 288     {
 289         p_sys->d_old_factor = d_factor;
 290         p_sys->i_old_wing   = i_filter_wing;
 291     }
 292     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 293     {
 294         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 295         {
 296             if( p_out_buf->i_buffer/i_bytes_per_frame <= i_out )
 297                 break;
 298
 299             assert( i_out < p_out_buf->i_buffer/i_bytes_per_frame );
 300             if( d_factor >= 1 )
 301             {
 302                 /* FilterFloatUP() is faster if we can use it */
 303
 304                 /* Perform left-wing inner product */
 305                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 306                                SMALL_FILTER_NWING, p_in, p_out,
 307                                p_sys->i_remainder,
 308                                p_filter->fmt_out.audio.i_rate,
 309                                -1, i_nb_channels );
 310
 311                 /* Perform right-wing inner product */
 312                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 313                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 314                                p_filter->fmt_out.audio.i_rate -
 315                                p_sys->i_remainder,
 316                                p_filter->fmt_out.audio.i_rate,
 317                                1, i_nb_channels );
 318
 319 #if 0
 320                 /* Normalize for unity filter gain */
 321                 for( int i = 0; i < i_nb_channels; i++ )
 322                 {
 323                     *(p_out+i) *= d_old_scale_factor;
 324                 }
 325 #endif
 326             }
 327             else
 328             {
 329                 /* Perform left-wing inner product */
 330                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 331                                SMALL_FILTER_NWING, p_in, p_out,
 332                                p_sys->i_remainder,
 333                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 334                                -1, i_nb_channels );
 335                 /* Perform right-wing inner product */
 336                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 337                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 338                                p_filter->fmt_out.audio.i_rate -
 339                                p_sys->i_remainder,
 340                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 341                                1, i_nb_channels );
 342             }
 343
 344             p_out += i_nb_channels;
 345             i_out++;
 346
 347             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 348         }
 349
 350         p_in += i_nb_channels;
 351         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 352     }
 353
 354     /* Finalize aout buffer */
 355     p_out_buf->i_nb_samples = i_out;
 356     p_out_buf->i_pts = date_Get( &p_sys->end_date );
 357     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 358                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 359
 360     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 361         i_nb_channels * sizeof(int32_t);
 362
 363     /* Buffer i_filter_wing * 2 samples for next time */
 364     if( p_sys->i_old_wing )
 365     {
 366         size_t newsize = p_sys->i_old_wing * 2 * i_bytes_per_frame;
 367         if( newsize > p_sys->i_buf_size )
 368         {
 369             free( p_sys->p_buf );
 370             p_sys->p_buf = malloc( newsize );
 371             if( p_sys->p_buf != NULL )
 372                 p_sys->i_buf_size = newsize;
 373             else
 374             {
 375                 p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
 376                 block_Release( p_in_buf );
 377                 return p_out_buf;
 378             }
 379         }
 380         memcpy( p_sys->p_buf,
 381                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 382                 i_nb_channels, (2 * p_sys->i_old_wing) *
 383                 p_filter->fmt_in.audio.i_bytes_per_frame );
 384     }
 385
 386 #if 0
 387     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 388              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 389 #endif
 390
 391     block_Release( p_in_buf );
 392     return p_out_buf;
 393 }
 394
 395 /*****************************************************************************
 396  * OpenFilter:
 397  *****************************************************************************/
 398 static int OpenFilter( vlc_object_t *p_this )
 399 {
 400     filter_t *p_filter = (filter_t *)p_this;
 401     filter_sys_t *p_sys;
 402     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 403
 404     if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
 405       || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
 406       || p_filter->fmt_in.audio.i_physical_channels
 407               != p_filter->fmt_out.audio.i_physical_channels
 408       || p_filter->fmt_in.audio.i_original_channels
 409               != p_filter->fmt_out.audio.i_original_channels
 410       || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
 411     {
 412         return VLC_EGENERIC;
 413     }
 414
 415 #if !defined( SYS_DARWIN )
 416     if( !var_InheritInteger( p_this, "hq-resampling" ) )
 417     {
 418         return VLC_EGENERIC;
 419     }
 420 #endif
 421
 422     /* Allocate the memory needed to store the module's structure */
 423     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 424     if( p_sys == NULL )
 425         return VLC_ENOMEM;
 426
 427     p_sys->p_buf = NULL;
 428     p_sys->i_buf_size = 0;
 429
 430     p_sys->i_old_wing = 0;
 431     p_sys->b_first = true;
 432     p_filter->pf_audio_filter = Resample;
 433
 434     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 435              (char *)&p_filter->fmt_in.i_codec,
 436              p_filter->fmt_in.audio.i_rate,
 437              p_filter->fmt_in.audio.i_channels,
 438              (char *)&p_filter->fmt_out.i_codec,
 439              p_filter->fmt_out.audio.i_rate,
 440              p_filter->fmt_out.audio.i_channels);
 441
 442     p_filter->fmt_out = p_filter->fmt_in;
 443     p_filter->fmt_out.audio.i_rate = i_out_rate;
 444
 445     return 0;
 446 }
 447
 448 /*****************************************************************************
 449  * CloseFilter : deallocate data structures
 450  *****************************************************************************/
 451 static void CloseFilter( vlc_object_t *p_this )
 452 {
 453     filter_t *p_filter = (filter_t *)p_this;
 454     free( p_filter->p_sys->p_buf );
 455     free( p_filter->p_sys );
 456 }
 457
 458 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 459                     float *p_out, uint32_t ui_remainder,
 460                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 461 {
 462     const float *Hp, *Hdp, *End;
 463     float t, temp;
 464     uint32_t ui_linear_remainder;
 465     int i;
 466
 467     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 468     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 469
 470     End = &Imp[Nwing];
 471
 472     ui_linear_remainder = (ui_remainder<<Nhc) -
 473                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 474
 475     if (Inc == 1)               /* If doing right wing...              */
 476     {                           /* ...drop extra coeff, so when Ph is  */
 477         End--;                  /*    0.5, we don't do too many mult's */
 478         if (ui_remainder == 0)  /* If the phase is zero...           */
 479         {                       /* ...then we've already skipped the */
 480             Hp += Npc;          /*    first sample, so we must also  */
 481             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 482         }
 483     }
 484
 485     while (Hp < End) {
 486         t = *Hp;                /* Get filter coeff */
 487                                 /* t is now interp'd filter coeff */
 488         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 489         for( i = 0; i < i_nb_channels; i++ )
 490         {
 491             temp = t;
 492             temp *= *(p_in+i);  /* Mult coeff by input sample */
 493             *(p_out+i) += temp; /* The filter output */
 494         }
 495         Hdp += Npc;             /* Filter coeff differences step */
 496         Hp += Npc;              /* Filter coeff step */
 497         p_in += (Inc * i_nb_channels); /* Input signal step */
 498     }
 499 }
 500
 501 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 502                     float *p_out, uint32_t ui_remainder,
 503                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 504                     int16_t Inc, int i_nb_channels )
 505 {
 506     const float *Hp, *Hdp, *End;
 507     float t, temp;
 508     uint32_t ui_linear_remainder;
 509     int i, ui_counter = 0;
 510
 511     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 512     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 513
 514     End = &Imp[Nwing];
 515
 516     if (Inc == 1)               /* If doing right wing...              */
 517     {                           /* ...drop extra coeff, so when Ph is  */
 518         End--;                  /*    0.5, we don't do too many mult's */
 519         if (ui_remainder == 0)  /* If the phase is zero...           */
 520         {                       /* ...then we've already skipped the */
 521             Hp = Imp +          /* first sample, so we must also  */
 522                   (ui_output_rate << Nhc) / ui_input_rate;
 523             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 524                   (ui_output_rate << Nhc) / ui_input_rate;
 525             ui_counter++;
 526         }
 527     }
 528
 529     while (Hp < End) {
 530         t = *Hp;                /* Get filter coeff */
 531                                 /* t is now interp'd filter coeff */
 532         ui_linear_remainder =
 533           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 534           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 535           ui_input_rate * ui_input_rate;
 536         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 537         for( i = 0; i < i_nb_channels; i++ )
 538         {
 539             temp = t;
 540             temp *= *(p_in+i);  /* Mult coeff by input sample */
 541             *(p_out+i) += temp; /* The filter output */
 542         }
 543
 544         ui_counter++;
 545
 546         /* Filter coeff step */
 547         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 548                     / ui_input_rate;
 549         /* Filter coeff differences step */
 550         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 551                      / ui_input_rate;
 552
 553         p_in += (Inc * i_nb_channels); /* Input signal step */
 554     }
 555 }