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 VLC authors and VideoLAN
   5  * $Id$
   6  *
   7  * Authors: Gildas Bazin <gbazin@netcourrier.com>
   8  *
   9  * This program is free software; you can redistribute it and/or modify it
  10  * under the terms of the GNU Lesser General Public License as published by
  11  * the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU Lesser General Public License
  20  * along with this program; if not, write to the Free Software Foundation,
  21  * 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     size_t 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 converter", 20 )
  91     set_callbacks( OpenFilter, CloseFilter )
  92
  93     add_submodule()
  94     set_capability( "audio resampler", 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
 155     if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
 156     {
 157         /* Continuity in sound samples has been broken, we'd better reset
 158          * everything. */
 159         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 160         p_sys->i_remainder = 0;
 161         date_Init( &p_sys->end_date, i_out_rate, 1 );
 162         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 163         p_sys->d_old_factor = 1;
 164         p_sys->i_old_wing   = 0;
 165         p_sys->b_first = false;
 166     }
 167
 168     size_t i_in_nb = p_in_buf->i_nb_samples;
 169     size_t i_in, i_out = 0;
 170     double d_factor, d_scale_factor, d_old_scale_factor;
 171     size_t i_filter_wing;
 172
 173 #if 0
 174     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 175              p_sys->i_old_rate, p_sys->d_old_factor,
 176              p_sys->i_old_wing, i_in_nb );
 177 #endif
 178
 179     /* Same format in and out... */
 180     assert( p_filter->fmt_in.audio.i_bytes_per_frame == i_bytes_per_frame );
 181
 182     /* Prepare the source buffer */
 183     if( p_sys->i_old_wing )
 184     {   /* Copy all our samples in p_in_buf */
 185         /* Normally, there should be enough room for the old wing in the
 186          * buffer head room. Otherwise, we need to copy memory anyway. */
 187         p_in_buf = block_Realloc( p_in_buf,
 188                                   p_sys->i_old_wing * 2 * i_bytes_per_frame,
 189                                   p_in_buf->i_buffer );
 190         if( unlikely(p_in_buf == NULL) )
 191             return NULL;
 192         memcpy( p_in_buf->p_buffer, p_sys->p_buf,
 193                 p_sys->i_old_wing * 2 * i_bytes_per_frame );
 194     }
 195     i_in_nb += (p_sys->i_old_wing * 2);
 196     float *p_in = (float *)p_in_buf->p_buffer;
 197     const float *p_in_orig = p_in;
 198
 199     /* Make sure the output buffer is reset */
 200     memset( p_out_buf->p_buffer, 0, p_out_buf->i_buffer );
 201
 202     /* Calculate the new length of the filter wing */
 203     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 204     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 205
 206     /* Account for increased filter gain when using factors less than 1 */
 207     d_old_scale_factor = SMALL_FILTER_SCALE *
 208         p_sys->d_old_factor + 0.5;
 209     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 210
 211     /* Apply the old rate until we have enough samples for the new one */
 212     i_in = p_sys->i_old_wing;
 213     p_in += p_sys->i_old_wing * i_nb_channels;
 214
 215     size_t i_old_in_end = 0;
 216     if( p_sys->i_old_wing <= i_in_nb )
 217         i_old_in_end = __MIN( i_filter_wing, i_in_nb - p_sys->i_old_wing );
 218
 219     ResampleFloat( p_filter,
 220                    &p_out_buf, &i_out, &p_in,
 221                    i_in, i_old_in_end,
 222                    p_sys->d_old_factor, true,
 223                    i_nb_channels, i_bytes_per_frame );
 224     i_in = __MAX( i_in, i_old_in_end );
 225
 226     /* Apply the new rate for the rest of the samples */
 227     if( i_in < i_in_nb - i_filter_wing )
 228     {
 229         p_sys->d_old_factor = d_factor;
 230         p_sys->i_old_wing   = i_filter_wing;
 231     }
 232     if( p_out_buf )
 233     {
 234         ResampleFloat( p_filter,
 235                        &p_out_buf, &i_out, &p_in,
 236                        i_in, i_in_nb - i_filter_wing,
 237                        d_factor, false,
 238                        i_nb_channels, i_bytes_per_frame );
 239
 240         /* Finalize aout buffer */
 241         p_out_buf->i_nb_samples = i_out;
 242         p_out_buf->i_dts =
 243         p_out_buf->i_pts = date_Get( &p_sys->end_date );
 244         p_out_buf->i_length = date_Increment( &p_sys->end_date,
 245                                       p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 246
 247         p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 248             i_nb_channels * sizeof(int32_t);
 249     }
 250
 251     /* Buffer i_filter_wing * 2 samples for next time */
 252     if( p_sys->i_old_wing )
 253     {
 254         size_t newsize = p_sys->i_old_wing * 2 * i_bytes_per_frame;
 255         if( newsize > p_sys->i_buf_size )
 256         {
 257             free( p_sys->p_buf );
 258             p_sys->p_buf = malloc( newsize );
 259             if( p_sys->p_buf != NULL )
 260                 p_sys->i_buf_size = newsize;
 261             else
 262             {
 263                 p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
 264                 block_Release( p_in_buf );
 265                 return p_out_buf;
 266             }
 267         }
 268         memcpy( p_sys->p_buf,
 269                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 270                 i_nb_channels, (2 * p_sys->i_old_wing) *
 271                 p_filter->fmt_in.audio.i_bytes_per_frame );
 272     }
 273
 274 #if 0
 275     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 276              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 277 #endif
 278
 279     block_Release( p_in_buf );
 280     return p_out_buf;
 281 }
 282
 283 /*****************************************************************************
 284  * OpenFilter:
 285  *****************************************************************************/
 286 static int OpenFilter( vlc_object_t *p_this )
 287 {
 288     filter_t *p_filter = (filter_t *)p_this;
 289     filter_sys_t *p_sys;
 290     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 291
 292     if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
 293       || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
 294       || p_filter->fmt_in.audio.i_physical_channels
 295               != p_filter->fmt_out.audio.i_physical_channels
 296       || p_filter->fmt_in.audio.i_original_channels
 297               != p_filter->fmt_out.audio.i_original_channels
 298       || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
 299     {
 300         return VLC_EGENERIC;
 301     }
 302
 303 #if !defined( SYS_DARWIN )
 304     if( !var_InheritBool( p_this, "hq-resampling" ) )
 305     {
 306         return VLC_EGENERIC;
 307     }
 308 #endif
 309
 310     /* Allocate the memory needed to store the module's structure */
 311     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 312     if( p_sys == NULL )
 313         return VLC_ENOMEM;
 314
 315     p_sys->p_buf = NULL;
 316     p_sys->i_buf_size = 0;
 317
 318     p_sys->i_old_wing = 0;
 319     p_sys->b_first = true;
 320     p_filter->pf_audio_filter = Resample;
 321
 322     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 323              (char *)&p_filter->fmt_in.i_codec,
 324              p_filter->fmt_in.audio.i_rate,
 325              p_filter->fmt_in.audio.i_channels,
 326              (char *)&p_filter->fmt_out.i_codec,
 327              p_filter->fmt_out.audio.i_rate,
 328              p_filter->fmt_out.audio.i_channels);
 329
 330     p_filter->fmt_out = p_filter->fmt_in;
 331     p_filter->fmt_out.audio.i_rate = i_out_rate;
 332
 333     return 0;
 334 }
 335
 336 /*****************************************************************************
 337  * CloseFilter : deallocate data structures
 338  *****************************************************************************/
 339 static void CloseFilter( vlc_object_t *p_this )
 340 {
 341     filter_t *p_filter = (filter_t *)p_this;
 342     free( p_filter->p_sys->p_buf );
 343     free( p_filter->p_sys );
 344 }
 345
 346 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 347                             float *p_out, uint32_t ui_remainder,
 348                             uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 349 {
 350     const float *Hp, *Hdp, *End;
 351     float t, temp;
 352     uint32_t ui_linear_remainder;
 353     int i;
 354
 355     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 356     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 357
 358     End = &Imp[Nwing];
 359
 360     ui_linear_remainder = (ui_remainder<<Nhc) -
 361                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 362
 363     if (Inc == 1)               /* If doing right wing...              */
 364     {                           /* ...drop extra coeff, so when Ph is  */
 365         End--;                  /*    0.5, we don't do too many mult's */
 366         if (ui_remainder == 0)  /* If the phase is zero...           */
 367         {                       /* ...then we've already skipped the */
 368             Hp += Npc;          /*    first sample, so we must also  */
 369             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 370         }
 371     }
 372
 373     while (Hp < End) {
 374         t = *Hp;                /* Get filter coeff */
 375                                 /* t is now interp'd filter coeff */
 376         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 377         for( i = 0; i < i_nb_channels; i++ )
 378         {
 379             temp = t;
 380             temp *= *(p_in+i);  /* Mult coeff by input sample */
 381             *(p_out+i) += temp; /* The filter output */
 382         }
 383         Hdp += Npc;             /* Filter coeff differences step */
 384         Hp += Npc;              /* Filter coeff step */
 385         p_in += (Inc * i_nb_channels); /* Input signal step */
 386     }
 387 }
 388
 389 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 390                            float *p_out, uint32_t ui_remainder,
 391                            uint32_t ui_output_rate, uint32_t ui_input_rate,
 392                            int16_t Inc, int i_nb_channels )
 393 {
 394     const float *Hp, *Hdp, *End;
 395     float t, temp;
 396     uint32_t ui_linear_remainder;
 397     int i, ui_counter = 0;
 398
 399     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 400     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 401
 402     End = &Imp[Nwing];
 403
 404     if (Inc == 1)               /* If doing right wing...              */
 405     {                           /* ...drop extra coeff, so when Ph is  */
 406         End--;                  /*    0.5, we don't do too many mult's */
 407         if (ui_remainder == 0)  /* If the phase is zero...           */
 408         {                       /* ...then we've already skipped the */
 409             Hp = Imp +          /* first sample, so we must also  */
 410                   (ui_output_rate << Nhc) / ui_input_rate;
 411             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 412                   (ui_output_rate << Nhc) / ui_input_rate;
 413             ui_counter++;
 414         }
 415     }
 416
 417     while (Hp < End) {
 418         t = *Hp;                /* Get filter coeff */
 419                                 /* t is now interp'd filter coeff */
 420         ui_linear_remainder =
 421           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 422           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 423           ui_input_rate * ui_input_rate;
 424         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 425         for( i = 0; i < i_nb_channels; i++ )
 426         {
 427             temp = t;
 428             temp *= *(p_in+i);  /* Mult coeff by input sample */
 429             *(p_out+i) += temp; /* The filter output */
 430         }
 431
 432         ui_counter++;
 433
 434         /* Filter coeff step */
 435         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 436                     / ui_input_rate;
 437         /* Filter coeff differences step */
 438         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 439                      / ui_input_rate;
 440
 441         p_in += (Inc * i_nb_channels); /* Input signal step */
 442     }
 443 }
 444
 445 static int ReallocBuffer( block_t **pp_out_buf,
 446                           float **pp_out, size_t i_out,
 447                           int i_nb_channels, int i_bytes_per_frame )
 448 {
 449     if( i_out < (*pp_out_buf)->i_buffer/i_bytes_per_frame )
 450         return VLC_SUCCESS;
 451
 452     /* It may happen when the wing size changes */
 453     const unsigned i_extra_frame = 256;
 454     *pp_out_buf = block_Realloc( *pp_out_buf, 0,
 455                                  (*pp_out_buf)->i_buffer +
 456                                     i_extra_frame * i_bytes_per_frame );
 457     if( !*pp_out_buf )
 458         return VLC_EGENERIC;
 459
 460     *pp_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
 461     memset( *pp_out, 0, i_extra_frame * i_bytes_per_frame );
 462     return VLC_SUCCESS;
 463 }
 464
 465 static void ResampleFloat( filter_t *p_filter,
 466                            block_t **pp_out_buf,  size_t *pi_out,
 467                            float **pp_in,
 468                            int i_in, int i_in_end,
 469                            double d_factor, bool b_factor_old,
 470                            int i_nb_channels, int i_bytes_per_frame )
 471 {
 472     filter_sys_t *p_sys = p_filter->p_sys;
 473
 474     float *p_in = *pp_in;
 475     size_t i_out = *pi_out;
 476     float *p_out = (float*)(*pp_out_buf)->p_buffer + i_out * i_nb_channels;
 477
 478     for( ; i_in < i_in_end; i_in++ )
 479     {
 480         if( b_factor_old && d_factor == 1 )
 481         {
 482             if( ReallocBuffer( pp_out_buf, &p_out,
 483                                i_out, i_nb_channels, i_bytes_per_frame ) )
 484                 return;
 485             /* Just copy the samples */
 486             memcpy( p_out, p_in, i_bytes_per_frame );
 487             p_in += i_nb_channels;
 488             p_out += i_nb_channels;
 489             i_out++;
 490             continue;
 491         }
 492
 493         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 494         {
 495             if( ReallocBuffer( pp_out_buf, &p_out,
 496                                i_out, i_nb_channels, i_bytes_per_frame ) )
 497                 return;
 498
 499             if( d_factor >= 1 )
 500             {
 501                 /* FilterFloatUP() is faster if we can use it */
 502
 503                 /* Perform left-wing inner product */
 504                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 505                                SMALL_FILTER_NWING, p_in, p_out,
 506                                p_sys->i_remainder,
 507                                p_filter->fmt_out.audio.i_rate,
 508                                -1, i_nb_channels );
 509                 /* Perform right-wing inner product */
 510                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 511                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 512                                p_filter->fmt_out.audio.i_rate -
 513                                p_sys->i_remainder,
 514                                p_filter->fmt_out.audio.i_rate,
 515                                1, i_nb_channels );
 516
 517 #if 0
 518                 /* Normalize for unity filter gain */
 519                 for( i = 0; i < i_nb_channels; i++ )
 520                 {
 521                     *(p_out+i) *= d_old_scale_factor;
 522                 }
 523 #endif
 524             }
 525             else
 526             {
 527                 /* Perform left-wing inner product */
 528                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 529                                SMALL_FILTER_NWING, p_in, p_out,
 530                                p_sys->i_remainder,
 531                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 532                                -1, i_nb_channels );
 533                 /* Perform right-wing inner product */
 534                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 535                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 536                                p_filter->fmt_out.audio.i_rate -
 537                                p_sys->i_remainder,
 538                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 539                                1, i_nb_channels );
 540             }
 541
 542             p_out += i_nb_channels;
 543             i_out++;
 544
 545             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 546         }
 547
 548         p_in += i_nb_channels;
 549         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 550     }
 551
 552     *pp_in  = p_in;
 553     *pi_out = i_out;
 554 }
 555
 556