git.sesse.net Git - vlc/blob - modules/audio_filter/resampler/bandlimited.c

   1 /*****************************************************************************
   2  * bandlimited.c : band-limited interpolation resampler
   3  *****************************************************************************
   4  * Copyright (C) 2002, 2006 the VideoLAN team
   5  * $Id$
   6  *
   7  * Authors: Gildas Bazin <gbazin@netcourrier.com>
   8  *
   9  * This program is free software; you can redistribute it and/or modify
  10  * it under the terms of the GNU General Public License as published by
  11  * the Free Software Foundation; either version 2 of the License, or
  12  * (at your option) any later version.
  13  *
  14  * This program is distributed in the hope that it will be useful,
  15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  17  * GNU General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU General Public License
  20  * along with this program; if not, write to the Free Software
  21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  22  *****************************************************************************/
  23
  24 /*****************************************************************************
  25  * Preamble:
  26  *
  27  * This implementation of the band-limited interpolationis based on the
  28  * following paper:
  29  * http://ccrma-www.stanford.edu/~jos/resample/resample.html
  30  *
  31  * It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
  32  * filter is 13 samples.
  33  *
  34  *****************************************************************************/
  35
  36 #ifdef HAVE_CONFIG_H
  37 # include "config.h"
  38 #endif
  39
  40 #include <vlc_common.h>
  41 #include <vlc_plugin.h>
  42 #include <vlc_aout.h>
  43 #include <vlc_filter.h>
  44 #include <vlc_block.h>
  45
  46 #include "bandlimited.h"
  47
  48 /*****************************************************************************
  49  * Local prototypes
  50  *****************************************************************************/
  51
  52 /* audio filter2 */
  53 static int  OpenFilter ( vlc_object_t * );
  54 static void CloseFilter( vlc_object_t * );
  55 static block_t *Resample( filter_t *, block_t * );
  56
  57
  58 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
  59                            float *f_in, float *f_out, uint32_t ui_remainder,
  60                            uint32_t ui_output_rate, int16_t Inc,
  61                            int i_nb_channels );
  62
  63 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
  64                            float *f_in, float *f_out, uint32_t ui_remainder,
  65                            uint32_t ui_output_rate, uint32_t ui_input_rate,
  66                            int16_t Inc, int i_nb_channels );
  67
  68 /*****************************************************************************
  69  * Local structures
  70  *****************************************************************************/
  71 struct filter_sys_t
  72 {
  73     int32_t *p_buf;                        /* this filter introduces a delay */
  74     int i_buf_size;
  75
  76     double d_old_factor;
  77     int i_old_wing;
  78
  79     unsigned int i_remainder;                /* remainder of previous sample */
  80     bool b_first;
  81
  82     date_t end_date;
  83 };
  84
  85 /*****************************************************************************
  86  * Module descriptor
  87  *****************************************************************************/
  88 vlc_module_begin ()
  89     set_category( CAT_AUDIO )
  90     set_subcategory( SUBCAT_AUDIO_MISC )
  91     set_description( N_("Audio filter for band-limited interpolation resampling") )
  92     set_capability( "audio filter2", 20 )
  93     set_callbacks( OpenFilter, CloseFilter )
  94 vlc_module_end ()
  95
  96 /*****************************************************************************
  97  * Resample: convert a buffer
  98  *****************************************************************************/
  99 static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
 100 {
 101     if( !p_in_buf || !p_in_buf->i_nb_samples )
 102     {
 103         if( p_in_buf )
 104             block_Release( p_in_buf );
 105         return NULL;
 106     }
 107
 108     filter_sys_t *p_sys = p_filter->p_sys;
 109     unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
 110     int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 111
 112     /* Check if we really need to run the resampler */
 113     if( i_out_rate == p_filter->fmt_in.audio.i_rate )
 114     {
 115         if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
 116             p_sys->i_old_wing )
 117         {
 118             /* output the whole thing with the samples from last time */
 119             p_in_buf = block_Realloc( p_in_buf,
 120                 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
 121                 p_in_buf->i_buffer );
 122             if( !p_in_buf )
 123                 return NULL;
 124             memcpy( p_in_buf->p_buffer, p_sys->p_buf +
 125                     i_nb_channels * p_sys->i_old_wing,
 126                     p_sys->i_old_wing *
 127                     p_filter->fmt_in.audio.i_bytes_per_frame );
 128
 129             p_in_buf->i_nb_samples += p_sys->i_old_wing;
 130
 131             p_in_buf->i_pts = date_Get( &p_sys->end_date );
 132             p_in_buf->i_length =
 133                 date_Increment( &p_sys->end_date,
 134                                 p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
 135         }
 136         p_sys->i_old_wing = 0;
 137         p_sys->b_first = true;
 138         return p_in_buf;
 139     }
 140
 141     unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 142                                  p_filter->fmt_out.audio.i_bitspersample / 8;
 143     size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
 144               p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
 145             + p_filter->p_sys->i_buf_size;
 146     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
 147     if( !p_out_buf )
 148         return NULL;
 149     float *p_out = (float *)p_out_buf->p_buffer;
 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     int i_in_nb = p_in_buf->i_nb_samples;
 165     int i_in, i_out = 0;
 166     double d_factor, d_scale_factor, d_old_scale_factor;
 167     int 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     /* Prepare the source buffer */
 176     i_in_nb += (p_sys->i_old_wing * 2);
 177
 178     float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
 179          *p_in = p_in_orig;
 180
 181     /* Copy all our samples in p_in */
 182     if( p_sys->i_old_wing )
 183     {
 184         vlc_memcpy( p_in, p_sys->p_buf,
 185                     p_sys->i_old_wing * 2 *
 186                       p_filter->fmt_in.audio.i_bytes_per_frame );
 187     }
 188     /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
 189     vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
 190                 p_in_buf->p_buffer,
 191                 p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
 192     block_Release( p_in_buf );
 193
 194     /* Make sure the output buffer is reset */
 195     memset( p_out, 0, p_out_buf->i_buffer );
 196
 197     /* Calculate the new length of the filter wing */
 198     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 199     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 200
 201     /* Account for increased filter gain when using factors less than 1 */
 202     d_old_scale_factor = SMALL_FILTER_SCALE *
 203         p_sys->d_old_factor + 0.5;
 204     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 205
 206     /* Apply the old rate until we have enough samples for the new one */
 207     i_in = p_sys->i_old_wing;
 208     p_in += p_sys->i_old_wing * i_nb_channels;
 209     for( ; i_in < i_filter_wing &&
 210            (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
 211     {
 212         if( p_sys->d_old_factor == 1 )
 213         {
 214             /* Just copy the samples */
 215             memcpy( p_out, p_in,
 216                     p_filter->fmt_in.audio.i_bytes_per_frame );
 217             p_in += i_nb_channels;
 218             p_out += i_nb_channels;
 219             i_out++;
 220             continue;
 221         }
 222
 223         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 224         {
 225
 226             if( p_sys->d_old_factor >= 1 )
 227             {
 228                 /* FilterFloatUP() is faster if we can use it */
 229
 230                 /* Perform left-wing inner product */
 231                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 232                                SMALL_FILTER_NWING, p_in, p_out,
 233                                p_sys->i_remainder,
 234                                p_filter->fmt_out.audio.i_rate,
 235                                -1, i_nb_channels );
 236                 /* Perform right-wing inner product */
 237                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 238                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 239                                p_filter->fmt_out.audio.i_rate -
 240                                p_sys->i_remainder,
 241                                p_filter->fmt_out.audio.i_rate,
 242                                1, i_nb_channels );
 243
 244 #if 0
 245                 /* Normalize for unity filter gain */
 246                 for( i = 0; i < i_nb_channels; i++ )
 247                 {
 248                     *(p_out+i) *= d_old_scale_factor;
 249                 }
 250 #endif
 251
 252                 /* Sanity check */
 253                 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
 254                     <= (unsigned int)i_out+1 )
 255                 {
 256                     p_out += i_nb_channels;
 257                     i_out++;
 258                     p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 259                     break;
 260                 }
 261             }
 262             else
 263             {
 264                 /* Perform left-wing inner product */
 265                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 266                                SMALL_FILTER_NWING, p_in, p_out,
 267                                p_sys->i_remainder,
 268                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 269                                -1, i_nb_channels );
 270                 /* Perform right-wing inner product */
 271                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 272                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 273                                p_filter->fmt_out.audio.i_rate -
 274                                p_sys->i_remainder,
 275                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 276                                1, i_nb_channels );
 277             }
 278
 279             p_out += i_nb_channels;
 280             i_out++;
 281
 282             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 283         }
 284
 285         p_in += i_nb_channels;
 286         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 287     }
 288
 289     /* Apply the new rate for the rest of the samples */
 290     if( i_in < i_in_nb - i_filter_wing )
 291     {
 292         p_sys->d_old_factor = d_factor;
 293         p_sys->i_old_wing   = i_filter_wing;
 294     }
 295     for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
 296     {
 297         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 298         {
 299
 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                 /* Sanity check */
 327                 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
 328                     <= (unsigned int)i_out+1 )
 329                 {
 330                     p_out += i_nb_channels;
 331                     i_out++;
 332                     p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 333                     break;
 334                 }
 335             }
 336             else
 337             {
 338                 /* Perform left-wing inner product */
 339                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 340                                SMALL_FILTER_NWING, p_in, p_out,
 341                                p_sys->i_remainder,
 342                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 343                                -1, i_nb_channels );
 344                 /* Perform right-wing inner product */
 345                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 346                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 347                                p_filter->fmt_out.audio.i_rate -
 348                                p_sys->i_remainder,
 349                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 350                                1, i_nb_channels );
 351             }
 352
 353             p_out += i_nb_channels;
 354             i_out++;
 355
 356             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 357         }
 358
 359         p_in += i_nb_channels;
 360         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 361     }
 362
 363     /* Buffer i_filter_wing * 2 samples for next time */
 364     if( p_sys->i_old_wing )
 365     {
 366         memcpy( p_sys->p_buf,
 367                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 368                 i_nb_channels, (2 * p_sys->i_old_wing) *
 369                 p_filter->fmt_in.audio.i_bytes_per_frame );
 370     }
 371
 372 #if 0
 373     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 374              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 375 #endif
 376
 377     /* Finalize aout buffer */
 378     p_out_buf->i_nb_samples = i_out;
 379     p_out_buf->i_pts = date_Get( &p_sys->end_date );
 380     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 381                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 382
 383     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 384         i_nb_channels * sizeof(int32_t);
 385     return p_out_buf;
 386 }
 387
 388 /*****************************************************************************
 389  * OpenFilter:
 390  *****************************************************************************/
 391 static int OpenFilter( vlc_object_t *p_this )
 392 {
 393     filter_t *p_filter = (filter_t *)p_this;
 394     filter_sys_t *p_sys;
 395     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 396     double d_factor;
 397     int i_filter_wing;
 398
 399     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 400         p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
 401     {
 402         return VLC_EGENERIC;
 403     }
 404
 405 #if !defined( SYS_DARWIN )
 406     if( !config_GetInt( p_this, "hq-resampling" ) )
 407     {
 408         return VLC_EGENERIC;
 409     }
 410 #endif
 411
 412     /* Allocate the memory needed to store the module's structure */
 413     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 414     if( p_sys == NULL )
 415         return VLC_ENOMEM;
 416
 417     /* Calculate worst case for the length of the filter wing */
 418     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 419     i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
 420                       * __MAX(1.0, 1.0/d_factor) + 10;
 421     p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
 422         sizeof(int32_t) * 2 * i_filter_wing;
 423
 424     /* Allocate enough memory to buffer previous samples */
 425     p_sys->p_buf = malloc( p_sys->i_buf_size );
 426     if( p_sys->p_buf == NULL )
 427     {
 428         free( p_sys );
 429         return VLC_ENOMEM;
 430     }
 431
 432     p_sys->i_old_wing = 0;
 433     p_sys->b_first = true;
 434     p_filter->pf_audio_filter = Resample;
 435
 436     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 437              (char *)&p_filter->fmt_in.i_codec,
 438              p_filter->fmt_in.audio.i_rate,
 439              p_filter->fmt_in.audio.i_channels,
 440              (char *)&p_filter->fmt_out.i_codec,
 441              p_filter->fmt_out.audio.i_rate,
 442              p_filter->fmt_out.audio.i_channels);
 443
 444     p_filter->fmt_out = p_filter->fmt_in;
 445     p_filter->fmt_out.audio.i_rate = i_out_rate;
 446
 447     return 0;
 448 }
 449
 450 /*****************************************************************************
 451  * CloseFilter : deallocate data structures
 452  *****************************************************************************/
 453 static void CloseFilter( vlc_object_t *p_this )
 454 {
 455     filter_t *p_filter = (filter_t *)p_this;
 456     free( p_filter->p_sys->p_buf );
 457     free( p_filter->p_sys );
 458 }
 459
 460 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 461                     float *p_out, uint32_t ui_remainder,
 462                     uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 463 {
 464     const float *Hp, *Hdp, *End;
 465     float t, temp;
 466     uint32_t ui_linear_remainder;
 467     int i;
 468
 469     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 470     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 471
 472     End = &Imp[Nwing];
 473
 474     ui_linear_remainder = (ui_remainder<<Nhc) -
 475                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 476
 477     if (Inc == 1)               /* If doing right wing...              */
 478     {                           /* ...drop extra coeff, so when Ph is  */
 479         End--;                  /*    0.5, we don't do too many mult's */
 480         if (ui_remainder == 0)  /* If the phase is zero...           */
 481         {                       /* ...then we've already skipped the */
 482             Hp += Npc;          /*    first sample, so we must also  */
 483             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 484         }
 485     }
 486
 487     while (Hp < End) {
 488         t = *Hp;                /* Get filter coeff */
 489                                 /* t is now interp'd filter coeff */
 490         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 491         for( i = 0; i < i_nb_channels; i++ )
 492         {
 493             temp = t;
 494             temp *= *(p_in+i);  /* Mult coeff by input sample */
 495             *(p_out+i) += temp; /* The filter output */
 496         }
 497         Hdp += Npc;             /* Filter coeff differences step */
 498         Hp += Npc;              /* Filter coeff step */
 499         p_in += (Inc * i_nb_channels); /* Input signal step */
 500     }
 501 }
 502
 503 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 504                     float *p_out, uint32_t ui_remainder,
 505                     uint32_t ui_output_rate, uint32_t ui_input_rate,
 506                     int16_t Inc, int i_nb_channels )
 507 {
 508     const float *Hp, *Hdp, *End;
 509     float t, temp;
 510     uint32_t ui_linear_remainder;
 511     int i, ui_counter = 0;
 512
 513     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 514     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 515
 516     End = &Imp[Nwing];
 517
 518     if (Inc == 1)               /* If doing right wing...              */
 519     {                           /* ...drop extra coeff, so when Ph is  */
 520         End--;                  /*    0.5, we don't do too many mult's */
 521         if (ui_remainder == 0)  /* If the phase is zero...           */
 522         {                       /* ...then we've already skipped the */
 523             Hp = Imp +          /* first sample, so we must also  */
 524                   (ui_output_rate << Nhc) / ui_input_rate;
 525             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 526                   (ui_output_rate << Nhc) / ui_input_rate;
 527             ui_counter++;
 528         }
 529     }
 530
 531     while (Hp < End) {
 532         t = *Hp;                /* Get filter coeff */
 533                                 /* t is now interp'd filter coeff */
 534         ui_linear_remainder =
 535           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 536           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 537           ui_input_rate * ui_input_rate;
 538         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 539         for( i = 0; i < i_nb_channels; i++ )
 540         {
 541             temp = t;
 542             temp *= *(p_in+i);  /* Mult coeff by input sample */
 543             *(p_out+i) += temp; /* The filter output */
 544         }
 545
 546         ui_counter++;
 547
 548         /* Filter coeff step */
 549         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 550                     / ui_input_rate;
 551         /* Filter coeff differences step */
 552         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 553                      / ui_input_rate;
 554
 555         p_in += (Inc * i_nb_channels); /* Input signal step */
 556     }
 557 }