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

   1 /*****************************************************************************
   2  * bandlimited.c : band-limited interpolation resampler
   3  *****************************************************************************
   4  * Copyright (C) 2002, 2006 the VideoLAN team
   5  * $Id$
   6  *
   7  * Authors: Gildas Bazin <gbazin@netcourrier.com>
   8  *
   9  * This program is free software; you can redistribute it and/or modify
  10  * it under the terms of the GNU General Public License as published by
  11  * the Free Software Foundation; either version 2 of the License, or
  12  * (at your option) any later version.
  13  *
  14  * This program is distributed in the hope that it will be useful,
  15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  17  * GNU General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU General Public License
  20  * along with this program; if not, write to the Free Software
  21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  22  *****************************************************************************/
  23
  24 /*****************************************************************************
  25  * Preamble:
  26  *
  27  * This implementation of the band-limited interpolationis based on the
  28  * following paper:
  29  * http://ccrma-www.stanford.edu/~jos/resample/resample.html
  30  *
  31  * It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
  32  * filter is 13 samples.
  33  *
  34  *****************************************************************************/
  35
  36 #ifdef HAVE_CONFIG_H
  37 # include "config.h"
  38 #endif
  39
  40 #include <vlc_common.h>
  41 #include <vlc_plugin.h>
  42 #include <vlc_aout.h>
  43 #include <vlc_filter.h>
  44 #include <vlc_block.h>
  45
  46 #include <assert.h>
  47
  48 #include "bandlimited.h"
  49
  50 /*****************************************************************************
  51  * Local prototypes
  52  *****************************************************************************/
  53
  54 /* audio filter */
  55 static int  OpenFilter ( vlc_object_t * );
  56 static void CloseFilter( vlc_object_t * );
  57 static block_t *Resample( filter_t *, block_t * );
  58
  59 static void ResampleFloat( filter_t *p_filter,
  60                            block_t *p_out_buf,  size_t *pi_out,
  61                            float **pp_in,
  62                            int i_in, int i_in_end,
  63                            double d_factor, bool b_factor_old,
  64                            int i_nb_channels, int i_bytes_per_frame );
  65
  66 /*****************************************************************************
  67  * Local structures
  68  *****************************************************************************/
  69 struct filter_sys_t
  70 {
  71     int32_t *p_buf;                        /* this filter introduces a delay */
  72     size_t i_buf_size;
  73
  74     double d_old_factor;
  75     int i_old_wing;
  76
  77     unsigned int i_remainder;                /* remainder of previous sample */
  78     bool b_first;
  79
  80     date_t end_date;
  81 };
  82
  83 /*****************************************************************************
  84  * Module descriptor
  85  *****************************************************************************/
  86 vlc_module_begin ()
  87     set_category( CAT_AUDIO )
  88     set_subcategory( SUBCAT_AUDIO_MISC )
  89     set_description( N_("Audio filter for band-limited interpolation resampling") )
  90     set_capability( "audio filter", 20 )
  91     set_callbacks( OpenFilter, CloseFilter )
  92 vlc_module_end ()
  93
  94 /*****************************************************************************
  95  * Resample: convert a buffer
  96  *****************************************************************************/
  97 static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
  98 {
  99     if( !p_in_buf || !p_in_buf->i_nb_samples )
 100     {
 101         if( p_in_buf )
 102             block_Release( p_in_buf );
 103         return NULL;
 104     }
 105
 106     filter_sys_t *p_sys = p_filter->p_sys;
 107     unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
 108     int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 109
 110     /* Check if we really need to run the resampler */
 111     if( i_out_rate == p_filter->fmt_in.audio.i_rate )
 112     {
 113         if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) &&
 114             p_sys->i_old_wing )
 115         {
 116             /* output the whole thing with the samples from last time */
 117             p_in_buf = block_Realloc( p_in_buf,
 118                 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
 119                 p_in_buf->i_buffer );
 120             if( !p_in_buf )
 121                 return NULL;
 122             memcpy( p_in_buf->p_buffer, p_sys->p_buf +
 123                     i_nb_channels * p_sys->i_old_wing,
 124                     p_sys->i_old_wing *
 125                     p_filter->fmt_in.audio.i_bytes_per_frame );
 126
 127             p_in_buf->i_nb_samples += p_sys->i_old_wing;
 128
 129             p_in_buf->i_pts = date_Get( &p_sys->end_date );
 130             p_in_buf->i_length =
 131                 date_Increment( &p_sys->end_date,
 132                                 p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
 133         }
 134         p_sys->i_old_wing = 0;
 135         p_sys->b_first = true;
 136         return p_in_buf;
 137     }
 138
 139     unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 140                                  p_filter->fmt_out.audio.i_bitspersample / 8;
 141     size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
 142               p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
 143             + p_filter->p_sys->i_buf_size;
 144     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
 145     if( !p_out_buf )
 146     {
 147         block_Release( p_in_buf );
 148         return NULL;
 149     }
 150     float *p_out = (float *)p_out_buf->p_buffer;
 151
 152     if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
 153     {
 154         /* Continuity in sound samples has been broken, we'd better reset
 155          * everything. */
 156         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 157         p_sys->i_remainder = 0;
 158         date_Init( &p_sys->end_date, i_out_rate, 1 );
 159         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 160         p_sys->d_old_factor = 1;
 161         p_sys->i_old_wing   = 0;
 162         p_sys->b_first = false;
 163     }
 164
 165     size_t i_in_nb = p_in_buf->i_nb_samples;
 166     size_t i_in, i_out = 0;
 167     double d_factor, d_scale_factor, d_old_scale_factor;
 168     size_t i_filter_wing;
 169
 170 #if 0
 171     msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
 172              p_sys->i_old_rate, p_sys->d_old_factor,
 173              p_sys->i_old_wing, i_in_nb );
 174 #endif
 175
 176     /* Same format in and out... */
 177     assert( p_filter->fmt_in.audio.i_bytes_per_frame == i_bytes_per_frame );
 178
 179     /* Prepare the source buffer */
 180     if( p_sys->i_old_wing )
 181     {   /* Copy all our samples in p_in_buf */
 182         /* Normally, there should be enough room for the old wing in the
 183          * buffer head room. Otherwise, we need to copy memory anyway. */
 184         p_in_buf = block_Realloc( p_in_buf,
 185                                   p_sys->i_old_wing * 2 * i_bytes_per_frame,
 186                                   p_in_buf->i_buffer );
 187         if( unlikely(p_in_buf == NULL) )
 188             return NULL;
 189         memcpy( p_in_buf->p_buffer, p_sys->p_buf,
 190                 p_sys->i_old_wing * 2 * i_bytes_per_frame );
 191     }
 192     i_in_nb += (p_sys->i_old_wing * 2);
 193     float *p_in = (float *)p_in_buf->p_buffer;
 194     const float *p_in_orig = p_in;
 195
 196     /* Make sure the output buffer is reset */
 197     memset( p_out, 0, p_out_buf->i_buffer );
 198
 199     /* Calculate the new length of the filter wing */
 200     d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
 201     i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
 202
 203     /* Account for increased filter gain when using factors less than 1 */
 204     d_old_scale_factor = SMALL_FILTER_SCALE *
 205         p_sys->d_old_factor + 0.5;
 206     d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
 207
 208     /* Apply the old rate until we have enough samples for the new one */
 209     i_in = p_sys->i_old_wing;
 210     p_in += p_sys->i_old_wing * i_nb_channels;
 211
 212     size_t i_old_in_end = 0;
 213     if( p_sys->i_old_wing <= i_in_nb )
 214         i_old_in_end = __MIN( i_filter_wing, i_in_nb - p_sys->i_old_wing );
 215
 216     ResampleFloat( p_filter,
 217                    p_out_buf, &i_out, &p_in,
 218                    i_in, i_old_in_end,
 219                    p_sys->d_old_factor, true,
 220                    i_nb_channels, i_bytes_per_frame );
 221     i_in = __MAX( i_in, i_old_in_end );
 222
 223     /* Apply the new rate for the rest of the samples */
 224     if( i_in < i_in_nb - i_filter_wing )
 225     {
 226         p_sys->d_old_factor = d_factor;
 227         p_sys->i_old_wing   = i_filter_wing;
 228     }
 229     ResampleFloat( p_filter,
 230                    p_out_buf, &i_out, &p_in,
 231                    i_in, i_in_nb - i_filter_wing,
 232                    d_factor, false,
 233                    i_nb_channels, i_bytes_per_frame );
 234
 235     /* Finalize aout buffer */
 236     p_out_buf->i_nb_samples = i_out;
 237     p_out_buf->i_pts = date_Get( &p_sys->end_date );
 238     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 239                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 240
 241     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 242         i_nb_channels * sizeof(int32_t);
 243
 244     /* Buffer i_filter_wing * 2 samples for next time */
 245     if( p_sys->i_old_wing )
 246     {
 247         size_t newsize = p_sys->i_old_wing * 2 * i_bytes_per_frame;
 248         if( newsize > p_sys->i_buf_size )
 249         {
 250             free( p_sys->p_buf );
 251             p_sys->p_buf = malloc( newsize );
 252             if( p_sys->p_buf != NULL )
 253                 p_sys->i_buf_size = newsize;
 254             else
 255             {
 256                 p_sys->i_buf_size = p_sys->i_old_wing = 0; /* oops! */
 257                 block_Release( p_in_buf );
 258                 return p_out_buf;
 259             }
 260         }
 261         memcpy( p_sys->p_buf,
 262                 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
 263                 i_nb_channels, (2 * p_sys->i_old_wing) *
 264                 p_filter->fmt_in.audio.i_bytes_per_frame );
 265     }
 266
 267 #if 0
 268     msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
 269              i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
 270 #endif
 271
 272     block_Release( p_in_buf );
 273     return p_out_buf;
 274 }
 275
 276 /*****************************************************************************
 277  * OpenFilter:
 278  *****************************************************************************/
 279 static int OpenFilter( vlc_object_t *p_this )
 280 {
 281     filter_t *p_filter = (filter_t *)p_this;
 282     filter_sys_t *p_sys;
 283     unsigned int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 284
 285     if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
 286       || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
 287       || p_filter->fmt_in.audio.i_physical_channels
 288               != p_filter->fmt_out.audio.i_physical_channels
 289       || p_filter->fmt_in.audio.i_original_channels
 290               != p_filter->fmt_out.audio.i_original_channels
 291       || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
 292     {
 293         return VLC_EGENERIC;
 294     }
 295
 296 #if !defined( SYS_DARWIN )
 297     if( !var_InheritInteger( p_this, "hq-resampling" ) )
 298     {
 299         return VLC_EGENERIC;
 300     }
 301 #endif
 302
 303     /* Allocate the memory needed to store the module's structure */
 304     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 305     if( p_sys == NULL )
 306         return VLC_ENOMEM;
 307
 308     p_sys->p_buf = NULL;
 309     p_sys->i_buf_size = 0;
 310
 311     p_sys->i_old_wing = 0;
 312     p_sys->b_first = true;
 313     p_filter->pf_audio_filter = Resample;
 314
 315     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 316              (char *)&p_filter->fmt_in.i_codec,
 317              p_filter->fmt_in.audio.i_rate,
 318              p_filter->fmt_in.audio.i_channels,
 319              (char *)&p_filter->fmt_out.i_codec,
 320              p_filter->fmt_out.audio.i_rate,
 321              p_filter->fmt_out.audio.i_channels);
 322
 323     p_filter->fmt_out = p_filter->fmt_in;
 324     p_filter->fmt_out.audio.i_rate = i_out_rate;
 325
 326     return 0;
 327 }
 328
 329 /*****************************************************************************
 330  * CloseFilter : deallocate data structures
 331  *****************************************************************************/
 332 static void CloseFilter( vlc_object_t *p_this )
 333 {
 334     filter_t *p_filter = (filter_t *)p_this;
 335     free( p_filter->p_sys->p_buf );
 336     free( p_filter->p_sys );
 337 }
 338
 339 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 340                             float *p_out, uint32_t ui_remainder,
 341                             uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
 342 {
 343     const float *Hp, *Hdp, *End;
 344     float t, temp;
 345     uint32_t ui_linear_remainder;
 346     int i;
 347
 348     Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
 349     Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
 350
 351     End = &Imp[Nwing];
 352
 353     ui_linear_remainder = (ui_remainder<<Nhc) -
 354                             (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
 355
 356     if (Inc == 1)               /* If doing right wing...              */
 357     {                           /* ...drop extra coeff, so when Ph is  */
 358         End--;                  /*    0.5, we don't do too many mult's */
 359         if (ui_remainder == 0)  /* If the phase is zero...           */
 360         {                       /* ...then we've already skipped the */
 361             Hp += Npc;          /*    first sample, so we must also  */
 362             Hdp += Npc;         /*    skip ahead in Imp[] and ImpD[] */
 363         }
 364     }
 365
 366     while (Hp < End) {
 367         t = *Hp;                /* Get filter coeff */
 368                                 /* t is now interp'd filter coeff */
 369         t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
 370         for( i = 0; i < i_nb_channels; i++ )
 371         {
 372             temp = t;
 373             temp *= *(p_in+i);  /* Mult coeff by input sample */
 374             *(p_out+i) += temp; /* The filter output */
 375         }
 376         Hdp += Npc;             /* Filter coeff differences step */
 377         Hp += Npc;              /* Filter coeff step */
 378         p_in += (Inc * i_nb_channels); /* Input signal step */
 379     }
 380 }
 381
 382 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
 383                            float *p_out, uint32_t ui_remainder,
 384                            uint32_t ui_output_rate, uint32_t ui_input_rate,
 385                            int16_t Inc, int i_nb_channels )
 386 {
 387     const float *Hp, *Hdp, *End;
 388     float t, temp;
 389     uint32_t ui_linear_remainder;
 390     int i, ui_counter = 0;
 391
 392     Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
 393     Hdp = ImpD  + (ui_remainder<<Nhc) / ui_input_rate;
 394
 395     End = &Imp[Nwing];
 396
 397     if (Inc == 1)               /* If doing right wing...              */
 398     {                           /* ...drop extra coeff, so when Ph is  */
 399         End--;                  /*    0.5, we don't do too many mult's */
 400         if (ui_remainder == 0)  /* If the phase is zero...           */
 401         {                       /* ...then we've already skipped the */
 402             Hp = Imp +          /* first sample, so we must also  */
 403                   (ui_output_rate << Nhc) / ui_input_rate;
 404             Hdp = ImpD +        /* skip ahead in Imp[] and ImpD[] */
 405                   (ui_output_rate << Nhc) / ui_input_rate;
 406             ui_counter++;
 407         }
 408     }
 409
 410     while (Hp < End) {
 411         t = *Hp;                /* Get filter coeff */
 412                                 /* t is now interp'd filter coeff */
 413         ui_linear_remainder =
 414           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
 415           ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
 416           ui_input_rate * ui_input_rate;
 417         t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
 418         for( i = 0; i < i_nb_channels; i++ )
 419         {
 420             temp = t;
 421             temp *= *(p_in+i);  /* Mult coeff by input sample */
 422             *(p_out+i) += temp; /* The filter output */
 423         }
 424
 425         ui_counter++;
 426
 427         /* Filter coeff step */
 428         Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 429                     / ui_input_rate;
 430         /* Filter coeff differences step */
 431         Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
 432                      / ui_input_rate;
 433
 434         p_in += (Inc * i_nb_channels); /* Input signal step */
 435     }
 436 }
 437
 438 static void ResampleFloat( filter_t *p_filter,
 439                            block_t *p_out_buf,  size_t *pi_out,
 440                            float **pp_in,
 441                            int i_in, int i_in_end,
 442                            double d_factor, bool b_factor_old,
 443                            int i_nb_channels, int i_bytes_per_frame )
 444 {
 445     filter_sys_t *p_sys = p_filter->p_sys;
 446
 447     float *p_in = *pp_in;
 448     size_t i_out = *pi_out;
 449     float *p_out = (float*)p_out_buf->p_buffer + i_out * i_nb_channels;
 450
 451     for( ; i_in < i_in_end; i_in++ )
 452     {
 453         if( b_factor_old && d_factor == 1 )
 454         {
 455             /* Just copy the samples */
 456             memcpy( p_out, p_in, i_bytes_per_frame );
 457             p_in += i_nb_channels;
 458             p_out += i_nb_channels;
 459             i_out++;
 460             continue;
 461         }
 462
 463         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 464         {
 465             if( p_out_buf->i_buffer/i_bytes_per_frame <= i_out )
 466                 break;
 467
 468             if( d_factor >= 1 )
 469             {
 470                 /* FilterFloatUP() is faster if we can use it */
 471
 472                 /* Perform left-wing inner product */
 473                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 474                                SMALL_FILTER_NWING, p_in, p_out,
 475                                p_sys->i_remainder,
 476                                p_filter->fmt_out.audio.i_rate,
 477                                -1, i_nb_channels );
 478                 /* Perform right-wing inner product */
 479                 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 480                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 481                                p_filter->fmt_out.audio.i_rate -
 482                                p_sys->i_remainder,
 483                                p_filter->fmt_out.audio.i_rate,
 484                                1, i_nb_channels );
 485
 486 #if 0
 487                 /* Normalize for unity filter gain */
 488                 for( i = 0; i < i_nb_channels; i++ )
 489                 {
 490                     *(p_out+i) *= d_old_scale_factor;
 491                 }
 492 #endif
 493             }
 494             else
 495             {
 496                 /* Perform left-wing inner product */
 497                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 498                                SMALL_FILTER_NWING, p_in, p_out,
 499                                p_sys->i_remainder,
 500                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 501                                -1, i_nb_channels );
 502                 /* Perform right-wing inner product */
 503                 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
 504                                SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
 505                                p_filter->fmt_out.audio.i_rate -
 506                                p_sys->i_remainder,
 507                                p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
 508                                1, i_nb_channels );
 509             }
 510
 511             p_out += i_nb_channels;
 512             i_out++;
 513
 514             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 515         }
 516
 517         p_in += i_nb_channels;
 518         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 519     }
 520
 521     *pp_in  = p_in;
 522     *pi_out = i_out;
 523 }
 524
 525