git.sesse.net Git - vlc/blob - modules/audio_filter/channel_mixer/mono.c

   1 /*****************************************************************************
   2  * mono.c : stereo2mono downmixsimple channel mixer plug-in
   3  *****************************************************************************
   4  * Copyright (C) 2006 M2X
   5  * $Id$
   6  *
   7  * Authors: Jean-Paul Saman <jpsaman at m2x dot nl>
   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 #include <math.h>                                        /* sqrt */
  28
  29 #ifdef HAVE_STDINT_H
  30 #   include <stdint.h>                                         /* int16_t .. */
  31 #elif defined(HAVE_INTTYPES_H)
  32 #   include <inttypes.h>                                       /* int16_t .. */
  33 #endif
  34
  35 #ifdef HAVE_UNISTD_H
  36 #   include <unistd.h>
  37 #endif
  38
  39 #include <vlc/vlc.h>
  40 #include <vlc_es.h>
  41 #include <vlc_block.h>
  42 #include <vlc_filter.h>
  43 #include <vlc_aout.h>
  44
  45 /*****************************************************************************
  46  * Local prototypes
  47  *****************************************************************************/
  48 static int  OpenFilter    ( vlc_object_t * );
  49 static void CloseFilter   ( vlc_object_t * );
  50
  51 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  52
  53 static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *,
  54                                     aout_buffer_t *, aout_buffer_t * );
  55 static unsigned int mono( aout_instance_t *, aout_filter_t *,
  56                                     aout_buffer_t *, aout_buffer_t * );
  57 static void stereo2mono_downmix( aout_instance_t *, aout_filter_t *,
  58                                  aout_buffer_t *, aout_buffer_t * );
  59
  60 /*****************************************************************************
  61  * Local structures
  62  *****************************************************************************/
  63 struct atomic_operation_t
  64 {
  65     int i_source_channel_offset;
  66     int i_dest_channel_offset;
  67     unsigned int i_delay;/* in sample unit */
  68     double d_amplitude_factor;
  69 };
  70
  71 struct filter_sys_t
  72 {
  73     vlc_bool_t b_downmix;
  74
  75     unsigned int i_nb_channels; /* number of int16_t per sample */
  76     int i_channel_selected;
  77     int i_bitspersample;
  78
  79     size_t i_overflow_buffer_size;/* in bytes */
  80     byte_t * p_overflow_buffer;
  81     unsigned int i_nb_atomic_operations;
  82     struct atomic_operation_t * p_atomic_operations;
  83 };
  84
  85 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
  86 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  87     "downmix algorithm that is used in the headphone channel mixer. It" \
  88     "gives the effect of standing in a room full of speakers." )
  89
  90 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  91 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  92     "except the selected channel. Choose one from (0=left, 1=right, " \
  93     "2=rear left, 3=rear right, 4=center, 5=left front)")
  94
  95 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
  96 static const char *ppsz_pos_descriptions[] =
  97 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
  98   N_("Left front") };
  99
 100 /* our internal channel order (WG-4 order) */
 101 static const uint32_t pi_channels_out[] =
 102 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
 103   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
 104
 105 #define MONO_CFG "sout-mono-"
 106 /*****************************************************************************
 107  * Module descriptor
 108  *****************************************************************************/
 109 vlc_module_begin();
 110     set_description( _("Audio filter for stereo to mono conversion") );
 111     set_capability( "audio filter2", 0 );
 112
 113     add_bool( MONO_CFG "downmix", VLC_FALSE, NULL, MONO_DOWNMIX_TEXT,
 114               MONO_DOWNMIX_LONGTEXT, VLC_FALSE );
 115     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 116         MONO_CHANNEL_LONGTEXT, VLC_FALSE );
 117         change_integer_list( pi_pos_values, ppsz_pos_descriptions, 0 );
 118
 119     set_category( CAT_AUDIO );
 120     set_subcategory( SUBCAT_AUDIO_MISC );
 121     set_callbacks( OpenFilter, CloseFilter );
 122     set_shortname( "Mono" );
 123 vlc_module_end();
 124
 125 /* Init() and ComputeChannelOperations() -
 126  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 127  * converted from float into int16_t based downmix
 128  * Written by Boris Dorès <babal@via.ecp.fr>
 129  */
 130
 131 /*****************************************************************************
 132  * Init: initialize internal data structures
 133  * and computes the needed atomic operations
 134  *****************************************************************************/
 135 /* x and z represent the coordinates of the virtual speaker
 136  *  relatively to the center of the listener's head, measured in meters :
 137  *
 138  *  left              right
 139  *Z
 140  *-
 141  *a          head
 142  *x
 143  *i
 144  *s
 145  *  rear left    rear right
 146  *
 147  *          x-axis
 148  *  */
 149 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 150         unsigned int i_rate, unsigned int i_next_atomic_operation,
 151         int i_source_channel_offset, double d_x, double d_z,
 152         double d_compensation_length, double d_channel_amplitude_factor )
 153 {
 154     double d_c = 340; /*sound celerity (unit: m/s)*/
 155     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 156
 157     /* Left ear */
 158     p_data->p_atomic_operations[i_next_atomic_operation]
 159         .i_source_channel_offset = i_source_channel_offset;
 160     p_data->p_atomic_operations[i_next_atomic_operation]
 161         .i_dest_channel_offset = 0;/* left */
 162     p_data->p_atomic_operations[i_next_atomic_operation]
 163         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 164                           / d_c * i_rate - d_compensation_delay );
 165     if( d_x < 0 )
 166     {
 167         p_data->p_atomic_operations[i_next_atomic_operation]
 168             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 169     }
 170     else if( d_x > 0 )
 171     {
 172         p_data->p_atomic_operations[i_next_atomic_operation]
 173             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 174     }
 175     else
 176     {
 177         p_data->p_atomic_operations[i_next_atomic_operation]
 178             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 179     }
 180
 181     /* Right ear */
 182     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 183         .i_source_channel_offset = i_source_channel_offset;
 184     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 185         .i_dest_channel_offset = 1;/* right */
 186     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 187         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 188                           / d_c * i_rate - d_compensation_delay );
 189     if( d_x < 0 )
 190     {
 191         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 192             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 193     }
 194     else if( d_x > 0 )
 195     {
 196         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 197             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 198     }
 199     else
 200     {
 201         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 202             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 203     }
 204 }
 205
 206 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 207                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 208                  unsigned int i_rate )
 209 {
 210     double d_x = config_GetInt( p_this, "headphone-dim" );
 211     double d_z = d_x;
 212     double d_z_rear = -d_x/3;
 213     double d_min = 0;
 214     unsigned int i_next_atomic_operation;
 215     int i_source_channel_offset;
 216     unsigned int i;
 217
 218     if( p_data == NULL )
 219     {
 220         msg_Dbg( p_this, "passing a null pointer as argument" );
 221         return 0;
 222     }
 223
 224     if( config_GetInt( p_this, "headphone-compensate" ) )
 225     {
 226         /* minimal distance to any speaker */
 227         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 228         {
 229             d_min = d_z_rear;
 230         }
 231         else
 232         {
 233             d_min = d_z;
 234         }
 235     }
 236
 237     /* Number of elementary operations */
 238     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 239     if( i_physical_channels & AOUT_CHAN_CENTER )
 240     {
 241         p_data->i_nb_atomic_operations += 2;
 242     }
 243     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 244             * p_data->i_nb_atomic_operations );
 245     if( p_data->p_atomic_operations == NULL )
 246     {
 247         msg_Err( p_this, "out of memory" );
 248         return -1;
 249     }
 250
 251     /* For each virtual speaker, computes elementary wave propagation time
 252      * to each ear */
 253     i_next_atomic_operation = 0;
 254     i_source_channel_offset = 0;
 255     if( i_physical_channels & AOUT_CHAN_LEFT )
 256     {
 257         ComputeChannelOperations( p_data , i_rate
 258                 , i_next_atomic_operation , i_source_channel_offset
 259                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 260         i_next_atomic_operation += 2;
 261         i_source_channel_offset++;
 262     }
 263     if( i_physical_channels & AOUT_CHAN_RIGHT )
 264     {
 265         ComputeChannelOperations( p_data , i_rate
 266                 , i_next_atomic_operation , i_source_channel_offset
 267                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 268         i_next_atomic_operation += 2;
 269         i_source_channel_offset++;
 270     }
 271     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 272     {
 273         ComputeChannelOperations( p_data , i_rate
 274                 , i_next_atomic_operation , i_source_channel_offset
 275                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 276         i_next_atomic_operation += 2;
 277         i_source_channel_offset++;
 278     }
 279     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 280     {
 281         ComputeChannelOperations( p_data , i_rate
 282                 , i_next_atomic_operation , i_source_channel_offset
 283                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 284         i_next_atomic_operation += 2;
 285         i_source_channel_offset++;
 286     }
 287     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 288     {
 289         ComputeChannelOperations( p_data , i_rate
 290                 , i_next_atomic_operation , i_source_channel_offset
 291                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 292         i_next_atomic_operation += 2;
 293         i_source_channel_offset++;
 294     }
 295     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 296     {
 297         ComputeChannelOperations( p_data , i_rate
 298                 , i_next_atomic_operation , i_source_channel_offset
 299                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 300         i_next_atomic_operation += 2;
 301         i_source_channel_offset++;
 302     }
 303     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 304     {
 305         ComputeChannelOperations( p_data , i_rate
 306                 , i_next_atomic_operation , i_source_channel_offset
 307                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 308         i_next_atomic_operation += 2;
 309         i_source_channel_offset++;
 310     }
 311     if( i_physical_channels & AOUT_CHAN_CENTER )
 312     {
 313         /* having two center channels increases the spatialization effect */
 314         ComputeChannelOperations( p_data , i_rate
 315                 , i_next_atomic_operation , i_source_channel_offset
 316                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 317         i_next_atomic_operation += 2;
 318         ComputeChannelOperations( p_data , i_rate
 319                 , i_next_atomic_operation , i_source_channel_offset
 320                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 321         i_next_atomic_operation += 2;
 322         i_source_channel_offset++;
 323     }
 324     if( i_physical_channels & AOUT_CHAN_LFE )
 325     {
 326         ComputeChannelOperations( p_data , i_rate
 327                 , i_next_atomic_operation , i_source_channel_offset
 328                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 329         i_next_atomic_operation += 2;
 330         i_source_channel_offset++;
 331     }
 332
 333     /* Initialize the overflow buffer
 334      * we need it because the process induce a delay in the samples */
 335     p_data->i_overflow_buffer_size = 0;
 336     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 337     {
 338         if( p_data->i_overflow_buffer_size
 339                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 340         {
 341             p_data->i_overflow_buffer_size
 342                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 343         }
 344     }
 345     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 346     if( p_data->p_atomic_operations == NULL )
 347     {
 348         msg_Err( p_this, "out of memory" );
 349         return -1;
 350     }
 351     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 352
 353     /* end */
 354     return 0;
 355 }
 356
 357 /*****************************************************************************
 358  * OpenFilter
 359  *****************************************************************************/
 360 static int OpenFilter( vlc_object_t *p_this )
 361 {
 362     filter_t * p_filter = (filter_t *)p_this;
 363     filter_sys_t *p_sys = NULL;
 364
 365     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 366     {
 367         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
 368         return VLC_EGENERIC;
 369     }
 370
 371     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
 372         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
 373     {
 374         msg_Err( p_this, "filter discarded (invalid format)" );
 375         return -1;
 376     }
 377
 378     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 379         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 380         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
 381         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
 382         (p_filter->fmt_in.audio.i_bitspersample !=
 383                                     p_filter->fmt_out.audio.i_bitspersample))
 384     {
 385         msg_Err( p_this, "couldn't load mono filter" );
 386         return VLC_EGENERIC;
 387     }
 388
 389     /* Allocate the memory needed to store the module's structure */
 390     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 391     if( p_sys == NULL )
 392     {
 393         msg_Err( p_filter, "out of memory" );
 394         return VLC_EGENERIC;
 395     }
 396
 397     var_Create( p_this, MONO_CFG "downmix",
 398                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 399     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 400
 401     var_Create( p_this, MONO_CFG "channel",
 402                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 403     p_sys->i_channel_selected =
 404             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 405
 406     if( p_sys->b_downmix )
 407     {
 408         msg_Dbg( p_this, "using stereo to mono downmix" );
 409         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 410         p_filter->fmt_out.audio.i_channels = 1;
 411     }
 412     else
 413     {
 414         msg_Dbg( p_this, "using pseudo mono" );
 415         p_filter->fmt_out.audio.i_physical_channels =
 416                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 417         p_filter->fmt_out.audio.i_channels = 2;
 418     }
 419
 420     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 421     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 422
 423     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 424     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 425
 426     p_sys->i_overflow_buffer_size = 0;
 427     p_sys->p_overflow_buffer = NULL;
 428     p_sys->i_nb_atomic_operations = 0;
 429     p_sys->p_atomic_operations = NULL;
 430
 431     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 432               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 433               p_filter->fmt_in.audio.i_physical_channels,
 434               p_filter->fmt_in.audio.i_rate ) < 0 )
 435     {
 436         return VLC_EGENERIC;
 437     }
 438
 439     p_filter->pf_audio_filter = Convert;
 440
 441     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 442              (char *)&p_filter->fmt_in.i_codec,
 443              (char *)&p_filter->fmt_out.i_codec,
 444              p_filter->fmt_in.audio.i_physical_channels,
 445              p_filter->fmt_out.audio.i_physical_channels,
 446              p_filter->fmt_in.audio.i_bitspersample,
 447              p_filter->fmt_out.audio.i_bitspersample );
 448
 449     return VLC_SUCCESS;
 450 }
 451
 452 /*****************************************************************************
 453  * CloseFilter
 454  *****************************************************************************/
 455 static void CloseFilter( vlc_object_t *p_this)
 456 {
 457     filter_t *p_filter = (filter_t *) p_this;
 458     filter_sys_t *p_sys = p_filter->p_sys;
 459
 460     var_Destroy( p_this, MONO_CFG "channel" );
 461     var_Destroy( p_this, MONO_CFG "downmix" );
 462     free( p_sys );
 463 }
 464
 465 /*****************************************************************************
 466  * Convert
 467  *****************************************************************************/
 468 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 469 {
 470     aout_filter_t aout_filter;
 471     aout_buffer_t in_buf, out_buf;
 472     block_t *p_out = NULL;
 473     unsigned int i_samples;
 474     int i_out_size;
 475
 476     if( !p_block || !p_block->i_samples )
 477     {
 478         if( p_block )
 479             p_block->pf_release( p_block );
 480         return NULL;
 481     }
 482
 483     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 484                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 485
 486     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 487     if( !p_out )
 488     {
 489         msg_Warn( p_filter, "can't get output buffer" );
 490         p_block->pf_release( p_block );
 491         return NULL;
 492     }
 493     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 494                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 495     p_out->i_dts = p_block->i_dts;
 496     p_out->i_pts = p_block->i_pts;
 497     p_out->i_length = p_block->i_length;
 498
 499     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 500     aout_filter.input = p_filter->fmt_in.audio;
 501     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 502     aout_filter.output = p_filter->fmt_out.audio;
 503     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 504
 505     in_buf.p_buffer = p_block->p_buffer;
 506     in_buf.i_nb_bytes = p_block->i_buffer;
 507     in_buf.i_nb_samples = p_block->i_samples;
 508
 509 #if 0
 510     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 511                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 512     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 513     {
 514         msg_Err( p_filter, "input buffer is not word aligned" );
 515         /* Fix output buffer to be word aligned */
 516     }
 517 #endif
 518
 519     out_buf.p_buffer = p_out->p_buffer;
 520     out_buf.i_nb_bytes = p_out->i_buffer;
 521     out_buf.i_nb_samples = p_out->i_samples;
 522
 523     memset( p_out->p_buffer, 0, i_out_size );
 524     if( p_filter->p_sys->b_downmix )
 525     {
 526         stereo2mono_downmix( (aout_instance_t *)p_filter, &aout_filter,
 527                              &in_buf, &out_buf );
 528         i_samples = mono( (aout_instance_t *)p_filter, &aout_filter,
 529                            &out_buf, &in_buf );
 530     }
 531     else
 532     {
 533         i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
 534                                     &out_buf, &in_buf );
 535     }
 536
 537     p_out->i_buffer = out_buf.i_nb_bytes;
 538     p_out->i_samples = out_buf.i_nb_samples;
 539
 540     p_block->pf_release( p_block );
 541     return p_out;
 542 }
 543
 544 /* stereo2mono_downmix - stereo channels into one mono channel.
 545  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 546  * converted from float into int16_t based downmix
 547  * Written by Boris Dorès <babal@via.ecp.fr>
 548  */
 549 static void stereo2mono_downmix( aout_instance_t * p_aout, aout_filter_t * p_filter,
 550                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 551 {
 552     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 553
 554     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 555     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 556
 557     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 558     byte_t * p_out;
 559     byte_t * p_overflow;
 560     byte_t * p_slide;
 561
 562     size_t i_overflow_size;     /* in bytes */
 563     size_t i_out_size;          /* in bytes */
 564
 565     unsigned int i, j;
 566
 567     int i_source_channel_offset;
 568     int i_dest_channel_offset;
 569     unsigned int i_delay;
 570     double d_amplitude_factor;
 571
 572     /* out buffer characterisitcs */
 573     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 574     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 575     p_out = p_out_buf->p_buffer;
 576     i_out_size = p_out_buf->i_nb_bytes;
 577
 578     if( p_sys != NULL )
 579     {
 580         /* Slide the overflow buffer */
 581         p_overflow = p_sys->p_overflow_buffer;
 582         i_overflow_size = p_sys->i_overflow_buffer_size;
 583
 584         if ( i_out_size > i_overflow_size )
 585             memcpy( p_out, p_overflow, i_overflow_size );
 586         else
 587             memcpy( p_out, p_overflow, i_out_size );
 588
 589         p_slide = p_sys->p_overflow_buffer;
 590         while( p_slide < p_overflow + i_overflow_size )
 591         {
 592             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 593             {
 594                 memset( p_slide, 0, i_out_size );
 595                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 596                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 597                 else
 598                     memcpy( p_slide, p_slide + i_out_size,
 599                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 600             }
 601             else
 602             {
 603                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 604             }
 605             p_slide += i_out_size;
 606         }
 607
 608         /* apply the atomic operations */
 609         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 610         {
 611             /* shorter variable names */
 612             i_source_channel_offset
 613                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 614             i_dest_channel_offset
 615                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 616             i_delay = p_sys->p_atomic_operations[i].i_delay;
 617             d_amplitude_factor
 618                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 619
 620             if( p_out_buf->i_nb_samples > i_delay )
 621             {
 622                 /* current buffer coefficients */
 623                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 624                 {
 625                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 626                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 627                            * d_amplitude_factor;
 628                 }
 629
 630                 /* overflow buffer coefficients */
 631                 for( j = 0; j < i_delay; j++ )
 632                 {
 633                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 634                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 635                            * i_input_nb + i_source_channel_offset ]
 636                            * d_amplitude_factor;
 637                 }
 638             }
 639             else
 640             {
 641                 /* overflow buffer coefficients only */
 642                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 643                 {
 644                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 645                         * i_output_nb + i_dest_channel_offset ]
 646                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 647                            * d_amplitude_factor;
 648                 }
 649             }
 650         }
 651     }
 652     else
 653     {
 654         memset( p_out, 0, i_out_size );
 655     }
 656 }
 657
 658 /* Simple stereo to mono mixing. */
 659 static unsigned int mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 660                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 661 {
 662     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 663     int16_t *p_in, *p_out;
 664     unsigned int n = 0, r = 0;
 665
 666     p_in = (int16_t *) p_input->p_buffer;
 667     p_out = (int16_t *) p_output->p_buffer;
 668
 669     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 670     {
 671         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 672         r++;
 673         n += 2;
 674     }
 675     return r;
 676 }
 677
 678 /* Simple stereo to mono mixing. */
 679 static unsigned int stereo_to_mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 680                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 681 {
 682     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 683     int16_t *p_in, *p_out;
 684     unsigned int n;
 685
 686     p_in = (int16_t *) p_input->p_buffer;
 687     p_out = (int16_t *) p_output->p_buffer;
 688
 689     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 690     {
 691         /* Fake real mono. */
 692         if( p_sys->i_channel_selected == -1)
 693         {
 694             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 695             n++;
 696         }
 697         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 698         {
 699             p_out[n] = p_out[n+1] = p_in[n];
 700         }
 701     }
 702     return n;
 703 }