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         change_safe();
 116
 117     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 118         MONO_CHANNEL_LONGTEXT, VLC_FALSE );
 119         change_integer_list( pi_pos_values, ppsz_pos_descriptions, 0 );
 120         change_safe();
 121
 122     set_category( CAT_AUDIO );
 123     set_subcategory( SUBCAT_AUDIO_MISC );
 124     set_callbacks( OpenFilter, CloseFilter );
 125     set_shortname( "Mono" );
 126 vlc_module_end();
 127
 128 /* Init() and ComputeChannelOperations() -
 129  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 130  * converted from float into int16_t based downmix
 131  * Written by Boris Dorès <babal@via.ecp.fr>
 132  */
 133
 134 /*****************************************************************************
 135  * Init: initialize internal data structures
 136  * and computes the needed atomic operations
 137  *****************************************************************************/
 138 /* x and z represent the coordinates of the virtual speaker
 139  *  relatively to the center of the listener's head, measured in meters :
 140  *
 141  *  left              right
 142  *Z
 143  *-
 144  *a          head
 145  *x
 146  *i
 147  *s
 148  *  rear left    rear right
 149  *
 150  *          x-axis
 151  *  */
 152 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 153         unsigned int i_rate, unsigned int i_next_atomic_operation,
 154         int i_source_channel_offset, double d_x, double d_z,
 155         double d_compensation_length, double d_channel_amplitude_factor )
 156 {
 157     double d_c = 340; /*sound celerity (unit: m/s)*/
 158     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 159
 160     /* Left ear */
 161     p_data->p_atomic_operations[i_next_atomic_operation]
 162         .i_source_channel_offset = i_source_channel_offset;
 163     p_data->p_atomic_operations[i_next_atomic_operation]
 164         .i_dest_channel_offset = 0;/* left */
 165     p_data->p_atomic_operations[i_next_atomic_operation]
 166         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 167                           / d_c * i_rate - d_compensation_delay );
 168     if( d_x < 0 )
 169     {
 170         p_data->p_atomic_operations[i_next_atomic_operation]
 171             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 172     }
 173     else if( d_x > 0 )
 174     {
 175         p_data->p_atomic_operations[i_next_atomic_operation]
 176             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 177     }
 178     else
 179     {
 180         p_data->p_atomic_operations[i_next_atomic_operation]
 181             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 182     }
 183
 184     /* Right ear */
 185     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 186         .i_source_channel_offset = i_source_channel_offset;
 187     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 188         .i_dest_channel_offset = 1;/* right */
 189     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 190         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 191                           / d_c * i_rate - d_compensation_delay );
 192     if( d_x < 0 )
 193     {
 194         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 195             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 196     }
 197     else if( d_x > 0 )
 198     {
 199         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 200             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 201     }
 202     else
 203     {
 204         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 205             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 206     }
 207 }
 208
 209 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 210                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 211                  unsigned int i_rate )
 212 {
 213     double d_x = config_GetInt( p_this, "headphone-dim" );
 214     double d_z = d_x;
 215     double d_z_rear = -d_x/3;
 216     double d_min = 0;
 217     unsigned int i_next_atomic_operation;
 218     int i_source_channel_offset;
 219     unsigned int i;
 220
 221     if( p_data == NULL )
 222     {
 223         msg_Dbg( p_this, "passing a null pointer as argument" );
 224         return 0;
 225     }
 226
 227     if( config_GetInt( p_this, "headphone-compensate" ) )
 228     {
 229         /* minimal distance to any speaker */
 230         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 231         {
 232             d_min = d_z_rear;
 233         }
 234         else
 235         {
 236             d_min = d_z;
 237         }
 238     }
 239
 240     /* Number of elementary operations */
 241     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 242     if( i_physical_channels & AOUT_CHAN_CENTER )
 243     {
 244         p_data->i_nb_atomic_operations += 2;
 245     }
 246     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 247             * p_data->i_nb_atomic_operations );
 248     if( p_data->p_atomic_operations == NULL )
 249     {
 250         msg_Err( p_this, "out of memory" );
 251         return -1;
 252     }
 253
 254     /* For each virtual speaker, computes elementary wave propagation time
 255      * to each ear */
 256     i_next_atomic_operation = 0;
 257     i_source_channel_offset = 0;
 258     if( i_physical_channels & AOUT_CHAN_LEFT )
 259     {
 260         ComputeChannelOperations( p_data , i_rate
 261                 , i_next_atomic_operation , i_source_channel_offset
 262                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 263         i_next_atomic_operation += 2;
 264         i_source_channel_offset++;
 265     }
 266     if( i_physical_channels & AOUT_CHAN_RIGHT )
 267     {
 268         ComputeChannelOperations( p_data , i_rate
 269                 , i_next_atomic_operation , i_source_channel_offset
 270                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 271         i_next_atomic_operation += 2;
 272         i_source_channel_offset++;
 273     }
 274     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 275     {
 276         ComputeChannelOperations( p_data , i_rate
 277                 , i_next_atomic_operation , i_source_channel_offset
 278                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 279         i_next_atomic_operation += 2;
 280         i_source_channel_offset++;
 281     }
 282     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 283     {
 284         ComputeChannelOperations( p_data , i_rate
 285                 , i_next_atomic_operation , i_source_channel_offset
 286                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 287         i_next_atomic_operation += 2;
 288         i_source_channel_offset++;
 289     }
 290     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 291     {
 292         ComputeChannelOperations( p_data , i_rate
 293                 , i_next_atomic_operation , i_source_channel_offset
 294                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 295         i_next_atomic_operation += 2;
 296         i_source_channel_offset++;
 297     }
 298     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 299     {
 300         ComputeChannelOperations( p_data , i_rate
 301                 , i_next_atomic_operation , i_source_channel_offset
 302                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 303         i_next_atomic_operation += 2;
 304         i_source_channel_offset++;
 305     }
 306     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 307     {
 308         ComputeChannelOperations( p_data , i_rate
 309                 , i_next_atomic_operation , i_source_channel_offset
 310                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 311         i_next_atomic_operation += 2;
 312         i_source_channel_offset++;
 313     }
 314     if( i_physical_channels & AOUT_CHAN_CENTER )
 315     {
 316         /* having two center channels increases the spatialization effect */
 317         ComputeChannelOperations( p_data , i_rate
 318                 , i_next_atomic_operation , i_source_channel_offset
 319                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 320         i_next_atomic_operation += 2;
 321         ComputeChannelOperations( p_data , i_rate
 322                 , i_next_atomic_operation , i_source_channel_offset
 323                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 324         i_next_atomic_operation += 2;
 325         i_source_channel_offset++;
 326     }
 327     if( i_physical_channels & AOUT_CHAN_LFE )
 328     {
 329         ComputeChannelOperations( p_data , i_rate
 330                 , i_next_atomic_operation , i_source_channel_offset
 331                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 332         i_next_atomic_operation += 2;
 333         i_source_channel_offset++;
 334     }
 335
 336     /* Initialize the overflow buffer
 337      * we need it because the process induce a delay in the samples */
 338     p_data->i_overflow_buffer_size = 0;
 339     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 340     {
 341         if( p_data->i_overflow_buffer_size
 342                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 343         {
 344             p_data->i_overflow_buffer_size
 345                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 346         }
 347     }
 348     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 349     if( p_data->p_atomic_operations == NULL )
 350     {
 351         msg_Err( p_this, "out of memory" );
 352         return -1;
 353     }
 354     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 355
 356     /* end */
 357     return 0;
 358 }
 359
 360 /*****************************************************************************
 361  * OpenFilter
 362  *****************************************************************************/
 363 static int OpenFilter( vlc_object_t *p_this )
 364 {
 365     filter_t * p_filter = (filter_t *)p_this;
 366     filter_sys_t *p_sys = NULL;
 367
 368     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 369     {
 370         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
 371         return VLC_EGENERIC;
 372     }
 373
 374     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
 375         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
 376     {
 377         msg_Err( p_this, "filter discarded (invalid format)" );
 378         return -1;
 379     }
 380
 381     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 382         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 383         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
 384         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
 385         (p_filter->fmt_in.audio.i_bitspersample !=
 386                                     p_filter->fmt_out.audio.i_bitspersample))
 387     {
 388         msg_Err( p_this, "couldn't load mono filter" );
 389         return VLC_EGENERIC;
 390     }
 391
 392     /* Allocate the memory needed to store the module's structure */
 393     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 394     if( p_sys == NULL )
 395     {
 396         msg_Err( p_filter, "out of memory" );
 397         return VLC_EGENERIC;
 398     }
 399
 400     var_Create( p_this, MONO_CFG "downmix",
 401                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 402     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 403
 404     var_Create( p_this, MONO_CFG "channel",
 405                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 406     p_sys->i_channel_selected =
 407             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 408
 409     if( p_sys->b_downmix )
 410     {
 411         msg_Dbg( p_this, "using stereo to mono downmix" );
 412         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 413         p_filter->fmt_out.audio.i_channels = 1;
 414     }
 415     else
 416     {
 417         msg_Dbg( p_this, "using pseudo mono" );
 418         p_filter->fmt_out.audio.i_physical_channels =
 419                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 420         p_filter->fmt_out.audio.i_channels = 2;
 421     }
 422
 423     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 424     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 425
 426     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 427     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 428
 429     p_sys->i_overflow_buffer_size = 0;
 430     p_sys->p_overflow_buffer = NULL;
 431     p_sys->i_nb_atomic_operations = 0;
 432     p_sys->p_atomic_operations = NULL;
 433
 434     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 435               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 436               p_filter->fmt_in.audio.i_physical_channels,
 437               p_filter->fmt_in.audio.i_rate ) < 0 )
 438     {
 439         return VLC_EGENERIC;
 440     }
 441
 442     p_filter->pf_audio_filter = Convert;
 443
 444     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 445              (char *)&p_filter->fmt_in.i_codec,
 446              (char *)&p_filter->fmt_out.i_codec,
 447              p_filter->fmt_in.audio.i_physical_channels,
 448              p_filter->fmt_out.audio.i_physical_channels,
 449              p_filter->fmt_in.audio.i_bitspersample,
 450              p_filter->fmt_out.audio.i_bitspersample );
 451
 452     return VLC_SUCCESS;
 453 }
 454
 455 /*****************************************************************************
 456  * CloseFilter
 457  *****************************************************************************/
 458 static void CloseFilter( vlc_object_t *p_this)
 459 {
 460     filter_t *p_filter = (filter_t *) p_this;
 461     filter_sys_t *p_sys = p_filter->p_sys;
 462
 463     var_Destroy( p_this, MONO_CFG "channel" );
 464     var_Destroy( p_this, MONO_CFG "downmix" );
 465     free( p_sys );
 466 }
 467
 468 /*****************************************************************************
 469  * Convert
 470  *****************************************************************************/
 471 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 472 {
 473     aout_filter_t aout_filter;
 474     aout_buffer_t in_buf, out_buf;
 475     block_t *p_out = NULL;
 476     unsigned int i_samples;
 477     int i_out_size;
 478
 479     if( !p_block || !p_block->i_samples )
 480     {
 481         if( p_block )
 482             p_block->pf_release( p_block );
 483         return NULL;
 484     }
 485
 486     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 487                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 488
 489     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 490     if( !p_out )
 491     {
 492         msg_Warn( p_filter, "can't get output buffer" );
 493         p_block->pf_release( p_block );
 494         return NULL;
 495     }
 496     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 497                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 498     p_out->i_dts = p_block->i_dts;
 499     p_out->i_pts = p_block->i_pts;
 500     p_out->i_length = p_block->i_length;
 501
 502     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 503     aout_filter.input = p_filter->fmt_in.audio;
 504     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 505     aout_filter.output = p_filter->fmt_out.audio;
 506     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 507
 508     in_buf.p_buffer = p_block->p_buffer;
 509     in_buf.i_nb_bytes = p_block->i_buffer;
 510     in_buf.i_nb_samples = p_block->i_samples;
 511
 512 #if 0
 513     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 514                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 515     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 516     {
 517         msg_Err( p_filter, "input buffer is not word aligned" );
 518         /* Fix output buffer to be word aligned */
 519     }
 520 #endif
 521
 522     out_buf.p_buffer = p_out->p_buffer;
 523     out_buf.i_nb_bytes = p_out->i_buffer;
 524     out_buf.i_nb_samples = p_out->i_samples;
 525
 526     memset( p_out->p_buffer, 0, i_out_size );
 527     if( p_filter->p_sys->b_downmix )
 528     {
 529         stereo2mono_downmix( (aout_instance_t *)p_filter, &aout_filter,
 530                              &in_buf, &out_buf );
 531         i_samples = mono( (aout_instance_t *)p_filter, &aout_filter,
 532                            &out_buf, &in_buf );
 533     }
 534     else
 535     {
 536         i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
 537                                     &out_buf, &in_buf );
 538     }
 539
 540     p_out->i_buffer = out_buf.i_nb_bytes;
 541     p_out->i_samples = out_buf.i_nb_samples;
 542
 543     p_block->pf_release( p_block );
 544     return p_out;
 545 }
 546
 547 /* stereo2mono_downmix - stereo channels into one mono channel.
 548  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 549  * converted from float into int16_t based downmix
 550  * Written by Boris Dorès <babal@via.ecp.fr>
 551  */
 552 static void stereo2mono_downmix( aout_instance_t * p_aout, aout_filter_t * p_filter,
 553                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 554 {
 555     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 556
 557     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 558     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 559
 560     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 561     byte_t * p_out;
 562     byte_t * p_overflow;
 563     byte_t * p_slide;
 564
 565     size_t i_overflow_size;     /* in bytes */
 566     size_t i_out_size;          /* in bytes */
 567
 568     unsigned int i, j;
 569
 570     int i_source_channel_offset;
 571     int i_dest_channel_offset;
 572     unsigned int i_delay;
 573     double d_amplitude_factor;
 574
 575     /* out buffer characterisitcs */
 576     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 577     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 578     p_out = p_out_buf->p_buffer;
 579     i_out_size = p_out_buf->i_nb_bytes;
 580
 581     if( p_sys != NULL )
 582     {
 583         /* Slide the overflow buffer */
 584         p_overflow = p_sys->p_overflow_buffer;
 585         i_overflow_size = p_sys->i_overflow_buffer_size;
 586
 587         if ( i_out_size > i_overflow_size )
 588             memcpy( p_out, p_overflow, i_overflow_size );
 589         else
 590             memcpy( p_out, p_overflow, i_out_size );
 591
 592         p_slide = p_sys->p_overflow_buffer;
 593         while( p_slide < p_overflow + i_overflow_size )
 594         {
 595             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 596             {
 597                 memset( p_slide, 0, i_out_size );
 598                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 599                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 600                 else
 601                     memcpy( p_slide, p_slide + i_out_size,
 602                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 603             }
 604             else
 605             {
 606                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 607             }
 608             p_slide += i_out_size;
 609         }
 610
 611         /* apply the atomic operations */
 612         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 613         {
 614             /* shorter variable names */
 615             i_source_channel_offset
 616                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 617             i_dest_channel_offset
 618                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 619             i_delay = p_sys->p_atomic_operations[i].i_delay;
 620             d_amplitude_factor
 621                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 622
 623             if( p_out_buf->i_nb_samples > i_delay )
 624             {
 625                 /* current buffer coefficients */
 626                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 627                 {
 628                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 629                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 630                            * d_amplitude_factor;
 631                 }
 632
 633                 /* overflow buffer coefficients */
 634                 for( j = 0; j < i_delay; j++ )
 635                 {
 636                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 637                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 638                            * i_input_nb + i_source_channel_offset ]
 639                            * d_amplitude_factor;
 640                 }
 641             }
 642             else
 643             {
 644                 /* overflow buffer coefficients only */
 645                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 646                 {
 647                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 648                         * i_output_nb + i_dest_channel_offset ]
 649                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 650                            * d_amplitude_factor;
 651                 }
 652             }
 653         }
 654     }
 655     else
 656     {
 657         memset( p_out, 0, i_out_size );
 658     }
 659 }
 660
 661 /* Simple stereo to mono mixing. */
 662 static unsigned int mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 663                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 664 {
 665     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 666     int16_t *p_in, *p_out;
 667     unsigned int n = 0, r = 0;
 668
 669     p_in = (int16_t *) p_input->p_buffer;
 670     p_out = (int16_t *) p_output->p_buffer;
 671
 672     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 673     {
 674         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 675         r++;
 676         n += 2;
 677     }
 678     return r;
 679 }
 680
 681 /* Simple stereo to mono mixing. */
 682 static unsigned int stereo_to_mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 683                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 684 {
 685     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 686     int16_t *p_in, *p_out;
 687     unsigned int n;
 688
 689     p_in = (int16_t *) p_input->p_buffer;
 690     p_out = (int16_t *) p_output->p_buffer;
 691
 692     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 693     {
 694         /* Fake real mono. */
 695         if( p_sys->i_channel_selected == -1)
 696         {
 697             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 698             n++;
 699         }
 700         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 701         {
 702             p_out[n] = p_out[n+1] = p_in[n];
 703         }
 704     }
 705     return n;
 706 }