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 <stdlib.h>                                      /* malloc(), free() */
  28 #include <string.h>
  29 #include <math.h>                                        /* sqrt */
  30
  31 #ifdef HAVE_STDINT_H
  32 #   include <stdint.h>                                         /* int16_t .. */
  33 #elif defined(HAVE_INTTYPES_H)
  34 #   include <inttypes.h>                                       /* int16_t .. */
  35 #endif
  36
  37 #ifdef HAVE_UNISTD_H
  38 #   include <unistd.h>
  39 #endif
  40
  41 #include <vlc/vlc.h>
  42 #include <vlc_es.h>
  43 #include <vlc_block.h>
  44 #include <vlc_filter.h>
  45 #include <vlc_aout.h>
  46
  47 /*****************************************************************************
  48  * Local prototypes
  49  *****************************************************************************/
  50 static int  OpenFilter    ( vlc_object_t * );
  51 static void CloseFilter   ( vlc_object_t * );
  52
  53 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  54
  55 static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *,
  56                                     aout_buffer_t *, aout_buffer_t * );
  57 static unsigned int mono( aout_instance_t *, aout_filter_t *,
  58                                     aout_buffer_t *, aout_buffer_t * );
  59 static void stereo2mono_downmix( aout_instance_t *, aout_filter_t *,
  60                                  aout_buffer_t *, aout_buffer_t * );
  61
  62 /*****************************************************************************
  63  * Local structures
  64  *****************************************************************************/
  65 struct atomic_operation_t
  66 {
  67     int i_source_channel_offset;
  68     int i_dest_channel_offset;
  69     unsigned int i_delay;/* in sample unit */
  70     double d_amplitude_factor;
  71 };
  72
  73 struct filter_sys_t
  74 {
  75     vlc_bool_t b_downmix;
  76
  77     unsigned int i_nb_channels; /* number of int16_t per sample */
  78     int i_channel_selected;
  79     int i_bitspersample;
  80
  81     size_t i_overflow_buffer_size;/* in bytes */
  82     byte_t * p_overflow_buffer;
  83     unsigned int i_nb_atomic_operations;
  84     struct atomic_operation_t * p_atomic_operations;
  85 };
  86
  87 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithme.")
  88 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  89     "downmix algorithm that is used in the headphone channel mixer. It" \
  90     "gives the effect of standing in a room full of speakers." )
  91
  92 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  93 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  94     "except the selected channel. Choose one from (0=left, 1=right " \
  95     "2=rear left, 3=rear right, 4=center, 5=left front)")
  96
  97 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
  98 static const char *ppsz_pos_descriptions[] =
  99 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
 100   N_("Left front") };
 101
 102 /* our internal channel order (WG-4 order) */
 103 static const uint32_t pi_channels_out[] =
 104 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
 105   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
 106
 107 #define MONO_CFG "sout-mono-"
 108 /*****************************************************************************
 109  * Module descriptor
 110  *****************************************************************************/
 111 vlc_module_begin();
 112     set_description( _("Audio filter for stereo to mono conversion") );
 113     set_capability( "audio filter2", 0 );
 114
 115     add_bool( MONO_CFG "downmix", VLC_FALSE, NULL, MONO_DOWNMIX_TEXT,
 116               MONO_DOWNMIX_LONGTEXT, VLC_FALSE );
 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
 121     set_category( CAT_AUDIO );
 122     set_subcategory( SUBCAT_AUDIO_MISC );
 123     set_callbacks( OpenFilter, CloseFilter );
 124     set_shortname( "Mono" );
 125 vlc_module_end();
 126
 127 /* Init() and ComputeChannelOperations() -
 128  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 129  * converted from float into int16_t based downmix
 130  * Written by Boris Dorès <babal@via.ecp.fr>
 131  */
 132
 133 /*****************************************************************************
 134  * Init: initialize internal data structures
 135  * and computes the needed atomic operations
 136  *****************************************************************************/
 137 /* x and z represent the coordinates of the virtual speaker
 138  *  relatively to the center of the listener's head, measured in meters :
 139  *
 140  *  left              right
 141  *Z
 142  *-
 143  *a          head
 144  *x
 145  *i
 146  *s
 147  *  rear left    rear right
 148  *
 149  *          x-axis
 150  *  */
 151 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 152         unsigned int i_rate, unsigned int i_next_atomic_operation,
 153         int i_source_channel_offset, double d_x, double d_z,
 154         double d_compensation_length, double d_channel_amplitude_factor )
 155 {
 156     double d_c = 340; /*sound celerity (unit: m/s)*/
 157     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 158
 159     /* Left ear */
 160     p_data->p_atomic_operations[i_next_atomic_operation]
 161         .i_source_channel_offset = i_source_channel_offset;
 162     p_data->p_atomic_operations[i_next_atomic_operation]
 163         .i_dest_channel_offset = 0;/* left */
 164     p_data->p_atomic_operations[i_next_atomic_operation]
 165         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 166                           / d_c * i_rate - d_compensation_delay );
 167     if( d_x < 0 )
 168     {
 169         p_data->p_atomic_operations[i_next_atomic_operation]
 170             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 171     }
 172     else if( d_x > 0 )
 173     {
 174         p_data->p_atomic_operations[i_next_atomic_operation]
 175             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 176     }
 177     else
 178     {
 179         p_data->p_atomic_operations[i_next_atomic_operation]
 180             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 181     }
 182
 183     /* Right ear */
 184     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 185         .i_source_channel_offset = i_source_channel_offset;
 186     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 187         .i_dest_channel_offset = 1;/* right */
 188     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 189         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 190                           / d_c * i_rate - d_compensation_delay );
 191     if( d_x < 0 )
 192     {
 193         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 194             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 195     }
 196     else if( d_x > 0 )
 197     {
 198         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 199             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 200     }
 201     else
 202     {
 203         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 204             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 205     }
 206 }
 207
 208 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 209                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 210                  unsigned int i_rate )
 211 {
 212     double d_x = config_GetInt( p_this, "headphone-dim" );
 213     double d_z = d_x;
 214     double d_z_rear = -d_x/3;
 215     double d_min = 0;
 216     unsigned int i_next_atomic_operation;
 217     int i_source_channel_offset;
 218     unsigned int i;
 219
 220     if( p_data == NULL )
 221     {
 222         msg_Dbg( p_this, "passing a null pointer as argument" );
 223         return 0;
 224     }
 225
 226     if( config_GetInt( p_this, "headphone-compensate" ) )
 227     {
 228         /* minimal distance to any speaker */
 229         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 230         {
 231             d_min = d_z_rear;
 232         }
 233         else
 234         {
 235             d_min = d_z;
 236         }
 237     }
 238
 239     /* Number of elementary operations */
 240     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 241     if( i_physical_channels & AOUT_CHAN_CENTER )
 242     {
 243         p_data->i_nb_atomic_operations += 2;
 244     }
 245     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 246             * p_data->i_nb_atomic_operations );
 247     if( p_data->p_atomic_operations == NULL )
 248     {
 249         msg_Err( p_this, "out of memory" );
 250         return -1;
 251     }
 252
 253     /* For each virtual speaker, computes elementary wave propagation time
 254      * to each ear */
 255     i_next_atomic_operation = 0;
 256     i_source_channel_offset = 0;
 257     if( i_physical_channels & AOUT_CHAN_LEFT )
 258     {
 259         ComputeChannelOperations( p_data , i_rate
 260                 , i_next_atomic_operation , i_source_channel_offset
 261                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 262         i_next_atomic_operation += 2;
 263         i_source_channel_offset++;
 264     }
 265     if( i_physical_channels & AOUT_CHAN_RIGHT )
 266     {
 267         ComputeChannelOperations( p_data , i_rate
 268                 , i_next_atomic_operation , i_source_channel_offset
 269                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 270         i_next_atomic_operation += 2;
 271         i_source_channel_offset++;
 272     }
 273     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 274     {
 275         ComputeChannelOperations( p_data , i_rate
 276                 , i_next_atomic_operation , i_source_channel_offset
 277                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 278         i_next_atomic_operation += 2;
 279         i_source_channel_offset++;
 280     }
 281     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 282     {
 283         ComputeChannelOperations( p_data , i_rate
 284                 , i_next_atomic_operation , i_source_channel_offset
 285                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 286         i_next_atomic_operation += 2;
 287         i_source_channel_offset++;
 288     }
 289     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 290     {
 291         ComputeChannelOperations( p_data , i_rate
 292                 , i_next_atomic_operation , i_source_channel_offset
 293                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 294         i_next_atomic_operation += 2;
 295         i_source_channel_offset++;
 296     }
 297     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 298     {
 299         ComputeChannelOperations( p_data , i_rate
 300                 , i_next_atomic_operation , i_source_channel_offset
 301                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 302         i_next_atomic_operation += 2;
 303         i_source_channel_offset++;
 304     }
 305     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 306     {
 307         ComputeChannelOperations( p_data , i_rate
 308                 , i_next_atomic_operation , i_source_channel_offset
 309                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 310         i_next_atomic_operation += 2;
 311         i_source_channel_offset++;
 312     }
 313     if( i_physical_channels & AOUT_CHAN_CENTER )
 314     {
 315         /* having two center channels increases the spatialization effect */
 316         ComputeChannelOperations( p_data , i_rate
 317                 , i_next_atomic_operation , i_source_channel_offset
 318                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 319         i_next_atomic_operation += 2;
 320         ComputeChannelOperations( p_data , i_rate
 321                 , i_next_atomic_operation , i_source_channel_offset
 322                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 323         i_next_atomic_operation += 2;
 324         i_source_channel_offset++;
 325     }
 326     if( i_physical_channels & AOUT_CHAN_LFE )
 327     {
 328         ComputeChannelOperations( p_data , i_rate
 329                 , i_next_atomic_operation , i_source_channel_offset
 330                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 331         i_next_atomic_operation += 2;
 332         i_source_channel_offset++;
 333     }
 334
 335     /* Initialize the overflow buffer
 336      * we need it because the process induce a delay in the samples */
 337     p_data->i_overflow_buffer_size = 0;
 338     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 339     {
 340         if( p_data->i_overflow_buffer_size
 341                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 342         {
 343             p_data->i_overflow_buffer_size
 344                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 345         }
 346     }
 347     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 348     if( p_data->p_atomic_operations == NULL )
 349     {
 350         msg_Err( p_this, "out of memory" );
 351         return -1;
 352     }
 353     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 354
 355     /* end */
 356     return 0;
 357 }
 358
 359 /*****************************************************************************
 360  * OpenFilter
 361  *****************************************************************************/
 362 static int OpenFilter( vlc_object_t *p_this )
 363 {
 364     filter_t * p_filter = (filter_t *)p_this;
 365     filter_sys_t *p_sys = NULL;
 366
 367     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 368     {
 369         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
 370         return VLC_EGENERIC;
 371     }
 372
 373     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
 374         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
 375     {
 376         msg_Err( p_this, "filter discarded (invalid format)" );
 377         return -1;
 378     }
 379
 380     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 381         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 382         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
 383         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
 384         (p_filter->fmt_in.audio.i_bitspersample !=
 385                                     p_filter->fmt_out.audio.i_bitspersample))
 386     {
 387         msg_Err( p_this, "couldn't load mono filter" );
 388         return VLC_EGENERIC;
 389     }
 390
 391     /* Allocate the memory needed to store the module's structure */
 392     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 393     if( p_sys == NULL )
 394     {
 395         msg_Err( p_filter, "out of memory" );
 396         return VLC_EGENERIC;
 397     }
 398
 399     var_Create( p_this, MONO_CFG "downmix",
 400                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 401     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 402
 403     var_Create( p_this, MONO_CFG "channel",
 404                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 405     p_sys->i_channel_selected =
 406             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 407
 408     if( p_sys->b_downmix )
 409     {
 410         msg_Dbg( p_this, "using stereo to mono downmix" );
 411         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 412         p_filter->fmt_out.audio.i_channels = 1;
 413     }
 414     else
 415     {
 416         msg_Dbg( p_this, "using pseudo mono" );
 417         p_filter->fmt_out.audio.i_physical_channels =
 418                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 419         p_filter->fmt_out.audio.i_channels = 2;
 420     }
 421
 422     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 423     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 424
 425     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 426     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 427
 428     p_sys->i_overflow_buffer_size = 0;
 429     p_sys->p_overflow_buffer = NULL;
 430     p_sys->i_nb_atomic_operations = 0;
 431     p_sys->p_atomic_operations = NULL;
 432
 433     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 434               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 435               p_filter->fmt_in.audio.i_physical_channels,
 436               p_filter->fmt_in.audio.i_rate ) < 0 )
 437     {
 438         return VLC_EGENERIC;
 439     }
 440
 441     p_filter->pf_audio_filter = Convert;
 442
 443     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 444              (char *)&p_filter->fmt_in.i_codec,
 445              (char *)&p_filter->fmt_out.i_codec,
 446              p_filter->fmt_in.audio.i_physical_channels,
 447              p_filter->fmt_out.audio.i_physical_channels,
 448              p_filter->fmt_in.audio.i_bitspersample,
 449              p_filter->fmt_out.audio.i_bitspersample );
 450
 451     return VLC_SUCCESS;
 452 }
 453
 454 /*****************************************************************************
 455  * CloseFilter
 456  *****************************************************************************/
 457 static void CloseFilter( vlc_object_t *p_this)
 458 {
 459     filter_t *p_filter = (filter_t *) p_this;
 460     filter_sys_t *p_sys = p_filter->p_sys;
 461
 462     var_Destroy( p_this, MONO_CFG "channel" );
 463     var_Destroy( p_this, MONO_CFG "downmix" );
 464     free( p_sys );
 465 }
 466
 467 /*****************************************************************************
 468  * Convert
 469  *****************************************************************************/
 470 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 471 {
 472     aout_filter_t aout_filter;
 473     aout_buffer_t in_buf, out_buf;
 474     block_t *p_out = NULL;
 475     unsigned int i_samples;
 476     int i_out_size;
 477
 478     if( !p_block || !p_block->i_samples )
 479     {
 480         if( p_block )
 481             p_block->pf_release( p_block );
 482         return NULL;
 483     }
 484
 485     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 486                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 487
 488     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 489     if( !p_out )
 490     {
 491         msg_Warn( p_filter, "can't get output buffer" );
 492         p_block->pf_release( p_block );
 493         return NULL;
 494     }
 495     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 496                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 497     p_out->i_dts = p_block->i_dts;
 498     p_out->i_pts = p_block->i_pts;
 499     p_out->i_length = p_block->i_length;
 500
 501     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 502     aout_filter.input = p_filter->fmt_in.audio;
 503     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 504     aout_filter.output = p_filter->fmt_out.audio;
 505     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 506
 507     in_buf.p_buffer = p_block->p_buffer;
 508     in_buf.i_nb_bytes = p_block->i_buffer;
 509     in_buf.i_nb_samples = p_block->i_samples;
 510
 511 #if 0
 512     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 513                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 514     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 515     {
 516         msg_Err( p_filter, "input buffer is not word aligned" );
 517         /* Fix output buffer to be word aligned */
 518     }
 519 #endif
 520
 521     out_buf.p_buffer = p_out->p_buffer;
 522     out_buf.i_nb_bytes = p_out->i_buffer;
 523     out_buf.i_nb_samples = p_out->i_samples;
 524
 525     memset( p_out->p_buffer, 0, i_out_size );
 526     if( p_filter->p_sys->b_downmix )
 527     {
 528         stereo2mono_downmix( (aout_instance_t *)p_filter, &aout_filter,
 529                              &in_buf, &out_buf );
 530         i_samples = mono( (aout_instance_t *)p_filter, &aout_filter,
 531                            &out_buf, &in_buf );
 532     }
 533     else
 534     {
 535         i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
 536                                     &out_buf, &in_buf );
 537     }
 538
 539     p_out->i_buffer = out_buf.i_nb_bytes;
 540     p_out->i_samples = out_buf.i_nb_samples;
 541
 542     p_block->pf_release( p_block );
 543     return p_out;
 544 }
 545
 546 /* stereo2mono_downmix - stereo channels into one mono channel.
 547  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 548  * converted from float into int16_t based downmix
 549  * Written by Boris Dorès <babal@via.ecp.fr>
 550  */
 551 static void stereo2mono_downmix( aout_instance_t * p_aout, aout_filter_t * p_filter,
 552                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 553 {
 554     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 555
 556     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 557     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 558
 559     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 560     byte_t * p_out;
 561     byte_t * p_overflow;
 562     byte_t * p_slide;
 563
 564     size_t i_overflow_size;     /* in bytes */
 565     size_t i_out_size;          /* in bytes */
 566
 567     unsigned int i, j;
 568
 569     int i_source_channel_offset;
 570     int i_dest_channel_offset;
 571     unsigned int i_delay;
 572     double d_amplitude_factor;
 573
 574     /* out buffer characterisitcs */
 575     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 576     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 577     p_out = p_out_buf->p_buffer;
 578     i_out_size = p_out_buf->i_nb_bytes;
 579
 580     if( p_sys != NULL )
 581     {
 582         /* Slide the overflow buffer */
 583         p_overflow = p_sys->p_overflow_buffer;
 584         i_overflow_size = p_sys->i_overflow_buffer_size;
 585
 586         if ( i_out_size > i_overflow_size )
 587             memcpy( p_out, p_overflow, i_overflow_size );
 588         else
 589             memcpy( p_out, p_overflow, i_out_size );
 590
 591         p_slide = p_sys->p_overflow_buffer;
 592         while( p_slide < p_overflow + i_overflow_size )
 593         {
 594             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 595             {
 596                 memset( p_slide, 0, i_out_size );
 597                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 598                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 599                 else
 600                     memcpy( p_slide, p_slide + i_out_size,
 601                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 602             }
 603             else
 604             {
 605                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 606             }
 607             p_slide += i_out_size;
 608         }
 609
 610         /* apply the atomic operations */
 611         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 612         {
 613             /* shorter variable names */
 614             i_source_channel_offset
 615                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 616             i_dest_channel_offset
 617                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 618             i_delay = p_sys->p_atomic_operations[i].i_delay;
 619             d_amplitude_factor
 620                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 621
 622             if( p_out_buf->i_nb_samples > i_delay )
 623             {
 624                 /* current buffer coefficients */
 625                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 626                 {
 627                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 628                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 629                            * d_amplitude_factor;
 630                 }
 631
 632                 /* overflow buffer coefficients */
 633                 for( j = 0; j < i_delay; j++ )
 634                 {
 635                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 636                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 637                            * i_input_nb + i_source_channel_offset ]
 638                            * d_amplitude_factor;
 639                 }
 640             }
 641             else
 642             {
 643                 /* overflow buffer coefficients only */
 644                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 645                 {
 646                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 647                         * i_output_nb + i_dest_channel_offset ]
 648                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 649                            * d_amplitude_factor;
 650                 }
 651             }
 652         }
 653     }
 654     else
 655     {
 656         memset( p_out, 0, i_out_size );
 657     }
 658 }
 659
 660 /* Simple stereo to mono mixing. */
 661 static unsigned int mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 662                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 663 {
 664     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 665     int16_t *p_in, *p_out;
 666     unsigned int n = 0, r = 0;
 667
 668     p_in = (int16_t *) p_input->p_buffer;
 669     p_out = (int16_t *) p_output->p_buffer;
 670
 671     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 672     {
 673         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 674         r++;
 675         n += 2;
 676     }
 677     return r;
 678 }
 679
 680 /* Simple stereo to mono mixing. */
 681 static unsigned int stereo_to_mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 682                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 683 {
 684     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 685     int16_t *p_in, *p_out;
 686     unsigned int n;
 687
 688     p_in = (int16_t *) p_input->p_buffer;
 689     p_out = (int16_t *) p_output->p_buffer;
 690
 691     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 692     {
 693         /* Fake real mono. */
 694         if( p_sys->i_channel_selected == -1)
 695         {
 696             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 697             n++;
 698         }
 699         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 700         {
 701             p_out[n] = p_out[n+1] = p_in[n];
 702         }
 703     }
 704     return n;
 705 }