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 #ifdef HAVE_CONFIG_H
  40 # include "config.h"
  41 #endif
  42
  43 #include <vlc/vlc.h>
  44 #include <vlc_es.h>
  45 #include <vlc_block.h>
  46 #include <vlc_filter.h>
  47 #include <vlc_aout.h>
  48
  49 /*****************************************************************************
  50  * Local prototypes
  51  *****************************************************************************/
  52 static int  OpenFilter    ( vlc_object_t * );
  53 static void CloseFilter   ( vlc_object_t * );
  54
  55 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  56
  57 static unsigned int stereo_to_mono( aout_instance_t *, aout_filter_t *,
  58                                     aout_buffer_t *, aout_buffer_t * );
  59 static unsigned int mono( aout_instance_t *, aout_filter_t *,
  60                                     aout_buffer_t *, aout_buffer_t * );
  61 static void stereo2mono_downmix( aout_instance_t *, aout_filter_t *,
  62                                  aout_buffer_t *, aout_buffer_t * );
  63
  64 /*****************************************************************************
  65  * Local structures
  66  *****************************************************************************/
  67 struct atomic_operation_t
  68 {
  69     int i_source_channel_offset;
  70     int i_dest_channel_offset;
  71     unsigned int i_delay;/* in sample unit */
  72     double d_amplitude_factor;
  73 };
  74
  75 struct filter_sys_t
  76 {
  77     vlc_bool_t b_downmix;
  78
  79     unsigned int i_nb_channels; /* number of int16_t per sample */
  80     int i_channel_selected;
  81     int i_bitspersample;
  82
  83     size_t i_overflow_buffer_size;/* in bytes */
  84     byte_t * p_overflow_buffer;
  85     unsigned int i_nb_atomic_operations;
  86     struct atomic_operation_t * p_atomic_operations;
  87 };
  88
  89 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
  90 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  91     "downmix algorithm that is used in the headphone channel mixer. It" \
  92     "gives the effect of standing in a room full of speakers." )
  93
  94 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  95 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  96     "except the selected channel. Choose one from (0=left, 1=right, " \
  97     "2=rear left, 3=rear right, 4=center, 5=left front)")
  98
  99 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
 100 static const char *ppsz_pos_descriptions[] =
 101 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
 102   N_("Left front") };
 103
 104 /* our internal channel order (WG-4 order) */
 105 static const uint32_t pi_channels_out[] =
 106 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
 107   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
 108
 109 #define MONO_CFG "sout-mono-"
 110 /*****************************************************************************
 111  * Module descriptor
 112  *****************************************************************************/
 113 vlc_module_begin();
 114     set_description( _("Audio filter for stereo to mono conversion") );
 115     set_capability( "audio filter2", 0 );
 116
 117     add_bool( MONO_CFG "downmix", VLC_FALSE, NULL, MONO_DOWNMIX_TEXT,
 118               MONO_DOWNMIX_LONGTEXT, VLC_FALSE );
 119     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 120         MONO_CHANNEL_LONGTEXT, VLC_FALSE );
 121         change_integer_list( pi_pos_values, ppsz_pos_descriptions, 0 );
 122
 123     set_category( CAT_AUDIO );
 124     set_subcategory( SUBCAT_AUDIO_MISC );
 125     set_callbacks( OpenFilter, CloseFilter );
 126     set_shortname( "Mono" );
 127 vlc_module_end();
 128
 129 /* Init() and ComputeChannelOperations() -
 130  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 131  * converted from float into int16_t based downmix
 132  * Written by Boris Dorès <babal@via.ecp.fr>
 133  */
 134
 135 /*****************************************************************************
 136  * Init: initialize internal data structures
 137  * and computes the needed atomic operations
 138  *****************************************************************************/
 139 /* x and z represent the coordinates of the virtual speaker
 140  *  relatively to the center of the listener's head, measured in meters :
 141  *
 142  *  left              right
 143  *Z
 144  *-
 145  *a          head
 146  *x
 147  *i
 148  *s
 149  *  rear left    rear right
 150  *
 151  *          x-axis
 152  *  */
 153 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 154         unsigned int i_rate, unsigned int i_next_atomic_operation,
 155         int i_source_channel_offset, double d_x, double d_z,
 156         double d_compensation_length, double d_channel_amplitude_factor )
 157 {
 158     double d_c = 340; /*sound celerity (unit: m/s)*/
 159     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 160
 161     /* Left ear */
 162     p_data->p_atomic_operations[i_next_atomic_operation]
 163         .i_source_channel_offset = i_source_channel_offset;
 164     p_data->p_atomic_operations[i_next_atomic_operation]
 165         .i_dest_channel_offset = 0;/* left */
 166     p_data->p_atomic_operations[i_next_atomic_operation]
 167         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 168                           / d_c * i_rate - d_compensation_delay );
 169     if( d_x < 0 )
 170     {
 171         p_data->p_atomic_operations[i_next_atomic_operation]
 172             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 173     }
 174     else if( d_x > 0 )
 175     {
 176         p_data->p_atomic_operations[i_next_atomic_operation]
 177             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 178     }
 179     else
 180     {
 181         p_data->p_atomic_operations[i_next_atomic_operation]
 182             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 183     }
 184
 185     /* Right ear */
 186     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 187         .i_source_channel_offset = i_source_channel_offset;
 188     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 189         .i_dest_channel_offset = 1;/* right */
 190     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 191         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 192                           / d_c * i_rate - d_compensation_delay );
 193     if( d_x < 0 )
 194     {
 195         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 196             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 197     }
 198     else if( d_x > 0 )
 199     {
 200         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 201             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 202     }
 203     else
 204     {
 205         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 206             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 207     }
 208 }
 209
 210 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 211                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 212                  unsigned int i_rate )
 213 {
 214     double d_x = config_GetInt( p_this, "headphone-dim" );
 215     double d_z = d_x;
 216     double d_z_rear = -d_x/3;
 217     double d_min = 0;
 218     unsigned int i_next_atomic_operation;
 219     int i_source_channel_offset;
 220     unsigned int i;
 221
 222     if( p_data == NULL )
 223     {
 224         msg_Dbg( p_this, "passing a null pointer as argument" );
 225         return 0;
 226     }
 227
 228     if( config_GetInt( p_this, "headphone-compensate" ) )
 229     {
 230         /* minimal distance to any speaker */
 231         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 232         {
 233             d_min = d_z_rear;
 234         }
 235         else
 236         {
 237             d_min = d_z;
 238         }
 239     }
 240
 241     /* Number of elementary operations */
 242     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 243     if( i_physical_channels & AOUT_CHAN_CENTER )
 244     {
 245         p_data->i_nb_atomic_operations += 2;
 246     }
 247     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 248             * p_data->i_nb_atomic_operations );
 249     if( p_data->p_atomic_operations == NULL )
 250     {
 251         msg_Err( p_this, "out of memory" );
 252         return -1;
 253     }
 254
 255     /* For each virtual speaker, computes elementary wave propagation time
 256      * to each ear */
 257     i_next_atomic_operation = 0;
 258     i_source_channel_offset = 0;
 259     if( i_physical_channels & AOUT_CHAN_LEFT )
 260     {
 261         ComputeChannelOperations( p_data , i_rate
 262                 , i_next_atomic_operation , i_source_channel_offset
 263                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 264         i_next_atomic_operation += 2;
 265         i_source_channel_offset++;
 266     }
 267     if( i_physical_channels & AOUT_CHAN_RIGHT )
 268     {
 269         ComputeChannelOperations( p_data , i_rate
 270                 , i_next_atomic_operation , i_source_channel_offset
 271                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 272         i_next_atomic_operation += 2;
 273         i_source_channel_offset++;
 274     }
 275     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 276     {
 277         ComputeChannelOperations( p_data , i_rate
 278                 , i_next_atomic_operation , i_source_channel_offset
 279                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 280         i_next_atomic_operation += 2;
 281         i_source_channel_offset++;
 282     }
 283     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 284     {
 285         ComputeChannelOperations( p_data , i_rate
 286                 , i_next_atomic_operation , i_source_channel_offset
 287                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 288         i_next_atomic_operation += 2;
 289         i_source_channel_offset++;
 290     }
 291     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 292     {
 293         ComputeChannelOperations( p_data , i_rate
 294                 , i_next_atomic_operation , i_source_channel_offset
 295                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 296         i_next_atomic_operation += 2;
 297         i_source_channel_offset++;
 298     }
 299     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 300     {
 301         ComputeChannelOperations( p_data , i_rate
 302                 , i_next_atomic_operation , i_source_channel_offset
 303                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 304         i_next_atomic_operation += 2;
 305         i_source_channel_offset++;
 306     }
 307     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 308     {
 309         ComputeChannelOperations( p_data , i_rate
 310                 , i_next_atomic_operation , i_source_channel_offset
 311                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 312         i_next_atomic_operation += 2;
 313         i_source_channel_offset++;
 314     }
 315     if( i_physical_channels & AOUT_CHAN_CENTER )
 316     {
 317         /* having two center channels increases the spatialization effect */
 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         ComputeChannelOperations( p_data , i_rate
 323                 , i_next_atomic_operation , i_source_channel_offset
 324                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 325         i_next_atomic_operation += 2;
 326         i_source_channel_offset++;
 327     }
 328     if( i_physical_channels & AOUT_CHAN_LFE )
 329     {
 330         ComputeChannelOperations( p_data , i_rate
 331                 , i_next_atomic_operation , i_source_channel_offset
 332                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 333         i_next_atomic_operation += 2;
 334         i_source_channel_offset++;
 335     }
 336
 337     /* Initialize the overflow buffer
 338      * we need it because the process induce a delay in the samples */
 339     p_data->i_overflow_buffer_size = 0;
 340     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 341     {
 342         if( p_data->i_overflow_buffer_size
 343                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 344         {
 345             p_data->i_overflow_buffer_size
 346                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 347         }
 348     }
 349     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 350     if( p_data->p_atomic_operations == NULL )
 351     {
 352         msg_Err( p_this, "out of memory" );
 353         return -1;
 354     }
 355     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 356
 357     /* end */
 358     return 0;
 359 }
 360
 361 /*****************************************************************************
 362  * OpenFilter
 363  *****************************************************************************/
 364 static int OpenFilter( vlc_object_t *p_this )
 365 {
 366     filter_t * p_filter = (filter_t *)p_this;
 367     filter_sys_t *p_sys = NULL;
 368
 369     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 370     {
 371         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
 372         return VLC_EGENERIC;
 373     }
 374
 375     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
 376         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
 377     {
 378         msg_Err( p_this, "filter discarded (invalid format)" );
 379         return -1;
 380     }
 381
 382     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 383         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 384         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
 385         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
 386         (p_filter->fmt_in.audio.i_bitspersample !=
 387                                     p_filter->fmt_out.audio.i_bitspersample))
 388     {
 389         msg_Err( p_this, "couldn't load mono filter" );
 390         return VLC_EGENERIC;
 391     }
 392
 393     /* Allocate the memory needed to store the module's structure */
 394     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 395     if( p_sys == NULL )
 396     {
 397         msg_Err( p_filter, "out of memory" );
 398         return VLC_EGENERIC;
 399     }
 400
 401     var_Create( p_this, MONO_CFG "downmix",
 402                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 403     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 404
 405     var_Create( p_this, MONO_CFG "channel",
 406                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 407     p_sys->i_channel_selected =
 408             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 409
 410     if( p_sys->b_downmix )
 411     {
 412         msg_Dbg( p_this, "using stereo to mono downmix" );
 413         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 414         p_filter->fmt_out.audio.i_channels = 1;
 415     }
 416     else
 417     {
 418         msg_Dbg( p_this, "using pseudo mono" );
 419         p_filter->fmt_out.audio.i_physical_channels =
 420                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 421         p_filter->fmt_out.audio.i_channels = 2;
 422     }
 423
 424     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 425     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 426
 427     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 428     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 429
 430     p_sys->i_overflow_buffer_size = 0;
 431     p_sys->p_overflow_buffer = NULL;
 432     p_sys->i_nb_atomic_operations = 0;
 433     p_sys->p_atomic_operations = NULL;
 434
 435     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 436               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 437               p_filter->fmt_in.audio.i_physical_channels,
 438               p_filter->fmt_in.audio.i_rate ) < 0 )
 439     {
 440         return VLC_EGENERIC;
 441     }
 442
 443     p_filter->pf_audio_filter = Convert;
 444
 445     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 446              (char *)&p_filter->fmt_in.i_codec,
 447              (char *)&p_filter->fmt_out.i_codec,
 448              p_filter->fmt_in.audio.i_physical_channels,
 449              p_filter->fmt_out.audio.i_physical_channels,
 450              p_filter->fmt_in.audio.i_bitspersample,
 451              p_filter->fmt_out.audio.i_bitspersample );
 452
 453     return VLC_SUCCESS;
 454 }
 455
 456 /*****************************************************************************
 457  * CloseFilter
 458  *****************************************************************************/
 459 static void CloseFilter( vlc_object_t *p_this)
 460 {
 461     filter_t *p_filter = (filter_t *) p_this;
 462     filter_sys_t *p_sys = p_filter->p_sys;
 463
 464     var_Destroy( p_this, MONO_CFG "channel" );
 465     var_Destroy( p_this, MONO_CFG "downmix" );
 466     free( p_sys );
 467 }
 468
 469 /*****************************************************************************
 470  * Convert
 471  *****************************************************************************/
 472 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 473 {
 474     aout_filter_t aout_filter;
 475     aout_buffer_t in_buf, out_buf;
 476     block_t *p_out = NULL;
 477     unsigned int i_samples;
 478     int i_out_size;
 479
 480     if( !p_block || !p_block->i_samples )
 481     {
 482         if( p_block )
 483             p_block->pf_release( p_block );
 484         return NULL;
 485     }
 486
 487     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 488                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 489
 490     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 491     if( !p_out )
 492     {
 493         msg_Warn( p_filter, "can't get output buffer" );
 494         p_block->pf_release( p_block );
 495         return NULL;
 496     }
 497     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 498                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 499     p_out->i_dts = p_block->i_dts;
 500     p_out->i_pts = p_block->i_pts;
 501     p_out->i_length = p_block->i_length;
 502
 503     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 504     aout_filter.input = p_filter->fmt_in.audio;
 505     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 506     aout_filter.output = p_filter->fmt_out.audio;
 507     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 508
 509     in_buf.p_buffer = p_block->p_buffer;
 510     in_buf.i_nb_bytes = p_block->i_buffer;
 511     in_buf.i_nb_samples = p_block->i_samples;
 512
 513 #if 0
 514     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 515                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 516     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 517     {
 518         msg_Err( p_filter, "input buffer is not word aligned" );
 519         /* Fix output buffer to be word aligned */
 520     }
 521 #endif
 522
 523     out_buf.p_buffer = p_out->p_buffer;
 524     out_buf.i_nb_bytes = p_out->i_buffer;
 525     out_buf.i_nb_samples = p_out->i_samples;
 526
 527     memset( p_out->p_buffer, 0, i_out_size );
 528     if( p_filter->p_sys->b_downmix )
 529     {
 530         stereo2mono_downmix( (aout_instance_t *)p_filter, &aout_filter,
 531                              &in_buf, &out_buf );
 532         i_samples = mono( (aout_instance_t *)p_filter, &aout_filter,
 533                            &out_buf, &in_buf );
 534     }
 535     else
 536     {
 537         i_samples = stereo_to_mono( (aout_instance_t *)p_filter, &aout_filter,
 538                                     &out_buf, &in_buf );
 539     }
 540
 541     p_out->i_buffer = out_buf.i_nb_bytes;
 542     p_out->i_samples = out_buf.i_nb_samples;
 543
 544     p_block->pf_release( p_block );
 545     return p_out;
 546 }
 547
 548 /* stereo2mono_downmix - stereo channels into one mono channel.
 549  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 550  * converted from float into int16_t based downmix
 551  * Written by Boris Dorès <babal@via.ecp.fr>
 552  */
 553 static void stereo2mono_downmix( aout_instance_t * p_aout, aout_filter_t * p_filter,
 554                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 555 {
 556     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 557
 558     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 559     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 560
 561     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 562     byte_t * p_out;
 563     byte_t * p_overflow;
 564     byte_t * p_slide;
 565
 566     size_t i_overflow_size;     /* in bytes */
 567     size_t i_out_size;          /* in bytes */
 568
 569     unsigned int i, j;
 570
 571     int i_source_channel_offset;
 572     int i_dest_channel_offset;
 573     unsigned int i_delay;
 574     double d_amplitude_factor;
 575
 576     /* out buffer characterisitcs */
 577     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 578     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 579     p_out = p_out_buf->p_buffer;
 580     i_out_size = p_out_buf->i_nb_bytes;
 581
 582     if( p_sys != NULL )
 583     {
 584         /* Slide the overflow buffer */
 585         p_overflow = p_sys->p_overflow_buffer;
 586         i_overflow_size = p_sys->i_overflow_buffer_size;
 587
 588         if ( i_out_size > i_overflow_size )
 589             memcpy( p_out, p_overflow, i_overflow_size );
 590         else
 591             memcpy( p_out, p_overflow, i_out_size );
 592
 593         p_slide = p_sys->p_overflow_buffer;
 594         while( p_slide < p_overflow + i_overflow_size )
 595         {
 596             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 597             {
 598                 memset( p_slide, 0, i_out_size );
 599                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 600                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 601                 else
 602                     memcpy( p_slide, p_slide + i_out_size,
 603                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 604             }
 605             else
 606             {
 607                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 608             }
 609             p_slide += i_out_size;
 610         }
 611
 612         /* apply the atomic operations */
 613         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 614         {
 615             /* shorter variable names */
 616             i_source_channel_offset
 617                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 618             i_dest_channel_offset
 619                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 620             i_delay = p_sys->p_atomic_operations[i].i_delay;
 621             d_amplitude_factor
 622                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 623
 624             if( p_out_buf->i_nb_samples > i_delay )
 625             {
 626                 /* current buffer coefficients */
 627                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 628                 {
 629                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 630                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 631                            * d_amplitude_factor;
 632                 }
 633
 634                 /* overflow buffer coefficients */
 635                 for( j = 0; j < i_delay; j++ )
 636                 {
 637                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 638                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 639                            * i_input_nb + i_source_channel_offset ]
 640                            * d_amplitude_factor;
 641                 }
 642             }
 643             else
 644             {
 645                 /* overflow buffer coefficients only */
 646                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 647                 {
 648                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 649                         * i_output_nb + i_dest_channel_offset ]
 650                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 651                            * d_amplitude_factor;
 652                 }
 653             }
 654         }
 655     }
 656     else
 657     {
 658         memset( p_out, 0, i_out_size );
 659     }
 660 }
 661
 662 /* Simple stereo to mono mixing. */
 663 static unsigned int mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 664                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 665 {
 666     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 667     int16_t *p_in, *p_out;
 668     unsigned int n = 0, r = 0;
 669
 670     p_in = (int16_t *) p_input->p_buffer;
 671     p_out = (int16_t *) p_output->p_buffer;
 672
 673     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 674     {
 675         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 676         r++;
 677         n += 2;
 678     }
 679     return r;
 680 }
 681
 682 /* Simple stereo to mono mixing. */
 683 static unsigned int stereo_to_mono( aout_instance_t * p_aout, aout_filter_t *p_filter,
 684                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 685 {
 686     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 687     int16_t *p_in, *p_out;
 688     unsigned int n;
 689
 690     p_in = (int16_t *) p_input->p_buffer;
 691     p_out = (int16_t *) p_output->p_buffer;
 692
 693     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 694     {
 695         /* Fake real mono. */
 696         if( p_sys->i_channel_selected == -1)
 697         {
 698             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 699             n++;
 700         }
 701         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 702         {
 703             p_out[n] = p_out[n+1] = p_in[n];
 704         }
 705     }
 706     return n;
 707 }