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 #ifdef HAVE_CONFIG_H
  28 # include "config.h"
  29 #endif
  30
  31 #include <math.h>                                        /* sqrt */
  32 #include <stdint.h>                                         /* int16_t .. */
  33
  34 #ifdef HAVE_UNISTD_H
  35 #   include <unistd.h>
  36 #endif
  37
  38 #include <vlc_common.h>
  39 #include <vlc_plugin.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_filter_t *, aout_buffer_t *,
  54                                     aout_buffer_t * );
  55 static unsigned int mono( aout_filter_t *, aout_buffer_t *, aout_buffer_t * );
  56 static void stereo2mono_downmix( aout_filter_t *, aout_buffer_t *,
  57                                  aout_buffer_t * );
  58
  59 /*****************************************************************************
  60  * Local structures
  61  *****************************************************************************/
  62 struct atomic_operation_t
  63 {
  64     int i_source_channel_offset;
  65     int i_dest_channel_offset;
  66     unsigned int i_delay;/* in sample unit */
  67     double d_amplitude_factor;
  68 };
  69
  70 struct filter_sys_t
  71 {
  72     bool b_downmix;
  73
  74     unsigned int i_nb_channels; /* number of int16_t per sample */
  75     int i_channel_selected;
  76     int i_bitspersample;
  77
  78     size_t i_overflow_buffer_size;/* in bytes */
  79     uint8_t * p_overflow_buffer;
  80     unsigned int i_nb_atomic_operations;
  81     struct atomic_operation_t * p_atomic_operations;
  82 };
  83
  84 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
  85 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  86     "downmix algorithm that is used in the headphone channel mixer. It " \
  87     "gives the effect of standing in a room full of speakers." )
  88
  89 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  90 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  91     "except the selected channel. Choose one from (0=left, 1=right, " \
  92     "2=rear left, 3=rear right, 4=center, 5=left front)")
  93
  94 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
  95 static const char *const ppsz_pos_descriptions[] =
  96 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
  97   N_("Left front") };
  98
  99 /* our internal channel order (WG-4 order) */
 100 static const uint32_t pi_channels_out[] =
 101 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
 102   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
 103
 104 #define MONO_CFG "sout-mono-"
 105 /*****************************************************************************
 106  * Module descriptor
 107  *****************************************************************************/
 108 vlc_module_begin ()
 109     set_description( N_("Audio filter for stereo to mono conversion") )
 110     set_capability( "audio filter2", 2 )
 111
 112     add_bool( MONO_CFG "downmix", true, NULL, MONO_DOWNMIX_TEXT,
 113               MONO_DOWNMIX_LONGTEXT, false )
 114     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 115         MONO_CHANNEL_LONGTEXT, false )
 116         change_integer_list( pi_pos_values, ppsz_pos_descriptions, NULL )
 117
 118     set_category( CAT_AUDIO )
 119     set_subcategory( SUBCAT_AUDIO_MISC )
 120     set_callbacks( OpenFilter, CloseFilter )
 121     set_shortname( "Mono" )
 122 vlc_module_end ()
 123
 124 /* Init() and ComputeChannelOperations() -
 125  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 126  * converted from float into int16_t based downmix
 127  * Written by Boris Dorès <babal@via.ecp.fr>
 128  */
 129
 130 /*****************************************************************************
 131  * Init: initialize internal data structures
 132  * and computes the needed atomic operations
 133  *****************************************************************************/
 134 /* x and z represent the coordinates of the virtual speaker
 135  *  relatively to the center of the listener's head, measured in meters :
 136  *
 137  *  left              right
 138  *Z
 139  *-
 140  *a          head
 141  *x
 142  *i
 143  *s
 144  *  rear left    rear right
 145  *
 146  *          x-axis
 147  *  */
 148 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 149         unsigned int i_rate, unsigned int i_next_atomic_operation,
 150         int i_source_channel_offset, double d_x, double d_z,
 151         double d_compensation_length, double d_channel_amplitude_factor )
 152 {
 153     double d_c = 340; /*sound celerity (unit: m/s)*/
 154     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 155
 156     /* Left ear */
 157     p_data->p_atomic_operations[i_next_atomic_operation]
 158         .i_source_channel_offset = i_source_channel_offset;
 159     p_data->p_atomic_operations[i_next_atomic_operation]
 160         .i_dest_channel_offset = 0;/* left */
 161     p_data->p_atomic_operations[i_next_atomic_operation]
 162         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 163                           / d_c * i_rate - d_compensation_delay );
 164     if( d_x < 0 )
 165     {
 166         p_data->p_atomic_operations[i_next_atomic_operation]
 167             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 168     }
 169     else if( d_x > 0 )
 170     {
 171         p_data->p_atomic_operations[i_next_atomic_operation]
 172             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 173     }
 174     else
 175     {
 176         p_data->p_atomic_operations[i_next_atomic_operation]
 177             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 178     }
 179
 180     /* Right ear */
 181     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 182         .i_source_channel_offset = i_source_channel_offset;
 183     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 184         .i_dest_channel_offset = 1;/* right */
 185     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 186         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 187                           / d_c * i_rate - d_compensation_delay );
 188     if( d_x < 0 )
 189     {
 190         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 191             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 192     }
 193     else if( d_x > 0 )
 194     {
 195         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 196             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 197     }
 198     else
 199     {
 200         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 201             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 202     }
 203 }
 204
 205 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 206                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 207                  unsigned int i_rate )
 208 {
 209     double d_x = config_GetInt( p_this, "headphone-dim" );
 210     double d_z = d_x;
 211     double d_z_rear = -d_x/3;
 212     double d_min = 0;
 213     unsigned int i_next_atomic_operation;
 214     int i_source_channel_offset;
 215     unsigned int i;
 216
 217     if( config_GetInt( p_this, "headphone-compensate" ) )
 218     {
 219         /* minimal distance to any speaker */
 220         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 221         {
 222             d_min = d_z_rear;
 223         }
 224         else
 225         {
 226             d_min = d_z;
 227         }
 228     }
 229
 230     /* Number of elementary operations */
 231     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 232     if( i_physical_channels & AOUT_CHAN_CENTER )
 233     {
 234         p_data->i_nb_atomic_operations += 2;
 235     }
 236     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 237             * p_data->i_nb_atomic_operations );
 238     if( p_data->p_atomic_operations == NULL )
 239         return -1;
 240
 241     /* For each virtual speaker, computes elementary wave propagation time
 242      * to each ear */
 243     i_next_atomic_operation = 0;
 244     i_source_channel_offset = 0;
 245     if( i_physical_channels & AOUT_CHAN_LEFT )
 246     {
 247         ComputeChannelOperations( p_data , i_rate
 248                 , i_next_atomic_operation , i_source_channel_offset
 249                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 250         i_next_atomic_operation += 2;
 251         i_source_channel_offset++;
 252     }
 253     if( i_physical_channels & AOUT_CHAN_RIGHT )
 254     {
 255         ComputeChannelOperations( p_data , i_rate
 256                 , i_next_atomic_operation , i_source_channel_offset
 257                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 258         i_next_atomic_operation += 2;
 259         i_source_channel_offset++;
 260     }
 261     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 262     {
 263         ComputeChannelOperations( p_data , i_rate
 264                 , i_next_atomic_operation , i_source_channel_offset
 265                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 266         i_next_atomic_operation += 2;
 267         i_source_channel_offset++;
 268     }
 269     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 270     {
 271         ComputeChannelOperations( p_data , i_rate
 272                 , i_next_atomic_operation , i_source_channel_offset
 273                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 274         i_next_atomic_operation += 2;
 275         i_source_channel_offset++;
 276     }
 277     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 278     {
 279         ComputeChannelOperations( p_data , i_rate
 280                 , i_next_atomic_operation , i_source_channel_offset
 281                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 282         i_next_atomic_operation += 2;
 283         i_source_channel_offset++;
 284     }
 285     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 286     {
 287         ComputeChannelOperations( p_data , i_rate
 288                 , i_next_atomic_operation , i_source_channel_offset
 289                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 290         i_next_atomic_operation += 2;
 291         i_source_channel_offset++;
 292     }
 293     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 294     {
 295         ComputeChannelOperations( p_data , i_rate
 296                 , i_next_atomic_operation , i_source_channel_offset
 297                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 298         i_next_atomic_operation += 2;
 299         i_source_channel_offset++;
 300     }
 301     if( i_physical_channels & AOUT_CHAN_CENTER )
 302     {
 303         /* having two center channels increases the spatialization effect */
 304         ComputeChannelOperations( p_data , i_rate
 305                 , i_next_atomic_operation , i_source_channel_offset
 306                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 307         i_next_atomic_operation += 2;
 308         ComputeChannelOperations( p_data , i_rate
 309                 , i_next_atomic_operation , i_source_channel_offset
 310                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 311         i_next_atomic_operation += 2;
 312         i_source_channel_offset++;
 313     }
 314     if( i_physical_channels & AOUT_CHAN_LFE )
 315     {
 316         ComputeChannelOperations( p_data , i_rate
 317                 , i_next_atomic_operation , i_source_channel_offset
 318                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 319         i_next_atomic_operation += 2;
 320         i_source_channel_offset++;
 321     }
 322
 323     /* Initialize the overflow buffer
 324      * we need it because the process induce a delay in the samples */
 325     p_data->i_overflow_buffer_size = 0;
 326     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 327     {
 328         if( p_data->i_overflow_buffer_size
 329                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 330         {
 331             p_data->i_overflow_buffer_size
 332                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 333         }
 334     }
 335     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 336     if( p_data->p_overflow_buffer == NULL )
 337     {
 338         free( p_data->p_atomic_operations );
 339         return -1;
 340     }
 341     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 342
 343     /* end */
 344     return 0;
 345 }
 346
 347 /*****************************************************************************
 348  * OpenFilter
 349  *****************************************************************************/
 350 static int OpenFilter( vlc_object_t *p_this )
 351 {
 352     filter_t * p_filter = (filter_t *)p_this;
 353     filter_sys_t *p_sys = NULL;
 354
 355     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 356     {
 357         /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
 358         return VLC_EGENERIC;
 359     }
 360
 361     if( (p_filter->fmt_in.i_codec != VLC_CODEC_S16N) ||
 362         (p_filter->fmt_out.i_codec != VLC_CODEC_S16N) )
 363     {
 364         /*msg_Err( p_this, "filter discarded (invalid format)" );*/
 365         return VLC_EGENERIC;
 366     }
 367
 368     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 369         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 370         (p_filter->fmt_in.audio.i_format != VLC_CODEC_S16N) &&
 371         (p_filter->fmt_out.audio.i_format != VLC_CODEC_S16N) &&
 372         (p_filter->fmt_in.audio.i_bitspersample !=
 373                                     p_filter->fmt_out.audio.i_bitspersample))
 374     {
 375         /*msg_Err( p_this, "couldn't load mono filter" );*/
 376         return VLC_EGENERIC;
 377     }
 378
 379     /* Allocate the memory needed to store the module's structure */
 380     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 381     if( p_sys == NULL )
 382         return VLC_EGENERIC;
 383
 384     var_Create( p_this, MONO_CFG "downmix",
 385                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 386     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 387
 388     var_Create( p_this, MONO_CFG "channel",
 389                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 390     p_sys->i_channel_selected =
 391             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 392
 393     if( p_sys->b_downmix )
 394     {
 395         msg_Dbg( p_this, "using stereo to mono downmix" );
 396         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 397         p_filter->fmt_out.audio.i_channels = 1;
 398     }
 399     else
 400     {
 401         msg_Dbg( p_this, "using pseudo mono" );
 402         p_filter->fmt_out.audio.i_physical_channels =
 403                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 404         p_filter->fmt_out.audio.i_channels = 2;
 405     }
 406
 407     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 408     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 409
 410     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 411     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 412
 413     p_sys->i_overflow_buffer_size = 0;
 414     p_sys->p_overflow_buffer = NULL;
 415     p_sys->i_nb_atomic_operations = 0;
 416     p_sys->p_atomic_operations = NULL;
 417
 418     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 419               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 420               p_filter->fmt_in.audio.i_physical_channels,
 421               p_filter->fmt_in.audio.i_rate ) < 0 )
 422     {
 423         var_Destroy( p_this, MONO_CFG "channel" );
 424         var_Destroy( p_this, MONO_CFG "downmix" );
 425         free( p_sys );
 426         return VLC_EGENERIC;
 427     }
 428
 429     p_filter->pf_audio_filter = Convert;
 430
 431     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 432              (char *)&p_filter->fmt_in.i_codec,
 433              (char *)&p_filter->fmt_out.i_codec,
 434              p_filter->fmt_in.audio.i_physical_channels,
 435              p_filter->fmt_out.audio.i_physical_channels,
 436              p_filter->fmt_in.audio.i_bitspersample,
 437              p_filter->fmt_out.audio.i_bitspersample );
 438
 439     return VLC_SUCCESS;
 440 }
 441
 442 /*****************************************************************************
 443  * CloseFilter
 444  *****************************************************************************/
 445 static void CloseFilter( vlc_object_t *p_this)
 446 {
 447     filter_t *p_filter = (filter_t *) p_this;
 448     filter_sys_t *p_sys = p_filter->p_sys;
 449
 450     var_Destroy( p_this, MONO_CFG "channel" );
 451     var_Destroy( p_this, MONO_CFG "downmix" );
 452     free( p_sys->p_atomic_operations );
 453     free( p_sys->p_overflow_buffer );
 454     free( p_sys );
 455 }
 456
 457 /*****************************************************************************
 458  * Convert
 459  *****************************************************************************/
 460 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 461 {
 462     aout_filter_t aout_filter;
 463     aout_buffer_t in_buf, out_buf;
 464     block_t *p_out = NULL;
 465     unsigned int i_samples;
 466     int i_out_size;
 467
 468     if( !p_block || !p_block->i_samples )
 469     {
 470         if( p_block )
 471             block_Release( p_block );
 472         return NULL;
 473     }
 474
 475     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 476                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 477
 478     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 479     if( !p_out )
 480     {
 481         msg_Warn( p_filter, "can't get output buffer" );
 482         block_Release( p_block );
 483         return NULL;
 484     }
 485     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 486                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 487     p_out->i_dts = p_block->i_dts;
 488     p_out->i_pts = p_block->i_pts;
 489     p_out->i_length = p_block->i_length;
 490
 491     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 492     aout_filter.input = p_filter->fmt_in.audio;
 493     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 494     aout_filter.output = p_filter->fmt_out.audio;
 495     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 496
 497     in_buf.p_buffer = p_block->p_buffer;
 498     in_buf.i_nb_bytes = p_block->i_buffer;
 499     in_buf.i_nb_samples = p_block->i_samples;
 500
 501 #if 0
 502     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 503                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 504     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 505     {
 506         msg_Err( p_filter, "input buffer is not word aligned" );
 507         /* Fix output buffer to be word aligned */
 508     }
 509 #endif
 510
 511     out_buf.p_buffer = p_out->p_buffer;
 512     out_buf.i_nb_bytes = p_out->i_buffer;
 513     out_buf.i_nb_samples = p_out->i_samples;
 514
 515     memset( p_out->p_buffer, 0, i_out_size );
 516     if( p_filter->p_sys->b_downmix )
 517     {
 518         stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
 519         i_samples = mono( &aout_filter, &out_buf, &in_buf );
 520     }
 521     else
 522     {
 523         i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
 524     }
 525
 526     p_out->i_buffer = out_buf.i_nb_bytes;
 527     p_out->i_samples = out_buf.i_nb_samples;
 528
 529     block_Release( p_block );
 530     return p_out;
 531 }
 532
 533 /* stereo2mono_downmix - stereo channels into one mono channel.
 534  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 535  * converted from float into int16_t based downmix
 536  * Written by Boris Dorès <babal@via.ecp.fr>
 537  */
 538 static void stereo2mono_downmix( aout_filter_t * p_filter,
 539                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 540 {
 541     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 542
 543     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 544     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 545
 546     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 547     uint8_t * p_out;
 548     uint8_t * p_overflow;
 549     uint8_t * p_slide;
 550
 551     size_t i_overflow_size;     /* in bytes */
 552     size_t i_out_size;          /* in bytes */
 553
 554     unsigned int i, j;
 555
 556     int i_source_channel_offset;
 557     int i_dest_channel_offset;
 558     unsigned int i_delay;
 559     double d_amplitude_factor;
 560
 561     /* out buffer characterisitcs */
 562     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 563     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 564     p_out = p_out_buf->p_buffer;
 565     i_out_size = p_out_buf->i_nb_bytes;
 566
 567     if( p_sys != NULL )
 568     {
 569         /* Slide the overflow buffer */
 570         p_overflow = p_sys->p_overflow_buffer;
 571         i_overflow_size = p_sys->i_overflow_buffer_size;
 572
 573         if ( i_out_size > i_overflow_size )
 574             memcpy( p_out, p_overflow, i_overflow_size );
 575         else
 576             memcpy( p_out, p_overflow, i_out_size );
 577
 578         p_slide = p_sys->p_overflow_buffer;
 579         while( p_slide < p_overflow + i_overflow_size )
 580         {
 581             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 582             {
 583                 memset( p_slide, 0, i_out_size );
 584                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 585                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 586                 else
 587                     memcpy( p_slide, p_slide + i_out_size,
 588                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 589             }
 590             else
 591             {
 592                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 593             }
 594             p_slide += i_out_size;
 595         }
 596
 597         /* apply the atomic operations */
 598         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 599         {
 600             /* shorter variable names */
 601             i_source_channel_offset
 602                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 603             i_dest_channel_offset
 604                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 605             i_delay = p_sys->p_atomic_operations[i].i_delay;
 606             d_amplitude_factor
 607                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 608
 609             if( p_out_buf->i_nb_samples > i_delay )
 610             {
 611                 /* current buffer coefficients */
 612                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 613                 {
 614                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 615                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 616                            * d_amplitude_factor;
 617                 }
 618
 619                 /* overflow buffer coefficients */
 620                 for( j = 0; j < i_delay; j++ )
 621                 {
 622                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 623                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 624                            * i_input_nb + i_source_channel_offset ]
 625                            * d_amplitude_factor;
 626                 }
 627             }
 628             else
 629             {
 630                 /* overflow buffer coefficients only */
 631                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 632                 {
 633                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 634                         * i_output_nb + i_dest_channel_offset ]
 635                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 636                            * d_amplitude_factor;
 637                 }
 638             }
 639         }
 640     }
 641     else
 642     {
 643         memset( p_out, 0, i_out_size );
 644     }
 645 }
 646
 647 /* Simple stereo to mono mixing. */
 648 static unsigned int mono( aout_filter_t *p_filter,
 649                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 650 {
 651     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 652     int16_t *p_in, *p_out;
 653     unsigned int n = 0, r = 0;
 654
 655     p_in = (int16_t *) p_input->p_buffer;
 656     p_out = (int16_t *) p_output->p_buffer;
 657
 658     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 659     {
 660         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 661         r++;
 662         n += 2;
 663     }
 664     return r;
 665 }
 666
 667 /* Simple stereo to mono mixing. */
 668 static unsigned int stereo_to_mono( aout_filter_t *p_filter,
 669                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 670 {
 671     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 672     int16_t *p_in, *p_out;
 673     unsigned int n;
 674
 675     p_in = (int16_t *) p_input->p_buffer;
 676     p_out = (int16_t *) p_output->p_buffer;
 677
 678     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 679     {
 680         /* Fake real mono. */
 681         if( p_sys->i_channel_selected == -1)
 682         {
 683             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 684             n++;
 685         }
 686         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 687         {
 688             p_out[n] = p_out[n+1] = p_in[n];
 689         }
 690     }
 691     return n;
 692 }