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_block.h>
  41 #include <vlc_filter.h>
  42 #include <vlc_aout.h>
  43
  44 /*****************************************************************************
  45  * Local prototypes
  46  *****************************************************************************/
  47 static int  OpenFilter    ( vlc_object_t * );
  48 static void CloseFilter   ( vlc_object_t * );
  49
  50 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  51
  52 static unsigned int stereo_to_mono( aout_filter_t *, aout_buffer_t *,
  53                                     aout_buffer_t * );
  54 static unsigned int mono( aout_filter_t *, aout_buffer_t *, aout_buffer_t * );
  55 static void stereo2mono_downmix( aout_filter_t *, aout_buffer_t *,
  56                                  aout_buffer_t * );
  57
  58 /*****************************************************************************
  59  * Local structures
  60  *****************************************************************************/
  61 struct atomic_operation_t
  62 {
  63     int i_source_channel_offset;
  64     int i_dest_channel_offset;
  65     unsigned int i_delay;/* in sample unit */
  66     double d_amplitude_factor;
  67 };
  68
  69 struct filter_sys_t
  70 {
  71     bool b_downmix;
  72
  73     unsigned int i_nb_channels; /* number of int16_t per sample */
  74     int i_channel_selected;
  75     int i_bitspersample;
  76
  77     size_t i_overflow_buffer_size;/* in bytes */
  78     uint8_t * p_overflow_buffer;
  79     unsigned int i_nb_atomic_operations;
  80     struct atomic_operation_t * p_atomic_operations;
  81 };
  82
  83 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
  84 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  85     "downmix algorithm that is used in the headphone channel mixer. It " \
  86     "gives the effect of standing in a room full of speakers." )
  87
  88 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  89 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  90     "except the selected channel. Choose one from (0=left, 1=right, " \
  91     "2=rear left, 3=rear right, 4=center, 5=left front)")
  92
  93 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
  94 static const char *const ppsz_pos_descriptions[] =
  95 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
  96   N_("Left front") };
  97
  98 /* our internal channel order (WG-4 order) */
  99 static const uint32_t pi_channels_out[] =
 100 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
 101   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
 102
 103 #define MONO_CFG "sout-mono-"
 104 /*****************************************************************************
 105  * Module descriptor
 106  *****************************************************************************/
 107 vlc_module_begin ()
 108     set_description( N_("Audio filter for stereo to mono conversion") )
 109     set_capability( "audio filter2", 2 )
 110     set_category( CAT_AUDIO )
 111     set_subcategory( SUBCAT_AUDIO_MISC )
 112     set_callbacks( OpenFilter, CloseFilter )
 113     set_shortname( "Mono" )
 114
 115     add_bool( MONO_CFG "downmix", true, NULL, MONO_DOWNMIX_TEXT,
 116               MONO_DOWNMIX_LONGTEXT, false )
 117     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 118         MONO_CHANNEL_LONGTEXT, false )
 119         change_integer_list( pi_pos_values, ppsz_pos_descriptions, NULL )
 120
 121 vlc_module_end ()
 122
 123 /* Init() and ComputeChannelOperations() -
 124  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 125  * converted from float into int16_t based downmix
 126  * Written by Boris Dorès <babal@via.ecp.fr>
 127  */
 128
 129 /*****************************************************************************
 130  * Init: initialize internal data structures
 131  * and computes the needed atomic operations
 132  *****************************************************************************/
 133 /* x and z represent the coordinates of the virtual speaker
 134  *  relatively to the center of the listener's head, measured in meters :
 135  *
 136  *  left              right
 137  *Z
 138  *-
 139  *a          head
 140  *x
 141  *i
 142  *s
 143  *  rear left    rear right
 144  *
 145  *          x-axis
 146  *  */
 147 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 148         unsigned int i_rate, unsigned int i_next_atomic_operation,
 149         int i_source_channel_offset, double d_x, double d_z,
 150         double d_compensation_length, double d_channel_amplitude_factor )
 151 {
 152     double d_c = 340; /*sound celerity (unit: m/s)*/
 153     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 154
 155     /* Left ear */
 156     p_data->p_atomic_operations[i_next_atomic_operation]
 157         .i_source_channel_offset = i_source_channel_offset;
 158     p_data->p_atomic_operations[i_next_atomic_operation]
 159         .i_dest_channel_offset = 0;/* left */
 160     p_data->p_atomic_operations[i_next_atomic_operation]
 161         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 162                           / d_c * i_rate - d_compensation_delay );
 163     if( d_x < 0 )
 164     {
 165         p_data->p_atomic_operations[i_next_atomic_operation]
 166             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 167     }
 168     else if( d_x > 0 )
 169     {
 170         p_data->p_atomic_operations[i_next_atomic_operation]
 171             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 172     }
 173     else
 174     {
 175         p_data->p_atomic_operations[i_next_atomic_operation]
 176             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 177     }
 178
 179     /* Right ear */
 180     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 181         .i_source_channel_offset = i_source_channel_offset;
 182     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 183         .i_dest_channel_offset = 1;/* right */
 184     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 185         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 186                           / d_c * i_rate - d_compensation_delay );
 187     if( d_x < 0 )
 188     {
 189         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 190             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 191     }
 192     else if( d_x > 0 )
 193     {
 194         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 195             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 196     }
 197     else
 198     {
 199         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 200             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 201     }
 202 }
 203
 204 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 205                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 206                  unsigned int i_rate )
 207 {
 208     double d_x = config_GetInt( p_this, "headphone-dim" );
 209     double d_z = d_x;
 210     double d_z_rear = -d_x/3;
 211     double d_min = 0;
 212     unsigned int i_next_atomic_operation;
 213     int i_source_channel_offset;
 214     unsigned int i;
 215
 216     if( config_GetInt( p_this, "headphone-compensate" ) )
 217     {
 218         /* minimal distance to any speaker */
 219         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 220         {
 221             d_min = d_z_rear;
 222         }
 223         else
 224         {
 225             d_min = d_z;
 226         }
 227     }
 228
 229     /* Number of elementary operations */
 230     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 231     if( i_physical_channels & AOUT_CHAN_CENTER )
 232     {
 233         p_data->i_nb_atomic_operations += 2;
 234     }
 235     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 236             * p_data->i_nb_atomic_operations );
 237     if( p_data->p_atomic_operations == NULL )
 238         return -1;
 239
 240     /* For each virtual speaker, computes elementary wave propagation time
 241      * to each ear */
 242     i_next_atomic_operation = 0;
 243     i_source_channel_offset = 0;
 244     if( i_physical_channels & AOUT_CHAN_LEFT )
 245     {
 246         ComputeChannelOperations( p_data , i_rate
 247                 , i_next_atomic_operation , i_source_channel_offset
 248                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 249         i_next_atomic_operation += 2;
 250         i_source_channel_offset++;
 251     }
 252     if( i_physical_channels & AOUT_CHAN_RIGHT )
 253     {
 254         ComputeChannelOperations( p_data , i_rate
 255                 , i_next_atomic_operation , i_source_channel_offset
 256                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 257         i_next_atomic_operation += 2;
 258         i_source_channel_offset++;
 259     }
 260     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 261     {
 262         ComputeChannelOperations( p_data , i_rate
 263                 , i_next_atomic_operation , i_source_channel_offset
 264                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 265         i_next_atomic_operation += 2;
 266         i_source_channel_offset++;
 267     }
 268     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 269     {
 270         ComputeChannelOperations( p_data , i_rate
 271                 , i_next_atomic_operation , i_source_channel_offset
 272                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 273         i_next_atomic_operation += 2;
 274         i_source_channel_offset++;
 275     }
 276     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 277     {
 278         ComputeChannelOperations( p_data , i_rate
 279                 , i_next_atomic_operation , i_source_channel_offset
 280                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 281         i_next_atomic_operation += 2;
 282         i_source_channel_offset++;
 283     }
 284     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 285     {
 286         ComputeChannelOperations( p_data , i_rate
 287                 , i_next_atomic_operation , i_source_channel_offset
 288                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 289         i_next_atomic_operation += 2;
 290         i_source_channel_offset++;
 291     }
 292     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 293     {
 294         ComputeChannelOperations( p_data , i_rate
 295                 , i_next_atomic_operation , i_source_channel_offset
 296                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 297         i_next_atomic_operation += 2;
 298         i_source_channel_offset++;
 299     }
 300     if( i_physical_channels & AOUT_CHAN_CENTER )
 301     {
 302         /* having two center channels increases the spatialization effect */
 303         ComputeChannelOperations( p_data , i_rate
 304                 , i_next_atomic_operation , i_source_channel_offset
 305                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 306         i_next_atomic_operation += 2;
 307         ComputeChannelOperations( p_data , i_rate
 308                 , i_next_atomic_operation , i_source_channel_offset
 309                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 310         i_next_atomic_operation += 2;
 311         i_source_channel_offset++;
 312     }
 313     if( i_physical_channels & AOUT_CHAN_LFE )
 314     {
 315         ComputeChannelOperations( p_data , i_rate
 316                 , i_next_atomic_operation , i_source_channel_offset
 317                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 318         i_next_atomic_operation += 2;
 319         i_source_channel_offset++;
 320     }
 321
 322     /* Initialize the overflow buffer
 323      * we need it because the process induce a delay in the samples */
 324     p_data->i_overflow_buffer_size = 0;
 325     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 326     {
 327         if( p_data->i_overflow_buffer_size
 328                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 329         {
 330             p_data->i_overflow_buffer_size
 331                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 332         }
 333     }
 334     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 335     if( p_data->p_overflow_buffer == NULL )
 336     {
 337         free( p_data->p_atomic_operations );
 338         return -1;
 339     }
 340     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 341
 342     /* end */
 343     return 0;
 344 }
 345
 346 /*****************************************************************************
 347  * OpenFilter
 348  *****************************************************************************/
 349 static int OpenFilter( vlc_object_t *p_this )
 350 {
 351     filter_t * p_filter = (filter_t *)p_this;
 352     filter_sys_t *p_sys = NULL;
 353
 354     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 355     {
 356         /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
 357         return VLC_EGENERIC;
 358     }
 359
 360     if( (p_filter->fmt_in.i_codec != VLC_CODEC_S16N) ||
 361         (p_filter->fmt_out.i_codec != VLC_CODEC_S16N) )
 362     {
 363         /*msg_Err( p_this, "filter discarded (invalid format)" );*/
 364         return VLC_EGENERIC;
 365     }
 366
 367     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 368         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 369         (p_filter->fmt_in.audio.i_format != VLC_CODEC_S16N) &&
 370         (p_filter->fmt_out.audio.i_format != VLC_CODEC_S16N) &&
 371         (p_filter->fmt_in.audio.i_bitspersample !=
 372                                     p_filter->fmt_out.audio.i_bitspersample))
 373     {
 374         /*msg_Err( p_this, "couldn't load mono filter" );*/
 375         return VLC_EGENERIC;
 376     }
 377
 378     /* Allocate the memory needed to store the module's structure */
 379     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 380     if( p_sys == NULL )
 381         return VLC_EGENERIC;
 382
 383     p_sys->b_downmix = var_CreateGetBool( p_this, MONO_CFG "downmix" );
 384     p_sys->i_channel_selected =
 385             (unsigned int) var_CreateGetInteger( p_this, MONO_CFG "channel" );
 386
 387     if( p_sys->b_downmix )
 388     {
 389         msg_Dbg( p_this, "using stereo to mono downmix" );
 390         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 391         p_filter->fmt_out.audio.i_channels = 1;
 392     }
 393     else
 394     {
 395         msg_Dbg( p_this, "using pseudo mono" );
 396         p_filter->fmt_out.audio.i_physical_channels =
 397                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 398         p_filter->fmt_out.audio.i_channels = 2;
 399     }
 400
 401     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 402     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 403
 404     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 405     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 406
 407     p_sys->i_overflow_buffer_size = 0;
 408     p_sys->p_overflow_buffer = NULL;
 409     p_sys->i_nb_atomic_operations = 0;
 410     p_sys->p_atomic_operations = NULL;
 411
 412     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 413               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 414               p_filter->fmt_in.audio.i_physical_channels,
 415               p_filter->fmt_in.audio.i_rate ) < 0 )
 416     {
 417         var_Destroy( p_this, MONO_CFG "channel" );
 418         var_Destroy( p_this, MONO_CFG "downmix" );
 419         free( p_sys );
 420         return VLC_EGENERIC;
 421     }
 422
 423     p_filter->pf_audio_filter = Convert;
 424
 425     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 426              (char *)&p_filter->fmt_in.i_codec,
 427              (char *)&p_filter->fmt_out.i_codec,
 428              p_filter->fmt_in.audio.i_physical_channels,
 429              p_filter->fmt_out.audio.i_physical_channels,
 430              p_filter->fmt_in.audio.i_bitspersample,
 431              p_filter->fmt_out.audio.i_bitspersample );
 432
 433     return VLC_SUCCESS;
 434 }
 435
 436 /*****************************************************************************
 437  * CloseFilter
 438  *****************************************************************************/
 439 static void CloseFilter( vlc_object_t *p_this)
 440 {
 441     filter_t *p_filter = (filter_t *) p_this;
 442     filter_sys_t *p_sys = p_filter->p_sys;
 443
 444     var_Destroy( p_this, MONO_CFG "channel" );
 445     var_Destroy( p_this, MONO_CFG "downmix" );
 446     free( p_sys->p_atomic_operations );
 447     free( p_sys->p_overflow_buffer );
 448     free( p_sys );
 449 }
 450
 451 /*****************************************************************************
 452  * Convert
 453  *****************************************************************************/
 454 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 455 {
 456     aout_filter_t aout_filter;
 457     aout_buffer_t in_buf, out_buf;
 458     block_t *p_out = NULL;
 459     unsigned int i_samples;
 460     int i_out_size;
 461
 462     if( !p_block || !p_block->i_nb_samples )
 463     {
 464         if( p_block )
 465             block_Release( p_block );
 466         return NULL;
 467     }
 468
 469     i_out_size = p_block->i_nb_samples * p_filter->p_sys->i_bitspersample/8 *
 470                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 471
 472     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 473     if( !p_out )
 474     {
 475         msg_Warn( p_filter, "can't get output buffer" );
 476         block_Release( p_block );
 477         return NULL;
 478     }
 479     p_out->i_nb_samples =
 480                   (p_block->i_nb_samples / p_filter->p_sys->i_nb_channels) *
 481                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 482     p_out->i_dts = p_block->i_dts;
 483     p_out->i_pts = p_block->i_pts;
 484     p_out->i_length = p_block->i_length;
 485
 486     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 487     aout_filter.fmt_in.audio = p_filter->fmt_in.audio;
 488     aout_filter.fmt_in.audio.i_format = p_filter->fmt_in.i_codec;
 489     aout_filter.fmt_out.audio = p_filter->fmt_out.audio;
 490     aout_filter.fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 491
 492     in_buf.p_buffer = p_block->p_buffer;
 493     in_buf.i_buffer = p_block->i_buffer;
 494     in_buf.i_nb_samples = p_block->i_nb_samples;
 495
 496 #if 0
 497     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 498                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 499     if( (in_buf.i_buffer != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 500     {
 501         msg_Err( p_filter, "input buffer is not word aligned" );
 502         /* Fix output buffer to be word aligned */
 503     }
 504 #endif
 505
 506     out_buf.p_buffer = p_out->p_buffer;
 507     out_buf.i_buffer = p_out->i_buffer;
 508     out_buf.i_nb_samples = p_out->i_nb_samples;
 509
 510     memset( p_out->p_buffer, 0, i_out_size );
 511     if( p_filter->p_sys->b_downmix )
 512     {
 513         stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
 514         i_samples = mono( &aout_filter, &out_buf, &in_buf );
 515     }
 516     else
 517     {
 518         i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
 519     }
 520
 521     p_out->i_buffer = out_buf.i_buffer;
 522     p_out->i_nb_samples = out_buf.i_nb_samples;
 523
 524     block_Release( p_block );
 525     return p_out;
 526 }
 527
 528 /* stereo2mono_downmix - stereo channels into one mono channel.
 529  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 530  * converted from float into int16_t based downmix
 531  * Written by Boris Dorès <babal@via.ecp.fr>
 532  */
 533 static void stereo2mono_downmix( aout_filter_t * p_filter,
 534                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 535 {
 536     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 537
 538     int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 539     int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio );
 540
 541     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 542     uint8_t * p_out;
 543     uint8_t * p_overflow;
 544     uint8_t * p_slide;
 545
 546     size_t i_overflow_size;     /* in bytes */
 547     size_t i_out_size;          /* in bytes */
 548
 549     unsigned int i, j;
 550
 551     int i_source_channel_offset;
 552     int i_dest_channel_offset;
 553     unsigned int i_delay;
 554     double d_amplitude_factor;
 555
 556     /* out buffer characterisitcs */
 557     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 558     p_out_buf->i_buffer = p_in_buf->i_buffer * i_output_nb / i_input_nb;
 559     p_out = p_out_buf->p_buffer;
 560     i_out_size = p_out_buf->i_buffer;
 561
 562     /* Slide the overflow buffer */
 563     p_overflow = p_sys->p_overflow_buffer;
 564     i_overflow_size = p_sys->i_overflow_buffer_size;
 565
 566     if ( i_out_size > i_overflow_size )
 567         memcpy( p_out, p_overflow, i_overflow_size );
 568     else
 569         memcpy( p_out, p_overflow, i_out_size );
 570
 571     p_slide = p_sys->p_overflow_buffer;
 572     while( p_slide < p_overflow + i_overflow_size )
 573     {
 574         if( p_slide + i_out_size < p_overflow + i_overflow_size )
 575         {
 576             memset( p_slide, 0, i_out_size );
 577             if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 578                 memcpy( p_slide, p_slide + i_out_size, i_out_size );
 579             else
 580                 memcpy( p_slide, p_slide + i_out_size,
 581                         p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 582         }
 583         else
 584         {
 585             memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 586         }
 587         p_slide += i_out_size;
 588     }
 589
 590     /* apply the atomic operations */
 591     for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 592     {
 593         /* shorter variable names */
 594         i_source_channel_offset
 595             = p_sys->p_atomic_operations[i].i_source_channel_offset;
 596         i_dest_channel_offset
 597             = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 598         i_delay = p_sys->p_atomic_operations[i].i_delay;
 599         d_amplitude_factor
 600             = p_sys->p_atomic_operations[i].d_amplitude_factor;
 601
 602         if( p_out_buf->i_nb_samples > i_delay )
 603         {
 604             /* current buffer coefficients */
 605             for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 606             {
 607                 ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 608                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 609                        * d_amplitude_factor;
 610             }
 611
 612             /* overflow buffer coefficients */
 613             for( j = 0; j < i_delay; j++ )
 614             {
 615                 ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 616                     += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 617                        * i_input_nb + i_source_channel_offset ]
 618                        * d_amplitude_factor;
 619             }
 620         }
 621         else
 622         {
 623             /* overflow buffer coefficients only */
 624             for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 625             {
 626                 ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 627                                         * 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     }
 633 }
 634
 635 /* Simple stereo to mono mixing. */
 636 static unsigned int mono( aout_filter_t *p_filter,
 637                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 638 {
 639     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 640     int16_t *p_in, *p_out;
 641     unsigned int n = 0, r = 0;
 642
 643     p_in = (int16_t *) p_input->p_buffer;
 644     p_out = (int16_t *) p_output->p_buffer;
 645
 646     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 647     {
 648         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 649         r++;
 650         n += 2;
 651     }
 652     return r;
 653 }
 654
 655 /* Simple stereo to mono mixing. */
 656 static unsigned int stereo_to_mono( aout_filter_t *p_filter,
 657                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 658 {
 659     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 660     int16_t *p_in, *p_out;
 661     unsigned int n;
 662
 663     p_in = (int16_t *) p_input->p_buffer;
 664     p_out = (int16_t *) p_output->p_buffer;
 665
 666     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 667     {
 668         /* Fake real mono. */
 669         if( p_sys->i_channel_selected == -1)
 670         {
 671             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 672             n++;
 673         }
 674         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 675         {
 676             p_out[n] = p_out[n+1] = p_in[n];
 677         }
 678     }
 679     return n;
 680 }