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
  33 #ifdef HAVE_STDINT_H
  34 #   include <stdint.h>                                         /* int16_t .. */
  35 #elif defined(HAVE_INTTYPES_H)
  36 #   include <inttypes.h>                                       /* int16_t .. */
  37 #endif
  38
  39 #ifdef HAVE_UNISTD_H
  40 #   include <unistd.h>
  41 #endif
  42
  43 #include <vlc_common.h>
  44 #include <vlc_plugin.h>
  45 #include <vlc_es.h>
  46 #include <vlc_block.h>
  47 #include <vlc_filter.h>
  48 #include <vlc_aout.h>
  49
  50 /*****************************************************************************
  51  * Local prototypes
  52  *****************************************************************************/
  53 static int  OpenFilter    ( vlc_object_t * );
  54 static void CloseFilter   ( vlc_object_t * );
  55
  56 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  57
  58 static unsigned int stereo_to_mono( aout_filter_t *, aout_buffer_t *,
  59                                     aout_buffer_t * );
  60 static unsigned int mono( aout_filter_t *, aout_buffer_t *, aout_buffer_t * );
  61 static void stereo2mono_downmix( aout_filter_t *, aout_buffer_t *,
  62                                  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     bool 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     uint8_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 *const 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( N_("Audio filter for stereo to mono conversion") )
 115     set_capability( "audio filter2", 2 )
 116
 117     add_bool( MONO_CFG "downmix", true, NULL, MONO_DOWNMIX_TEXT,
 118               MONO_DOWNMIX_LONGTEXT, false );
 119     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
 120         MONO_CHANNEL_LONGTEXT, false );
 121         change_integer_list( pi_pos_values, ppsz_pos_descriptions, NULL );
 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( config_GetInt( p_this, "headphone-compensate" ) )
 223     {
 224         /* minimal distance to any speaker */
 225         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 226         {
 227             d_min = d_z_rear;
 228         }
 229         else
 230         {
 231             d_min = d_z;
 232         }
 233     }
 234
 235     /* Number of elementary operations */
 236     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 237     if( i_physical_channels & AOUT_CHAN_CENTER )
 238     {
 239         p_data->i_nb_atomic_operations += 2;
 240     }
 241     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 242             * p_data->i_nb_atomic_operations );
 243     if( p_data->p_atomic_operations == NULL )
 244         return -1;
 245
 246     /* For each virtual speaker, computes elementary wave propagation time
 247      * to each ear */
 248     i_next_atomic_operation = 0;
 249     i_source_channel_offset = 0;
 250     if( i_physical_channels & AOUT_CHAN_LEFT )
 251     {
 252         ComputeChannelOperations( p_data , i_rate
 253                 , i_next_atomic_operation , i_source_channel_offset
 254                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 255         i_next_atomic_operation += 2;
 256         i_source_channel_offset++;
 257     }
 258     if( i_physical_channels & AOUT_CHAN_RIGHT )
 259     {
 260         ComputeChannelOperations( p_data , i_rate
 261                 , i_next_atomic_operation , i_source_channel_offset
 262                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 263         i_next_atomic_operation += 2;
 264         i_source_channel_offset++;
 265     }
 266     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 267     {
 268         ComputeChannelOperations( p_data , i_rate
 269                 , i_next_atomic_operation , i_source_channel_offset
 270                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 271         i_next_atomic_operation += 2;
 272         i_source_channel_offset++;
 273     }
 274     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 275     {
 276         ComputeChannelOperations( p_data , i_rate
 277                 , i_next_atomic_operation , i_source_channel_offset
 278                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 279         i_next_atomic_operation += 2;
 280         i_source_channel_offset++;
 281     }
 282     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 283     {
 284         ComputeChannelOperations( p_data , i_rate
 285                 , i_next_atomic_operation , i_source_channel_offset
 286                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 287         i_next_atomic_operation += 2;
 288         i_source_channel_offset++;
 289     }
 290     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 291     {
 292         ComputeChannelOperations( p_data , i_rate
 293                 , i_next_atomic_operation , i_source_channel_offset
 294                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 295         i_next_atomic_operation += 2;
 296         i_source_channel_offset++;
 297     }
 298     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 299     {
 300         ComputeChannelOperations( p_data , i_rate
 301                 , i_next_atomic_operation , i_source_channel_offset
 302                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 303         i_next_atomic_operation += 2;
 304         i_source_channel_offset++;
 305     }
 306     if( i_physical_channels & AOUT_CHAN_CENTER )
 307     {
 308         /* having two center channels increases the spatialization effect */
 309         ComputeChannelOperations( p_data , i_rate
 310                 , i_next_atomic_operation , i_source_channel_offset
 311                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 312         i_next_atomic_operation += 2;
 313         ComputeChannelOperations( p_data , i_rate
 314                 , i_next_atomic_operation , i_source_channel_offset
 315                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 316         i_next_atomic_operation += 2;
 317         i_source_channel_offset++;
 318     }
 319     if( i_physical_channels & AOUT_CHAN_LFE )
 320     {
 321         ComputeChannelOperations( p_data , i_rate
 322                 , i_next_atomic_operation , i_source_channel_offset
 323                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 324         i_next_atomic_operation += 2;
 325         i_source_channel_offset++;
 326     }
 327
 328     /* Initialize the overflow buffer
 329      * we need it because the process induce a delay in the samples */
 330     p_data->i_overflow_buffer_size = 0;
 331     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 332     {
 333         if( p_data->i_overflow_buffer_size
 334                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 335         {
 336             p_data->i_overflow_buffer_size
 337                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 338         }
 339     }
 340     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 341     if( p_data->p_overflow_buffer == NULL )
 342     {
 343         free( p_data->p_atomic_operations );
 344         return -1;
 345     }
 346     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 347
 348     /* end */
 349     return 0;
 350 }
 351
 352 /*****************************************************************************
 353  * OpenFilter
 354  *****************************************************************************/
 355 static int OpenFilter( vlc_object_t *p_this )
 356 {
 357     filter_t * p_filter = (filter_t *)p_this;
 358     filter_sys_t *p_sys = NULL;
 359
 360     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 361     {
 362         /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
 363         return VLC_EGENERIC;
 364     }
 365
 366     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
 367         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
 368     {
 369         /*msg_Err( p_this, "filter discarded (invalid format)" );*/
 370         return VLC_EGENERIC;
 371     }
 372
 373     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
 374         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
 375         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
 376         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
 377         (p_filter->fmt_in.audio.i_bitspersample !=
 378                                     p_filter->fmt_out.audio.i_bitspersample))
 379     {
 380         /*msg_Err( p_this, "couldn't load mono filter" );*/
 381         return VLC_EGENERIC;
 382     }
 383
 384     /* Allocate the memory needed to store the module's structure */
 385     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 386     if( p_sys == NULL )
 387         return VLC_EGENERIC;
 388
 389     var_Create( p_this, MONO_CFG "downmix",
 390                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
 391     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
 392
 393     var_Create( p_this, MONO_CFG "channel",
 394                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
 395     p_sys->i_channel_selected =
 396             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
 397
 398     if( p_sys->b_downmix )
 399     {
 400         msg_Dbg( p_this, "using stereo to mono downmix" );
 401         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 402         p_filter->fmt_out.audio.i_channels = 1;
 403     }
 404     else
 405     {
 406         msg_Dbg( p_this, "using pseudo mono" );
 407         p_filter->fmt_out.audio.i_physical_channels =
 408                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
 409         p_filter->fmt_out.audio.i_channels = 2;
 410     }
 411
 412     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 413     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
 414
 415     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 416     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 417
 418     p_sys->i_overflow_buffer_size = 0;
 419     p_sys->p_overflow_buffer = NULL;
 420     p_sys->i_nb_atomic_operations = 0;
 421     p_sys->p_atomic_operations = NULL;
 422
 423     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 424               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 425               p_filter->fmt_in.audio.i_physical_channels,
 426               p_filter->fmt_in.audio.i_rate ) < 0 )
 427     {
 428         var_Destroy( p_this, MONO_CFG "channel" );
 429         var_Destroy( p_this, MONO_CFG "downmix" );
 430         free( p_sys );
 431         return VLC_EGENERIC;
 432     }
 433
 434     p_filter->pf_audio_filter = Convert;
 435
 436     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 437              (char *)&p_filter->fmt_in.i_codec,
 438              (char *)&p_filter->fmt_out.i_codec,
 439              p_filter->fmt_in.audio.i_physical_channels,
 440              p_filter->fmt_out.audio.i_physical_channels,
 441              p_filter->fmt_in.audio.i_bitspersample,
 442              p_filter->fmt_out.audio.i_bitspersample );
 443
 444     return VLC_SUCCESS;
 445 }
 446
 447 /*****************************************************************************
 448  * CloseFilter
 449  *****************************************************************************/
 450 static void CloseFilter( vlc_object_t *p_this)
 451 {
 452     filter_t *p_filter = (filter_t *) p_this;
 453     filter_sys_t *p_sys = p_filter->p_sys;
 454
 455     var_Destroy( p_this, MONO_CFG "channel" );
 456     var_Destroy( p_this, MONO_CFG "downmix" );
 457     free( p_sys->p_atomic_operations );
 458     free( p_sys->p_overflow_buffer );
 459     free( p_sys );
 460 }
 461
 462 /*****************************************************************************
 463  * Convert
 464  *****************************************************************************/
 465 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 466 {
 467     aout_filter_t aout_filter;
 468     aout_buffer_t in_buf, out_buf;
 469     block_t *p_out = NULL;
 470     unsigned int i_samples;
 471     int i_out_size;
 472
 473     if( !p_block || !p_block->i_samples )
 474     {
 475         if( p_block )
 476             block_Release( p_block );
 477         return NULL;
 478     }
 479
 480     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
 481                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 482
 483     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 484     if( !p_out )
 485     {
 486         msg_Warn( p_filter, "can't get output buffer" );
 487         block_Release( p_block );
 488         return NULL;
 489     }
 490     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
 491                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 492     p_out->i_dts = p_block->i_dts;
 493     p_out->i_pts = p_block->i_pts;
 494     p_out->i_length = p_block->i_length;
 495
 496     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 497     aout_filter.input = p_filter->fmt_in.audio;
 498     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
 499     aout_filter.output = p_filter->fmt_out.audio;
 500     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
 501
 502     in_buf.p_buffer = p_block->p_buffer;
 503     in_buf.i_nb_bytes = p_block->i_buffer;
 504     in_buf.i_nb_samples = p_block->i_samples;
 505
 506 #if 0
 507     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 508                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 509     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 510     {
 511         msg_Err( p_filter, "input buffer is not word aligned" );
 512         /* Fix output buffer to be word aligned */
 513     }
 514 #endif
 515
 516     out_buf.p_buffer = p_out->p_buffer;
 517     out_buf.i_nb_bytes = p_out->i_buffer;
 518     out_buf.i_nb_samples = p_out->i_samples;
 519
 520     memset( p_out->p_buffer, 0, i_out_size );
 521     if( p_filter->p_sys->b_downmix )
 522     {
 523         stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
 524         i_samples = mono( &aout_filter, &out_buf, &in_buf );
 525     }
 526     else
 527     {
 528         i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
 529     }
 530
 531     p_out->i_buffer = out_buf.i_nb_bytes;
 532     p_out->i_samples = out_buf.i_nb_samples;
 533
 534     block_Release( p_block );
 535     return p_out;
 536 }
 537
 538 /* stereo2mono_downmix - stereo channels into one mono channel.
 539  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 540  * converted from float into int16_t based downmix
 541  * Written by Boris Dorès <babal@via.ecp.fr>
 542  */
 543 static void stereo2mono_downmix( aout_filter_t * p_filter,
 544                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 545 {
 546     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 547
 548     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
 549     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
 550
 551     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 552     uint8_t * p_out;
 553     uint8_t * p_overflow;
 554     uint8_t * p_slide;
 555
 556     size_t i_overflow_size;     /* in bytes */
 557     size_t i_out_size;          /* in bytes */
 558
 559     unsigned int i, j;
 560
 561     int i_source_channel_offset;
 562     int i_dest_channel_offset;
 563     unsigned int i_delay;
 564     double d_amplitude_factor;
 565
 566     /* out buffer characterisitcs */
 567     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 568     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
 569     p_out = p_out_buf->p_buffer;
 570     i_out_size = p_out_buf->i_nb_bytes;
 571
 572     if( p_sys != NULL )
 573     {
 574         /* Slide the overflow buffer */
 575         p_overflow = p_sys->p_overflow_buffer;
 576         i_overflow_size = p_sys->i_overflow_buffer_size;
 577
 578         if ( i_out_size > i_overflow_size )
 579             memcpy( p_out, p_overflow, i_overflow_size );
 580         else
 581             memcpy( p_out, p_overflow, i_out_size );
 582
 583         p_slide = p_sys->p_overflow_buffer;
 584         while( p_slide < p_overflow + i_overflow_size )
 585         {
 586             if( p_slide + i_out_size < p_overflow + i_overflow_size )
 587             {
 588                 memset( p_slide, 0, i_out_size );
 589                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 590                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
 591                 else
 592                     memcpy( p_slide, p_slide + i_out_size,
 593                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 594             }
 595             else
 596             {
 597                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 598             }
 599             p_slide += i_out_size;
 600         }
 601
 602         /* apply the atomic operations */
 603         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 604         {
 605             /* shorter variable names */
 606             i_source_channel_offset
 607                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
 608             i_dest_channel_offset
 609                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 610             i_delay = p_sys->p_atomic_operations[i].i_delay;
 611             d_amplitude_factor
 612                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
 613
 614             if( p_out_buf->i_nb_samples > i_delay )
 615             {
 616                 /* current buffer coefficients */
 617                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 618                 {
 619                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 620                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 621                            * d_amplitude_factor;
 622                 }
 623
 624                 /* overflow buffer coefficients */
 625                 for( j = 0; j < i_delay; j++ )
 626                 {
 627                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 628                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 629                            * i_input_nb + i_source_channel_offset ]
 630                            * d_amplitude_factor;
 631                 }
 632             }
 633             else
 634             {
 635                 /* overflow buffer coefficients only */
 636                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 637                 {
 638                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 639                         * i_output_nb + i_dest_channel_offset ]
 640                         += p_in[ j * i_input_nb + i_source_channel_offset ]
 641                            * d_amplitude_factor;
 642                 }
 643             }
 644         }
 645     }
 646     else
 647     {
 648         memset( p_out, 0, i_out_size );
 649     }
 650 }
 651
 652 /* Simple stereo to mono mixing. */
 653 static unsigned int mono( aout_filter_t *p_filter,
 654                           aout_buffer_t *p_output, aout_buffer_t *p_input )
 655 {
 656     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 657     int16_t *p_in, *p_out;
 658     unsigned int n = 0, r = 0;
 659
 660     p_in = (int16_t *) p_input->p_buffer;
 661     p_out = (int16_t *) p_output->p_buffer;
 662
 663     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 664     {
 665         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 666         r++;
 667         n += 2;
 668     }
 669     return r;
 670 }
 671
 672 /* Simple stereo to mono mixing. */
 673 static unsigned int stereo_to_mono( aout_filter_t *p_filter,
 674                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
 675 {
 676     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 677     int16_t *p_in, *p_out;
 678     unsigned int n;
 679
 680     p_in = (int16_t *) p_input->p_buffer;
 681     p_out = (int16_t *) p_output->p_buffer;
 682
 683     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 684     {
 685         /* Fake real mono. */
 686         if( p_sys->i_channel_selected == -1)
 687         {
 688             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 689             n++;
 690         }
 691         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 692         {
 693             p_out[n] = p_out[n+1] = p_in[n];
 694         }
 695     }
 696     return n;
 697 }