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( filter_t *, block_t *, block_t * );
  53 static unsigned int mono( filter_t *, block_t *, block_t * );
  54 static void stereo2mono_downmix( filter_t *, block_t *, block_t * );
  55
  56 /*****************************************************************************
  57  * Local structures
  58  *****************************************************************************/
  59 struct atomic_operation_t
  60 {
  61     int i_source_channel_offset;
  62     int i_dest_channel_offset;
  63     unsigned int i_delay;/* in sample unit */
  64     double d_amplitude_factor;
  65 };
  66
  67 struct filter_sys_t
  68 {
  69     bool b_downmix;
  70
  71     unsigned int i_nb_channels; /* number of int16_t per sample */
  72     int i_channel_selected;
  73     int i_bitspersample;
  74
  75     size_t i_overflow_buffer_size;/* in bytes */
  76     uint8_t * p_overflow_buffer;
  77     unsigned int i_nb_atomic_operations;
  78     struct atomic_operation_t * p_atomic_operations;
  79 };
  80
  81 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
  82 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  83     "downmix algorithm that is used in the headphone channel mixer. It " \
  84     "gives the effect of standing in a room full of speakers." )
  85
  86 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  87 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  88     "except the selected channel.")
  89
  90 static const int pi_pos_values[] = { 0, 1, 4, 5, 7, 8, 2, 3, 6 };
  91 static const char *const ppsz_pos_descriptions[] =
  92 { N_("Left"), N_("Right"),
  93   N_("Rear left"), N_("Rear right"),
  94   N_("Center"), N_("Low-frequency effects"),
  95   N_("Side left"), N_("Side right"), N_("Rear center") };
  96
  97 #define MONO_CFG "sout-mono-"
  98 /*****************************************************************************
  99  * Module descriptor
 100  *****************************************************************************/
 101 vlc_module_begin ()
 102     set_description( N_("Audio filter for stereo to mono conversion") )
 103     set_capability( "audio filter", 0 )
 104     set_category( CAT_AUDIO )
 105     set_subcategory( SUBCAT_AUDIO_AFILTER )
 106     set_callbacks( OpenFilter, CloseFilter )
 107     set_shortname( "Mono" )
 108
 109     add_bool( MONO_CFG "downmix", true, MONO_DOWNMIX_TEXT,
 110               MONO_DOWNMIX_LONGTEXT, false )
 111     add_integer( MONO_CFG "channel", -1, MONO_CHANNEL_TEXT,
 112         MONO_CHANNEL_LONGTEXT, false )
 113         change_integer_list( pi_pos_values, ppsz_pos_descriptions )
 114
 115 vlc_module_end ()
 116
 117 /* Init() and ComputeChannelOperations() -
 118  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 119  * converted from float into int16_t based downmix
 120  * Written by Boris Dorès <babal@via.ecp.fr>
 121  */
 122
 123 /*****************************************************************************
 124  * Init: initialize internal data structures
 125  * and computes the needed atomic operations
 126  *****************************************************************************/
 127 /* x and z represent the coordinates of the virtual speaker
 128  *  relatively to the center of the listener's head, measured in meters :
 129  *
 130  *  left              right
 131  *Z
 132  *-
 133  *a          head
 134  *x
 135  *i
 136  *s
 137  *  rear left    rear right
 138  *
 139  *          x-axis
 140  *  */
 141 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 142         unsigned int i_rate, unsigned int i_next_atomic_operation,
 143         int i_source_channel_offset, double d_x, double d_z,
 144         double d_compensation_length, double d_channel_amplitude_factor )
 145 {
 146     double d_c = 340; /*sound celerity (unit: m/s)*/
 147     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 148
 149     /* Left ear */
 150     p_data->p_atomic_operations[i_next_atomic_operation]
 151         .i_source_channel_offset = i_source_channel_offset;
 152     p_data->p_atomic_operations[i_next_atomic_operation]
 153         .i_dest_channel_offset = 0;/* left */
 154     p_data->p_atomic_operations[i_next_atomic_operation]
 155         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 156                           / d_c * i_rate - d_compensation_delay );
 157     if( d_x < 0 )
 158     {
 159         p_data->p_atomic_operations[i_next_atomic_operation]
 160             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 161     }
 162     else if( d_x > 0 )
 163     {
 164         p_data->p_atomic_operations[i_next_atomic_operation]
 165             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 166     }
 167     else
 168     {
 169         p_data->p_atomic_operations[i_next_atomic_operation]
 170             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 171     }
 172
 173     /* Right ear */
 174     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 175         .i_source_channel_offset = i_source_channel_offset;
 176     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 177         .i_dest_channel_offset = 1;/* right */
 178     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 179         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 180                           / d_c * i_rate - d_compensation_delay );
 181     if( d_x < 0 )
 182     {
 183         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 184             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 185     }
 186     else if( d_x > 0 )
 187     {
 188         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 189             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 190     }
 191     else
 192     {
 193         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 194             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 195     }
 196 }
 197
 198 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 199                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 200                  unsigned int i_rate )
 201 {
 202     double d_x = var_InheritInteger( p_this, "headphone-dim" );
 203     double d_z = d_x;
 204     double d_z_rear = -d_x/3;
 205     double d_min = 0;
 206     unsigned int i_next_atomic_operation;
 207     int i_source_channel_offset;
 208     unsigned int i;
 209
 210     if( var_InheritBool( p_this, "headphone-compensate" ) )
 211     {
 212         /* minimal distance to any speaker */
 213         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 214         {
 215             d_min = d_z_rear;
 216         }
 217         else
 218         {
 219             d_min = d_z;
 220         }
 221     }
 222
 223     /* Number of elementary operations */
 224     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 225     if( i_physical_channels & AOUT_CHAN_CENTER )
 226     {
 227         p_data->i_nb_atomic_operations += 2;
 228     }
 229     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 230             * p_data->i_nb_atomic_operations );
 231     if( p_data->p_atomic_operations == NULL )
 232         return -1;
 233
 234     /* For each virtual speaker, computes elementary wave propagation time
 235      * to each ear */
 236     i_next_atomic_operation = 0;
 237     i_source_channel_offset = 0;
 238     if( i_physical_channels & AOUT_CHAN_LEFT )
 239     {
 240         ComputeChannelOperations( p_data , i_rate
 241                 , i_next_atomic_operation , i_source_channel_offset
 242                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 243         i_next_atomic_operation += 2;
 244         i_source_channel_offset++;
 245     }
 246     if( i_physical_channels & AOUT_CHAN_RIGHT )
 247     {
 248         ComputeChannelOperations( p_data , i_rate
 249                 , i_next_atomic_operation , i_source_channel_offset
 250                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 251         i_next_atomic_operation += 2;
 252         i_source_channel_offset++;
 253     }
 254     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 255     {
 256         ComputeChannelOperations( p_data , i_rate
 257                 , i_next_atomic_operation , i_source_channel_offset
 258                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 259         i_next_atomic_operation += 2;
 260         i_source_channel_offset++;
 261     }
 262     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 263     {
 264         ComputeChannelOperations( p_data , i_rate
 265                 , i_next_atomic_operation , i_source_channel_offset
 266                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 267         i_next_atomic_operation += 2;
 268         i_source_channel_offset++;
 269     }
 270     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 271     {
 272         ComputeChannelOperations( p_data , i_rate
 273                 , i_next_atomic_operation , i_source_channel_offset
 274                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 275         i_next_atomic_operation += 2;
 276         i_source_channel_offset++;
 277     }
 278     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 279     {
 280         ComputeChannelOperations( p_data , i_rate
 281                 , i_next_atomic_operation , i_source_channel_offset
 282                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 283         i_next_atomic_operation += 2;
 284         i_source_channel_offset++;
 285     }
 286     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 287     {
 288         ComputeChannelOperations( p_data , i_rate
 289                 , i_next_atomic_operation , i_source_channel_offset
 290                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 291         i_next_atomic_operation += 2;
 292         i_source_channel_offset++;
 293     }
 294     if( i_physical_channels & AOUT_CHAN_CENTER )
 295     {
 296         /* having two center channels increases the spatialization effect */
 297         ComputeChannelOperations( p_data , i_rate
 298                 , i_next_atomic_operation , i_source_channel_offset
 299                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 300         i_next_atomic_operation += 2;
 301         ComputeChannelOperations( p_data , i_rate
 302                 , i_next_atomic_operation , i_source_channel_offset
 303                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 304         i_next_atomic_operation += 2;
 305         i_source_channel_offset++;
 306     }
 307     if( i_physical_channels & AOUT_CHAN_LFE )
 308     {
 309         ComputeChannelOperations( p_data , i_rate
 310                 , i_next_atomic_operation , i_source_channel_offset
 311                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 312         i_next_atomic_operation += 2;
 313         i_source_channel_offset++;
 314     }
 315
 316     /* Initialize the overflow buffer
 317      * we need it because the process induce a delay in the samples */
 318     p_data->i_overflow_buffer_size = 0;
 319     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 320     {
 321         if( p_data->i_overflow_buffer_size
 322                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 323         {
 324             p_data->i_overflow_buffer_size
 325                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 326         }
 327     }
 328     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 329     if( p_data->p_overflow_buffer == NULL )
 330     {
 331         free( p_data->p_atomic_operations );
 332         return -1;
 333     }
 334     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 335
 336     /* end */
 337     return 0;
 338 }
 339
 340 /*****************************************************************************
 341  * OpenFilter
 342  *****************************************************************************/
 343 static int OpenFilter( vlc_object_t *p_this )
 344 {
 345     filter_t * p_filter = (filter_t *)p_this;
 346     filter_sys_t *p_sys = NULL;
 347
 348     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 349     {
 350         /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
 351         return VLC_EGENERIC;
 352     }
 353
 354     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) ||
 355         (p_filter->fmt_in.audio.i_format != VLC_CODEC_S16N) ||
 356         (p_filter->fmt_out.audio.i_format != VLC_CODEC_S16N) )
 357     {
 358         /*msg_Err( p_this, "couldn't load mono filter" );*/
 359         return VLC_EGENERIC;
 360     }
 361
 362     /* Allocate the memory needed to store the module's structure */
 363     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 364     if( p_sys == NULL )
 365         return VLC_EGENERIC;
 366
 367     p_sys->b_downmix = var_CreateGetBool( p_this, MONO_CFG "downmix" );
 368     p_sys->i_channel_selected =
 369             (unsigned int) var_CreateGetInteger( p_this, MONO_CFG "channel" );
 370
 371     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 372     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 373
 374     p_sys->i_overflow_buffer_size = 0;
 375     p_sys->p_overflow_buffer = NULL;
 376     p_sys->i_nb_atomic_operations = 0;
 377     p_sys->p_atomic_operations = NULL;
 378
 379     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 380               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 381               p_filter->fmt_in.audio.i_physical_channels,
 382               p_filter->fmt_in.audio.i_rate ) < 0 )
 383     {
 384         var_Destroy( p_this, MONO_CFG "channel" );
 385         var_Destroy( p_this, MONO_CFG "downmix" );
 386         free( p_sys );
 387         return VLC_EGENERIC;
 388     }
 389
 390     if( p_sys->b_downmix )
 391     {
 392         msg_Dbg( p_this, "using stereo to mono downmix" );
 393         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 394         p_filter->fmt_out.audio.i_channels = 1;
 395     }
 396     else
 397     {
 398         msg_Dbg( p_this, "using pseudo mono" );
 399         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHANS_STEREO;
 400         p_filter->fmt_out.audio.i_channels = 2;
 401     }
 402     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 403     p_filter->pf_audio_filter = Convert;
 404
 405     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 406              (char *)&p_filter->fmt_in.i_codec,
 407              (char *)&p_filter->fmt_out.i_codec,
 408              p_filter->fmt_in.audio.i_physical_channels,
 409              p_filter->fmt_out.audio.i_physical_channels,
 410              p_filter->fmt_in.audio.i_bitspersample,
 411              p_filter->fmt_out.audio.i_bitspersample );
 412
 413     return VLC_SUCCESS;
 414 }
 415
 416 /*****************************************************************************
 417  * CloseFilter
 418  *****************************************************************************/
 419 static void CloseFilter( vlc_object_t *p_this)
 420 {
 421     filter_t *p_filter = (filter_t *) p_this;
 422     filter_sys_t *p_sys = p_filter->p_sys;
 423
 424     var_Destroy( p_this, MONO_CFG "channel" );
 425     var_Destroy( p_this, MONO_CFG "downmix" );
 426     free( p_sys->p_atomic_operations );
 427     free( p_sys->p_overflow_buffer );
 428     free( p_sys );
 429 }
 430
 431 /*****************************************************************************
 432  * Convert
 433  *****************************************************************************/
 434 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 435 {
 436     block_t *p_out;
 437     int i_out_size;
 438
 439     if( !p_block || !p_block->i_nb_samples )
 440     {
 441         if( p_block )
 442             block_Release( p_block );
 443         return NULL;
 444     }
 445
 446     i_out_size = p_block->i_nb_samples * p_filter->p_sys->i_bitspersample/8 *
 447                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 448
 449     p_out = block_Alloc( i_out_size );
 450     if( !p_out )
 451     {
 452         msg_Warn( p_filter, "can't get output buffer" );
 453         block_Release( p_block );
 454         return NULL;
 455     }
 456     p_out->i_nb_samples =
 457                   (p_block->i_nb_samples / p_filter->p_sys->i_nb_channels) *
 458                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 459
 460 #if 0
 461     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 462                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 463     if( (in_buf.i_buffer != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 464     {
 465         msg_Err( p_filter, "input buffer is not word aligned" );
 466         /* Fix output buffer to be word aligned */
 467     }
 468 #endif
 469
 470     memset( p_out->p_buffer, 0, i_out_size );
 471     if( p_filter->p_sys->b_downmix )
 472     {
 473         stereo2mono_downmix( p_filter, p_block, p_out );
 474         mono( p_filter, p_out, p_block );
 475     }
 476     else
 477     {
 478         stereo_to_mono( p_filter, p_out, p_block );
 479     }
 480
 481     block_Release( p_block );
 482     return p_out;
 483 }
 484
 485 /* stereo2mono_downmix - stereo channels into one mono channel.
 486  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 487  * converted from float into int16_t based downmix
 488  * Written by Boris Dorès <babal@via.ecp.fr>
 489  */
 490 static void stereo2mono_downmix( filter_t * p_filter,
 491                                  block_t * p_in_buf, block_t * p_out_buf )
 492 {
 493     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 494
 495     int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 496     int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio );
 497
 498     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 499     uint8_t * p_out;
 500     uint8_t * p_overflow;
 501     uint8_t * p_slide;
 502
 503     size_t i_overflow_size;     /* in bytes */
 504     size_t i_out_size;          /* in bytes */
 505
 506     unsigned int i, j;
 507
 508     int i_source_channel_offset;
 509     int i_dest_channel_offset;
 510     unsigned int i_delay;
 511     double d_amplitude_factor;
 512
 513     /* out buffer characterisitcs */
 514     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 515     p_out_buf->i_buffer = p_in_buf->i_buffer * i_output_nb / i_input_nb;
 516     p_out = p_out_buf->p_buffer;
 517     i_out_size = p_out_buf->i_buffer;
 518
 519     /* Slide the overflow buffer */
 520     p_overflow = p_sys->p_overflow_buffer;
 521     i_overflow_size = p_sys->i_overflow_buffer_size;
 522
 523     if ( i_out_size > i_overflow_size )
 524         memcpy( p_out, p_overflow, i_overflow_size );
 525     else
 526         memcpy( p_out, p_overflow, i_out_size );
 527
 528     p_slide = p_sys->p_overflow_buffer;
 529     while( p_slide < p_overflow + i_overflow_size )
 530     {
 531         if( p_slide + i_out_size < p_overflow + i_overflow_size )
 532         {
 533             memset( p_slide, 0, i_out_size );
 534             if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 535                 memcpy( p_slide, p_slide + i_out_size, i_out_size );
 536             else
 537                 memcpy( p_slide, p_slide + i_out_size,
 538                         p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 539         }
 540         else
 541         {
 542             memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 543         }
 544         p_slide += i_out_size;
 545     }
 546
 547     /* apply the atomic operations */
 548     for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 549     {
 550         /* shorter variable names */
 551         i_source_channel_offset
 552             = p_sys->p_atomic_operations[i].i_source_channel_offset;
 553         i_dest_channel_offset
 554             = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 555         i_delay = p_sys->p_atomic_operations[i].i_delay;
 556         d_amplitude_factor
 557             = p_sys->p_atomic_operations[i].d_amplitude_factor;
 558
 559         if( p_out_buf->i_nb_samples > i_delay )
 560         {
 561             /* current buffer coefficients */
 562             for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 563             {
 564                 ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 565                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 566                        * d_amplitude_factor;
 567             }
 568
 569             /* overflow buffer coefficients */
 570             for( j = 0; j < i_delay; j++ )
 571             {
 572                 ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 573                     += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 574                        * i_input_nb + i_source_channel_offset ]
 575                        * d_amplitude_factor;
 576             }
 577         }
 578         else
 579         {
 580             /* overflow buffer coefficients only */
 581             for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 582             {
 583                 ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 584                                         * i_output_nb + i_dest_channel_offset ]
 585                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 586                        * d_amplitude_factor;
 587             }
 588         }
 589     }
 590 }
 591
 592 /* Simple stereo to mono mixing. */
 593 static unsigned int mono( filter_t *p_filter,
 594                           block_t *p_output, block_t *p_input )
 595 {
 596     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 597     int16_t *p_in, *p_out;
 598     unsigned int n = 0, r = 0;
 599
 600     p_in = (int16_t *) p_input->p_buffer;
 601     p_out = (int16_t *) p_output->p_buffer;
 602
 603     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 604     {
 605         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 606         r++;
 607         n += 2;
 608     }
 609     return r;
 610 }
 611
 612 /* Simple stereo to mono mixing. */
 613 static unsigned int stereo_to_mono( filter_t *p_filter,
 614                                     block_t *p_output, block_t *p_input )
 615 {
 616     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 617     int16_t *p_in, *p_out;
 618     unsigned int n;
 619
 620     p_in = (int16_t *) p_input->p_buffer;
 621     p_out = (int16_t *) p_output->p_buffer;
 622
 623     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 624     {
 625         /* Fake real mono. */
 626         if( p_sys->i_channel_selected == -1)
 627         {
 628             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 629             n++;
 630         }
 631         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 632         {
 633             p_out[n] = p_out[n+1] = p_in[n];
 634         }
 635     }
 636     return n;
 637 }