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 #define VLC_MODULE_LICENSE VLC_LICENSE_GPL_2_PLUS
  35 #include <vlc_common.h>
  36 #include <vlc_plugin.h>
  37 #include <vlc_block.h>
  38 #include <vlc_filter.h>
  39 #include <vlc_aout.h>
  40
  41 /*****************************************************************************
  42  * Local prototypes
  43  *****************************************************************************/
  44 static int  OpenFilter    ( vlc_object_t * );
  45 static void CloseFilter   ( vlc_object_t * );
  46
  47 static block_t *Convert( filter_t *p_filter, block_t *p_block );
  48
  49 static unsigned int stereo_to_mono( filter_t *, block_t *, block_t * );
  50 static unsigned int mono( filter_t *, block_t *, block_t * );
  51 static void stereo2mono_downmix( filter_t *, block_t *, block_t * );
  52
  53 /*****************************************************************************
  54  * Local structures
  55  *****************************************************************************/
  56 struct atomic_operation_t
  57 {
  58     int i_source_channel_offset;
  59     int i_dest_channel_offset;
  60     unsigned int i_delay;/* in sample unit */
  61     double d_amplitude_factor;
  62 };
  63
  64 struct filter_sys_t
  65 {
  66     bool b_downmix;
  67
  68     unsigned int i_nb_channels; /* number of int16_t per sample */
  69     int i_channel_selected;
  70     int i_bitspersample;
  71
  72     size_t i_overflow_buffer_size;/* in bytes */
  73     uint8_t * p_overflow_buffer;
  74     unsigned int i_nb_atomic_operations;
  75     struct atomic_operation_t * p_atomic_operations;
  76 };
  77
  78 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
  79 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
  80     "downmix algorithm that is used in the headphone channel mixer. It " \
  81     "gives the effect of standing in a room full of speakers." )
  82
  83 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
  84 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
  85     "except the selected channel.")
  86
  87 static const int pi_pos_values[] = { 0, 1, 4, 5, 7, 8, 2, 3, 6 };
  88 static const char *const ppsz_pos_descriptions[] =
  89 { N_("Left"), N_("Right"),
  90   N_("Rear left"), N_("Rear right"),
  91   N_("Center"), N_("Low-frequency effects"),
  92   N_("Side left"), N_("Side right"), N_("Rear center") };
  93
  94 #define MONO_CFG "sout-mono-"
  95 /*****************************************************************************
  96  * Module descriptor
  97  *****************************************************************************/
  98 vlc_module_begin ()
  99     set_description( N_("Stereo to mono downmixer") )
 100     set_capability( "audio filter", 0 )
 101     set_category( CAT_AUDIO )
 102     set_subcategory( SUBCAT_AUDIO_AFILTER )
 103     set_callbacks( OpenFilter, CloseFilter )
 104     set_shortname( "Mono" )
 105
 106     add_bool( MONO_CFG "downmix", true, MONO_DOWNMIX_TEXT,
 107               MONO_DOWNMIX_LONGTEXT, false )
 108     add_integer( MONO_CFG "channel", -1, MONO_CHANNEL_TEXT,
 109         MONO_CHANNEL_LONGTEXT, false )
 110         change_integer_list( pi_pos_values, ppsz_pos_descriptions )
 111
 112 vlc_module_end ()
 113
 114 /* Init() and ComputeChannelOperations() -
 115  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 116  * converted from float into int16_t based downmix
 117  * Written by Boris Dorès <babal@via.ecp.fr>
 118  */
 119
 120 /*****************************************************************************
 121  * Init: initialize internal data structures
 122  * and computes the needed atomic operations
 123  *****************************************************************************/
 124 /* x and z represent the coordinates of the virtual speaker
 125  *  relatively to the center of the listener's head, measured in meters :
 126  *
 127  *  left              right
 128  *Z
 129  *-
 130  *a          head
 131  *x
 132  *i
 133  *s
 134  *  rear left    rear right
 135  *
 136  *          x-axis
 137  *  */
 138 static void ComputeChannelOperations( struct filter_sys_t * p_data,
 139         unsigned int i_rate, unsigned int i_next_atomic_operation,
 140         int i_source_channel_offset, double d_x, double d_z,
 141         double d_compensation_length, double d_channel_amplitude_factor )
 142 {
 143     double d_c = 340; /*sound celerity (unit: m/s)*/
 144     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 145
 146     /* Left ear */
 147     p_data->p_atomic_operations[i_next_atomic_operation]
 148         .i_source_channel_offset = i_source_channel_offset;
 149     p_data->p_atomic_operations[i_next_atomic_operation]
 150         .i_dest_channel_offset = 0;/* left */
 151     p_data->p_atomic_operations[i_next_atomic_operation]
 152         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 153                           / d_c * i_rate - d_compensation_delay );
 154     if( d_x < 0 )
 155     {
 156         p_data->p_atomic_operations[i_next_atomic_operation]
 157             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 158     }
 159     else if( d_x > 0 )
 160     {
 161         p_data->p_atomic_operations[i_next_atomic_operation]
 162             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 163     }
 164     else
 165     {
 166         p_data->p_atomic_operations[i_next_atomic_operation]
 167             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 168     }
 169
 170     /* Right ear */
 171     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 172         .i_source_channel_offset = i_source_channel_offset;
 173     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 174         .i_dest_channel_offset = 1;/* right */
 175     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 176         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 177                           / d_c * i_rate - d_compensation_delay );
 178     if( d_x < 0 )
 179     {
 180         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 181             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 182     }
 183     else if( d_x > 0 )
 184     {
 185         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 186             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 187     }
 188     else
 189     {
 190         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 191             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 192     }
 193 }
 194
 195 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
 196                  unsigned int i_nb_channels, uint32_t i_physical_channels,
 197                  unsigned int i_rate )
 198 {
 199     double d_x = var_InheritInteger( p_this, "headphone-dim" );
 200     double d_z = d_x;
 201     double d_z_rear = -d_x/3;
 202     double d_min = 0;
 203     unsigned int i_next_atomic_operation;
 204     int i_source_channel_offset;
 205     unsigned int i;
 206
 207     if( var_InheritBool( p_this, "headphone-compensate" ) )
 208     {
 209         /* minimal distance to any speaker */
 210         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 211         {
 212             d_min = d_z_rear;
 213         }
 214         else
 215         {
 216             d_min = d_z;
 217         }
 218     }
 219
 220     /* Number of elementary operations */
 221     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 222     if( i_physical_channels & AOUT_CHAN_CENTER )
 223     {
 224         p_data->i_nb_atomic_operations += 2;
 225     }
 226     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 227             * p_data->i_nb_atomic_operations );
 228     if( p_data->p_atomic_operations == NULL )
 229         return -1;
 230
 231     /* For each virtual speaker, computes elementary wave propagation time
 232      * to each ear */
 233     i_next_atomic_operation = 0;
 234     i_source_channel_offset = 0;
 235     if( i_physical_channels & AOUT_CHAN_LEFT )
 236     {
 237         ComputeChannelOperations( p_data , i_rate
 238                 , i_next_atomic_operation , i_source_channel_offset
 239                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 240         i_next_atomic_operation += 2;
 241         i_source_channel_offset++;
 242     }
 243     if( i_physical_channels & AOUT_CHAN_RIGHT )
 244     {
 245         ComputeChannelOperations( p_data , i_rate
 246                 , i_next_atomic_operation , i_source_channel_offset
 247                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 248         i_next_atomic_operation += 2;
 249         i_source_channel_offset++;
 250     }
 251     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 252     {
 253         ComputeChannelOperations( p_data , i_rate
 254                 , i_next_atomic_operation , i_source_channel_offset
 255                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 256         i_next_atomic_operation += 2;
 257         i_source_channel_offset++;
 258     }
 259     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 260     {
 261         ComputeChannelOperations( p_data , i_rate
 262                 , i_next_atomic_operation , i_source_channel_offset
 263                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 264         i_next_atomic_operation += 2;
 265         i_source_channel_offset++;
 266     }
 267     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 268     {
 269         ComputeChannelOperations( p_data , i_rate
 270                 , i_next_atomic_operation , i_source_channel_offset
 271                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 272         i_next_atomic_operation += 2;
 273         i_source_channel_offset++;
 274     }
 275     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 276     {
 277         ComputeChannelOperations( p_data , i_rate
 278                 , i_next_atomic_operation , i_source_channel_offset
 279                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 280         i_next_atomic_operation += 2;
 281         i_source_channel_offset++;
 282     }
 283     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 284     {
 285         ComputeChannelOperations( p_data , i_rate
 286                 , i_next_atomic_operation , i_source_channel_offset
 287                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 288         i_next_atomic_operation += 2;
 289         i_source_channel_offset++;
 290     }
 291     if( i_physical_channels & AOUT_CHAN_CENTER )
 292     {
 293         /* having two center channels increases the spatialization effect */
 294         ComputeChannelOperations( p_data , i_rate
 295                 , i_next_atomic_operation , i_source_channel_offset
 296                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 297         i_next_atomic_operation += 2;
 298         ComputeChannelOperations( p_data , i_rate
 299                 , i_next_atomic_operation , i_source_channel_offset
 300                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 301         i_next_atomic_operation += 2;
 302         i_source_channel_offset++;
 303     }
 304     if( i_physical_channels & AOUT_CHAN_LFE )
 305     {
 306         ComputeChannelOperations( p_data , i_rate
 307                 , i_next_atomic_operation , i_source_channel_offset
 308                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 309         i_next_atomic_operation += 2;
 310         i_source_channel_offset++;
 311     }
 312
 313     /* Initialize the overflow buffer
 314      * we need it because the process induce a delay in the samples */
 315     p_data->i_overflow_buffer_size = 0;
 316     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 317     {
 318         if( p_data->i_overflow_buffer_size
 319                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
 320         {
 321             p_data->i_overflow_buffer_size
 322                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
 323         }
 324     }
 325     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
 326     if( p_data->p_overflow_buffer == NULL )
 327     {
 328         free( p_data->p_atomic_operations );
 329         return -1;
 330     }
 331     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
 332
 333     /* end */
 334     return 0;
 335 }
 336
 337 /*****************************************************************************
 338  * OpenFilter
 339  *****************************************************************************/
 340 static int OpenFilter( vlc_object_t *p_this )
 341 {
 342     filter_t * p_filter = (filter_t *)p_this;
 343     filter_sys_t *p_sys = NULL;
 344
 345     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
 346     {
 347         /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
 348         return VLC_EGENERIC;
 349     }
 350
 351     p_filter->fmt_in.audio.i_format = VLC_CODEC_S16N;
 352     p_filter->fmt_out.audio.i_format = VLC_CODEC_S16N;
 353
 354     /* Allocate the memory needed to store the module's structure */
 355     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
 356     if( p_sys == NULL )
 357         return VLC_EGENERIC;
 358
 359     p_sys->b_downmix = var_InheritBool( p_this, MONO_CFG "downmix" );
 360     p_sys->i_channel_selected = var_InheritInteger( p_this, MONO_CFG "channel" );
 361
 362     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 363     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
 364
 365     p_sys->i_overflow_buffer_size = 0;
 366     p_sys->p_overflow_buffer = NULL;
 367     p_sys->i_nb_atomic_operations = 0;
 368     p_sys->p_atomic_operations = NULL;
 369
 370     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
 371               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
 372               p_filter->fmt_in.audio.i_physical_channels,
 373               p_filter->fmt_in.audio.i_rate ) < 0 )
 374     {
 375         free( p_sys );
 376         return VLC_EGENERIC;
 377     }
 378
 379     if( p_sys->b_downmix )
 380     {
 381         msg_Dbg( p_this, "using stereo to mono downmix" );
 382         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
 383         p_filter->fmt_out.audio.i_channels = 1;
 384     }
 385     else
 386     {
 387         msg_Dbg( p_this, "using pseudo mono" );
 388         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHANS_STEREO;
 389         p_filter->fmt_out.audio.i_channels = 2;
 390     }
 391     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 392     p_filter->pf_audio_filter = Convert;
 393
 394     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
 395              (char *)&p_filter->fmt_in.i_codec,
 396              (char *)&p_filter->fmt_out.i_codec,
 397              p_filter->fmt_in.audio.i_physical_channels,
 398              p_filter->fmt_out.audio.i_physical_channels,
 399              p_filter->fmt_in.audio.i_bitspersample,
 400              p_filter->fmt_out.audio.i_bitspersample );
 401
 402     return VLC_SUCCESS;
 403 }
 404
 405 /*****************************************************************************
 406  * CloseFilter
 407  *****************************************************************************/
 408 static void CloseFilter( vlc_object_t *p_this)
 409 {
 410     filter_t *p_filter = (filter_t *) p_this;
 411     filter_sys_t *p_sys = p_filter->p_sys;
 412
 413     free( p_sys->p_atomic_operations );
 414     free( p_sys->p_overflow_buffer );
 415     free( p_sys );
 416 }
 417
 418 /*****************************************************************************
 419  * Convert
 420  *****************************************************************************/
 421 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 422 {
 423     block_t *p_out;
 424     int i_out_size;
 425
 426     if( !p_block || !p_block->i_nb_samples )
 427     {
 428         if( p_block )
 429             block_Release( p_block );
 430         return NULL;
 431     }
 432
 433     i_out_size = p_block->i_nb_samples * p_filter->p_sys->i_bitspersample/8 *
 434                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 435
 436     p_out = block_Alloc( i_out_size );
 437     if( !p_out )
 438     {
 439         msg_Warn( p_filter, "can't get output buffer" );
 440         block_Release( p_block );
 441         return NULL;
 442     }
 443     p_out->i_nb_samples =
 444                   (p_block->i_nb_samples / p_filter->p_sys->i_nb_channels) *
 445                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
 446
 447 #if 0
 448     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
 449                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 450     if( (in_buf.i_buffer != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
 451     {
 452         msg_Err( p_filter, "input buffer is not word aligned" );
 453         /* Fix output buffer to be word aligned */
 454     }
 455 #endif
 456
 457     memset( p_out->p_buffer, 0, i_out_size );
 458     if( p_filter->p_sys->b_downmix )
 459     {
 460         stereo2mono_downmix( p_filter, p_block, p_out );
 461         mono( p_filter, p_out, p_block );
 462     }
 463     else
 464     {
 465         stereo_to_mono( p_filter, p_out, p_block );
 466     }
 467
 468     block_Release( p_block );
 469     return p_out;
 470 }
 471
 472 /* stereo2mono_downmix - stereo channels into one mono channel.
 473  * Code taken from modules/audio_filter/channel_mixer/headphone.c
 474  * converted from float into int16_t based downmix
 475  * Written by Boris Dorès <babal@via.ecp.fr>
 476  */
 477 static void stereo2mono_downmix( filter_t * p_filter,
 478                                  block_t * p_in_buf, block_t * p_out_buf )
 479 {
 480     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 481
 482     int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 483     int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio );
 484
 485     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
 486     uint8_t * p_out;
 487     uint8_t * p_overflow;
 488     uint8_t * p_slide;
 489
 490     size_t i_overflow_size;     /* in bytes */
 491     size_t i_out_size;          /* in bytes */
 492
 493     unsigned int i, j;
 494
 495     int i_source_channel_offset;
 496     int i_dest_channel_offset;
 497     unsigned int i_delay;
 498     double d_amplitude_factor;
 499
 500     /* out buffer characterisitcs */
 501     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 502     p_out_buf->i_buffer = p_in_buf->i_buffer * i_output_nb / i_input_nb;
 503     p_out = p_out_buf->p_buffer;
 504     i_out_size = p_out_buf->i_buffer;
 505
 506     /* Slide the overflow buffer */
 507     p_overflow = p_sys->p_overflow_buffer;
 508     i_overflow_size = p_sys->i_overflow_buffer_size;
 509
 510     if ( i_out_size > i_overflow_size )
 511         memcpy( p_out, p_overflow, i_overflow_size );
 512     else
 513         memcpy( p_out, p_overflow, i_out_size );
 514
 515     p_slide = p_sys->p_overflow_buffer;
 516     while( p_slide < p_overflow + i_overflow_size )
 517     {
 518         if( p_slide + i_out_size < p_overflow + i_overflow_size )
 519         {
 520             memset( p_slide, 0, i_out_size );
 521             if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
 522                 memcpy( p_slide, p_slide + i_out_size, i_out_size );
 523             else
 524                 memcpy( p_slide, p_slide + i_out_size,
 525                         p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
 526         }
 527         else
 528         {
 529             memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
 530         }
 531         p_slide += i_out_size;
 532     }
 533
 534     /* apply the atomic operations */
 535     for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 536     {
 537         /* shorter variable names */
 538         i_source_channel_offset
 539             = p_sys->p_atomic_operations[i].i_source_channel_offset;
 540         i_dest_channel_offset
 541             = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 542         i_delay = p_sys->p_atomic_operations[i].i_delay;
 543         d_amplitude_factor
 544             = p_sys->p_atomic_operations[i].d_amplitude_factor;
 545
 546         if( p_out_buf->i_nb_samples > i_delay )
 547         {
 548             /* current buffer coefficients */
 549             for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 550             {
 551                 ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 552                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 553                        * d_amplitude_factor;
 554             }
 555
 556             /* overflow buffer coefficients */
 557             for( j = 0; j < i_delay; j++ )
 558             {
 559                 ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 560                     += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 561                        * i_input_nb + i_source_channel_offset ]
 562                        * d_amplitude_factor;
 563             }
 564         }
 565         else
 566         {
 567             /* overflow buffer coefficients only */
 568             for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 569             {
 570                 ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 571                                         * i_output_nb + i_dest_channel_offset ]
 572                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 573                        * d_amplitude_factor;
 574             }
 575         }
 576     }
 577 }
 578
 579 /* Simple stereo to mono mixing. */
 580 static unsigned int mono( filter_t *p_filter,
 581                           block_t *p_output, block_t *p_input )
 582 {
 583     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 584     int16_t *p_in, *p_out;
 585     unsigned int n = 0, r = 0;
 586
 587     p_in = (int16_t *) p_input->p_buffer;
 588     p_out = (int16_t *) p_output->p_buffer;
 589
 590     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
 591     {
 592         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
 593         r++;
 594         n += 2;
 595     }
 596     return r;
 597 }
 598
 599 /* Simple stereo to mono mixing. */
 600 static unsigned int stereo_to_mono( filter_t *p_filter,
 601                                     block_t *p_output, block_t *p_input )
 602 {
 603     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 604     int16_t *p_in, *p_out;
 605     unsigned int n;
 606
 607     p_in = (int16_t *) p_input->p_buffer;
 608     p_out = (int16_t *) p_output->p_buffer;
 609
 610     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
 611     {
 612         /* Fake real mono. */
 613         if( p_sys->i_channel_selected == -1)
 614         {
 615             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
 616             n++;
 617         }
 618         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
 619         {
 620             p_out[n] = p_out[n+1] = p_in[n];
 621         }
 622     }
 623     return n;
 624 }