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