mono.c 23.1 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26
/*****************************************************************************
 * mono.c : stereo2mono downmixsimple channel mixer plug-in
 *****************************************************************************
 * Copyright (C) 2006 M2X
 * $Id$
 *
 * Authors: Jean-Paul Saman <jpsaman at m2x dot nl>
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
 *****************************************************************************/

/*****************************************************************************
 * Preamble
 *****************************************************************************/
27 28 29 30
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif

31
#include <math.h>                                        /* sqrt */
32
#include <stdint.h>                                         /* int16_t .. */
33 34 35 36 37

#ifdef HAVE_UNISTD_H
#   include <unistd.h>
#endif

38
#include <vlc_common.h>
39
#include <vlc_plugin.h>
40 41
#include <vlc_block.h>
#include <vlc_filter.h>
Clément Stenac's avatar
Clément Stenac committed
42
#include <vlc_aout.h>
43 44 45 46 47 48 49 50 51

/*****************************************************************************
 * Local prototypes
 *****************************************************************************/
static int  OpenFilter    ( vlc_object_t * );
static void CloseFilter   ( vlc_object_t * );

static block_t *Convert( filter_t *p_filter, block_t *p_block );

52
static unsigned int stereo_to_mono( filter_t *, aout_buffer_t *,
Rafaël Carré's avatar
Rafaël Carré committed
53
                                    aout_buffer_t * );
54 55
static unsigned int mono( filter_t *, aout_buffer_t *, aout_buffer_t * );
static void stereo2mono_downmix( filter_t *, aout_buffer_t *,
Rafaël Carré's avatar
Rafaël Carré committed
56
                                 aout_buffer_t * );
57 58 59 60

/*****************************************************************************
 * Local structures
 *****************************************************************************/
61 62 63 64 65 66 67 68
struct atomic_operation_t
{
    int i_source_channel_offset;
    int i_dest_channel_offset;
    unsigned int i_delay;/* in sample unit */
    double d_amplitude_factor;
};

69 70
struct filter_sys_t
{
71
    bool b_downmix;
72

73
    unsigned int i_nb_channels; /* number of int16_t per sample */
74
    int i_channel_selected;
75
    int i_bitspersample;
76 77

    size_t i_overflow_buffer_size;/* in bytes */
78
    uint8_t * p_overflow_buffer;
79 80
    unsigned int i_nb_atomic_operations;
    struct atomic_operation_t * p_atomic_operations;
81 82
};

Christophe Mutricy's avatar
Typo  
Christophe Mutricy committed
83
#define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
84
#define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
Rémi Denis-Courmont's avatar
Rémi Denis-Courmont committed
85
    "downmix algorithm that is used in the headphone channel mixer. It " \
86 87
    "gives the effect of standing in a room full of speakers." )

88 89
#define MONO_CHANNEL_TEXT N_("Select channel to keep")
#define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
90
    "except the selected channel. Choose one from (0=left, 1=right, " \
Jean-Paul Saman's avatar
Jean-Paul Saman committed
91
    "2=rear left, 3=rear right, 4=center, 5=left front)")
92

93
static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
94
static const char *const ppsz_pos_descriptions[] =
95 96 97 98 99 100 101 102
{ N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
  N_("Left front") };

/* our internal channel order (WG-4 order) */
static const uint32_t pi_channels_out[] =
{ AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
  AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };

Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
103
#define MONO_CFG "sout-mono-"
104 105 106
/*****************************************************************************
 * Module descriptor
 *****************************************************************************/
107 108
vlc_module_begin ()
    set_description( N_("Audio filter for stereo to mono conversion") )
109
    set_capability( "audio filter", 0 )
110
    set_category( CAT_AUDIO )
111
    set_subcategory( SUBCAT_AUDIO_AFILTER )
112 113
    set_callbacks( OpenFilter, CloseFilter )
    set_shortname( "Mono" )
114

115
    add_bool( MONO_CFG "downmix", true, MONO_DOWNMIX_TEXT,
Rémi Denis-Courmont's avatar
Rémi Denis-Courmont committed
116
              MONO_DOWNMIX_LONGTEXT, false )
117
    add_integer( MONO_CFG "channel", -1, MONO_CHANNEL_TEXT,
Rémi Denis-Courmont's avatar
Rémi Denis-Courmont committed
118
        MONO_CHANNEL_LONGTEXT, false )
119
        change_integer_list( pi_pos_values, ppsz_pos_descriptions )
120

121
vlc_module_end ()
122

123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207
/* Init() and ComputeChannelOperations() -
 * Code taken from modules/audio_filter/channel_mixer/headphone.c
 * converted from float into int16_t based downmix
 * Written by Boris Dorès <babal@via.ecp.fr>
 */

/*****************************************************************************
 * Init: initialize internal data structures
 * and computes the needed atomic operations
 *****************************************************************************/
/* x and z represent the coordinates of the virtual speaker
 *  relatively to the center of the listener's head, measured in meters :
 *
 *  left              right
 *Z
 *-
 *a          head
 *x
 *i
 *s
 *  rear left    rear right
 *
 *          x-axis
 *  */
static void ComputeChannelOperations( struct filter_sys_t * p_data,
        unsigned int i_rate, unsigned int i_next_atomic_operation,
        int i_source_channel_offset, double d_x, double d_z,
        double d_compensation_length, double d_channel_amplitude_factor )
{
    double d_c = 340; /*sound celerity (unit: m/s)*/
    double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;

    /* Left ear */
    p_data->p_atomic_operations[i_next_atomic_operation]
        .i_source_channel_offset = i_source_channel_offset;
    p_data->p_atomic_operations[i_next_atomic_operation]
        .i_dest_channel_offset = 0;/* left */
    p_data->p_atomic_operations[i_next_atomic_operation]
        .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
                          / d_c * i_rate - d_compensation_delay );
    if( d_x < 0 )
    {
        p_data->p_atomic_operations[i_next_atomic_operation]
            .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
    }
    else if( d_x > 0 )
    {
        p_data->p_atomic_operations[i_next_atomic_operation]
            .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
    }
    else
    {
        p_data->p_atomic_operations[i_next_atomic_operation]
            .d_amplitude_factor = d_channel_amplitude_factor / 2;
    }

    /* Right ear */
    p_data->p_atomic_operations[i_next_atomic_operation + 1]
        .i_source_channel_offset = i_source_channel_offset;
    p_data->p_atomic_operations[i_next_atomic_operation + 1]
        .i_dest_channel_offset = 1;/* right */
    p_data->p_atomic_operations[i_next_atomic_operation + 1]
        .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
                          / d_c * i_rate - d_compensation_delay );
    if( d_x < 0 )
    {
        p_data->p_atomic_operations[i_next_atomic_operation + 1]
            .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
    }
    else if( d_x > 0 )
    {
        p_data->p_atomic_operations[i_next_atomic_operation + 1]
            .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
    }
    else
    {
        p_data->p_atomic_operations[i_next_atomic_operation + 1]
            .d_amplitude_factor = d_channel_amplitude_factor / 2;
    }
}

static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data,
                 unsigned int i_nb_channels, uint32_t i_physical_channels,
                 unsigned int i_rate )
{
208
    double d_x = var_InheritInteger( p_this, "headphone-dim" );
209 210 211 212 213 214 215
    double d_z = d_x;
    double d_z_rear = -d_x/3;
    double d_min = 0;
    unsigned int i_next_atomic_operation;
    int i_source_channel_offset;
    unsigned int i;

216
    if( var_InheritBool( p_this, "headphone-compensate" ) )
217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334
    {
        /* minimal distance to any speaker */
        if( i_physical_channels & AOUT_CHAN_REARCENTER )
        {
            d_min = d_z_rear;
        }
        else
        {
            d_min = d_z;
        }
    }

    /* Number of elementary operations */
    p_data->i_nb_atomic_operations = i_nb_channels * 2;
    if( i_physical_channels & AOUT_CHAN_CENTER )
    {
        p_data->i_nb_atomic_operations += 2;
    }
    p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
            * p_data->i_nb_atomic_operations );
    if( p_data->p_atomic_operations == NULL )
        return -1;

    /* For each virtual speaker, computes elementary wave propagation time
     * to each ear */
    i_next_atomic_operation = 0;
    i_source_channel_offset = 0;
    if( i_physical_channels & AOUT_CHAN_LEFT )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , -d_x , d_z , d_min , 2.0 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_RIGHT )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , d_x , d_z , d_min , 2.0 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , -d_x , 0 , d_min , 1.5 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , d_x , 0 , d_min , 1.5 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_REARLEFT )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_REARRIGHT )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_REARCENTER )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , 0 , -d_z , d_min , 1.5 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_CENTER )
    {
        /* having two center channels increases the spatialization effect */
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
        i_next_atomic_operation += 2;
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }
    if( i_physical_channels & AOUT_CHAN_LFE )
    {
        ComputeChannelOperations( p_data , i_rate
                , i_next_atomic_operation , i_source_channel_offset
                , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
        i_next_atomic_operation += 2;
        i_source_channel_offset++;
    }

    /* Initialize the overflow buffer
     * we need it because the process induce a delay in the samples */
    p_data->i_overflow_buffer_size = 0;
    for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
    {
        if( p_data->i_overflow_buffer_size
                < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
        {
            p_data->i_overflow_buffer_size
                = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
        }
    }
    p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
335 336 337
    if( p_data->p_overflow_buffer == NULL )
    {
        free( p_data->p_atomic_operations );
338
        return -1;
339
    }
340
    memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
341 342 343 344 345

    /* end */
    return 0;
}

346 347 348 349 350 351 352 353 354 355
/*****************************************************************************
 * OpenFilter
 *****************************************************************************/
static int OpenFilter( vlc_object_t *p_this )
{
    filter_t * p_filter = (filter_t *)p_this;
    filter_sys_t *p_sys = NULL;

    if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
    {
356
        /*msg_Dbg( p_filter, "filter discarded (incompatible format)" );*/
357 358 359
        return VLC_EGENERIC;
    }

360 361
    if( (p_filter->fmt_in.i_codec != VLC_CODEC_S16N) ||
        (p_filter->fmt_out.i_codec != VLC_CODEC_S16N) )
362
    {
363 364
        /*msg_Err( p_this, "filter discarded (invalid format)" );*/
        return VLC_EGENERIC;
365 366 367 368
    }

    if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
        (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
369 370
        (p_filter->fmt_in.audio.i_format != VLC_CODEC_S16N) &&
        (p_filter->fmt_out.audio.i_format != VLC_CODEC_S16N) &&
Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
371 372
        (p_filter->fmt_in.audio.i_bitspersample !=
                                    p_filter->fmt_out.audio.i_bitspersample))
373
    {
374
        /*msg_Err( p_this, "couldn't load mono filter" );*/
375 376 377 378 379 380 381 382
        return VLC_EGENERIC;
    }

    /* Allocate the memory needed to store the module's structure */
    p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
    if( p_sys == NULL )
        return VLC_EGENERIC;

383
    p_sys->b_downmix = var_CreateGetBool( p_this, MONO_CFG "downmix" );
384
    p_sys->i_channel_selected =
385
            (unsigned int) var_CreateGetInteger( p_this, MONO_CFG "channel" );
386

387 388
    if( p_sys->b_downmix )
    {
Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
389
        msg_Dbg( p_this, "using stereo to mono downmix" );
390
        p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
391
        p_filter->fmt_out.audio.i_channels = 1;
392 393 394
    }
    else
    {
Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
395
        msg_Dbg( p_this, "using pseudo mono" );
396
        p_filter->fmt_out.audio.i_physical_channels =
397
                            (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
398
        p_filter->fmt_out.audio.i_channels = 2;
399 400
    }

401 402 403
    p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
    p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;

404
    p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
405 406
    p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;

407 408 409 410 411 412 413 414 415 416
    p_sys->i_overflow_buffer_size = 0;
    p_sys->p_overflow_buffer = NULL;
    p_sys->i_nb_atomic_operations = 0;
    p_sys->p_atomic_operations = NULL;

    if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
              aout_FormatNbChannels( &p_filter->fmt_in.audio ),
              p_filter->fmt_in.audio.i_physical_channels,
              p_filter->fmt_in.audio.i_rate ) < 0 )
    {
417 418 419
        var_Destroy( p_this, MONO_CFG "channel" );
        var_Destroy( p_this, MONO_CFG "downmix" );
        free( p_sys );
420 421 422
        return VLC_EGENERIC;
    }

423 424
    p_filter->pf_audio_filter = Convert;

425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443
    msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
             (char *)&p_filter->fmt_in.i_codec,
             (char *)&p_filter->fmt_out.i_codec,
             p_filter->fmt_in.audio.i_physical_channels,
             p_filter->fmt_out.audio.i_physical_channels,
             p_filter->fmt_in.audio.i_bitspersample,
             p_filter->fmt_out.audio.i_bitspersample );

    return VLC_SUCCESS;
}

/*****************************************************************************
 * CloseFilter
 *****************************************************************************/
static void CloseFilter( vlc_object_t *p_this)
{
    filter_t *p_filter = (filter_t *) p_this;
    filter_sys_t *p_sys = p_filter->p_sys;

Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
444 445
    var_Destroy( p_this, MONO_CFG "channel" );
    var_Destroy( p_this, MONO_CFG "downmix" );
446 447
    free( p_sys->p_atomic_operations );
    free( p_sys->p_overflow_buffer );
448 449 450 451 452 453 454 455
    free( p_sys );
}

/*****************************************************************************
 * Convert
 *****************************************************************************/
static block_t *Convert( filter_t *p_filter, block_t *p_block )
{
456
    block_t *p_out;
457
    int i_out_size;
458

459
    if( !p_block || !p_block->i_nb_samples )
460 461
    {
        if( p_block )
462
            block_Release( p_block );
463 464 465
        return NULL;
    }

466
    i_out_size = p_block->i_nb_samples * p_filter->p_sys->i_bitspersample/8 *
467
                 aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
468

469
    p_out = block_Alloc( i_out_size );
470 471 472
    if( !p_out )
    {
        msg_Warn( p_filter, "can't get output buffer" );
473
        block_Release( p_block );
474 475
        return NULL;
    }
476 477
    p_out->i_nb_samples =
                  (p_block->i_nb_samples / p_filter->p_sys->i_nb_channels) *
478
                       aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
479 480

#if 0
481 482
    unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
                             aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
Rémi Denis-Courmont's avatar
Rémi Denis-Courmont committed
483
    if( (in_buf.i_buffer != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
484
    {
485 486
        msg_Err( p_filter, "input buffer is not word aligned" );
        /* Fix output buffer to be word aligned */
487 488 489
    }
#endif

Jean-Paul Saman's avatar
Cleanup  
Jean-Paul Saman committed
490
    memset( p_out->p_buffer, 0, i_out_size );
491 492
    if( p_filter->p_sys->b_downmix )
    {
493 494
        stereo2mono_downmix( p_filter, p_block, p_out );
        mono( p_filter, p_out, p_block );
495 496 497
    }
    else
    {
498
        stereo_to_mono( p_filter, p_out, p_block );
499
    }
500

501
    block_Release( p_block );
502 503 504
    return p_out;
}

505 506 507 508
/* stereo2mono_downmix - stereo channels into one mono channel.
 * Code taken from modules/audio_filter/channel_mixer/headphone.c
 * converted from float into int16_t based downmix
 * Written by Boris Dorès <babal@via.ecp.fr>
509
 */
510
static void stereo2mono_downmix( filter_t * p_filter,
511 512 513 514
                            aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;

515 516
    int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio );
    int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio );
517 518

    int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
519 520 521
    uint8_t * p_out;
    uint8_t * p_overflow;
    uint8_t * p_slide;
522 523 524 525 526 527 528 529 530 531 532 533 534

    size_t i_overflow_size;     /* in bytes */
    size_t i_out_size;          /* in bytes */

    unsigned int i, j;

    int i_source_channel_offset;
    int i_dest_channel_offset;
    unsigned int i_delay;
    double d_amplitude_factor;

    /* out buffer characterisitcs */
    p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
Rémi Denis-Courmont's avatar
Rémi Denis-Courmont committed
535
    p_out_buf->i_buffer = p_in_buf->i_buffer * i_output_nb / i_input_nb;
536
    p_out = p_out_buf->p_buffer;
Rémi Denis-Courmont's avatar
Rémi Denis-Courmont committed
537
    i_out_size = p_out_buf->i_buffer;
538

539 540 541 542 543 544 545 546
    /* Slide the overflow buffer */
    p_overflow = p_sys->p_overflow_buffer;
    i_overflow_size = p_sys->i_overflow_buffer_size;

    if ( i_out_size > i_overflow_size )
        memcpy( p_out, p_overflow, i_overflow_size );
    else
        memcpy( p_out, p_overflow, i_out_size );
547

548 549 550 551 552 553 554 555 556 557 558 559
    p_slide = p_sys->p_overflow_buffer;
    while( p_slide < p_overflow + i_overflow_size )
    {
        if( p_slide + i_out_size < p_overflow + i_overflow_size )
        {
            memset( p_slide, 0, i_out_size );
            if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
                memcpy( p_slide, p_slide + i_out_size, i_out_size );
            else
                memcpy( p_slide, p_slide + i_out_size,
                        p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
        }
560
        else
561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576 577
        {
            memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
        }
        p_slide += i_out_size;
    }

    /* apply the atomic operations */
    for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
    {
        /* shorter variable names */
        i_source_channel_offset
            = p_sys->p_atomic_operations[i].i_source_channel_offset;
        i_dest_channel_offset
            = p_sys->p_atomic_operations[i].i_dest_channel_offset;
        i_delay = p_sys->p_atomic_operations[i].i_delay;
        d_amplitude_factor
            = p_sys->p_atomic_operations[i].d_amplitude_factor;
578

579
        if( p_out_buf->i_nb_samples > i_delay )
580
        {
581 582
            /* current buffer coefficients */
            for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
583
            {
584 585 586
                ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
                    += p_in[ j * i_input_nb + i_source_channel_offset ]
                       * d_amplitude_factor;
587
            }
588 589 590

            /* overflow buffer coefficients */
            for( j = 0; j < i_delay; j++ )
591
            {
592 593 594 595
                ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
                    += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
                       * i_input_nb + i_source_channel_offset ]
                       * d_amplitude_factor;
596 597
            }
        }
598
        else
599
        {
600 601
            /* overflow buffer coefficients only */
            for( j = 0; j < p_out_buf->i_nb_samples; j++ )
602
            {
603 604 605 606
                ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
                                        * i_output_nb + i_dest_channel_offset ]
                    += p_in[ j * i_input_nb + i_source_channel_offset ]
                       * d_amplitude_factor;
607 608 609 610 611
            }
        }
    }
}

612
/* Simple stereo to mono mixing. */
613
static unsigned int mono( filter_t *p_filter,
614 615 616 617 618 619 620 621 622 623 624 625 626 627 628 629 630 631
                          aout_buffer_t *p_output, aout_buffer_t *p_input )
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    int16_t *p_in, *p_out;
    unsigned int n = 0, r = 0;

    p_in = (int16_t *) p_input->p_buffer;
    p_out = (int16_t *) p_output->p_buffer;

    while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
    {
        p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
        r++;
        n += 2;
    }
    return r;
}

632
/* Simple stereo to mono mixing. */
633
static unsigned int stereo_to_mono( filter_t *p_filter,
634
                                    aout_buffer_t *p_output, aout_buffer_t *p_input )
635 636 637
{
    filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
    int16_t *p_in, *p_out;
638
    unsigned int n;
639

640 641
    p_in = (int16_t *) p_input->p_buffer;
    p_out = (int16_t *) p_output->p_buffer;
642

643
    for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
644
    {
645
        /* Fake real mono. */
646 647 648 649 650
        if( p_sys->i_channel_selected == -1)
        {
            p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
            n++;
        }
651
        else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
652 653 654
        {
            p_out[n] = p_out[n+1] = p_in[n];
        }
655 656 657
    }
    return n;
}