Prefer mono over trivial (in case trivial somehow wants to work).
[vlc.git] / modules / audio_filter / channel_mixer / mono.c
1 /*****************************************************************************
2  * mono.c : stereo2mono downmixsimple channel mixer plug-in
3  *****************************************************************************
4  * Copyright (C) 2006 M2X
5  * $Id$
6  *
7  * Authors: Jean-Paul Saman <jpsaman at m2x dot nl>
8  *
9  * This program is free software; you can redistribute it and/or modify
10  * it under the terms of the GNU General Public License as published by
11  * the Free Software Foundation; either version 2 of the License, or
12  * (at your option) any later version.
13  *
14  * This program is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
17  * GNU General Public License for more details.
18  *
19  * You should have received a copy of the GNU General Public License
20  * along with this program; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22  *****************************************************************************/
23
24 /*****************************************************************************
25  * Preamble
26  *****************************************************************************/
27 #ifdef HAVE_CONFIG_H
28 # include "config.h"
29 #endif
30
31 #include <math.h>                                        /* sqrt */
32
33 #ifdef HAVE_STDINT_H
34 #   include <stdint.h>                                         /* int16_t .. */
35 #elif defined(HAVE_INTTYPES_H)
36 #   include <inttypes.h>                                       /* int16_t .. */
37 #endif
38
39 #ifdef HAVE_UNISTD_H
40 #   include <unistd.h>
41 #endif
42
43 #include <vlc_common.h>
44 #include <vlc_plugin.h>
45 #include <vlc_es.h>
46 #include <vlc_block.h>
47 #include <vlc_filter.h>
48 #include <vlc_aout.h>
49
50 /*****************************************************************************
51  * Local prototypes
52  *****************************************************************************/
53 static int  OpenFilter    ( vlc_object_t * );
54 static void CloseFilter   ( vlc_object_t * );
55
56 static block_t *Convert( filter_t *p_filter, block_t *p_block );
57
58 static unsigned int stereo_to_mono( aout_filter_t *, aout_buffer_t *,
59                                     aout_buffer_t * );
60 static unsigned int mono( aout_filter_t *, aout_buffer_t *, aout_buffer_t * );
61 static void stereo2mono_downmix( aout_filter_t *, aout_buffer_t *,
62                                  aout_buffer_t * );
63
64 /*****************************************************************************
65  * Local structures
66  *****************************************************************************/
67 struct atomic_operation_t
68 {
69     int i_source_channel_offset;
70     int i_dest_channel_offset;
71     unsigned int i_delay;/* in sample unit */
72     double d_amplitude_factor;
73 };
74
75 struct filter_sys_t
76 {
77     bool b_downmix;
78
79     unsigned int i_nb_channels; /* number of int16_t per sample */
80     int i_channel_selected;
81     int i_bitspersample;
82
83     size_t i_overflow_buffer_size;/* in bytes */
84     uint8_t * p_overflow_buffer;
85     unsigned int i_nb_atomic_operations;
86     struct atomic_operation_t * p_atomic_operations;
87 };
88
89 #define MONO_DOWNMIX_TEXT N_("Use downmix algorithm")
90 #define MONO_DOWNMIX_LONGTEXT N_("This option selects a stereo to mono " \
91     "downmix algorithm that is used in the headphone channel mixer. It " \
92     "gives the effect of standing in a room full of speakers." )
93
94 #define MONO_CHANNEL_TEXT N_("Select channel to keep")
95 #define MONO_CHANNEL_LONGTEXT N_("This option silences all other channels " \
96     "except the selected channel. Choose one from (0=left, 1=right, " \
97     "2=rear left, 3=rear right, 4=center, 5=left front)")
98
99 static const int pi_pos_values[] = { 0, 1, 2, 4, 8, 5 };
100 static const char *const ppsz_pos_descriptions[] =
101 { N_("Left"), N_("Right"), N_("Left rear"), N_("Right rear"), N_("Center"),
102   N_("Left front") };
103
104 /* our internal channel order (WG-4 order) */
105 static const uint32_t pi_channels_out[] =
106 { AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
107   AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
108
109 #define MONO_CFG "sout-mono-"
110 /*****************************************************************************
111  * Module descriptor
112  *****************************************************************************/
113 vlc_module_begin();
114     set_description( N_("Audio filter for stereo to mono conversion") );
115     set_capability( "audio filter2", 2 );
116
117     add_bool( MONO_CFG "downmix", true, NULL, MONO_DOWNMIX_TEXT,
118               MONO_DOWNMIX_LONGTEXT, false );
119     add_integer( MONO_CFG "channel", -1, NULL, MONO_CHANNEL_TEXT,
120         MONO_CHANNEL_LONGTEXT, false );
121         change_integer_list( pi_pos_values, ppsz_pos_descriptions, 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( config_GetInt( p_this, "headphone-compensate" ) )
223     {
224         /* minimal distance to any speaker */
225         if( i_physical_channels & AOUT_CHAN_REARCENTER )
226         {
227             d_min = d_z_rear;
228         }
229         else
230         {
231             d_min = d_z;
232         }
233     }
234
235     /* Number of elementary operations */
236     p_data->i_nb_atomic_operations = i_nb_channels * 2;
237     if( i_physical_channels & AOUT_CHAN_CENTER )
238     {
239         p_data->i_nb_atomic_operations += 2;
240     }
241     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
242             * p_data->i_nb_atomic_operations );
243     if( p_data->p_atomic_operations == NULL )
244         return -1;
245
246     /* For each virtual speaker, computes elementary wave propagation time
247      * to each ear */
248     i_next_atomic_operation = 0;
249     i_source_channel_offset = 0;
250     if( i_physical_channels & AOUT_CHAN_LEFT )
251     {
252         ComputeChannelOperations( p_data , i_rate
253                 , i_next_atomic_operation , i_source_channel_offset
254                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
255         i_next_atomic_operation += 2;
256         i_source_channel_offset++;
257     }
258     if( i_physical_channels & AOUT_CHAN_RIGHT )
259     {
260         ComputeChannelOperations( p_data , i_rate
261                 , i_next_atomic_operation , i_source_channel_offset
262                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
263         i_next_atomic_operation += 2;
264         i_source_channel_offset++;
265     }
266     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
267     {
268         ComputeChannelOperations( p_data , i_rate
269                 , i_next_atomic_operation , i_source_channel_offset
270                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
271         i_next_atomic_operation += 2;
272         i_source_channel_offset++;
273     }
274     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
275     {
276         ComputeChannelOperations( p_data , i_rate
277                 , i_next_atomic_operation , i_source_channel_offset
278                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
279         i_next_atomic_operation += 2;
280         i_source_channel_offset++;
281     }
282     if( i_physical_channels & AOUT_CHAN_REARLEFT )
283     {
284         ComputeChannelOperations( p_data , i_rate
285                 , i_next_atomic_operation , i_source_channel_offset
286                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
287         i_next_atomic_operation += 2;
288         i_source_channel_offset++;
289     }
290     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
291     {
292         ComputeChannelOperations( p_data , i_rate
293                 , i_next_atomic_operation , i_source_channel_offset
294                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
295         i_next_atomic_operation += 2;
296         i_source_channel_offset++;
297     }
298     if( i_physical_channels & AOUT_CHAN_REARCENTER )
299     {
300         ComputeChannelOperations( p_data , i_rate
301                 , i_next_atomic_operation , i_source_channel_offset
302                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
303         i_next_atomic_operation += 2;
304         i_source_channel_offset++;
305     }
306     if( i_physical_channels & AOUT_CHAN_CENTER )
307     {
308         /* having two center channels increases the spatialization effect */
309         ComputeChannelOperations( p_data , i_rate
310                 , i_next_atomic_operation , i_source_channel_offset
311                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
312         i_next_atomic_operation += 2;
313         ComputeChannelOperations( p_data , i_rate
314                 , i_next_atomic_operation , i_source_channel_offset
315                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
316         i_next_atomic_operation += 2;
317         i_source_channel_offset++;
318     }
319     if( i_physical_channels & AOUT_CHAN_LFE )
320     {
321         ComputeChannelOperations( p_data , i_rate
322                 , i_next_atomic_operation , i_source_channel_offset
323                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
324         i_next_atomic_operation += 2;
325         i_source_channel_offset++;
326     }
327
328     /* Initialize the overflow buffer
329      * we need it because the process induce a delay in the samples */
330     p_data->i_overflow_buffer_size = 0;
331     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
332     {
333         if( p_data->i_overflow_buffer_size
334                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
335         {
336             p_data->i_overflow_buffer_size
337                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
338         }
339     }
340     p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
341     if( p_data->p_overflow_buffer == NULL )
342     {
343         free( p_data->p_atomic_operations );
344         return -1;
345     }
346     memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
347
348     /* end */
349     return 0;
350 }
351
352 /*****************************************************************************
353  * OpenFilter
354  *****************************************************************************/
355 static int OpenFilter( vlc_object_t *p_this )
356 {
357     filter_t * p_filter = (filter_t *)p_this;
358     filter_sys_t *p_sys = NULL;
359
360     if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
361     {
362         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
363         return VLC_EGENERIC;
364     }
365
366     if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
367         (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
368     {
369         msg_Err( p_this, "filter discarded (invalid format)" );
370         return -1;
371     }
372
373     if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
374         (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
375         (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
376         (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
377         (p_filter->fmt_in.audio.i_bitspersample !=
378                                     p_filter->fmt_out.audio.i_bitspersample))
379     {
380         msg_Err( p_this, "couldn't load mono filter" );
381         return VLC_EGENERIC;
382     }
383
384     /* Allocate the memory needed to store the module's structure */
385     p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
386     if( p_sys == NULL )
387         return VLC_EGENERIC;
388
389     var_Create( p_this, MONO_CFG "downmix",
390                 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
391     p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
392
393     var_Create( p_this, MONO_CFG "channel",
394                 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
395     p_sys->i_channel_selected =
396             (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
397
398     if( p_sys->b_downmix )
399     {
400         msg_Dbg( p_this, "using stereo to mono downmix" );
401         p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
402         p_filter->fmt_out.audio.i_channels = 1;
403     }
404     else
405     {
406         msg_Dbg( p_this, "using pseudo mono" );
407         p_filter->fmt_out.audio.i_physical_channels =
408                             (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
409         p_filter->fmt_out.audio.i_channels = 2;
410     }
411
412     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
413     p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
414
415     p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
416     p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
417
418     p_sys->i_overflow_buffer_size = 0;
419     p_sys->p_overflow_buffer = NULL;
420     p_sys->i_nb_atomic_operations = 0;
421     p_sys->p_atomic_operations = NULL;
422
423     if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
424               aout_FormatNbChannels( &p_filter->fmt_in.audio ),
425               p_filter->fmt_in.audio.i_physical_channels,
426               p_filter->fmt_in.audio.i_rate ) < 0 )
427     {
428         var_Destroy( p_this, MONO_CFG "channel" );
429         var_Destroy( p_this, MONO_CFG "downmix" );
430         free( p_sys );
431         return VLC_EGENERIC;
432     }
433
434     p_filter->pf_audio_filter = Convert;
435
436     msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
437              (char *)&p_filter->fmt_in.i_codec,
438              (char *)&p_filter->fmt_out.i_codec,
439              p_filter->fmt_in.audio.i_physical_channels,
440              p_filter->fmt_out.audio.i_physical_channels,
441              p_filter->fmt_in.audio.i_bitspersample,
442              p_filter->fmt_out.audio.i_bitspersample );
443
444     return VLC_SUCCESS;
445 }
446
447 /*****************************************************************************
448  * CloseFilter
449  *****************************************************************************/
450 static void CloseFilter( vlc_object_t *p_this)
451 {
452     filter_t *p_filter = (filter_t *) p_this;
453     filter_sys_t *p_sys = p_filter->p_sys;
454
455     var_Destroy( p_this, MONO_CFG "channel" );
456     var_Destroy( p_this, MONO_CFG "downmix" );
457     free( p_sys->p_atomic_operations );
458     free( p_sys->p_overflow_buffer );
459     free( p_sys );
460 }
461
462 /*****************************************************************************
463  * Convert
464  *****************************************************************************/
465 static block_t *Convert( filter_t *p_filter, block_t *p_block )
466 {
467     aout_filter_t aout_filter;
468     aout_buffer_t in_buf, out_buf;
469     block_t *p_out = NULL;
470     unsigned int i_samples;
471     int i_out_size;
472
473     if( !p_block || !p_block->i_samples )
474     {
475         if( p_block )
476             block_Release( p_block );
477         return NULL;
478     }
479
480     i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
481                  aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
482
483     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
484     if( !p_out )
485     {
486         msg_Warn( p_filter, "can't get output buffer" );
487         block_Release( p_block );
488         return NULL;
489     }
490     p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
491                        aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
492     p_out->i_dts = p_block->i_dts;
493     p_out->i_pts = p_block->i_pts;
494     p_out->i_length = p_block->i_length;
495
496     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
497     aout_filter.input = p_filter->fmt_in.audio;
498     aout_filter.input.i_format = p_filter->fmt_in.i_codec;
499     aout_filter.output = p_filter->fmt_out.audio;
500     aout_filter.output.i_format = p_filter->fmt_out.i_codec;
501
502     in_buf.p_buffer = p_block->p_buffer;
503     in_buf.i_nb_bytes = p_block->i_buffer;
504     in_buf.i_nb_samples = p_block->i_samples;
505
506 #if 0
507     unsigned int i_in_size = in_buf.i_nb_samples  * (p_filter->p_sys->i_bitspersample/8) *
508                              aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
509     if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
510     {
511         msg_Err( p_filter, "input buffer is not word aligned" );
512         /* Fix output buffer to be word aligned */
513     }
514 #endif
515
516     out_buf.p_buffer = p_out->p_buffer;
517     out_buf.i_nb_bytes = p_out->i_buffer;
518     out_buf.i_nb_samples = p_out->i_samples;
519
520     memset( p_out->p_buffer, 0, i_out_size );
521     if( p_filter->p_sys->b_downmix )
522     {
523         stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
524         i_samples = mono( &aout_filter, &out_buf, &in_buf );
525     }
526     else
527     {
528         i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
529     }
530
531     p_out->i_buffer = out_buf.i_nb_bytes;
532     p_out->i_samples = out_buf.i_nb_samples;
533
534     block_Release( p_block );
535     return p_out;
536 }
537
538 /* stereo2mono_downmix - stereo channels into one mono channel.
539  * Code taken from modules/audio_filter/channel_mixer/headphone.c
540  * converted from float into int16_t based downmix
541  * Written by Boris Dorès <babal@via.ecp.fr>
542  */
543 static void stereo2mono_downmix( aout_filter_t * p_filter,
544                             aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
545 {
546     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
547
548     int i_input_nb = aout_FormatNbChannels( &p_filter->input );
549     int i_output_nb = aout_FormatNbChannels( &p_filter->output );
550
551     int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
552     uint8_t * p_out;
553     uint8_t * p_overflow;
554     uint8_t * p_slide;
555
556     size_t i_overflow_size;     /* in bytes */
557     size_t i_out_size;          /* in bytes */
558
559     unsigned int i, j;
560
561     int i_source_channel_offset;
562     int i_dest_channel_offset;
563     unsigned int i_delay;
564     double d_amplitude_factor;
565
566     /* out buffer characterisitcs */
567     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
568     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
569     p_out = p_out_buf->p_buffer;
570     i_out_size = p_out_buf->i_nb_bytes;
571
572     if( p_sys != NULL )
573     {
574         /* Slide the overflow buffer */
575         p_overflow = p_sys->p_overflow_buffer;
576         i_overflow_size = p_sys->i_overflow_buffer_size;
577
578         if ( i_out_size > i_overflow_size )
579             memcpy( p_out, p_overflow, i_overflow_size );
580         else
581             memcpy( p_out, p_overflow, i_out_size );
582
583         p_slide = p_sys->p_overflow_buffer;
584         while( p_slide < p_overflow + i_overflow_size )
585         {
586             if( p_slide + i_out_size < p_overflow + i_overflow_size )
587             {
588                 memset( p_slide, 0, i_out_size );
589                 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
590                     memcpy( p_slide, p_slide + i_out_size, i_out_size );
591                 else
592                     memcpy( p_slide, p_slide + i_out_size,
593                             p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
594             }
595             else
596             {
597                 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
598             }
599             p_slide += i_out_size;
600         }
601
602         /* apply the atomic operations */
603         for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
604         {
605             /* shorter variable names */
606             i_source_channel_offset
607                 = p_sys->p_atomic_operations[i].i_source_channel_offset;
608             i_dest_channel_offset
609                 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
610             i_delay = p_sys->p_atomic_operations[i].i_delay;
611             d_amplitude_factor
612                 = p_sys->p_atomic_operations[i].d_amplitude_factor;
613
614             if( p_out_buf->i_nb_samples > i_delay )
615             {
616                 /* current buffer coefficients */
617                 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
618                 {
619                     ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
620                         += p_in[ j * i_input_nb + i_source_channel_offset ]
621                            * d_amplitude_factor;
622                 }
623
624                 /* overflow buffer coefficients */
625                 for( j = 0; j < i_delay; j++ )
626                 {
627                     ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
628                         += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
629                            * i_input_nb + i_source_channel_offset ]
630                            * d_amplitude_factor;
631                 }
632             }
633             else
634             {
635                 /* overflow buffer coefficients only */
636                 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
637                 {
638                     ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
639                         * i_output_nb + i_dest_channel_offset ]
640                         += p_in[ j * i_input_nb + i_source_channel_offset ]
641                            * d_amplitude_factor;
642                 }
643             }
644         }
645     }
646     else
647     {
648         memset( p_out, 0, i_out_size );
649     }
650 }
651
652 /* Simple stereo to mono mixing. */
653 static unsigned int mono( aout_filter_t *p_filter,
654                           aout_buffer_t *p_output, aout_buffer_t *p_input )
655 {
656     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
657     int16_t *p_in, *p_out;
658     unsigned int n = 0, r = 0;
659
660     p_in = (int16_t *) p_input->p_buffer;
661     p_out = (int16_t *) p_output->p_buffer;
662
663     while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
664     {
665         p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
666         r++;
667         n += 2;
668     }
669     return r;
670 }
671
672 /* Simple stereo to mono mixing. */
673 static unsigned int stereo_to_mono( aout_filter_t *p_filter,
674                                     aout_buffer_t *p_output, aout_buffer_t *p_input )
675 {
676     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
677     int16_t *p_in, *p_out;
678     unsigned int n;
679
680     p_in = (int16_t *) p_input->p_buffer;
681     p_out = (int16_t *) p_output->p_buffer;
682
683     for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
684     {
685         /* Fake real mono. */
686         if( p_sys->i_channel_selected == -1)
687         {
688             p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
689             n++;
690         }
691         else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
692         {
693             p_out[n] = p_out[n+1] = p_in[n];
694         }
695     }
696     return n;
697 }