1 /*****************************************************************************
2 * mono.c : stereo2mono downmixsimple channel mixer plug-in
3 *****************************************************************************
4 * Copyright (C) 2006 M2X
7 * Authors: Jean-Paul Saman <jpsaman at m2x dot nl>
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.
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.
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 *****************************************************************************/
24 /*****************************************************************************
26 *****************************************************************************/
27 #include <math.h> /* sqrt */
30 # include <stdint.h> /* int16_t .. */
31 #elif defined(HAVE_INTTYPES_H)
32 # include <inttypes.h> /* int16_t .. */
44 #include <vlc_plugin.h>
46 #include <vlc_block.h>
47 #include <vlc_filter.h>
50 /*****************************************************************************
52 *****************************************************************************/
53 static int OpenFilter ( vlc_object_t * );
54 static void CloseFilter ( vlc_object_t * );
56 static block_t *Convert( filter_t *p_filter, block_t *p_block );
58 static unsigned int stereo_to_mono( aout_filter_t *, 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 *,
64 /*****************************************************************************
66 *****************************************************************************/
67 struct atomic_operation_t
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;
79 unsigned int i_nb_channels; /* number of int16_t per sample */
80 int i_channel_selected;
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;
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." )
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)")
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"),
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 };
109 #define MONO_CFG "sout-mono-"
110 /*****************************************************************************
112 *****************************************************************************/
114 set_description( N_("Audio filter for stereo to mono conversion") );
115 set_capability( "audio filter2", 0 );
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 );
123 set_category( CAT_AUDIO );
124 set_subcategory( SUBCAT_AUDIO_MISC );
125 set_callbacks( OpenFilter, CloseFilter );
126 set_shortname( "Mono" );
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>
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 :
149 * rear left rear right
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 )
158 double d_c = 340; /*sound celerity (unit: m/s)*/
159 double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
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 );
171 p_data->p_atomic_operations[i_next_atomic_operation]
172 .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
176 p_data->p_atomic_operations[i_next_atomic_operation]
177 .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
181 p_data->p_atomic_operations[i_next_atomic_operation]
182 .d_amplitude_factor = d_channel_amplitude_factor / 2;
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 );
195 p_data->p_atomic_operations[i_next_atomic_operation + 1]
196 .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
200 p_data->p_atomic_operations[i_next_atomic_operation + 1]
201 .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
205 p_data->p_atomic_operations[i_next_atomic_operation + 1]
206 .d_amplitude_factor = d_channel_amplitude_factor / 2;
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 )
214 double d_x = config_GetInt( p_this, "headphone-dim" );
216 double d_z_rear = -d_x/3;
218 unsigned int i_next_atomic_operation;
219 int i_source_channel_offset;
224 msg_Dbg( p_this, "passing a null pointer as argument" );
228 if( config_GetInt( p_this, "headphone-compensate" ) )
230 /* minimal distance to any speaker */
231 if( i_physical_channels & AOUT_CHAN_REARCENTER )
241 /* Number of elementary operations */
242 p_data->i_nb_atomic_operations = i_nb_channels * 2;
243 if( i_physical_channels & AOUT_CHAN_CENTER )
245 p_data->i_nb_atomic_operations += 2;
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 )
251 msg_Err( p_this, "out of memory" );
255 /* For each virtual speaker, computes elementary wave propagation time
257 i_next_atomic_operation = 0;
258 i_source_channel_offset = 0;
259 if( i_physical_channels & AOUT_CHAN_LEFT )
261 ComputeChannelOperations( p_data , i_rate
262 , i_next_atomic_operation , i_source_channel_offset
263 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
264 i_next_atomic_operation += 2;
265 i_source_channel_offset++;
267 if( i_physical_channels & AOUT_CHAN_RIGHT )
269 ComputeChannelOperations( p_data , i_rate
270 , i_next_atomic_operation , i_source_channel_offset
271 , d_x , d_z , d_min , 2.0 / i_nb_channels );
272 i_next_atomic_operation += 2;
273 i_source_channel_offset++;
275 if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
277 ComputeChannelOperations( p_data , i_rate
278 , i_next_atomic_operation , i_source_channel_offset
279 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
280 i_next_atomic_operation += 2;
281 i_source_channel_offset++;
283 if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
285 ComputeChannelOperations( p_data , i_rate
286 , i_next_atomic_operation , i_source_channel_offset
287 , d_x , 0 , d_min , 1.5 / i_nb_channels );
288 i_next_atomic_operation += 2;
289 i_source_channel_offset++;
291 if( i_physical_channels & AOUT_CHAN_REARLEFT )
293 ComputeChannelOperations( p_data , i_rate
294 , i_next_atomic_operation , i_source_channel_offset
295 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
296 i_next_atomic_operation += 2;
297 i_source_channel_offset++;
299 if( i_physical_channels & AOUT_CHAN_REARRIGHT )
301 ComputeChannelOperations( p_data , i_rate
302 , i_next_atomic_operation , i_source_channel_offset
303 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
304 i_next_atomic_operation += 2;
305 i_source_channel_offset++;
307 if( i_physical_channels & AOUT_CHAN_REARCENTER )
309 ComputeChannelOperations( p_data , i_rate
310 , i_next_atomic_operation , i_source_channel_offset
311 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
312 i_next_atomic_operation += 2;
313 i_source_channel_offset++;
315 if( i_physical_channels & AOUT_CHAN_CENTER )
317 /* having two center channels increases the spatialization effect */
318 ComputeChannelOperations( p_data , i_rate
319 , i_next_atomic_operation , i_source_channel_offset
320 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
321 i_next_atomic_operation += 2;
322 ComputeChannelOperations( p_data , i_rate
323 , i_next_atomic_operation , i_source_channel_offset
324 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
325 i_next_atomic_operation += 2;
326 i_source_channel_offset++;
328 if( i_physical_channels & AOUT_CHAN_LFE )
330 ComputeChannelOperations( p_data , i_rate
331 , i_next_atomic_operation , i_source_channel_offset
332 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
333 i_next_atomic_operation += 2;
334 i_source_channel_offset++;
337 /* Initialize the overflow buffer
338 * we need it because the process induce a delay in the samples */
339 p_data->i_overflow_buffer_size = 0;
340 for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
342 if( p_data->i_overflow_buffer_size
343 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
345 p_data->i_overflow_buffer_size
346 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
349 p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
350 if( p_data->p_atomic_operations == NULL )
352 msg_Err( p_this, "out of memory" );
355 memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
361 /*****************************************************************************
363 *****************************************************************************/
364 static int OpenFilter( vlc_object_t *p_this )
366 filter_t * p_filter = (filter_t *)p_this;
367 filter_sys_t *p_sys = NULL;
369 if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
371 msg_Dbg( p_filter, "filter discarded (incompatible format)" );
375 if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
376 (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
378 msg_Err( p_this, "filter discarded (invalid format)" );
382 if( (p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format) &&
383 (p_filter->fmt_in.audio.i_rate != p_filter->fmt_out.audio.i_rate) &&
384 (p_filter->fmt_in.audio.i_format != AOUT_FMT_S16_NE) &&
385 (p_filter->fmt_out.audio.i_format != AOUT_FMT_S16_NE) &&
386 (p_filter->fmt_in.audio.i_bitspersample !=
387 p_filter->fmt_out.audio.i_bitspersample))
389 msg_Err( p_this, "couldn't load mono filter" );
393 /* Allocate the memory needed to store the module's structure */
394 p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
397 msg_Err( p_filter, "out of memory" );
401 var_Create( p_this, MONO_CFG "downmix",
402 VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
403 p_sys->b_downmix = var_GetBool( p_this, MONO_CFG "downmix" );
405 var_Create( p_this, MONO_CFG "channel",
406 VLC_VAR_INTEGER | VLC_VAR_DOINHERIT );
407 p_sys->i_channel_selected =
408 (unsigned int) var_GetInteger( p_this, MONO_CFG "channel" );
410 if( p_sys->b_downmix )
412 msg_Dbg( p_this, "using stereo to mono downmix" );
413 p_filter->fmt_out.audio.i_physical_channels = AOUT_CHAN_CENTER;
414 p_filter->fmt_out.audio.i_channels = 1;
418 msg_Dbg( p_this, "using pseudo mono" );
419 p_filter->fmt_out.audio.i_physical_channels =
420 (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT);
421 p_filter->fmt_out.audio.i_channels = 2;
424 p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
425 p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
427 p_sys->i_nb_channels = aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
428 p_sys->i_bitspersample = p_filter->fmt_out.audio.i_bitspersample;
430 p_sys->i_overflow_buffer_size = 0;
431 p_sys->p_overflow_buffer = NULL;
432 p_sys->i_nb_atomic_operations = 0;
433 p_sys->p_atomic_operations = NULL;
435 if( Init( VLC_OBJECT(p_filter), p_filter->p_sys,
436 aout_FormatNbChannels( &p_filter->fmt_in.audio ),
437 p_filter->fmt_in.audio.i_physical_channels,
438 p_filter->fmt_in.audio.i_rate ) < 0 )
443 p_filter->pf_audio_filter = Convert;
445 msg_Dbg( p_this, "%4.4s->%4.4s, channels %d->%d, bits per sample: %i->%i",
446 (char *)&p_filter->fmt_in.i_codec,
447 (char *)&p_filter->fmt_out.i_codec,
448 p_filter->fmt_in.audio.i_physical_channels,
449 p_filter->fmt_out.audio.i_physical_channels,
450 p_filter->fmt_in.audio.i_bitspersample,
451 p_filter->fmt_out.audio.i_bitspersample );
456 /*****************************************************************************
458 *****************************************************************************/
459 static void CloseFilter( vlc_object_t *p_this)
461 filter_t *p_filter = (filter_t *) p_this;
462 filter_sys_t *p_sys = p_filter->p_sys;
464 var_Destroy( p_this, MONO_CFG "channel" );
465 var_Destroy( p_this, MONO_CFG "downmix" );
469 /*****************************************************************************
471 *****************************************************************************/
472 static block_t *Convert( filter_t *p_filter, block_t *p_block )
474 aout_filter_t aout_filter;
475 aout_buffer_t in_buf, out_buf;
476 block_t *p_out = NULL;
477 unsigned int i_samples;
480 if( !p_block || !p_block->i_samples )
483 p_block->pf_release( p_block );
487 i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
488 aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
490 p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
493 msg_Warn( p_filter, "can't get output buffer" );
494 p_block->pf_release( p_block );
497 p_out->i_samples = (p_block->i_samples / p_filter->p_sys->i_nb_channels) *
498 aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
499 p_out->i_dts = p_block->i_dts;
500 p_out->i_pts = p_block->i_pts;
501 p_out->i_length = p_block->i_length;
503 aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
504 aout_filter.input = p_filter->fmt_in.audio;
505 aout_filter.input.i_format = p_filter->fmt_in.i_codec;
506 aout_filter.output = p_filter->fmt_out.audio;
507 aout_filter.output.i_format = p_filter->fmt_out.i_codec;
509 in_buf.p_buffer = p_block->p_buffer;
510 in_buf.i_nb_bytes = p_block->i_buffer;
511 in_buf.i_nb_samples = p_block->i_samples;
514 unsigned int i_in_size = in_buf.i_nb_samples * (p_filter->p_sys->i_bitspersample/8) *
515 aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
516 if( (in_buf.i_nb_bytes != i_in_size) && ((i_in_size % 32) != 0) ) /* is it word aligned?? */
518 msg_Err( p_filter, "input buffer is not word aligned" );
519 /* Fix output buffer to be word aligned */
523 out_buf.p_buffer = p_out->p_buffer;
524 out_buf.i_nb_bytes = p_out->i_buffer;
525 out_buf.i_nb_samples = p_out->i_samples;
527 memset( p_out->p_buffer, 0, i_out_size );
528 if( p_filter->p_sys->b_downmix )
530 stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
531 i_samples = mono( &aout_filter, &out_buf, &in_buf );
535 i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
538 p_out->i_buffer = out_buf.i_nb_bytes;
539 p_out->i_samples = out_buf.i_nb_samples;
541 p_block->pf_release( p_block );
545 /* stereo2mono_downmix - stereo channels into one mono channel.
546 * Code taken from modules/audio_filter/channel_mixer/headphone.c
547 * converted from float into int16_t based downmix
548 * Written by Boris Dorès <babal@via.ecp.fr>
550 static void stereo2mono_downmix( aout_filter_t * p_filter,
551 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
553 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
555 int i_input_nb = aout_FormatNbChannels( &p_filter->input );
556 int i_output_nb = aout_FormatNbChannels( &p_filter->output );
558 int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
560 uint8_t * p_overflow;
563 size_t i_overflow_size; /* in bytes */
564 size_t i_out_size; /* in bytes */
568 int i_source_channel_offset;
569 int i_dest_channel_offset;
570 unsigned int i_delay;
571 double d_amplitude_factor;
573 /* out buffer characterisitcs */
574 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
575 p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes * i_output_nb / i_input_nb;
576 p_out = p_out_buf->p_buffer;
577 i_out_size = p_out_buf->i_nb_bytes;
581 /* Slide the overflow buffer */
582 p_overflow = p_sys->p_overflow_buffer;
583 i_overflow_size = p_sys->i_overflow_buffer_size;
585 if ( i_out_size > i_overflow_size )
586 memcpy( p_out, p_overflow, i_overflow_size );
588 memcpy( p_out, p_overflow, i_out_size );
590 p_slide = p_sys->p_overflow_buffer;
591 while( p_slide < p_overflow + i_overflow_size )
593 if( p_slide + i_out_size < p_overflow + i_overflow_size )
595 memset( p_slide, 0, i_out_size );
596 if( p_slide + 2 * i_out_size < p_overflow + i_overflow_size )
597 memcpy( p_slide, p_slide + i_out_size, i_out_size );
599 memcpy( p_slide, p_slide + i_out_size,
600 p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
604 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
606 p_slide += i_out_size;
609 /* apply the atomic operations */
610 for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
612 /* shorter variable names */
613 i_source_channel_offset
614 = p_sys->p_atomic_operations[i].i_source_channel_offset;
615 i_dest_channel_offset
616 = p_sys->p_atomic_operations[i].i_dest_channel_offset;
617 i_delay = p_sys->p_atomic_operations[i].i_delay;
619 = p_sys->p_atomic_operations[i].d_amplitude_factor;
621 if( p_out_buf->i_nb_samples > i_delay )
623 /* current buffer coefficients */
624 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
626 ((int16_t*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
627 += p_in[ j * i_input_nb + i_source_channel_offset ]
628 * d_amplitude_factor;
631 /* overflow buffer coefficients */
632 for( j = 0; j < i_delay; j++ )
634 ((int16_t*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
635 += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
636 * i_input_nb + i_source_channel_offset ]
637 * d_amplitude_factor;
642 /* overflow buffer coefficients only */
643 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
645 ((int16_t*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
646 * i_output_nb + i_dest_channel_offset ]
647 += p_in[ j * i_input_nb + i_source_channel_offset ]
648 * d_amplitude_factor;
655 memset( p_out, 0, i_out_size );
659 /* Simple stereo to mono mixing. */
660 static unsigned int mono( aout_filter_t *p_filter,
661 aout_buffer_t *p_output, aout_buffer_t *p_input )
663 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
664 int16_t *p_in, *p_out;
665 unsigned int n = 0, r = 0;
667 p_in = (int16_t *) p_input->p_buffer;
668 p_out = (int16_t *) p_output->p_buffer;
670 while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
672 p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
679 /* Simple stereo to mono mixing. */
680 static unsigned int stereo_to_mono( aout_filter_t *p_filter,
681 aout_buffer_t *p_output, aout_buffer_t *p_input )
683 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
684 int16_t *p_in, *p_out;
687 p_in = (int16_t *) p_input->p_buffer;
688 p_out = (int16_t *) p_output->p_buffer;
690 for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
692 /* Fake real mono. */
693 if( p_sys->i_channel_selected == -1)
695 p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
698 else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
700 p_out[n] = p_out[n+1] = p_in[n];