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 *****************************************************************************/
31 #include <math.h> /* sqrt */
34 # include <stdint.h> /* int16_t .. */
35 #elif defined(HAVE_INTTYPES_H)
36 # include <inttypes.h> /* int16_t .. */
43 #include <vlc_common.h>
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 )
252 /* For each virtual speaker, computes elementary wave propagation time
254 i_next_atomic_operation = 0;
255 i_source_channel_offset = 0;
256 if( i_physical_channels & AOUT_CHAN_LEFT )
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++;
264 if( i_physical_channels & AOUT_CHAN_RIGHT )
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++;
272 if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
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++;
280 if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
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++;
288 if( i_physical_channels & AOUT_CHAN_REARLEFT )
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++;
296 if( i_physical_channels & AOUT_CHAN_REARRIGHT )
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++;
304 if( i_physical_channels & AOUT_CHAN_REARCENTER )
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++;
312 if( i_physical_channels & AOUT_CHAN_CENTER )
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++;
325 if( i_physical_channels & AOUT_CHAN_LFE )
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++;
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++ )
339 if( p_data->i_overflow_buffer_size
340 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t) )
342 p_data->i_overflow_buffer_size
343 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (int16_t);
346 p_data->p_overflow_buffer = malloc( p_data->i_overflow_buffer_size );
347 if( p_data->p_atomic_operations == NULL )
349 memset( p_data->p_overflow_buffer, 0, p_data->i_overflow_buffer_size );
355 /*****************************************************************************
357 *****************************************************************************/
358 static int OpenFilter( vlc_object_t *p_this )
360 filter_t * p_filter = (filter_t *)p_this;
361 filter_sys_t *p_sys = NULL;
363 if( aout_FormatNbChannels( &(p_filter->fmt_in.audio) ) == 1 )
365 msg_Dbg( p_filter, "filter discarded (incompatible format)" );
369 if( (p_filter->fmt_in.i_codec != AOUT_FMT_S16_NE) ||
370 (p_filter->fmt_out.i_codec != AOUT_FMT_S16_NE) )
372 msg_Err( p_this, "filter discarded (invalid format)" );
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))
383 msg_Err( p_this, "couldn't load mono filter" );
387 /* Allocate the memory needed to store the module's structure */
388 p_sys = p_filter->p_sys = malloc( sizeof(filter_sys_t) );
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" );
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" );
401 if( p_sys->b_downmix )
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;
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;
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;
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;
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;
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 )
434 p_filter->pf_audio_filter = Convert;
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 );
447 /*****************************************************************************
449 *****************************************************************************/
450 static void CloseFilter( vlc_object_t *p_this)
452 filter_t *p_filter = (filter_t *) p_this;
453 filter_sys_t *p_sys = p_filter->p_sys;
455 var_Destroy( p_this, MONO_CFG "channel" );
456 var_Destroy( p_this, MONO_CFG "downmix" );
460 /*****************************************************************************
462 *****************************************************************************/
463 static block_t *Convert( filter_t *p_filter, block_t *p_block )
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;
471 if( !p_block || !p_block->i_samples )
474 block_Release( p_block );
478 i_out_size = p_block->i_samples * p_filter->p_sys->i_bitspersample/8 *
479 aout_FormatNbChannels( &(p_filter->fmt_out.audio) );
481 p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
484 msg_Warn( p_filter, "can't get output buffer" );
485 block_Release( p_block );
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;
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;
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;
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?? */
509 msg_Err( p_filter, "input buffer is not word aligned" );
510 /* Fix output buffer to be word aligned */
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;
518 memset( p_out->p_buffer, 0, i_out_size );
519 if( p_filter->p_sys->b_downmix )
521 stereo2mono_downmix( &aout_filter, &in_buf, &out_buf );
522 i_samples = mono( &aout_filter, &out_buf, &in_buf );
526 i_samples = stereo_to_mono( &aout_filter, &out_buf, &in_buf );
529 p_out->i_buffer = out_buf.i_nb_bytes;
530 p_out->i_samples = out_buf.i_nb_samples;
532 block_Release( p_block );
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>
541 static void stereo2mono_downmix( aout_filter_t * p_filter,
542 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
544 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
546 int i_input_nb = aout_FormatNbChannels( &p_filter->input );
547 int i_output_nb = aout_FormatNbChannels( &p_filter->output );
549 int16_t * p_in = (int16_t*) p_in_buf->p_buffer;
551 uint8_t * p_overflow;
554 size_t i_overflow_size; /* in bytes */
555 size_t i_out_size; /* in bytes */
559 int i_source_channel_offset;
560 int i_dest_channel_offset;
561 unsigned int i_delay;
562 double d_amplitude_factor;
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;
572 /* Slide the overflow buffer */
573 p_overflow = p_sys->p_overflow_buffer;
574 i_overflow_size = p_sys->i_overflow_buffer_size;
576 if ( i_out_size > i_overflow_size )
577 memcpy( p_out, p_overflow, i_overflow_size );
579 memcpy( p_out, p_overflow, i_out_size );
581 p_slide = p_sys->p_overflow_buffer;
582 while( p_slide < p_overflow + i_overflow_size )
584 if( p_slide + i_out_size < p_overflow + i_overflow_size )
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 );
590 memcpy( p_slide, p_slide + i_out_size,
591 p_overflow + i_overflow_size - ( p_slide + i_out_size ) );
595 memset( p_slide, 0, p_overflow + i_overflow_size - p_slide );
597 p_slide += i_out_size;
600 /* apply the atomic operations */
601 for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
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;
610 = p_sys->p_atomic_operations[i].d_amplitude_factor;
612 if( p_out_buf->i_nb_samples > i_delay )
614 /* current buffer coefficients */
615 for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
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;
622 /* overflow buffer coefficients */
623 for( j = 0; j < i_delay; j++ )
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;
633 /* overflow buffer coefficients only */
634 for( j = 0; j < p_out_buf->i_nb_samples; j++ )
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;
646 memset( p_out, 0, i_out_size );
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 )
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;
658 p_in = (int16_t *) p_input->p_buffer;
659 p_out = (int16_t *) p_output->p_buffer;
661 while( n < (p_input->i_nb_samples * p_sys->i_nb_channels) )
663 p_out[r] = (p_in[n] + p_in[n+1]) >> 1;
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 )
674 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
675 int16_t *p_in, *p_out;
678 p_in = (int16_t *) p_input->p_buffer;
679 p_out = (int16_t *) p_output->p_buffer;
681 for( n = 0; n < (p_input->i_nb_samples * p_sys->i_nb_channels); n++ )
683 /* Fake real mono. */
684 if( p_sys->i_channel_selected == -1)
686 p_out[n] = p_out[n+1] = (p_in[n] + p_in[n+1]) >> 1;
689 else if( (n % p_sys->i_nb_channels) == (unsigned int) p_sys->i_channel_selected )
691 p_out[n] = p_out[n+1] = p_in[n];