git.sesse.net Git - vlc/blob - modules/audio_filter/channel_mixer/headphone.c

   1 /*****************************************************************************
   2  * headphone.c : headphone virtual spatialization channel mixer module
   3  *               -> gives the feeling of a real room with a simple headphone
   4  *****************************************************************************
   5  * Copyright (C) 2002-2006 the VideoLAN team
   6  * $Id$
   7  *
   8  * Authors: Boris Dorès <babal@via.ecp.fr>
   9  *
  10  * This program is free software; you can redistribute it and/or modify it
  11  * under the terms of the GNU General Public License as published by the Free
  12  * Software Foundation; either version 2 of the License, or (at your option)
  13  * any later version.
  14  *
  15  * This program is distributed in the hope that it will be useful, but WITHOUT
  16  * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
  17  * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
  18  * more details.
  19  *
  20  * You should have received a copy of the GNU General Public License along with
  21  * this program; if not, write to the Free Software Foundation, Inc., 51
  22  * Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  23  *****************************************************************************/
  24
  25 /*****************************************************************************
  26  * Preamble
  27  *****************************************************************************/
  28
  29 #ifdef HAVE_CONFIG_H
  30 # include "config.h"
  31 #endif
  32
  33 #include <math.h>                                        /* sqrt */
  34
  35 #include <vlc_common.h>
  36 #include <vlc_plugin.h>
  37 #include <vlc_aout.h>
  38 #include <vlc_filter.h>
  39 #include <vlc_block.h>
  40
  41 /*****************************************************************************
  42  * Local prototypes
  43  *****************************************************************************/
  44 static int  OpenFilter ( vlc_object_t * );
  45 static void CloseFilter( vlc_object_t * );
  46 static block_t *Convert( filter_t *, block_t * );
  47
  48 /*****************************************************************************
  49  * Module descriptor
  50  *****************************************************************************/
  51 #define MODULE_DESCRIPTION N_ ( \
  52      "This effect gives you the feeling that you are standing in a room " \
  53      "with a complete 7.1 speaker set when using only a headphone, " \
  54      "providing a more realistic sound experience. It should also be " \
  55      "more comfortable and less tiring when listening to music for " \
  56      "long periods of time.\nIt works with any source format from mono " \
  57      "to 7.1.")
  58
  59 #define HEADPHONE_DIM_TEXT N_("Characteristic dimension")
  60 #define HEADPHONE_DIM_LONGTEXT N_( \
  61      "Distance between front left speaker and listener in meters.")
  62
  63 #define HEADPHONE_COMPENSATE_TEXT N_("Compensate delay")
  64 #define HEADPHONE_COMPENSATE_LONGTEXT N_( \
  65      "The delay which is introduced by the physical algorithm may "\
  66      "sometimes be disturbing for the synchronization between lips-movement "\
  67      "and speech. In case, turn this on to compensate.")
  68
  69 #define HEADPHONE_DOLBY_TEXT N_("No decoding of Dolby Surround")
  70 #define HEADPHONE_DOLBY_LONGTEXT N_( \
  71      "Dolby Surround encoded streams won't be decoded before being " \
  72      "processed by this filter. Enabling this setting is not recommended.")
  73
  74 vlc_module_begin ()
  75     set_description( N_("Headphone virtual spatialization effect") )
  76     set_shortname( N_("Headphone effect") )
  77     set_help( MODULE_DESCRIPTION )
  78     set_category( CAT_AUDIO )
  79     set_subcategory( SUBCAT_AUDIO_AFILTER )
  80
  81     add_integer( "headphone-dim", 10, HEADPHONE_DIM_TEXT,
  82                  HEADPHONE_DIM_LONGTEXT, false )
  83     add_bool( "headphone-compensate", false, HEADPHONE_COMPENSATE_TEXT,
  84               HEADPHONE_COMPENSATE_LONGTEXT, true )
  85     add_bool( "headphone-dolby", false, HEADPHONE_DOLBY_TEXT,
  86               HEADPHONE_DOLBY_LONGTEXT, true )
  87
  88     set_capability( "audio filter", 0 )
  89     set_callbacks( OpenFilter, CloseFilter )
  90     add_shortcut( "headphone" )
  91 vlc_module_end ()
  92
  93
  94 /*****************************************************************************
  95  * Internal data structures
  96  *****************************************************************************/
  97 struct atomic_operation_t
  98 {
  99     int i_source_channel_offset;
 100     int i_dest_channel_offset;
 101     unsigned int i_delay;/* in sample unit */
 102     double d_amplitude_factor;
 103 };
 104
 105 struct filter_sys_t
 106 {
 107     size_t i_overflow_buffer_size;/* in bytes */
 108     float * p_overflow_buffer;
 109     unsigned int i_nb_atomic_operations;
 110     struct atomic_operation_t * p_atomic_operations;
 111 };
 112
 113 /*****************************************************************************
 114  * Init: initialize internal data structures
 115  * and computes the needed atomic operations
 116  *****************************************************************************/
 117 /* x and z represent the coordinates of the virtual speaker
 118  *  relatively to the center of the listener's head, measured in meters :
 119  *
 120  *  left              right
 121  *Z
 122  *-
 123  *a          head
 124  *x
 125  *i
 126  *s
 127  *  rear left    rear right
 128  *
 129  *          x-axis
 130  *  */
 131 static void ComputeChannelOperations( struct filter_sys_t * p_data
 132         , unsigned int i_rate, unsigned int i_next_atomic_operation
 133         , int i_source_channel_offset, double d_x, double d_z
 134         , double d_compensation_length, double d_channel_amplitude_factor )
 135 {
 136     double d_c = 340; /*sound celerity (unit: m/s)*/
 137     double d_compensation_delay = (d_compensation_length-0.1) / d_c * i_rate;
 138
 139     /* Left ear */
 140     p_data->p_atomic_operations[i_next_atomic_operation]
 141         .i_source_channel_offset = i_source_channel_offset;
 142     p_data->p_atomic_operations[i_next_atomic_operation]
 143         .i_dest_channel_offset = 0;/* left */
 144     p_data->p_atomic_operations[i_next_atomic_operation]
 145         .i_delay = (int)( sqrt( (-0.1-d_x)*(-0.1-d_x) + (0-d_z)*(0-d_z) )
 146                           / d_c * i_rate - d_compensation_delay );
 147     if( d_x < 0 )
 148     {
 149         p_data->p_atomic_operations[i_next_atomic_operation]
 150             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 151     }
 152     else if( d_x > 0 )
 153     {
 154         p_data->p_atomic_operations[i_next_atomic_operation]
 155             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 156     }
 157     else
 158     {
 159         p_data->p_atomic_operations[i_next_atomic_operation]
 160             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 161     }
 162
 163     /* Right ear */
 164     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 165         .i_source_channel_offset = i_source_channel_offset;
 166     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 167         .i_dest_channel_offset = 1;/* right */
 168     p_data->p_atomic_operations[i_next_atomic_operation + 1]
 169         .i_delay = (int)( sqrt( (0.1-d_x)*(0.1-d_x) + (0-d_z)*(0-d_z) )
 170                           / d_c * i_rate - d_compensation_delay );
 171     if( d_x < 0 )
 172     {
 173         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 174             .d_amplitude_factor = d_channel_amplitude_factor * 0.9 / 2;
 175     }
 176     else if( d_x > 0 )
 177     {
 178         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 179             .d_amplitude_factor = d_channel_amplitude_factor * 1.1 / 2;
 180     }
 181     else
 182     {
 183         p_data->p_atomic_operations[i_next_atomic_operation + 1]
 184             .d_amplitude_factor = d_channel_amplitude_factor / 2;
 185     }
 186 }
 187
 188 static int Init( vlc_object_t *p_this, struct filter_sys_t * p_data
 189         , unsigned int i_nb_channels, uint32_t i_physical_channels
 190         , unsigned int i_rate )
 191 {
 192     double d_x = var_InheritInteger( p_this, "headphone-dim" );
 193     double d_z = d_x;
 194     double d_z_rear = -d_x/3;
 195     double d_min = 0;
 196     unsigned int i_next_atomic_operation;
 197     int i_source_channel_offset;
 198     unsigned int i;
 199
 200     if( var_InheritBool( p_this, "headphone-compensate" ) )
 201     {
 202         /* minimal distance to any speaker */
 203         if( i_physical_channels & AOUT_CHAN_REARCENTER )
 204         {
 205             d_min = d_z_rear;
 206         }
 207         else
 208         {
 209             d_min = d_z;
 210         }
 211     }
 212
 213     /* Number of elementary operations */
 214     p_data->i_nb_atomic_operations = i_nb_channels * 2;
 215     if( i_physical_channels & AOUT_CHAN_CENTER )
 216     {
 217         p_data->i_nb_atomic_operations += 2;
 218     }
 219     p_data->p_atomic_operations = malloc( sizeof(struct atomic_operation_t)
 220             * p_data->i_nb_atomic_operations );
 221     if( p_data->p_atomic_operations == NULL )
 222         return -1;
 223
 224     /* For each virtual speaker, computes elementary wave propagation time
 225      * to each ear */
 226     i_next_atomic_operation = 0;
 227     i_source_channel_offset = 0;
 228     if( i_physical_channels & AOUT_CHAN_LEFT )
 229     {
 230         ComputeChannelOperations( p_data , i_rate
 231                 , i_next_atomic_operation , i_source_channel_offset
 232                 , -d_x , d_z , d_min , 2.0 / i_nb_channels );
 233         i_next_atomic_operation += 2;
 234         i_source_channel_offset++;
 235     }
 236     if( i_physical_channels & AOUT_CHAN_RIGHT )
 237     {
 238         ComputeChannelOperations( p_data , i_rate
 239                 , i_next_atomic_operation , i_source_channel_offset
 240                 , d_x , d_z , d_min , 2.0 / i_nb_channels );
 241         i_next_atomic_operation += 2;
 242         i_source_channel_offset++;
 243     }
 244     if( i_physical_channels & AOUT_CHAN_MIDDLELEFT )
 245     {
 246         ComputeChannelOperations( p_data , i_rate
 247                 , i_next_atomic_operation , i_source_channel_offset
 248                 , -d_x , 0 , d_min , 1.5 / i_nb_channels );
 249         i_next_atomic_operation += 2;
 250         i_source_channel_offset++;
 251     }
 252     if( i_physical_channels & AOUT_CHAN_MIDDLERIGHT )
 253     {
 254         ComputeChannelOperations( p_data , i_rate
 255                 , i_next_atomic_operation , i_source_channel_offset
 256                 , d_x , 0 , d_min , 1.5 / i_nb_channels );
 257         i_next_atomic_operation += 2;
 258         i_source_channel_offset++;
 259     }
 260     if( i_physical_channels & AOUT_CHAN_REARLEFT )
 261     {
 262         ComputeChannelOperations( p_data , i_rate
 263                 , i_next_atomic_operation , i_source_channel_offset
 264                 , -d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 265         i_next_atomic_operation += 2;
 266         i_source_channel_offset++;
 267     }
 268     if( i_physical_channels & AOUT_CHAN_REARRIGHT )
 269     {
 270         ComputeChannelOperations( p_data , i_rate
 271                 , i_next_atomic_operation , i_source_channel_offset
 272                 , d_x , d_z_rear , d_min , 1.5 / i_nb_channels );
 273         i_next_atomic_operation += 2;
 274         i_source_channel_offset++;
 275     }
 276     if( i_physical_channels & AOUT_CHAN_REARCENTER )
 277     {
 278         ComputeChannelOperations( p_data , i_rate
 279                 , i_next_atomic_operation , i_source_channel_offset
 280                 , 0 , -d_z , d_min , 1.5 / i_nb_channels );
 281         i_next_atomic_operation += 2;
 282         i_source_channel_offset++;
 283     }
 284     if( i_physical_channels & AOUT_CHAN_CENTER )
 285     {
 286         /* having two center channels increases the spatialization effect */
 287         ComputeChannelOperations( p_data , i_rate
 288                 , i_next_atomic_operation , i_source_channel_offset
 289                 , d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 290         i_next_atomic_operation += 2;
 291         ComputeChannelOperations( p_data , i_rate
 292                 , i_next_atomic_operation , i_source_channel_offset
 293                 , -d_x / 5.0 , d_z , d_min , 0.75 / i_nb_channels );
 294         i_next_atomic_operation += 2;
 295         i_source_channel_offset++;
 296     }
 297     if( i_physical_channels & AOUT_CHAN_LFE )
 298     {
 299         ComputeChannelOperations( p_data , i_rate
 300                 , i_next_atomic_operation , i_source_channel_offset
 301                 , 0 , d_z_rear , d_min , 5.0 / i_nb_channels );
 302         i_next_atomic_operation += 2;
 303         i_source_channel_offset++;
 304     }
 305
 306     /* Initialize the overflow buffer
 307      * we need it because the process induce a delay in the samples */
 308     p_data->i_overflow_buffer_size = 0;
 309     for( i = 0 ; i < p_data->i_nb_atomic_operations ; i++ )
 310     {
 311         if( p_data->i_overflow_buffer_size
 312                 < p_data->p_atomic_operations[i].i_delay * 2 * sizeof (float) )
 313         {
 314             p_data->i_overflow_buffer_size
 315                 = p_data->p_atomic_operations[i].i_delay * 2 * sizeof (float);
 316         }
 317     }
 318     p_data->p_overflow_buffer = (float *)malloc( p_data->i_overflow_buffer_size );
 319     if( p_data->p_overflow_buffer == NULL )
 320     {
 321         free( p_data->p_atomic_operations );
 322         return -1;
 323     }
 324     memset( p_data->p_overflow_buffer, 0 , p_data->i_overflow_buffer_size );
 325
 326     return 0;
 327 }
 328
 329 /*****************************************************************************
 330  * DoWork: convert a buffer
 331  *****************************************************************************/
 332 static void DoWork( filter_t * p_filter,
 333                     block_t * p_in_buf, block_t * p_out_buf )
 334 {
 335     filter_sys_t *p_sys = p_filter->p_sys;
 336     int i_input_nb = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 337     int i_output_nb = aout_FormatNbChannels( &p_filter->fmt_out.audio );
 338
 339     float * p_in = (float*) p_in_buf->p_buffer;
 340     float * p_out;
 341     uint8_t * p_overflow;
 342     uint8_t * p_end_overflow;
 343     uint8_t * p_slide;
 344
 345     size_t i_overflow_size;     /* in bytes */
 346     size_t i_out_size;          /* in bytes */
 347
 348     unsigned int i, j;
 349
 350     int i_source_channel_offset;
 351     int i_dest_channel_offset;
 352     unsigned int i_delay;
 353     double d_amplitude_factor;
 354
 355     p_out = (float *)p_out_buf->p_buffer;
 356     i_out_size = p_out_buf->i_buffer;
 357
 358     /* Slide the overflow buffer */
 359     p_overflow = (uint8_t *) p_sys->p_overflow_buffer;
 360     i_overflow_size = p_sys->i_overflow_buffer_size;
 361     p_end_overflow = p_overflow + i_overflow_size;
 362
 363     memset( p_out, 0, i_out_size );
 364     memcpy( p_out, p_overflow, __MIN( i_out_size, i_overflow_size ) );
 365
 366     p_slide = (uint8_t *) p_sys->p_overflow_buffer;
 367     while( p_slide < p_end_overflow )
 368     {
 369         size_t i_bytes_copied;
 370
 371         if( p_slide + i_out_size < p_end_overflow )
 372         {
 373             memset( p_slide, 0, i_out_size );
 374             if( p_slide + 2 * i_out_size < p_end_overflow )
 375                 i_bytes_copied = i_out_size;
 376             else
 377                 i_bytes_copied = p_end_overflow - ( p_slide + i_out_size );
 378             memcpy( p_slide, p_slide + i_out_size, i_bytes_copied );
 379         }
 380         else
 381         {
 382             i_bytes_copied = p_end_overflow - p_slide;
 383             memset( p_slide, 0, i_bytes_copied );
 384         }
 385         p_slide += i_bytes_copied;
 386     }
 387
 388     /* apply the atomic operations */
 389     for( i = 0; i < p_sys->i_nb_atomic_operations; i++ )
 390     {
 391         /* shorter variable names */
 392         i_source_channel_offset
 393             = p_sys->p_atomic_operations[i].i_source_channel_offset;
 394         i_dest_channel_offset
 395             = p_sys->p_atomic_operations[i].i_dest_channel_offset;
 396         i_delay = p_sys->p_atomic_operations[i].i_delay;
 397         d_amplitude_factor
 398             = p_sys->p_atomic_operations[i].d_amplitude_factor;
 399
 400         if( p_out_buf->i_nb_samples > i_delay )
 401         {
 402             /* current buffer coefficients */
 403             for( j = 0; j < p_out_buf->i_nb_samples - i_delay; j++ )
 404             {
 405                 ((float*)p_out)[ (i_delay+j)*i_output_nb + i_dest_channel_offset ]
 406                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 407                        * d_amplitude_factor;
 408             }
 409
 410             /* overflow buffer coefficients */
 411             for( j = 0; j < i_delay; j++ )
 412             {
 413                 ((float*)p_overflow)[ j*i_output_nb + i_dest_channel_offset ]
 414                     += p_in[ (p_out_buf->i_nb_samples - i_delay + j)
 415                        * i_input_nb + i_source_channel_offset ]
 416                        * d_amplitude_factor;
 417             }
 418         }
 419         else
 420         {
 421             /* overflow buffer coefficients only */
 422             for( j = 0; j < p_out_buf->i_nb_samples; j++ )
 423             {
 424                 ((float*)p_overflow)[ (i_delay - p_out_buf->i_nb_samples + j)
 425                                         * i_output_nb + i_dest_channel_offset ]
 426                     += p_in[ j * i_input_nb + i_source_channel_offset ]
 427                        * d_amplitude_factor;
 428             }
 429         }
 430     }
 431 }
 432
 433 /*
 434  * Audio filter 2
 435  */
 436 /*****************************************************************************
 437  * OpenFilter:
 438  *****************************************************************************/
 439 static int OpenFilter( vlc_object_t *p_this )
 440 {
 441     filter_t *p_filter = (filter_t *)p_this;
 442     filter_sys_t *p_sys;
 443
 444     /* Activate this filter only with stereo devices */
 445     if( p_filter->fmt_out.audio.i_physical_channels
 446             != (AOUT_CHAN_LEFT|AOUT_CHAN_RIGHT) )
 447     {
 448         msg_Dbg( p_filter, "filter discarded (incompatible format)" );
 449         return VLC_EGENERIC;
 450     }
 451
 452     /* Allocate the memory needed to store the module's structure */
 453     p_sys = p_filter->p_sys = malloc( sizeof(struct filter_sys_t) );
 454     if( p_sys == NULL )
 455         return VLC_ENOMEM;
 456     p_sys->i_overflow_buffer_size = 0;
 457     p_sys->p_overflow_buffer = NULL;
 458     p_sys->i_nb_atomic_operations = 0;
 459     p_sys->p_atomic_operations = NULL;
 460
 461     if( Init( VLC_OBJECT(p_filter), p_sys
 462                 , aout_FormatNbChannels ( &(p_filter->fmt_in.audio) )
 463                 , p_filter->fmt_in.audio.i_physical_channels
 464                 , p_filter->fmt_in.audio.i_rate ) < 0 )
 465     {
 466         free( p_sys );
 467         return VLC_EGENERIC;
 468     }
 469
 470     /* Request a specific format if not already compatible */
 471     p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
 472     p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
 473     p_filter->fmt_out.audio.i_rate = p_filter->fmt_in.audio.i_rate;
 474     p_filter->fmt_in.audio.i_original_channels =
 475                                    p_filter->fmt_out.audio.i_original_channels;
 476     if( p_filter->fmt_in.audio.i_physical_channels == AOUT_CHANS_STEREO
 477      && (p_filter->fmt_in.audio.i_original_channels & AOUT_CHAN_DOLBYSTEREO)
 478      && !var_InheritBool( p_filter, "headphone-dolby" ) )
 479     {
 480         p_filter->fmt_in.audio.i_physical_channels = AOUT_CHANS_5_0;
 481     }
 482     p_filter->pf_audio_filter = Convert;
 483
 484     return VLC_SUCCESS;
 485 }
 486
 487 /*****************************************************************************
 488  * CloseFilter : deallocate data structures
 489  *****************************************************************************/
 490 static void CloseFilter( vlc_object_t *p_this )
 491 {
 492     filter_t *p_filter = (filter_t *)p_this;
 493
 494     free( p_filter->p_sys->p_overflow_buffer );
 495     free( p_filter->p_sys->p_atomic_operations );
 496     free( p_filter->p_sys );
 497 }
 498
 499 static block_t *Convert( filter_t *p_filter, block_t *p_block )
 500 {
 501     if( !p_block || !p_block->i_nb_samples )
 502     {
 503         if( p_block )
 504             block_Release( p_block );
 505         return NULL;
 506     }
 507
 508     size_t i_out_size = p_block->i_buffer *
 509         aout_FormatNbChannels( &(p_filter->fmt_out.audio) ) /
 510         aout_FormatNbChannels( &(p_filter->fmt_in.audio) );
 511
 512     block_t *p_out = block_Alloc( i_out_size );
 513     if( !p_out )
 514     {
 515         msg_Warn( p_filter, "can't get output buffer" );
 516         block_Release( p_block );
 517         return NULL;
 518     }
 519
 520     p_out->i_nb_samples = p_block->i_nb_samples;
 521     p_out->i_dts = p_block->i_dts;
 522     p_out->i_pts = p_block->i_pts;
 523     p_out->i_length = p_block->i_length;
 524
 525     DoWork( p_filter, p_block, p_out );
 526
 527     block_Release( p_block );
 528     return p_out;
 529 }