git.sesse.net Git - vlc/blob - src/audio_output/filters.c

   1 /*****************************************************************************
   2  * filters.c : audio output filters management
   3  *****************************************************************************
   4  * Copyright (C) 2002-2007 the VideoLAN team
   5  * $Id$
   6  *
   7  * Authors: Christophe Massiot <massiot@via.ecp.fr>
   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 <assert.h>
  32
  33 #include <vlc_common.h>
  34 #include <vlc_dialog.h>
  35 #include <vlc_modules.h>
  36
  37 #include <vlc_aout.h>
  38 #include <vlc_filter.h>
  39 #include "aout_internal.h"
  40 #include <libvlc.h>
  41
  42 block_t *aout_FilterBufferNew( filter_t *p_filter, int size )
  43 {
  44     (void) p_filter;
  45     return block_Alloc( size );
  46 }
  47
  48 /*****************************************************************************
  49  * FindFilter: find an audio filter for a specific transformation
  50  *****************************************************************************/
  51 static filter_t * FindFilter( aout_instance_t * p_aout,
  52                               const audio_sample_format_t * p_input_format,
  53                               const audio_sample_format_t * p_output_format )
  54 {
  55     static const char typename[] = "audio filter";
  56     filter_t * p_filter;
  57
  58     p_filter = vlc_custom_create( p_aout, sizeof(*p_filter),
  59                                   VLC_OBJECT_GENERIC, typename );
  60
  61     if ( p_filter == NULL ) return NULL;
  62     vlc_object_attach( p_filter, p_aout );
  63
  64     memcpy( &p_filter->fmt_in.audio, p_input_format,
  65             sizeof(audio_sample_format_t) );
  66     p_filter->fmt_in.i_codec = p_input_format->i_format;
  67     memcpy( &p_filter->fmt_out.audio, p_output_format,
  68             sizeof(audio_sample_format_t) );
  69     p_filter->fmt_out.i_codec = p_output_format->i_format;
  70     p_filter->pf_audio_buffer_new = aout_FilterBufferNew;
  71
  72     p_filter->p_module = module_need( p_filter, "audio filter", NULL, false );
  73     if ( p_filter->p_module == NULL )
  74     {
  75         vlc_object_release( p_filter );
  76         return NULL;
  77     }
  78
  79     assert( p_filter->pf_audio_filter );
  80     return p_filter;
  81 }
  82
  83 /*****************************************************************************
  84  * SplitConversion: split a conversion in two parts
  85  *****************************************************************************
  86  * Returns the number of conversions required by the first part - 0 if only
  87  * one conversion was asked.
  88  * Beware : p_output_format can be modified during this function if the
  89  * developer passed SplitConversion( toto, titi, titi, ... ). That is legal.
  90  * SplitConversion( toto, titi, toto, ... ) isn't.
  91  *****************************************************************************/
  92 static int SplitConversion( const audio_sample_format_t * p_input_format,
  93                             const audio_sample_format_t * p_output_format,
  94                             audio_sample_format_t * p_middle_format )
  95 {
  96     bool b_format =
  97              (p_input_format->i_format != p_output_format->i_format);
  98     bool b_rate = (p_input_format->i_rate != p_output_format->i_rate);
  99     bool b_channels =
 100         (p_input_format->i_physical_channels
 101           != p_output_format->i_physical_channels)
 102      || (p_input_format->i_original_channels
 103           != p_output_format->i_original_channels);
 104     int i_nb_conversions = b_format + b_rate + b_channels;
 105
 106     if ( i_nb_conversions <= 1 ) return 0;
 107
 108     memcpy( p_middle_format, p_output_format, sizeof(audio_sample_format_t) );
 109
 110     if ( i_nb_conversions == 2 )
 111     {
 112         if ( !b_format || !b_channels )
 113         {
 114             p_middle_format->i_rate = p_input_format->i_rate;
 115             aout_FormatPrepare( p_middle_format );
 116             return 1;
 117         }
 118
 119         /* !b_rate */
 120         p_middle_format->i_physical_channels
 121              = p_input_format->i_physical_channels;
 122         p_middle_format->i_original_channels
 123              = p_input_format->i_original_channels;
 124         aout_FormatPrepare( p_middle_format );
 125         return 1;
 126     }
 127
 128     /* i_nb_conversion == 3 */
 129     p_middle_format->i_rate = p_input_format->i_rate;
 130     aout_FormatPrepare( p_middle_format );
 131     return 2;
 132 }
 133
 134 static void ReleaseFilter( filter_t * p_filter )
 135 {
 136     module_unneed( p_filter, p_filter->p_module );
 137     vlc_object_release( p_filter );
 138 }
 139
 140 /*****************************************************************************
 141  * aout_FiltersCreatePipeline: create a filters pipeline to transform a sample
 142  *                             format to another
 143  *****************************************************************************
 144  * pi_nb_filters must be initialized before calling this function
 145  *****************************************************************************/
 146 int aout_FiltersCreatePipeline( aout_instance_t * p_aout,
 147                                 filter_t ** pp_filters_start,
 148                                 int * pi_nb_filters,
 149                                 const audio_sample_format_t * p_input_format,
 150                                 const audio_sample_format_t * p_output_format )
 151 {
 152     filter_t** pp_filters = pp_filters_start + *pi_nb_filters;
 153     audio_sample_format_t temp_format;
 154     int i_nb_conversions;
 155
 156     if ( AOUT_FMTS_IDENTICAL( p_input_format, p_output_format ) )
 157     {
 158         msg_Dbg( p_aout, "no need for any filter" );
 159         return 0;
 160     }
 161
 162     aout_FormatsPrint( p_aout, "filter(s)", p_input_format, p_output_format );
 163
 164     if( *pi_nb_filters + 1 > AOUT_MAX_FILTERS )
 165     {
 166         msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
 167         dialog_Fatal( p_aout, _("Audio filtering failed"),
 168                       _("The maximum number of filters (%d) was reached."),
 169                       AOUT_MAX_FILTERS );
 170         return -1;
 171     }
 172
 173     /* Try to find a filter to do the whole conversion. */
 174     pp_filters[0] = FindFilter( p_aout, p_input_format, p_output_format );
 175     if ( pp_filters[0] != NULL )
 176     {
 177         msg_Dbg( p_aout, "found a filter for the whole conversion" );
 178         ++*pi_nb_filters;
 179         return 0;
 180     }
 181
 182     /* We'll have to split the conversion. We always do the downmixing
 183      * before the resampling, because the audio decoder can probably do it
 184      * for us. */
 185     i_nb_conversions = SplitConversion( p_input_format,
 186                                         p_output_format, &temp_format );
 187     if ( !i_nb_conversions )
 188     {
 189         /* There was only one conversion to do, and we already failed. */
 190         msg_Err( p_aout, "couldn't find a filter for the conversion "
 191                 "%4.4s -> %4.4s",
 192                 &p_input_format->i_format, &p_output_format->i_format );
 193         return -1;
 194     }
 195
 196     pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format );
 197     if ( pp_filters[0] == NULL && i_nb_conversions == 2 )
 198     {
 199         /* Try with only one conversion. */
 200         SplitConversion( p_input_format, &temp_format, &temp_format );
 201         pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format );
 202     }
 203     if ( pp_filters[0] == NULL )
 204     {
 205         msg_Err( p_aout,
 206               "couldn't find a filter for the first part of the conversion" );
 207         return -1;
 208     }
 209
 210     /* We have the first stage of the conversion. Find a filter for
 211      * the rest. */
 212     if( *pi_nb_filters + 2 > AOUT_MAX_FILTERS )
 213     {
 214         ReleaseFilter( pp_filters[0] );
 215         msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
 216         dialog_Fatal( p_aout, _("Audio filtering failed"),
 217                       _("The maximum number of filters (%d) was reached."),
 218                       AOUT_MAX_FILTERS );
 219         return -1;
 220     }
 221     pp_filters[1] = FindFilter( p_aout, &pp_filters[0]->fmt_out.audio,
 222                                 p_output_format );
 223     if ( pp_filters[1] == NULL )
 224     {
 225         /* Try to split the conversion. */
 226         i_nb_conversions = SplitConversion( &pp_filters[0]->fmt_out.audio,
 227                                            p_output_format, &temp_format );
 228         if ( !i_nb_conversions )
 229         {
 230             ReleaseFilter( pp_filters[0] );
 231             msg_Err( p_aout,
 232               "couldn't find a filter for the second part of the conversion" );
 233             return -1;
 234         }
 235         if( *pi_nb_filters + 3 > AOUT_MAX_FILTERS )
 236         {
 237             ReleaseFilter( pp_filters[0] );
 238             msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS );
 239             dialog_Fatal( p_aout, _("Audio filtering failed"),
 240                           _("The maximum number of filters (%d) was reached."),
 241                           AOUT_MAX_FILTERS );
 242             return -1;
 243         }
 244         pp_filters[1] = FindFilter( p_aout, &pp_filters[0]->fmt_out.audio,
 245                                     &temp_format );
 246         pp_filters[2] = FindFilter( p_aout, &temp_format,
 247                                     p_output_format );
 248
 249         if ( pp_filters[1] == NULL || pp_filters[2] == NULL )
 250         {
 251             ReleaseFilter( pp_filters[0] );
 252             if ( pp_filters[1] != NULL )
 253             {
 254                 ReleaseFilter( pp_filters[1] );
 255             }
 256             if ( pp_filters[2] != NULL )
 257             {
 258                 ReleaseFilter( pp_filters[2] );
 259             }
 260             msg_Err( p_aout,
 261                "couldn't find filters for the second part of the conversion" );
 262             return -1;
 263         }
 264         *pi_nb_filters += 3;
 265         msg_Dbg( p_aout, "found 3 filters for the whole conversion" );
 266     }
 267     else
 268     {
 269         *pi_nb_filters += 2;
 270         msg_Dbg( p_aout, "found 2 filters for the whole conversion" );
 271     }
 272
 273     return 0;
 274 }
 275
 276 /*****************************************************************************
 277  * aout_FiltersDestroyPipeline: deallocate a filters pipeline
 278  *****************************************************************************/
 279 void aout_FiltersDestroyPipeline( aout_instance_t * p_aout,
 280                                   filter_t ** pp_filters,
 281                                   int i_nb_filters )
 282 {
 283     int i;
 284     (void)p_aout;
 285
 286     for ( i = 0; i < i_nb_filters; i++ )
 287     {
 288         filter_t *p_filter = pp_filters[i];
 289
 290         module_unneed( p_filter, p_filter->p_module );
 291         free( p_filter->p_owner );
 292         vlc_object_release( p_filter );
 293     }
 294 }
 295
 296 /*****************************************************************************
 297  * aout_FiltersHintBuffers: fill in aout_alloc_t structures to optimize
 298  *                          buffer allocations
 299  *****************************************************************************/
 300 void aout_FiltersHintBuffers( aout_instance_t * p_aout,
 301                               filter_t ** pp_filters,
 302                               int i_nb_filters, aout_alloc_t * p_first_alloc )
 303 {
 304     int i;
 305
 306     (void)p_aout; /* unused */
 307
 308     for ( i = i_nb_filters - 1; i >= 0; i-- )
 309     {
 310         filter_t * p_filter = pp_filters[i];
 311
 312         int i_output_size = p_filter->fmt_out.audio.i_bytes_per_frame
 313                          * p_filter->fmt_out.audio.i_rate * AOUT_MAX_INPUT_RATE
 314                          / p_filter->fmt_out.audio.i_frame_length;
 315         int i_input_size = p_filter->fmt_in.audio.i_bytes_per_frame
 316                          * p_filter->fmt_in.audio.i_rate * AOUT_MAX_INPUT_RATE
 317                          / p_filter->fmt_in.audio.i_frame_length;
 318
 319         if( i_output_size > p_first_alloc->i_bytes_per_sec )
 320             p_first_alloc->i_bytes_per_sec = i_output_size;
 321         if( i_input_size > p_first_alloc->i_bytes_per_sec )
 322             p_first_alloc->i_bytes_per_sec = i_input_size;
 323     }
 324 }
 325
 326 /*****************************************************************************
 327  * aout_FiltersPlay: play a buffer
 328  *****************************************************************************/
 329 void aout_FiltersPlay( filter_t ** pp_filters,
 330                        unsigned i_nb_filters, block_t ** pp_block )
 331 {
 332     block_t *p_block = *pp_block;
 333
 334     /* TODO: use filter chain */
 335     for( unsigned i = 0; i < i_nb_filters; i++ )
 336     {
 337         filter_t * p_filter = pp_filters[i];
 338
 339         /* Please note that p_block->i_nb_samples & i_buffer
 340          * shall be set by the filter plug-in. */
 341         p_block = p_filter->pf_audio_filter( p_filter, p_block );
 342     }
 343     *pp_block = p_block;
 344 }
 345