git.sesse.net Git - vlc/blob - modules/audio_filter/resampler/linear.c

   1 /*****************************************************************************
   2  * linear.c : linear interpolation resampler
   3  *****************************************************************************
   4  * Copyright (C) 2002, 2006 the VideoLAN team
   5  * $Id$
   6  *
   7  * Authors: Gildas Bazin <gbazin@netcourrier.com>
   8  *          Sigmund Augdal Helberg <dnumgis@videolan.org>
   9  *
  10  * This program is free software; you can redistribute it and/or modify
  11  * it under the terms of the GNU General Public License as published by
  12  * the Free Software Foundation; either version 2 of the License, or
  13  * (at your option) any later version.
  14  *
  15  * This program is distributed in the hope that it will be useful,
  16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  18  * GNU General Public License for more details.
  19  *
  20  * You should have received a copy of the GNU General Public License
  21  * along with this program; if not, write to the Free Software
  22  * Foundation, Inc., 51 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 <vlc_common.h>
  34 #include <vlc_plugin.h>
  35 #include <vlc_aout.h>
  36 #include <vlc_filter.h>
  37 #include <vlc_block.h>
  38
  39 /*****************************************************************************
  40  * Local prototypes
  41  *****************************************************************************/
  42 static int  Create    ( vlc_object_t * );
  43 static void Close     ( vlc_object_t * );
  44 static void DoWork    ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
  45                         aout_buffer_t * );
  46
  47 static int  OpenFilter ( vlc_object_t * );
  48 static void CloseFilter( vlc_object_t * );
  49 static block_t *Resample( filter_t *, block_t * );
  50
  51 /*****************************************************************************
  52  * Local structures
  53  *****************************************************************************/
  54 struct filter_sys_t
  55 {
  56     int32_t *p_prev_sample;       /* this filter introduces a 1 sample delay */
  57
  58     unsigned int i_remainder;                /* remainder of previous sample */
  59
  60     date_t       end_date;
  61 };
  62
  63 /*****************************************************************************
  64  * Module descriptor
  65  *****************************************************************************/
  66 vlc_module_begin ()
  67     set_description( N_("Audio filter for linear interpolation resampling") )
  68     set_category( CAT_AUDIO )
  69     set_subcategory( SUBCAT_AUDIO_MISC )
  70     set_capability( "audio filter", 5 )
  71     set_callbacks( Create, Close )
  72
  73     add_submodule ()
  74     set_description( N_("Audio filter for linear interpolation resampling") )
  75     set_capability( "audio filter2", 5 )
  76     set_callbacks( OpenFilter, CloseFilter )
  77 vlc_module_end ()
  78
  79 /*****************************************************************************
  80  * Create: allocate linear resampler
  81  *****************************************************************************/
  82 static int Create( vlc_object_t *p_this )
  83 {
  84     aout_filter_t * p_filter = (aout_filter_t *)p_this;
  85     struct filter_sys_t * p_sys;
  86
  87     if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
  88           || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
  89           || p_filter->fmt_in.audio.i_physical_channels
  90               != p_filter->fmt_out.audio.i_physical_channels
  91           || p_filter->fmt_in.audio.i_original_channels
  92               != p_filter->fmt_out.audio.i_original_channels
  93           || p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 )
  94     {
  95         return VLC_EGENERIC;
  96     }
  97
  98     /* Allocate the memory needed to store the module's structure */
  99     p_sys = malloc( sizeof(filter_sys_t) );
 100     p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
 101     if( p_sys == NULL )
 102         return VLC_ENOMEM;
 103     p_sys->p_prev_sample = malloc(
 104         p_filter->fmt_in.audio.i_channels * sizeof(int32_t) );
 105     if( p_sys->p_prev_sample == NULL )
 106     {
 107         free( p_sys );
 108         return VLC_ENOMEM;
 109     }
 110     date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
 111
 112     p_filter->pf_do_work = DoWork;
 113
 114     /* We don't want a new buffer to be created because we're not sure we'll
 115      * actually need to resample anything. */
 116     p_filter->b_in_place = true;
 117
 118     return VLC_SUCCESS;
 119 }
 120
 121 /*****************************************************************************
 122  * Close: free our resources
 123  *****************************************************************************/
 124 static void Close( vlc_object_t * p_this )
 125 {
 126     aout_filter_t * p_filter = (aout_filter_t *)p_this;
 127     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 128
 129     free( p_sys->p_prev_sample );
 130     free( p_sys );
 131 }
 132
 133 /*****************************************************************************
 134  * DoWork: convert a buffer
 135  *****************************************************************************/
 136 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
 137                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 138 {
 139     filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
 140     float *p_out = (float *)p_out_buf->p_buffer;
 141     float *p_prev_sample = (float *)p_sys->p_prev_sample;
 142
 143     int i_nb_channels = p_filter->fmt_in.audio.i_channels;
 144     int i_in_nb = p_in_buf->i_nb_samples;
 145     int i_chan, i_in, i_out = 0;
 146
 147     /* Check if we really need to run the resampler */
 148     if( p_aout->mixer_format.i_rate == p_filter->fmt_in.audio.i_rate )
 149     {
 150 #if 0   /* FIXME: needs audio filter2 for block_Realloc */
 151         if( p_filter->b_continuity )
 152         {
 153             p_in_buf = block_Realloc( p_in_buf, sizeof(float) * i_nb_channels,
 154                                       p_in_buf->i_buffer );
 155             if( !p_in_buf )
 156                 abort();
 157
 158             memcpy( p_in_buf->p_buffer, p_prev_sample,
 159                     i_nb_channels * sizeof(float) );
 160         }
 161 #endif
 162         p_filter->b_continuity = false;
 163         return;
 164     }
 165
 166     float p_in_orig[p_in_buf->i_buffer / 4], *p_in = p_in_orig;
 167
 168     vlc_memcpy( p_in, p_in_buf->p_buffer, p_in_buf->i_buffer );
 169
 170     /* Take care of the previous input sample (if any) */
 171     if( !p_filter->b_continuity )
 172     {
 173         p_filter->b_continuity = true;
 174         p_sys->i_remainder = 0;
 175         date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
 176     }
 177     else
 178     {
 179         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 180         {
 181             for( i_chan = i_nb_channels ; i_chan ; )
 182             {
 183                 i_chan--;
 184                 p_out[i_chan] = p_prev_sample[i_chan];
 185                 p_out[i_chan] += ( ( p_in[i_chan] - p_prev_sample[i_chan] )
 186                                    * p_sys->i_remainder
 187                                    / p_filter->fmt_out.audio.i_rate );
 188             }
 189             p_out += i_nb_channels;
 190               i_out++;
 191
 192             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 193         }
 194         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 195     }
 196
 197     /* Take care of the current input samples (minus last one) */
 198     for( i_in = 0; i_in < i_in_nb - 1; i_in++ )
 199     {
 200         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 201         {
 202             for( i_chan = i_nb_channels ; i_chan ; )
 203             {
 204                 i_chan--;
 205                 p_out[i_chan] = p_in[i_chan];
 206                 p_out[i_chan] += ( ( p_in[i_chan + i_nb_channels]
 207                     - p_in[i_chan] )
 208                     * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate );
 209             }
 210             p_out += i_nb_channels;
 211               i_out++;
 212
 213             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 214         }
 215
 216         p_in += i_nb_channels;
 217         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 218     }
 219
 220     /* Backup the last input sample for next time */
 221     for( i_chan = i_nb_channels ; i_chan ; )
 222     {
 223         i_chan--;
 224         p_prev_sample[i_chan] = p_in[i_chan];
 225     }
 226
 227     p_out_buf->i_nb_samples = i_out;
 228     p_out_buf->i_pts = p_in_buf->i_pts;
 229
 230     if( p_in_buf->i_pts !=
 231         date_Get( &p_sys->end_date ) )
 232     {
 233         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 234     }
 235
 236     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 237                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 238
 239     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 240         i_nb_channels * sizeof(int32_t);
 241 }
 242
 243 /*****************************************************************************
 244  * OpenFilter:
 245  *****************************************************************************/
 246 static int OpenFilter( vlc_object_t *p_this )
 247 {
 248     filter_t *p_filter = (filter_t *)p_this;
 249     filter_sys_t *p_sys;
 250     int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 251
 252     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 253         p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
 254     {
 255         return VLC_EGENERIC;
 256     }
 257
 258     /* Allocate the memory needed to store the module's structure */
 259     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 260     if( p_sys == NULL )
 261         return VLC_ENOMEM;
 262
 263     p_sys->p_prev_sample = malloc(
 264         p_filter->fmt_in.audio.i_channels * sizeof(int32_t) );
 265     if( p_sys->p_prev_sample == NULL )
 266     {
 267         free( p_sys );
 268         return VLC_ENOMEM;
 269     }
 270     date_Init( &p_sys->end_date, p_filter->fmt_in.audio.i_rate, 1 );
 271
 272     p_filter->pf_audio_filter = Resample;
 273
 274     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 275              (char *)&p_filter->fmt_in.i_codec,
 276              p_filter->fmt_in.audio.i_rate,
 277              p_filter->fmt_in.audio.i_channels,
 278              (char *)&p_filter->fmt_out.i_codec,
 279              p_filter->fmt_out.audio.i_rate,
 280              p_filter->fmt_out.audio.i_channels);
 281
 282     p_filter->fmt_out = p_filter->fmt_in;
 283     p_filter->fmt_out.audio.i_rate = i_out_rate;
 284
 285     return 0;
 286 }
 287
 288 /*****************************************************************************
 289  * CloseFilter : deallocate data structures
 290  *****************************************************************************/
 291 static void CloseFilter( vlc_object_t *p_this )
 292 {
 293     filter_t *p_filter = (filter_t *)p_this;
 294     free( p_filter->p_sys->p_prev_sample );
 295     free( p_filter->p_sys );
 296 }
 297
 298 /*****************************************************************************
 299  * Resample
 300  *****************************************************************************/
 301 static block_t *Resample( filter_t *p_filter, block_t *p_block )
 302 {
 303     aout_filter_t aout_filter;
 304     aout_buffer_t in_buf, out_buf;
 305     block_t *p_out;
 306     int i_out_size;
 307     int i_bytes_per_frame;
 308
 309     if( !p_block || !p_block->i_nb_samples )
 310     {
 311         if( p_block )
 312             block_Release( p_block );
 313         return NULL;
 314     }
 315
 316     i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 317                   p_filter->fmt_out.audio.i_bitspersample / 8;
 318
 319     i_out_size = i_bytes_per_frame * ( 1 + (p_block->i_nb_samples *
 320         p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
 321
 322     p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
 323     if( !p_out )
 324     {
 325         msg_Warn( p_filter, "can't get output buffer" );
 326         block_Release( p_block );
 327         return NULL;
 328     }
 329
 330     p_out->i_nb_samples = i_out_size / i_bytes_per_frame;
 331     p_out->i_dts = p_block->i_dts;
 332     p_out->i_pts = p_block->i_pts;
 333     p_out->i_length = p_block->i_length;
 334
 335     aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
 336     aout_filter.fmt_in.audio = p_filter->fmt_in.audio;
 337     aout_filter.fmt_out.audio = p_filter->fmt_out.audio;
 338     aout_filter.b_continuity = false;
 339
 340     in_buf.p_buffer = p_block->p_buffer;
 341     in_buf.i_buffer = p_block->i_buffer;
 342     in_buf.i_nb_samples = p_block->i_nb_samples;
 343     out_buf.p_buffer = p_out->p_buffer;
 344     out_buf.i_buffer = p_out->i_buffer;
 345     out_buf.i_nb_samples = p_out->i_nb_samples;
 346
 347     DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
 348
 349     block_Release( p_block );
 350
 351     p_out->i_buffer = out_buf.i_buffer;
 352     p_out->i_nb_samples = out_buf.i_nb_samples;
 353
 354     return p_out;
 355 }