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  OpenFilter ( vlc_object_t * );
  43 static void CloseFilter( vlc_object_t * );
  44 static block_t *Resample( filter_t *, block_t * );
  45
  46 /*****************************************************************************
  47  * Local structures
  48  *****************************************************************************/
  49 struct filter_sys_t
  50 {
  51     int32_t *p_prev_sample;       /* this filter introduces a 1 sample delay */
  52
  53     unsigned int i_remainder;                /* remainder of previous sample */
  54
  55     date_t       end_date;
  56 };
  57
  58 /*****************************************************************************
  59  * Module descriptor
  60  *****************************************************************************/
  61 vlc_module_begin ()
  62     set_description( N_("Audio filter for linear interpolation resampling") )
  63     set_category( CAT_AUDIO )
  64     set_subcategory( SUBCAT_AUDIO_MISC )
  65     set_capability( "audio filter2", 5 )
  66     set_callbacks( OpenFilter, CloseFilter )
  67 vlc_module_end ()
  68
  69 /*****************************************************************************
  70  * Resample: convert a buffer
  71  *****************************************************************************/
  72 static block_t *Resample( filter_t *p_filter, block_t *p_in_buf )
  73 {
  74     if( !p_in_buf || !p_in_buf->i_nb_samples )
  75     {
  76         if( p_in_buf )
  77             block_Release( p_in_buf );
  78         return NULL;
  79     }
  80
  81     filter_sys_t *p_sys = p_filter->p_sys;
  82     unsigned i_nb_channels = p_filter->fmt_in.audio.i_channels;
  83     float *p_prev_sample = (float *)p_sys->p_prev_sample;
  84
  85     /* Check if we really need to run the resampler */
  86     if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
  87     {
  88         if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) )
  89         {
  90             p_in_buf = block_Realloc( p_in_buf, sizeof(float) * i_nb_channels,
  91                                       p_in_buf->i_buffer );
  92             if( !p_in_buf )
  93                 return NULL;
  94
  95             memcpy( p_in_buf->p_buffer, p_prev_sample,
  96                     i_nb_channels * sizeof(float) );
  97         }
  98         return p_in_buf;
  99     }
 100
 101     unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 102                                  p_filter->fmt_out.audio.i_bitspersample / 8;
 103
 104     size_t i_out_size = i_bytes_per_frame * (1 + (p_in_buf->i_nb_samples *
 105               p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
 106     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
 107     if( !p_out_buf )
 108         goto out;
 109
 110     float *p_out = (float *)p_out_buf->p_buffer;
 111
 112     unsigned i_in_nb = p_in_buf->i_nb_samples;
 113     unsigned i_out = 0;
 114     const float *p_in = (float *)p_in_buf->p_buffer;
 115
 116     /* Take care of the previous input sample (if any) */
 117     if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
 118     {
 119         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 120         p_sys->i_remainder = 0;
 121         date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
 122     }
 123     else
 124     {
 125         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 126         {
 127             for( unsigned i_chan = i_nb_channels ; i_chan ; )
 128             {
 129                 i_chan--;
 130                 p_out[i_chan] = p_prev_sample[i_chan];
 131                 p_out[i_chan] += ( ( p_in[i_chan] - p_prev_sample[i_chan] )
 132                                    * p_sys->i_remainder
 133                                    / p_filter->fmt_out.audio.i_rate );
 134             }
 135             p_out += i_nb_channels;
 136             i_out++;
 137
 138             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 139         }
 140         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 141     }
 142
 143     /* Take care of the current input samples (minus last one) */
 144     for( unsigned i_in = 0; i_in < i_in_nb - 1; i_in++ )
 145     {
 146         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 147         {
 148             for( unsigned i_chan = i_nb_channels ; i_chan ; )
 149             {
 150                 i_chan--;
 151                 p_out[i_chan] = p_in[i_chan];
 152                 p_out[i_chan] += ( ( p_in[i_chan + i_nb_channels]
 153                     - p_in[i_chan] )
 154                     * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate );
 155             }
 156             p_out += i_nb_channels;
 157             i_out++;
 158
 159             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 160         }
 161
 162         p_in += i_nb_channels;
 163         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 164     }
 165
 166     /* Backup the last input sample for next time */
 167     for( unsigned i_chan = i_nb_channels ; i_chan ; )
 168     {
 169         i_chan--;
 170         p_prev_sample[i_chan] = p_in[i_chan];
 171     }
 172
 173     p_out_buf->i_nb_samples = i_out;
 174     p_out_buf->i_pts = p_in_buf->i_pts;
 175
 176     if( p_in_buf->i_pts !=
 177         date_Get( &p_sys->end_date ) )
 178     {
 179         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 180     }
 181
 182     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 183                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 184
 185     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 186         i_nb_channels * sizeof(int32_t);
 187 out:
 188     block_Release( p_in_buf );
 189     return p_out_buf;
 190 }
 191
 192 /*****************************************************************************
 193  * OpenFilter:
 194  *****************************************************************************/
 195 static int OpenFilter( vlc_object_t *p_this )
 196 {
 197     filter_t *p_filter = (filter_t *)p_this;
 198     filter_sys_t *p_sys;
 199     int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 200
 201     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 202         p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
 203     {
 204         return VLC_EGENERIC;
 205     }
 206
 207     /* Allocate the memory needed to store the module's structure */
 208     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 209     if( p_sys == NULL )
 210         return VLC_ENOMEM;
 211
 212     p_sys->p_prev_sample = malloc(
 213         p_filter->fmt_in.audio.i_channels * sizeof(int32_t) );
 214     if( p_sys->p_prev_sample == NULL )
 215     {
 216         free( p_sys );
 217         return VLC_ENOMEM;
 218     }
 219     date_Init( &p_sys->end_date, p_filter->fmt_in.audio.i_rate, 1 );
 220     p_sys->i_remainder = 0;
 221
 222     p_filter->pf_audio_filter = Resample;
 223
 224     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 225              (char *)&p_filter->fmt_in.i_codec,
 226              p_filter->fmt_in.audio.i_rate,
 227              p_filter->fmt_in.audio.i_channels,
 228              (char *)&p_filter->fmt_out.i_codec,
 229              p_filter->fmt_out.audio.i_rate,
 230              p_filter->fmt_out.audio.i_channels);
 231
 232     p_filter->fmt_out = p_filter->fmt_in;
 233     p_filter->fmt_out.audio.i_rate = i_out_rate;
 234
 235     return 0;
 236 }
 237
 238 /*****************************************************************************
 239  * CloseFilter : deallocate data structures
 240  *****************************************************************************/
 241 static void CloseFilter( vlc_object_t *p_this )
 242 {
 243     filter_t *p_filter = (filter_t *)p_this;
 244     free( p_filter->p_sys->p_prev_sample );
 245     free( p_filter->p_sys );
 246 }