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