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 #include <vlc_cpu.h>
  39
  40 /*****************************************************************************
  41  * Local prototypes
  42  *****************************************************************************/
  43 static int  OpenFilter ( vlc_object_t * );
  44 static void CloseFilter( vlc_object_t * );
  45 static block_t *Resample( filter_t *, block_t * );
  46
  47 #if HAVE_FPU
  48 typedef float sample_t;
  49 # define VLC_CODEC_NATIVE VLC_CODEC_FL32
  50 #else
  51 typedef int32_t sample_t;
  52 # define VLC_CODEC_NATIVE VLC_CODEC_FI32
  53 #endif
  54
  55 /*****************************************************************************
  56  * Local structures
  57  *****************************************************************************/
  58 struct filter_sys_t
  59 {
  60     sample_t *p_prev_sample;      /* this filter introduces a 1 sample delay */
  61
  62     unsigned int i_remainder;                /* remainder of previous sample */
  63
  64     date_t       end_date;
  65 };
  66
  67 /*****************************************************************************
  68  * Module descriptor
  69  *****************************************************************************/
  70 vlc_module_begin ()
  71     set_description( N_("Audio filter for linear interpolation resampling") )
  72     set_category( CAT_AUDIO )
  73     set_subcategory( SUBCAT_AUDIO_MISC )
  74     set_capability( "audio filter", 5 )
  75     set_callbacks( OpenFilter, CloseFilter )
  76 vlc_module_end ()
  77
  78 /*****************************************************************************
  79  * Resample: convert a buffer
  80  *****************************************************************************/
  81 static block_t *Resample( filter_t *p_filter, block_t *p_in_buf )
  82 {
  83     if( !p_in_buf || !p_in_buf->i_nb_samples )
  84     {
  85         if( p_in_buf )
  86             block_Release( p_in_buf );
  87         return NULL;
  88     }
  89
  90     filter_sys_t *p_sys = p_filter->p_sys;
  91     unsigned i_nb_channels = p_filter->fmt_in.audio.i_channels;
  92     sample_t *p_prev_sample = p_sys->p_prev_sample;
  93
  94     /* Check if we really need to run the resampler */
  95     if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
  96     {
  97         if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) )
  98         {
  99             p_in_buf = block_Realloc( p_in_buf,
 100                                       sizeof(sample_t) * i_nb_channels,
 101                                       p_in_buf->i_buffer );
 102             if( !p_in_buf )
 103                 return NULL;
 104
 105             memcpy( p_in_buf->p_buffer, p_prev_sample,
 106                     i_nb_channels * sizeof(sample_t) );
 107         }
 108         return p_in_buf;
 109     }
 110
 111     unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
 112                                  p_filter->fmt_out.audio.i_bitspersample / 8;
 113
 114     size_t i_out_size = i_bytes_per_frame * (1 + (p_in_buf->i_nb_samples *
 115               p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
 116     block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
 117     if( !p_out_buf )
 118         goto out;
 119
 120     sample_t *p_out = (sample_t *)p_out_buf->p_buffer;
 121
 122     unsigned i_in_nb = p_in_buf->i_nb_samples;
 123     unsigned i_out = 0;
 124     const sample_t *p_in = (sample_t *)p_in_buf->p_buffer;
 125
 126     /* Take care of the previous input sample (if any) */
 127     if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
 128     {
 129         p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
 130         p_sys->i_remainder = 0;
 131         date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
 132     }
 133     else
 134     {
 135         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 136         {
 137             for( unsigned i = 0; i < i_nb_channels ; i++ )
 138             {
 139                 p_out[i] = p_prev_sample[i];
 140 #if HAVE_FPU
 141                 p_out[i] += (p_in[i] - p_prev_sample[i])
 142 #else
 143                 p_out[i] += (int64_t)(p_in[i] - p_prev_sample[i])
 144 #endif
 145                     * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
 146             }
 147             p_out += i_nb_channels;
 148             i_out++;
 149
 150             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 151         }
 152         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 153     }
 154
 155     /* Take care of the current input samples (minus last one) */
 156     for( unsigned i_in = 0; i_in < i_in_nb - 1; i_in++ )
 157     {
 158         while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
 159         {
 160             for( unsigned i = 0; i < i_nb_channels ; i++ )
 161             {
 162                 p_out[i] = p_in[i];
 163 #if HAVE_FPU
 164                 p_out[i] += (p_in[i + i_nb_channels] - p_in[i])
 165 #else
 166                 p_out[i] += (int64_t)(p_in[i + i_nb_channels] - p_in[i])
 167 #endif
 168                     * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
 169             }
 170             p_out += i_nb_channels;
 171             i_out++;
 172
 173             p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
 174         }
 175
 176         p_in += i_nb_channels;
 177         p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
 178     }
 179
 180     /* Backup the last input sample for next time */
 181     memcpy( p_prev_sample, p_in, i_nb_channels * sizeof(sample_t) );
 182
 183     p_out_buf->i_nb_samples = i_out;
 184     p_out_buf->i_pts = p_in_buf->i_pts;
 185
 186     if( p_in_buf->i_pts !=
 187         date_Get( &p_sys->end_date ) )
 188     {
 189         date_Set( &p_sys->end_date, p_in_buf->i_pts );
 190     }
 191
 192     p_out_buf->i_length = date_Increment( &p_sys->end_date,
 193                                   p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
 194
 195     p_out_buf->i_buffer = p_out_buf->i_nb_samples *
 196         i_nb_channels * sizeof(sample_t);
 197 out:
 198     block_Release( p_in_buf );
 199     return p_out_buf;
 200 }
 201
 202 /*****************************************************************************
 203  * OpenFilter:
 204  *****************************************************************************/
 205 static int OpenFilter( vlc_object_t *p_this )
 206 {
 207     filter_t *p_filter = (filter_t *)p_this;
 208     filter_sys_t *p_sys;
 209     int i_out_rate  = p_filter->fmt_out.audio.i_rate;
 210
 211     if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
 212         p_filter->fmt_in.i_codec != VLC_CODEC_NATIVE )
 213     {
 214         return VLC_EGENERIC;
 215     }
 216
 217     /* Allocate the memory needed to store the module's structure */
 218     p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
 219     if( p_sys == NULL )
 220         return VLC_ENOMEM;
 221
 222     p_sys->p_prev_sample = malloc(
 223         p_filter->fmt_in.audio.i_channels * sizeof(sample_t) );
 224     if( p_sys->p_prev_sample == NULL )
 225     {
 226         free( p_sys );
 227         return VLC_ENOMEM;
 228     }
 229     date_Init( &p_sys->end_date, p_filter->fmt_in.audio.i_rate, 1 );
 230     p_sys->i_remainder = 0;
 231
 232     p_filter->pf_audio_filter = Resample;
 233
 234     msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
 235              (char *)&p_filter->fmt_in.i_codec,
 236              p_filter->fmt_in.audio.i_rate,
 237              p_filter->fmt_in.audio.i_channels,
 238              (char *)&p_filter->fmt_out.i_codec,
 239              p_filter->fmt_out.audio.i_rate,
 240              p_filter->fmt_out.audio.i_channels);
 241
 242     p_filter->fmt_out = p_filter->fmt_in;
 243     p_filter->fmt_out.audio.i_rate = i_out_rate;
 244
 245     return 0;
 246 }
 247
 248 /*****************************************************************************
 249  * CloseFilter : deallocate data structures
 250  *****************************************************************************/
 251 static void CloseFilter( vlc_object_t *p_this )
 252 {
 253     filter_t *p_filter = (filter_t *)p_this;
 254     free( p_filter->p_sys->p_prev_sample );
 255     free( p_filter->p_sys );
 256 }