modules/audio_filter/normvol.c

   1 /*****************************************************************************
   2  * normvol.c: volume normalizer
   3  *****************************************************************************
   4  * Copyright (C) 2001, 2006 VLC authors and VideoLAN
   5  *
   6  * Authors: Clément Stenac <zorglub@videolan.org>
   7  *
   8  * This program is free software; you can redistribute it and/or modify it
   9  * under the terms of the GNU Lesser General Public License as published by
  10  * the Free Software Foundation; either version 2.1 of the License, or
  11  * (at your option) any later version.
  12  *
  13  * This program is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
  16  * GNU Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public License
  19  * along with this program; if not, write to the Free Software Foundation,
  20  * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  21  *****************************************************************************/
  22
  23 /*
  24  * TODO:
  25  *
  26  * We should detect fast power increases and react faster to these
  27  * This way, we can increase the buffer size to get a more stable filter */
  28
  29
  30 /*****************************************************************************
  31  * Preamble
  32  *****************************************************************************/
  33
  34 #ifdef HAVE_CONFIG_H
  35 # include "config.h"
  36 #endif
  37
  38 #include <math.h>
  39
  40 #include <vlc_common.h>
  41 #include <vlc_plugin.h>
  42
  43 #include <vlc_aout.h>
  44 #include <vlc_filter.h>
  45
  46 /*****************************************************************************
  47  * Local prototypes
  48  *****************************************************************************/
  49
  50 static int  Open     ( vlc_object_t * );
  51 static void Close    ( filter_t * );
  52 static block_t *DoWork( filter_t *, block_t * );
  53
  54 typedef struct
  55 {
  56     int i_nb;
  57     float *p_last;
  58     float f_max;
  59 } filter_sys_t;
  60
  61 /*****************************************************************************
  62  * Module descriptor
  63  *****************************************************************************/
  64 #define BUFF_TEXT N_("Number of audio buffers" )
  65 #define BUFF_LONGTEXT N_("This is the number of audio buffers on which the " \
  66                 "power measurement is made. A higher number of buffers will " \
  67                 "increase the response time of the filter to a spike " \
  68                 "but will make it less sensitive to short variations." )
  69
  70 #define LEVEL_TEXT N_("Maximal volume level" )
  71 #define LEVEL_LONGTEXT N_("If the average power over the last N buffers " \
  72                "is higher than this value, the volume will be normalized. " \
  73                "This value is a positive floating point number. A value " \
  74                "between 0.5 and 10 seems sensible." )
  75
  76 vlc_module_begin ()
  77     set_description( N_("Volume normalizer") )
  78     set_shortname( N_("Volume normalizer") )
  79     set_category( CAT_AUDIO )
  80     set_subcategory( SUBCAT_AUDIO_AFILTER )
  81     add_shortcut( "volnorm" )
  82     add_integer( "norm-buff-size", 20  ,BUFF_TEXT, BUFF_LONGTEXT,
  83                  true )
  84     add_float( "norm-max-level", 2.0, LEVEL_TEXT,
  85                LEVEL_LONGTEXT, true )
  86     set_capability( "audio filter", 0 )
  87     set_callback( Open )
  88 vlc_module_end ()
  89
  90 /*****************************************************************************
  91  * Open: initialize and create stuff
  92  *****************************************************************************/
  93 static int Open( vlc_object_t *p_this )
  94 {
  95     filter_t *p_filter = (filter_t*)p_this;
  96     unsigned i_channels;
  97     filter_sys_t *p_sys;
  98
  99     i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 100
 101     p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
 102     if( !p_sys )
 103         return VLC_ENOMEM;
 104     p_sys->i_nb = var_CreateGetInteger( vlc_object_parent(p_filter),
 105                                         "norm-buff-size" );
 106     p_sys->f_max = var_CreateGetFloat( vlc_object_parent(p_filter),
 107                                        "norm-max-level" );
 108
 109     if( p_sys->f_max <= 0 ) p_sys->f_max = 0.01;
 110
 111     /* We need to store (nb_buffers+1)*nb_channels floats */
 112     p_sys->p_last = calloc( i_channels * (p_sys->i_nb + 2), sizeof(float) );
 113     if( !p_sys->p_last )
 114     {
 115         free( p_sys );
 116         return VLC_ENOMEM;
 117     }
 118
 119     p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
 120     aout_FormatPrepare(&p_filter->fmt_in.audio);
 121     p_filter->fmt_out.audio = p_filter->fmt_in.audio;
 122     static const struct vlc_filter_operations filter_ops =
 123     {
 124         .filter_audio = DoWork, .close = Close,
 125     };
 126     p_filter->ops = &filter_ops;
 127
 128     return VLC_SUCCESS;
 129 }
 130
 131 /*****************************************************************************
 132  * DoWork : normalizes and sends a buffer
 133  *****************************************************************************/
 134 static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
 135 {
 136     float *pf_sum;
 137     float *pf_gain;
 138     float f_average = 0;
 139     int i, i_chan;
 140
 141     int i_samples = p_in_buf->i_nb_samples;
 142     int i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 143     float *p_out = (float*)p_in_buf->p_buffer;
 144     float *p_in =  (float*)p_in_buf->p_buffer;
 145
 146     filter_sys_t *p_sys = p_filter->p_sys;
 147
 148     pf_sum = calloc( i_channels, sizeof(float) );
 149     if( !pf_sum )
 150         goto out;
 151
 152     pf_gain = vlc_alloc( i_channels, sizeof(float) );
 153     if( !pf_gain )
 154     {
 155         free( pf_sum );
 156         goto out;
 157     }
 158
 159     /* Calculate the average power level on this buffer */
 160     for( i = 0 ; i < i_samples; i++ )
 161     {
 162         for( i_chan = 0; i_chan < i_channels; i_chan++ )
 163         {
 164             float f_sample = p_in[i_chan];
 165             pf_sum[i_chan] += f_sample * f_sample;
 166         }
 167         p_in += i_channels;
 168     }
 169
 170     /* sum now contains for each channel the sigma(value²) */
 171     for( i_chan = 0; i_chan < i_channels; i_chan++ )
 172     {
 173         /* Shift our lastbuff */
 174         memmove( &p_sys->p_last[ i_chan * p_sys->i_nb],
 175                         &p_sys->p_last[i_chan * p_sys->i_nb + 1],
 176                  (p_sys->i_nb-1) * sizeof( float ) );
 177
 178         /* Insert the new average : sqrt(sigma(value²)) */
 179         p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] =
 180                 sqrt( pf_sum[i_chan] );
 181
 182         pf_sum[i_chan] = 0;
 183
 184         /* Get the average power on the lastbuff */
 185         f_average = 0;
 186         for( i = 0; i < p_sys->i_nb ; i++)
 187         {
 188             f_average += p_sys->p_last[ i_chan * p_sys->i_nb + i ];
 189         }
 190         f_average = f_average / p_sys->i_nb;
 191
 192         /* Seuil arbitraire */
 193         p_sys->f_max = var_GetFloat( vlc_object_parent(p_filter),
 194                                      "norm-max-level" );
 195
 196         //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
 197         if( f_average > p_sys->f_max )
 198         {
 199              pf_gain[i_chan] = f_average / p_sys->f_max;
 200         }
 201         else
 202         {
 203            pf_gain[i_chan] = 1;
 204         }
 205     }
 206
 207     /* Apply gain */
 208     for( i = 0; i < i_samples; i++)
 209     {
 210         for( i_chan = 0; i_chan < i_channels; i_chan++ )
 211         {
 212             p_out[i_chan] /= pf_gain[i_chan];
 213         }
 214         p_out += i_channels;
 215     }
 216
 217     free( pf_sum );
 218     free( pf_gain );
 219
 220     return p_in_buf;
 221 out:
 222     block_Release( p_in_buf );
 223     return NULL;
 224 }
 225
 226 /**********************************************************************
 227  * Close
 228  **********************************************************************/
 229 static void Close( filter_t *p_filter )
 230 {
 231     filter_sys_t *p_sys = p_filter->p_sys;
 232
 233     free( p_sys->p_last );
 234     free( p_sys );
 235 }