modules/audio_filter/normvol.c

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