modules/audio_filter/normvol.c

   1 /*****************************************************************************
   2  * normvol.c: volume normalizer
   3  *****************************************************************************
   4  * Copyright (C) 2001, 2006 the VideoLAN team
   5  * $Id$
   6  *
   7  * Authors: Clément Stenac <zorglub@videolan.org>
   8  *
   9  * This program is free software; you can redistribute it and/or modify
  10  * it under the terms of the GNU General Public License as published by
  11  * the Free Software Foundation; either version 2 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 General Public License for more details.
  18  *
  19  * You should have received a copy of the GNU General Public License
  20  * along with this program; if not, write to the Free Software
  21  * Foundation, 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 struct filter_sys_t
  56 {
  57     int i_nb;
  58     float *p_last;
  59     float f_max;
  60 };
  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 ,NULL ,BUFF_TEXT, BUFF_LONGTEXT,
  84                  true )
  85     add_float( "norm-max-level", 2.0, NULL, 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     if( p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ||
 101         p_filter->fmt_out.audio.i_format != VLC_CODEC_FL32 )
 102     {
 103         p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
 104         p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
 105         msg_Warn( p_filter, "bad input or output format" );
 106         return VLC_EGENERIC;
 107     }
 108
 109     if ( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio, &p_filter->fmt_out.audio ) )
 110     {
 111         memcpy( &p_filter->fmt_out.audio, &p_filter->fmt_in.audio,
 112                 sizeof(audio_sample_format_t) );
 113         msg_Warn( p_filter, "input and output formats are not similar" );
 114         return VLC_EGENERIC;
 115     }
 116
 117     p_filter->pf_audio_filter = DoWork;
 118
 119     i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 120
 121     p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
 122     if( !p_sys )
 123         return VLC_ENOMEM;
 124     p_sys->i_nb = var_CreateGetInteger( p_filter->p_parent, "norm-buff-size" );
 125     p_sys->f_max = var_CreateGetFloat( p_filter->p_parent, "norm-max-level" );
 126
 127     if( p_sys->f_max <= 0 ) p_sys->f_max = 0.01;
 128
 129     /* We need to store (nb_buffers+1)*nb_channels floats */
 130     p_sys->p_last = calloc( i_channels * (p_filter->p_sys->i_nb + 2), sizeof(float) );
 131     if( !p_sys->p_last )
 132     {
 133         free( p_sys );
 134         return VLC_ENOMEM;
 135     }
 136
 137     return VLC_SUCCESS;
 138 }
 139
 140 /*****************************************************************************
 141  * DoWork : normalizes and sends a buffer
 142  *****************************************************************************/
 143 static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
 144 {
 145     float *pf_sum;
 146     float *pf_gain;
 147     float f_average = 0;
 148     int i, i_chan;
 149
 150     int i_samples = p_in_buf->i_nb_samples;
 151     int i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
 152     float *p_out = (float*)p_in_buf->p_buffer;
 153     float *p_in =  (float*)p_in_buf->p_buffer;
 154
 155     struct filter_sys_t *p_sys = p_filter->p_sys;
 156
 157     pf_sum = calloc( i_channels, sizeof(float) );
 158     if( !pf_sum )
 159         goto out;
 160
 161     pf_gain = malloc( sizeof(float) * i_channels );
 162     if( !pf_gain )
 163     {
 164         free( pf_sum );
 165         goto out;
 166     }
 167
 168     /* Calculate the average power level on this buffer */
 169     for( i = 0 ; i < i_samples; i++ )
 170     {
 171         for( i_chan = 0; i_chan < i_channels; i_chan++ )
 172         {
 173             float f_sample = p_in[i_chan];
 174             float f_square = pow( f_sample, 2 );
 175             pf_sum[i_chan] += f_square;
 176         }
 177         p_in += i_channels;
 178     }
 179
 180     /* sum now contains for each channel the sigma(value²) */
 181     for( i_chan = 0; i_chan < i_channels; i_chan++ )
 182     {
 183         /* Shift our lastbuff */
 184         memmove( &p_sys->p_last[ i_chan * p_sys->i_nb],
 185                         &p_sys->p_last[i_chan * p_sys->i_nb + 1],
 186                  (p_sys->i_nb-1) * sizeof( float ) );
 187
 188         /* Insert the new average : sqrt(sigma(value²)) */
 189         p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] =
 190                 sqrt( pf_sum[i_chan] );
 191
 192         pf_sum[i_chan] = 0;
 193
 194         /* Get the average power on the lastbuff */
 195         f_average = 0;
 196         for( i = 0; i < p_sys->i_nb ; i++)
 197         {
 198             f_average += p_sys->p_last[ i_chan * p_sys->i_nb + i ];
 199         }
 200         f_average = f_average / p_sys->i_nb;
 201
 202         /* Seuil arbitraire */
 203         p_sys->f_max = var_GetFloat( p_filter->p_parent, "norm-max-level" );
 204
 205         //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
 206         if( f_average > p_sys->f_max )
 207         {
 208              pf_gain[i_chan] = f_average / p_sys->f_max;
 209         }
 210         else
 211         {
 212            pf_gain[i_chan] = 1;
 213         }
 214     }
 215
 216     /* Apply gain */
 217     for( i = 0; i < i_samples; i++)
 218     {
 219         for( i_chan = 0; i_chan < i_channels; i_chan++ )
 220         {
 221             p_out[i_chan] /= pf_gain[i_chan];
 222         }
 223         p_out += i_channels;
 224     }
 225
 226     free( pf_sum );
 227     free( pf_gain );
 228
 229     return p_in_buf;
 230 out:
 231     block_Release( p_in_buf );
 232     return NULL;
 233 }
 234
 235 /**********************************************************************
 236  * Close
 237  **********************************************************************/
 238 static void Close( vlc_object_t *p_this )
 239 {
 240     filter_t *p_filter = (filter_t*)p_this;
 241     filter_sys_t *p_sys = p_filter->p_sys;
 242
 243     free( p_sys->p_last );
 244     free( p_sys );
 245 }