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 #include <stdlib.h>                                      /* malloc(), free() */
  35 #include <string.h>
  36
  37 #include <errno.h>                                                 /* ENOMEM */
  38 #include <stdio.h>
  39 #include <ctype.h>
  40 #include <signal.h>
  41
  42 #include <math.h>
  43
  44
  45 #include <vlc/vlc.h>
  46
  47 #include <vlc_aout.h>
  48
  49 /*****************************************************************************
  50  * Local prototypes
  51  *****************************************************************************/
  52
  53 static int  Open     ( vlc_object_t * );
  54 static void Close    ( vlc_object_t * );
  55 static void DoWork   ( aout_instance_t * , aout_filter_t *,
  56                 aout_buffer_t * , aout_buffer_t *);
  57
  58 typedef struct aout_filter_sys_t
  59 {
  60     int i_nb;
  61     float *p_last;
  62     float f_max;
  63 } aout_filter_sys_t;
  64
  65 /*****************************************************************************
  66  * Module descriptor
  67  *****************************************************************************/
  68 #define BUFF_TEXT N_("Number of audio buffers" )
  69 #define BUFF_LONGTEXT N_("This is the number of audio buffers on which the " \
  70                 "power measurement is made. A higher number of buffers will " \
  71                 "increase the response time of the filter to a spike " \
  72                 "but will make it less sensitive to short variations." )
  73
  74 #define LEVEL_TEXT N_("Max level" )
  75 #define LEVEL_LONGTEXT N_("If the average power over the last N buffers " \
  76                "is higher than this value, the volume will be normalized. " \
  77                "This value is a positive floating point number. A value " \
  78                "between 0.5 and 10 seems sensible." )
  79
  80 vlc_module_begin();
  81     set_description( _("Volume normalizer") );
  82     set_shortname( _("Volume normalizer") );
  83     set_category( CAT_AUDIO );
  84     set_subcategory( SUBCAT_AUDIO_AFILTER );
  85     add_shortcut( "volnorm" );
  86     add_integer( "norm-buff-size", 20 ,NULL ,BUFF_TEXT, BUFF_LONGTEXT,
  87                  VLC_TRUE);
  88     add_float( "norm-max-level", 2.0, NULL, LEVEL_TEXT,
  89                LEVEL_LONGTEXT, VLC_TRUE );
  90     set_capability( "audio filter", 0 );
  91     set_callbacks( Open, Close );
  92 vlc_module_end();
  93
  94 /*****************************************************************************
  95  * Open: initialize and create stuff
  96  *****************************************************************************/
  97 static int Open( vlc_object_t *p_this )
  98 {
  99     aout_filter_t *p_filter = (aout_filter_t*)p_this;
 100     vlc_bool_t b_fit = VLC_TRUE;
 101     int i_channels;
 102     aout_filter_sys_t *p_sys;
 103
 104     if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||
 105         p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )
 106     {
 107         b_fit = VLC_FALSE;
 108         p_filter->input.i_format = VLC_FOURCC('f','l','3','2');
 109         p_filter->output.i_format = VLC_FOURCC('f','l','3','2');
 110         msg_Warn( p_filter, "bad input or output format" );
 111     }
 112
 113     if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
 114     {
 115         b_fit = VLC_FALSE;
 116         memcpy( &p_filter->output, &p_filter->input,
 117                 sizeof(audio_sample_format_t) );
 118         msg_Warn( p_filter, "input and output formats are not similar" );
 119     }
 120
 121     if ( ! b_fit )
 122     {
 123         return VLC_EGENERIC;
 124     }
 125
 126     p_filter->pf_do_work = DoWork;
 127     p_filter->b_in_place = VLC_TRUE;
 128
 129     i_channels = aout_FormatNbChannels( &p_filter->input );
 130
 131     p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
 132     p_sys->i_nb = var_CreateGetInteger( p_filter->p_parent, "norm-buff-size" );
 133     p_sys->f_max = var_CreateGetFloat( p_filter->p_parent, "norm-max-level" );
 134
 135     if( p_sys->f_max <= 0 ) p_sys->f_max = 0.01;
 136
 137     /* We need to store (nb_buffers+1)*nb_channels floats */
 138     p_sys->p_last = malloc( sizeof( float ) * (i_channels) *
 139                             (p_filter->p_sys->i_nb + 2) );
 140     memset( p_sys->p_last, 0 ,sizeof( float ) * (i_channels) *
 141             (p_filter->p_sys->i_nb + 2) );
 142     return VLC_SUCCESS;
 143 }
 144
 145 /*****************************************************************************
 146  * DoWork : normalizes and sends a buffer
 147  *****************************************************************************/
 148  static void DoWork( aout_instance_t *p_aout, aout_filter_t *p_filter,
 149                      aout_buffer_t *p_in_buf, aout_buffer_t *p_out_buf )
 150 {
 151     float *pf_sum;
 152     float *pf_gain;
 153     float f_average = 0;
 154     int i, i_chan;
 155
 156     int i_samples = p_in_buf->i_nb_samples;
 157     int i_channels = aout_FormatNbChannels( &p_filter->input );
 158     float *p_out = (float*)p_out_buf->p_buffer;
 159     float *p_in =  (float*)p_in_buf->p_buffer;
 160
 161     struct aout_filter_sys_t *p_sys = p_filter->p_sys;
 162
 163     pf_sum = (float *)malloc( sizeof(float) * i_channels );
 164     memset( pf_sum, 0, sizeof(float) * i_channels );
 165
 166     pf_gain = (float *)malloc( sizeof(float) * i_channels );
 167
 168     p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
 169     p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
 170
 171     /* Calculate the average power level on this buffer */
 172     for( i = 0 ; i < i_samples; i++ )
 173     {
 174         for( i_chan = 0; i_chan < i_channels; i_chan++ )
 175         {
 176             float f_sample = p_in[i_chan];
 177             float f_square = pow( f_sample, 2 );
 178             pf_sum[i_chan] += f_square;
 179         }
 180         p_in += i_channels;
 181     }
 182
 183     /* sum now contains for each channel the sigma(value²) */
 184     for( i_chan = 0; i_chan < i_channels; i_chan++ )
 185     {
 186         /* Shift our lastbuff */
 187         memmove( &p_sys->p_last[ i_chan * p_sys->i_nb],
 188                         &p_sys->p_last[i_chan * p_sys->i_nb + 1],
 189                  (p_sys->i_nb-1) * sizeof( float ) );
 190
 191         /* Insert the new average : sqrt(sigma(value²)) */
 192         p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] =
 193                 sqrt( pf_sum[i_chan] );
 194
 195         pf_sum[i_chan] = 0;
 196
 197         /* Get the average power on the lastbuff */
 198         f_average = 0;
 199         for( i = 0; i < p_sys->i_nb ; i++)
 200         {
 201             f_average += p_sys->p_last[ i_chan * p_sys->i_nb + i ];
 202         }
 203         f_average = f_average / p_sys->i_nb;
 204
 205         /* Seuil arbitraire */
 206         p_sys->f_max = var_GetFloat( p_aout, "norm-max-level" );
 207
 208         //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
 209         if( f_average > p_sys->f_max )
 210         {
 211              pf_gain[i_chan] = f_average / p_sys->f_max;
 212         }
 213         else
 214         {
 215            pf_gain[i_chan] = 1;
 216         }
 217     }
 218
 219     /* Apply gain */
 220     for( i = 0; i < i_samples; i++)
 221     {
 222         for( i_chan = 0; i_chan < i_channels; i_chan++ )
 223         {
 224             p_out[i_chan] /= pf_gain[i_chan];
 225         }
 226         p_out += i_channels;
 227     }
 228
 229     free( pf_sum );
 230     free( pf_gain );
 231
 232     return;
 233 }
 234
 235 /**********************************************************************
 236  * Close
 237  **********************************************************************/
 238 static void Close( vlc_object_t *p_this )
 239 {
 240     aout_filter_t *p_filter = (aout_filter_t*)p_this;
 241     aout_filter_sys_t *p_sys = p_filter->p_sys;
 242
 243     if( p_sys )
 244     {
 245         if( p_sys->p_last) free( p_sys->p_last );
 246         free( p_sys );
 247     }
 248 }