/*****************************************************************************
- * normvol.c : volume normalizer
+ * normvol.c: volume normalizer
*****************************************************************************
- * Copyright (C) 2001 VideoLAN
- * $Id: equalizer.c,v 1.21 2003/07/31 23:44:49 zorglub Exp $
+ * Copyright (C) 2001, 2006 the VideoLAN team
+ * $Id$
*
- * Authors: Clément Stenac <zorglub@videolan.org>
+ * Authors: Clément Stenac <zorglub@videolan.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*
/*****************************************************************************
* Preamble
*****************************************************************************/
-#include <stdlib.h> /* malloc(), free() */
-#include <string.h>
#include <errno.h> /* ENOMEM */
-#include <stdio.h>
#include <ctype.h>
#include <signal.h>
#include <math.h>
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
#include <vlc/vlc.h>
-#include <vlc/aout.h>
-#include <aout_internal.h>
+#include <vlc_aout.h>
/*****************************************************************************
* Local prototypes
static void DoWork ( aout_instance_t * , aout_filter_t *,
aout_buffer_t * , aout_buffer_t *);
-struct aout_filter_sys_t {
+typedef struct aout_filter_sys_t
+{
int i_nb;
float *p_last;
float f_max;
-};
+} aout_filter_sys_t;
/*****************************************************************************
* Module descriptor
#define BUFF_TEXT N_("Number of audio buffers" )
#define BUFF_LONGTEXT N_("This is the number of audio buffers on which the " \
"power measurement is made. A higher number of buffers will " \
- "increase the response time of the filter to a high " \
- "power but will make it less sensitive to short variations " )
+ "increase the response time of the filter to a spike " \
+ "but will make it less sensitive to short variations." )
#define LEVEL_TEXT N_("Max level" )
#define LEVEL_LONGTEXT N_("If the average power over the last N buffers " \
- "is higher than this value, the volume will be normalized." \
+ "is higher than this value, the volume will be normalized. " \
"This value is a positive floating point number. A value " \
"between 0.5 and 10 seems sensible." )
vlc_module_begin();
set_description( _("Volume normalizer") );
- add_shortcut("volnorm");
- add_integer("norm-buff-size" , 20 , NULL , BUFF_TEXT, BUFF_LONGTEXT,
- VLC_TRUE);
- add_float("norm-max-level",2.0,NULL,LEVEL_TEXT,LEVEL_LONGTEXT,VLC_TRUE);
+ set_shortname( _("Volume normalizer") );
+ set_category( CAT_AUDIO );
+ set_subcategory( SUBCAT_AUDIO_AFILTER );
+ add_shortcut( "volnorm" );
+ add_integer( "norm-buff-size", 20 ,NULL ,BUFF_TEXT, BUFF_LONGTEXT,
+ VLC_TRUE);
+ add_float( "norm-max-level", 2.0, NULL, LEVEL_TEXT,
+ LEVEL_LONGTEXT, VLC_TRUE );
set_capability( "audio filter", 0 );
set_callbacks( Open, Close );
vlc_module_end();
static int Open( vlc_object_t *p_this )
{
aout_filter_t *p_filter = (aout_filter_t*)p_this;
-
- struct aout_filter_sys_t *p_sys = p_filter->p_sys =
- malloc( sizeof( struct aout_filter_sys_t ) );
+ vlc_bool_t b_fit = VLC_TRUE;
+ int i_channels;
+ aout_filter_sys_t *p_sys;
if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||
p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )
{
- msg_Warn( p_filter, "Bad input or output format" );
- return VLC_EGENERIC;
+ b_fit = VLC_FALSE;
+ p_filter->input.i_format = VLC_FOURCC('f','l','3','2');
+ p_filter->output.i_format = VLC_FOURCC('f','l','3','2');
+ msg_Warn( p_filter, "bad input or output format" );
}
if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
{
+ b_fit = VLC_FALSE;
+ memcpy( &p_filter->output, &p_filter->input,
+ sizeof(audio_sample_format_t) );
msg_Warn( p_filter, "input and output formats are not similar" );
+ }
+
+ if ( ! b_fit )
+ {
return VLC_EGENERIC;
}
p_filter->pf_do_work = DoWork;
p_filter->b_in_place = VLC_TRUE;
- int i_channels = aout_FormatNbChannels( &p_filter->input );
+ i_channels = aout_FormatNbChannels( &p_filter->input );
- p_sys->i_nb = var_CreateGetInteger( p_filter->p_parent, "norm-buff-size" );
+ p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
+ p_sys->i_nb = var_CreateGetInteger( p_filter->p_parent, "norm-buff-size" );
p_sys->f_max = var_CreateGetFloat( p_filter->p_parent, "norm-max-level" );
- if( p_sys->f_max <= 0 )
- {
- p_sys->f_max = 0.01;
- }
+ if( p_sys->f_max <= 0 ) p_sys->f_max = 0.01;
/* We need to store (nb_buffers+1)*nb_channels floats */
p_sys->p_last = malloc( sizeof( float ) * (i_channels) *
(p_filter->p_sys->i_nb + 2) );
- memset( p_sys->p_last, 0 , sizeof( float ) * (i_channels) *
- (p_filter->p_sys->i_nb + 2) );
+ memset( p_sys->p_last, 0 ,sizeof( float ) * (i_channels) *
+ (p_filter->p_sys->i_nb + 2) );
return VLC_SUCCESS;
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
- /* Calculate the average power level on this buffer */
- for( i = 0 ; i < i_samples; i++ )
- {
+ /* Calculate the average power level on this buffer */
+ for( i = 0 ; i < i_samples; i++ )
+ {
for( i_chan = 0; i_chan < i_channels; i_chan++ )
{
float f_sample = p_in[i_chan];
pf_sum[i_chan] += f_square;
}
p_in += i_channels;
- }
+ }
- /* sum now contains for each channel the sigma(value²) */
- for( i_chan = 0; i_chan < i_channels; i_chan++ )
- {
+ /* sum now contains for each channel the sigma(value²) */
+ for( i_chan = 0; i_chan < i_channels; i_chan++ )
+ {
/* Shift our lastbuff */
memmove( &p_sys->p_last[ i_chan * p_sys->i_nb],
&p_sys->p_last[i_chan * p_sys->i_nb + 1],
(p_sys->i_nb-1) * sizeof( float ) );
- /* Insert the new average : sqrt(sigma(value²)) */
+ /* Insert the new average : sqrt(sigma(value²)) */
p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] =
sqrt( pf_sum[i_chan] );
/* Seuil arbitraire */
p_sys->f_max = var_GetFloat( p_aout, "norm-max-level" );
-// fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
+ //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
if( f_average > p_sys->f_max )
{
pf_gain[i_chan] = f_average / p_sys->f_max;
}
/* Apply gain */
- for( i = 0 ; i < i_samples ; i++)
+ for( i = 0; i < i_samples; i++)
{
for( i_chan = 0; i_chan < i_channels; i_chan++ )
{
static void Close( vlc_object_t *p_this )
{
aout_filter_t *p_filter = (aout_filter_t*)p_this;
- struct aout_filter_sys_t *p_sys = p_filter->p_sys;
+ aout_filter_sys_t *p_sys = p_filter->p_sys;
if( p_sys )
{
- if( p_sys->p_last)
- {
- free( p_sys->p_last );
- }
+ free( p_sys->p_last );
free( p_sys );
}
}