1 /*****************************************************************************
2 * normvol.c: volume normalizer
3 *****************************************************************************
4 * Copyright (C) 2001, 2006 VLC authors and VideoLAN
7 * Authors: Clément Stenac <zorglub@videolan.org>
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.
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.
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 *****************************************************************************/
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 */
31 /*****************************************************************************
33 *****************************************************************************/
41 #include <vlc_common.h>
42 #include <vlc_plugin.h>
45 #include <vlc_filter.h>
47 /*****************************************************************************
49 *****************************************************************************/
51 static int Open ( vlc_object_t
* );
52 static void Close ( vlc_object_t
* );
53 static block_t
*DoWork( filter_t
*, block_t
* );
62 /*****************************************************************************
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." )
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." )
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
,
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
)
91 /*****************************************************************************
92 * Open: initialize and create stuff
93 *****************************************************************************/
94 static int Open( vlc_object_t
*p_this
)
96 filter_t
*p_filter
= (filter_t
*)p_this
;
100 i_channels
= aout_FormatNbChannels( &p_filter
->fmt_in
.audio
);
102 p_sys
= p_filter
->p_sys
= malloc( sizeof( *p_sys
) );
105 p_sys
->i_nb
= var_CreateGetInteger( p_filter
->obj
.parent
,
107 p_sys
->f_max
= var_CreateGetFloat( p_filter
->obj
.parent
,
110 if( p_sys
->f_max
<= 0 ) p_sys
->f_max
= 0.01;
112 /* We need to store (nb_buffers+1)*nb_channels floats */
113 p_sys
->p_last
= calloc( i_channels
* (p_filter
->p_sys
->i_nb
+ 2), sizeof(float) );
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
;
128 /*****************************************************************************
129 * DoWork : normalizes and sends a buffer
130 *****************************************************************************/
131 static block_t
*DoWork( filter_t
*p_filter
, block_t
*p_in_buf
)
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
;
143 struct filter_sys_t
*p_sys
= p_filter
->p_sys
;
145 pf_sum
= calloc( i_channels
, sizeof(float) );
149 pf_gain
= vlc_alloc( i_channels
, sizeof(float) );
156 /* Calculate the average power level on this buffer */
157 for( i
= 0 ; i
< i_samples
; i
++ )
159 for( i_chan
= 0; i_chan
< i_channels
; i_chan
++ )
161 float f_sample
= p_in
[i_chan
];
162 pf_sum
[i_chan
] += f_sample
* f_sample
;
167 /* sum now contains for each channel the sigma(value²) */
168 for( i_chan
= 0; i_chan
< i_channels
; i_chan
++ )
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 ) );
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
] );
181 /* Get the average power on the lastbuff */
183 for( i
= 0; i
< p_sys
->i_nb
; i
++)
185 f_average
+= p_sys
->p_last
[ i_chan
* p_sys
->i_nb
+ i
];
187 f_average
= f_average
/ p_sys
->i_nb
;
189 /* Seuil arbitraire */
190 p_sys
->f_max
= var_GetFloat( p_filter
->obj
.parent
,
193 //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
194 if( f_average
> p_sys
->f_max
)
196 pf_gain
[i_chan
] = f_average
/ p_sys
->f_max
;
205 for( i
= 0; i
< i_samples
; i
++)
207 for( i_chan
= 0; i_chan
< i_channels
; i_chan
++ )
209 p_out
[i_chan
] /= pf_gain
[i_chan
];
219 block_Release( p_in_buf
);
223 /**********************************************************************
225 **********************************************************************/
226 static void Close( vlc_object_t
*p_this
)
228 filter_t
*p_filter
= (filter_t
*)p_this
;
229 filter_sys_t
*p_sys
= p_filter
->p_sys
;
231 free( p_sys
->p_last
);