1 /*****************************************************************************
2 * normvol.c: volume normalizer
3 *****************************************************************************
4 * Copyright (C) 2001, 2006 the VideoLAN team
7 * Authors: Clément Stenac <zorglub@videolan.org>
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.
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.
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 *****************************************************************************/
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 ,NULL
,BUFF_TEXT
, BUFF_LONGTEXT
,
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
)
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 if( p_filter
->fmt_in
.audio
.i_format
!= VLC_CODEC_FL32
||
101 p_filter
->fmt_out
.audio
.i_format
!= VLC_CODEC_FL32
)
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" );
109 if ( !AOUT_FMTS_SIMILAR( &p_filter
->fmt_in
.audio
, &p_filter
->fmt_out
.audio
) )
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" );
117 p_filter
->pf_audio_filter
= DoWork
;
119 i_channels
= aout_FormatNbChannels( &p_filter
->fmt_in
.audio
);
121 p_sys
= p_filter
->p_sys
= malloc( sizeof( *p_sys
) );
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" );
127 if( p_sys
->f_max
<= 0 ) p_sys
->f_max
= 0.01;
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) );
140 /*****************************************************************************
141 * DoWork : normalizes and sends a buffer
142 *****************************************************************************/
143 static block_t
*DoWork( filter_t
*p_filter
, block_t
*p_in_buf
)
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
;
155 struct filter_sys_t
*p_sys
= p_filter
->p_sys
;
157 pf_sum
= calloc( i_channels
, sizeof(float) );
161 pf_gain
= malloc( sizeof(float) * i_channels
);
168 /* Calculate the average power level on this buffer */
169 for( i
= 0 ; i
< i_samples
; i
++ )
171 for( i_chan
= 0; i_chan
< i_channels
; i_chan
++ )
173 float f_sample
= p_in
[i_chan
];
174 float f_square
= pow( f_sample
, 2 );
175 pf_sum
[i_chan
] += f_square
;
180 /* sum now contains for each channel the sigma(value²) */
181 for( i_chan
= 0; i_chan
< i_channels
; i_chan
++ )
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 ) );
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
] );
194 /* Get the average power on the lastbuff */
196 for( i
= 0; i
< p_sys
->i_nb
; i
++)
198 f_average
+= p_sys
->p_last
[ i_chan
* p_sys
->i_nb
+ i
];
200 f_average
= f_average
/ p_sys
->i_nb
;
202 /* Seuil arbitraire */
203 p_sys
->f_max
= var_GetFloat( p_filter
->p_parent
, "norm-max-level" );
205 //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
206 if( f_average
> p_sys
->f_max
)
208 pf_gain
[i_chan
] = f_average
/ p_sys
->f_max
;
217 for( i
= 0; i
< i_samples
; i
++)
219 for( i_chan
= 0; i_chan
< i_channels
; i_chan
++ )
221 p_out
[i_chan
] /= pf_gain
[i_chan
];
231 block_Release( p_in_buf
);
235 /**********************************************************************
237 **********************************************************************/
238 static void Close( vlc_object_t
*p_this
)
240 filter_t
*p_filter
= (filter_t
*)p_this
;
241 filter_sys_t
*p_sys
= p_filter
->p_sys
;
243 free( p_sys
->p_last
);