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->p_parent, "norm-buff-size" );
106 p_sys->f_max = var_CreateGetFloat( p_filter->p_parent, "norm-max-level" );
108 if( p_sys->f_max <= 0 ) p_sys->f_max = 0.01;
110 /* We need to store (nb_buffers+1)*nb_channels floats */
111 p_sys->p_last = calloc( i_channels * (p_filter->p_sys->i_nb + 2), sizeof(float) );
118 p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
119 p_filter->fmt_out.audio = p_filter->fmt_in.audio;
120 p_filter->pf_audio_filter = DoWork;
125 /*****************************************************************************
126 * DoWork : normalizes and sends a buffer
127 *****************************************************************************/
128 static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
135 int i_samples = p_in_buf->i_nb_samples;
136 int i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
137 float *p_out = (float*)p_in_buf->p_buffer;
138 float *p_in = (float*)p_in_buf->p_buffer;
140 struct filter_sys_t *p_sys = p_filter->p_sys;
142 pf_sum = calloc( i_channels, sizeof(float) );
146 pf_gain = malloc( sizeof(float) * i_channels );
153 /* Calculate the average power level on this buffer */
154 for( i = 0 ; i < i_samples; i++ )
156 for( i_chan = 0; i_chan < i_channels; i_chan++ )
158 float f_sample = p_in[i_chan];
159 pf_sum[i_chan] += f_sample * f_sample;
164 /* sum now contains for each channel the sigma(value²) */
165 for( i_chan = 0; i_chan < i_channels; i_chan++ )
167 /* Shift our lastbuff */
168 memmove( &p_sys->p_last[ i_chan * p_sys->i_nb],
169 &p_sys->p_last[i_chan * p_sys->i_nb + 1],
170 (p_sys->i_nb-1) * sizeof( float ) );
172 /* Insert the new average : sqrt(sigma(value²)) */
173 p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] =
174 sqrt( pf_sum[i_chan] );
178 /* Get the average power on the lastbuff */
180 for( i = 0; i < p_sys->i_nb ; i++)
182 f_average += p_sys->p_last[ i_chan * p_sys->i_nb + i ];
184 f_average = f_average / p_sys->i_nb;
186 /* Seuil arbitraire */
187 p_sys->f_max = var_GetFloat( p_filter->p_parent, "norm-max-level" );
189 //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
190 if( f_average > p_sys->f_max )
192 pf_gain[i_chan] = f_average / p_sys->f_max;
201 for( i = 0; i < i_samples; i++)
203 for( i_chan = 0; i_chan < i_channels; i_chan++ )
205 p_out[i_chan] /= pf_gain[i_chan];
215 block_Release( p_in_buf );
219 /**********************************************************************
221 **********************************************************************/
222 static void Close( vlc_object_t *p_this )
224 filter_t *p_filter = (filter_t*)p_this;
225 filter_sys_t *p_sys = p_filter->p_sys;
227 free( p_sys->p_last );