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 *****************************************************************************/
39 #include <errno.h> /* ENOMEM */
46 #include <vlc_common.h>
47 #include <vlc_plugin.h>
50 #include <vlc_filter.h>
52 /*****************************************************************************
54 *****************************************************************************/
56 static int Open ( vlc_object_t * );
57 static void Close ( vlc_object_t * );
58 static block_t *DoWork( filter_t *, block_t * );
67 /*****************************************************************************
69 *****************************************************************************/
70 #define BUFF_TEXT N_("Number of audio buffers" )
71 #define BUFF_LONGTEXT N_("This is the number of audio buffers on which the " \
72 "power measurement is made. A higher number of buffers will " \
73 "increase the response time of the filter to a spike " \
74 "but will make it less sensitive to short variations." )
76 #define LEVEL_TEXT N_("Max level" )
77 #define LEVEL_LONGTEXT N_("If the average power over the last N buffers " \
78 "is higher than this value, the volume will be normalized. " \
79 "This value is a positive floating point number. A value " \
80 "between 0.5 and 10 seems sensible." )
83 set_description( N_("Volume normalizer") )
84 set_shortname( N_("Volume normalizer") )
85 set_category( CAT_AUDIO )
86 set_subcategory( SUBCAT_AUDIO_AFILTER )
87 add_shortcut( "volnorm" )
88 add_integer( "norm-buff-size", 20 ,NULL ,BUFF_TEXT, BUFF_LONGTEXT,
90 add_float( "norm-max-level", 2.0, NULL, LEVEL_TEXT,
91 LEVEL_LONGTEXT, true )
92 set_capability( "audio filter", 0 )
93 set_callbacks( Open, Close )
96 /*****************************************************************************
97 * Open: initialize and create stuff
98 *****************************************************************************/
99 static int Open( vlc_object_t *p_this )
101 filter_t *p_filter = (filter_t*)p_this;
105 if( p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ||
106 p_filter->fmt_out.audio.i_format != VLC_CODEC_FL32 )
108 p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
109 p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
110 msg_Warn( p_filter, "bad input or output format" );
114 if ( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio, &p_filter->fmt_out.audio ) )
116 memcpy( &p_filter->fmt_out.audio, &p_filter->fmt_in.audio,
117 sizeof(audio_sample_format_t) );
118 msg_Warn( p_filter, "input and output formats are not similar" );
122 p_filter->pf_audio_filter = DoWork;
124 i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
126 p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
129 p_sys->i_nb = var_CreateGetInteger( p_filter->p_parent, "norm-buff-size" );
130 p_sys->f_max = var_CreateGetFloat( p_filter->p_parent, "norm-max-level" );
132 if( p_sys->f_max <= 0 ) p_sys->f_max = 0.01;
134 /* We need to store (nb_buffers+1)*nb_channels floats */
135 p_sys->p_last = calloc( i_channels * (p_filter->p_sys->i_nb + 2), sizeof(float) );
145 /*****************************************************************************
146 * DoWork : normalizes and sends a buffer
147 *****************************************************************************/
148 static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
155 int i_samples = p_in_buf->i_nb_samples;
156 int i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
157 float *p_out = (float*)p_in_buf->p_buffer;
158 float *p_in = (float*)p_in_buf->p_buffer;
160 struct filter_sys_t *p_sys = p_filter->p_sys;
162 pf_sum = calloc( i_channels, sizeof(float) );
166 pf_gain = malloc( sizeof(float) * i_channels );
173 /* Calculate the average power level on this buffer */
174 for( i = 0 ; i < i_samples; i++ )
176 for( i_chan = 0; i_chan < i_channels; i_chan++ )
178 float f_sample = p_in[i_chan];
179 float f_square = pow( f_sample, 2 );
180 pf_sum[i_chan] += f_square;
185 /* sum now contains for each channel the sigma(value²) */
186 for( i_chan = 0; i_chan < i_channels; i_chan++ )
188 /* Shift our lastbuff */
189 memmove( &p_sys->p_last[ i_chan * p_sys->i_nb],
190 &p_sys->p_last[i_chan * p_sys->i_nb + 1],
191 (p_sys->i_nb-1) * sizeof( float ) );
193 /* Insert the new average : sqrt(sigma(value²)) */
194 p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] =
195 sqrt( pf_sum[i_chan] );
199 /* Get the average power on the lastbuff */
201 for( i = 0; i < p_sys->i_nb ; i++)
203 f_average += p_sys->p_last[ i_chan * p_sys->i_nb + i ];
205 f_average = f_average / p_sys->i_nb;
207 /* Seuil arbitraire */
208 p_sys->f_max = var_GetFloat( p_filter->p_parent, "norm-max-level" );
210 //fprintf(stderr,"Average %f, max %f\n", f_average, p_sys->f_max );
211 if( f_average > p_sys->f_max )
213 pf_gain[i_chan] = f_average / p_sys->f_max;
222 for( i = 0; i < i_samples; i++)
224 for( i_chan = 0; i_chan < i_channels; i_chan++ )
226 p_out[i_chan] /= pf_gain[i_chan];
236 block_Release( p_in_buf );
240 /**********************************************************************
242 **********************************************************************/
243 static void Close( vlc_object_t *p_this )
245 filter_t *p_filter = (filter_t*)p_this;
246 filter_sys_t *p_sys = p_filter->p_sys;
248 free( p_sys->p_last );