📄 normvol.c
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/***************************************************************************** * normvol.c: volume normalizer ***************************************************************************** * Copyright (C) 2001, 2006 the VideoLAN team * $Id: normvol.c 17012 2006-10-09 22:11:32Z xtophe $ * * 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 * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * 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., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************//* * TODO: * * We should detect fast power increases and react faster to these * This way, we can increase the buffer size to get a more stable filter *//***************************************************************************** * 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>#include <vlc/vlc.h>#include <vlc/aout.h>#include <aout_internal.h>/***************************************************************************** * Local prototypes *****************************************************************************/static int Open ( vlc_object_t * );static void Close ( vlc_object_t * );static void DoWork ( aout_instance_t * , aout_filter_t *, aout_buffer_t * , aout_buffer_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 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. " \ "This value is a positive floating point number. A value " \ "between 0.5 and 10 seems sensible." )vlc_module_begin(); set_description( _("Volume normalizer") ); 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();/***************************************************************************** * Open: initialize and create stuff *****************************************************************************/static int Open( vlc_object_t *p_this ){ aout_filter_t *p_filter = (aout_filter_t*)p_this; vlc_bool_t b_fit = VLC_TRUE; int i_channels; aout_filter_sys_t *p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) ); if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) || p_filter->output.i_format != VLC_FOURCC('f','l','3','2') ) { 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; i_channels = aout_FormatNbChannels( &p_filter->input ); 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; /* 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) ); return VLC_SUCCESS;}/***************************************************************************** * DoWork : normalizes and sends a buffer *****************************************************************************/ static void DoWork( aout_instance_t *p_aout, aout_filter_t *p_filter, aout_buffer_t *p_in_buf, aout_buffer_t *p_out_buf ){ float *pf_sum; float *pf_gain; float f_average = 0; int i, i_chan; int i_samples = p_in_buf->i_nb_samples; int i_channels = aout_FormatNbChannels( &p_filter->input ); float *p_out = (float*)p_out_buf->p_buffer; float *p_in = (float*)p_in_buf->p_buffer; struct aout_filter_sys_t *p_sys = p_filter->p_sys; pf_sum = (float *)malloc( sizeof(float) * i_channels ); memset( pf_sum, 0, sizeof(float) * i_channels ); pf_gain = (float *)malloc( sizeof(float) * i_channels ); 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++ ) { for( i_chan = 0; i_chan < i_channels; i_chan++ ) { float f_sample = p_in[i_chan]; float f_square = pow( f_sample, 2 ); 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++ ) { /* 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)) */ p_sys->p_last[ i_chan * p_sys->i_nb + p_sys->i_nb - 1] = sqrt( pf_sum[i_chan] ); pf_sum[i_chan] = 0; /* Get the average power on the lastbuff */ f_average = 0; for( i = 0; i < p_sys->i_nb ; i++) { f_average += p_sys->p_last[ i_chan * p_sys->i_nb + i ]; } f_average = f_average / p_sys->i_nb; /* 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 ); if( f_average > p_sys->f_max ) { pf_gain[i_chan] = f_average / p_sys->f_max; } else { pf_gain[i_chan] = 1; } } /* Apply gain */ for( i = 0; i < i_samples; i++) { for( i_chan = 0; i_chan < i_channels; i_chan++ ) { p_out[i_chan] /= pf_gain[i_chan]; } p_out += i_channels; } free( pf_sum ); free( pf_gain ); return;}/********************************************************************** * Close **********************************************************************/static void Close( vlc_object_t *p_this ){ aout_filter_t *p_filter = (aout_filter_t*)p_this; 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 ); }}
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