Skip to content

V
vlc
Commit 1f78d643 authored by Laurent Aimar's avatar Laurent Aimar
Browse files
Options

* equalizer: simple (and slow) 10 bands equalizer using a IIR filter.

parent 69bbdda9
Show whitespace changes
Inline Side-by-side
Showing with 571 additions and 0 deletions
modules/audio_filter/equalizer.c 0 → 100644
View file @ 1f78d643
/*****************************************************************************
* equalizer.c:
*****************************************************************************
* Copyright (C) 2004 VideoLAN
* $Id: goom.c 8019 2004-06-22 19:35:01Z fenrir $
*
* Authors: Laurent Aimar <fenrir@via.ecp.fr>
*
* 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., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#include <stdlib.h> /* malloc(), free() */
#include <math.h>
#include <vlc/vlc.h>
#include <vlc/aout.h>
#include "aout_internal.h"
/* TODO:
* - add tables for other rates ( 22500, 11250, ...)
* - optimize a bit (you can hardly do slower ;)
* - add tables for more bands (15 and 32 would be cool), maybe with auto coeffs
* computation (not too hard once the Q is found).
* - support for external preset
* - callback to handle preset changes on the fly
* - ...
*/
/*****************************************************************************
* Module descriptor
*****************************************************************************/
static int Open ( vlc_object_t * );
static void Close( vlc_object_t * );
#define PRESET_TEXT N_( "Equalizer preset" )
#define PRESET_LONGTEXT PRESET_TEXT
#define BANDS_TEXT N_( "Bands gain")
#define BANDS_LONGTEXT N_( "Override preset bands gain in dB (-20 ... 20)" )
#define TWOPASS_TEXT N_( "Two pass" )
#define TWOPASS_LONGTEXT N_( "Filter twice the audio" )
#define PREAMP_TEXT N_("Global gain" )
#define PREAMP_LONGTEXT N_("Set the global gain in dB (-20 ... 20)" )
static char *preset_list[] = {
"flat", "classical", "club", "dance", "fullbass", "fullbasstreeble",
"fulltreeble", "headphones","largehall", "live", "party", "pop", "reggae",
"rock", "ska", "soft", "softrock", "techno"
};
static char *preset_list_text[] = {
N_("Flat"), N_("Classical"), N_("Club"), N_("Dance"), N_("Full bass"),
N_("Full bass and treeble"), N_("Full treeble"), N_("Laptop speakers and Headphones"),
N_("Large Hall"), N_("Live"), N_("Party"), N_("Pop"), N_("Reggae"),
N_("Rock"), N_("Ska"), N_("Soft"), N_("Soft rock"), N_("Techno"),
};
vlc_module_begin();
set_description( _("Equalizer 10 bands") );
set_capability( "audio filter", 0 );
add_string( "equalizer-preset", "flat", NULL, PRESET_TEXT, PRESET_LONGTEXT, VLC_TRUE );
change_string_list( preset_list, preset_list_text, 0 );
add_string( "equalizer-bands", NULL, NULL, BANDS_TEXT, BANDS_LONGTEXT, VLC_TRUE );
add_bool( "equalizer-2pass", 0, NULL, TWOPASS_TEXT, TWOPASS_LONGTEXT, VLC_TRUE );
add_float( "equalizer-preamp", 0.0, NULL, PREAMP_TEXT, PREAMP_LONGTEXT, VLC_TRUE );
set_callbacks( Open, Close );
add_shortcut( "equalizer" );
vlc_module_end();
/*****************************************************************************
* Local prototypes
*****************************************************************************/
typedef struct aout_filter_sys_t
{
/* Filter static config */
int i_band;
float *f_alpha;
float *f_beta;
float *f_gamma;
/* Filter dyn config */
float *f_amp; /* Per band amp */
float f_gamp; /* Global preamp */
vlc_bool_t b_2eqz;
/* Filter state */
float x[32][2];
float y[32][128][2];
/* Second filter state */
float x2[32][2];
float y2[32][128][2];
} aout_filter_sys_t;
static void DoWork( aout_instance_t *, aout_filter_t *,
aout_buffer_t *, aout_buffer_t * );
#define EQZ_IN_FACTOR (0.25)
static int EqzInit( aout_filter_t *, int i_rate );
static void EqzFilter( aout_filter_t *, float *out, float *in, int i_samples, int i_channels );
static void EqzClean( aout_filter_t * );
/*****************************************************************************
* Open:
*****************************************************************************/
static int Open( vlc_object_t *p_this )
{
aout_filter_t *p_filter = (aout_filter_t *)p_this;
aout_filter_sys_t *p_sys;
if( 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" );
return VLC_EGENERIC;
}
if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
{
msg_Warn( p_filter, "input and output formats are not similar" );
return VLC_EGENERIC;
}
p_filter->pf_do_work = DoWork;
p_filter->b_in_place = VLC_TRUE;
/* Allocate structure */
p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
if( EqzInit( p_filter, p_filter->input.i_rate ) )
return VLC_EGENERIC;
return VLC_SUCCESS;
}
/*****************************************************************************
* Close: close the plugin
*****************************************************************************/
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;
EqzClean( p_filter );
free( p_sys );
}
/*****************************************************************************
* DoWork: process samples 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 )
{
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
EqzFilter( p_filter, (float*)p_out_buf->p_buffer, (float*)p_in_buf->p_buffer,
p_in_buf->i_nb_samples, aout_FormatNbChannels( &p_filter->input ) );
}
/*****************************************************************************
* Equalizer stuff
*****************************************************************************/
typedef struct
{
int i_band;
struct
{
float f_frequency;
float f_alpha;
float f_beta;
float f_gamma;
} band[];
} eqz_config_t;
/* Value from equ-xmms */
static const eqz_config_t eqz_config_44100_10b =
{
10,
{
{ 60, 0.003013, 0.993973, 1.993901 },
{ 170, 0.008490, 0.983019, 1.982437 },
{ 310, 0.015374, 0.969252, 1.967331 },
{ 600, 0.029328, 0.941343, 1.934254 },
{ 1000, 0.047918, 0.904163, 1.884869 },
{ 3000, 0.130408, 0.739184, 1.582718 },
{ 6000, 0.226555, 0.546889, 1.015267 },
{ 12000, 0.344937, 0.310127, -0.181410 },
{ 14000, 0.366438, 0.267123, -0.521151 },
{ 16000, 0.379009, 0.241981, -0.808451 },
}
};
static const eqz_config_t eqz_config_48000_10b =
{
10,
{
{ 60, 0.002769, 0.994462, 1.994400 },
{ 170, 0.007806, 0.984388, 1.983897 },
{ 310, 0.014143, 0.971714, 1.970091 },
{ 600, 0.027011, 0.945978, 1.939979 },
{ 1000, 0.044203, 0.911595, 1.895241 },
{ 3000, 0.121223, 0.757553, 1.623767 },
{ 6000, 0.212888, 0.574224, 1.113145 },
{ 12000, 0.331347, 0.337307, 0.000000 },
{ 14000, 0.355263, 0.289473, -0.333740 },
{ 16000, 0.371900, 0.256201, -0.628100 }
}
};
typedef struct
{
char *psz_name;
int i_band;
float f_amp[];
} eqz_preset_t;
static const eqz_preset_t eqz_preset_flat_10b=
{
"flat", 10,
{ 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0 },
};
static const eqz_preset_t eqz_preset_classical_10b=
{
"classical", 10,
{ -1.11022e-15, -1.11022e-15, -1.11022e-15, -1.11022e-15, -1.11022e-15, -1.11022e-15, -7.2, -7.2, -7.2, -9.6 }
};
static const eqz_preset_t eqz_preset_club_10b=
{
"club", 10,
{ -1.11022e-15, -1.11022e-15, 8, 5.6, 5.6, 5.6, 3.2, -1.11022e-15, -1.11022e-15, -1.11022e-15 }
};
static const eqz_preset_t eqz_preset_dance_10b=
{
"dance", 10,
{ 9.6, 7.2, 2.4, -1.11022e-15, -1.11022e-15, -5.6, -7.2, -7.2, -1.11022e-15, -1.11022e-15 }
};
static const eqz_preset_t eqz_preset_fullbass_10b=
{
"fullbass", 10,
{ -8, 9.6, 9.6, 5.6, 1.6, -4, -8, -10.4, -11.2, -11.2 }
};
static const eqz_preset_t eqz_preset_fullbasstreeble_10b=
{
"fullbasstreeble", 10,
{ 7.2, 5.6, -1.11022e-15, -7.2, -4.8, 1.6, 8, 11.2, 12, 12 }
};
static const eqz_preset_t eqz_preset_fulltreeble_10b=
{
"fulltreeble", 10,
{ -9.6, -9.6, -9.6, -4, 2.4, 11.2, 16, 16, 16, 16.8 }
};
static const eqz_preset_t eqz_preset_headphones_10b=
{
"headphones", 10,
{ 4.8, 11.2, 5.6, -3.2, -2.4, 1.6, 4.8, 9.6, 12.8, 14.4 }
};
static const eqz_preset_t eqz_preset_largehall_10b=
{
"largehall", 10,
{ 10.4, 10.4, 5.6, 5.6, -1.11022e-15, -4.8, -4.8, -4.8, -1.11022e-15, -1.11022e-15 }
};
static const eqz_preset_t eqz_preset_live_10b=
{
"live", 10,
{ -4.8, -1.11022e-15, 4, 5.6, 5.6, 5.6, 4, 2.4, 2.4, 2.4 }
};
static const eqz_preset_t eqz_preset_party_10b=
{
"party", 10,
{ 7.2, 7.2, -1.11022e-15, -1.11022e-15, -1.11022e-15, -1.11022e-15, -1.11022e-15, -1.11022e-15, 7.2, 7.2 }
};
static const eqz_preset_t eqz_preset_pop_10b=
{
"pop", 10,
{ -1.6, 4.8, 7.2, 8, 5.6, -1.11022e-15, -2.4, -2.4, -1.6, -1.6 }
};
static const eqz_preset_t eqz_preset_reggae_10b=
{
"reggae", 10,
{ -1.11022e-15, -1.11022e-15, -1.11022e-15, -5.6, -1.11022e-15, 6.4, 6.4, -1.11022e-15, -1.11022e-15, -1.11022e-15 }
};
static const eqz_preset_t eqz_preset_rock_10b=
{
"rock", 10,
{ 8, 4.8, -5.6, -8, -3.2, 4, 8.8, 11.2, 11.2, 11.2 }
};
static const eqz_preset_t eqz_preset_ska_10b=
{
"ska", 10,
{ -2.4, -4.8, -4, -1.11022e-15, 4, 5.6, 8.8, 9.6, 11.2, 9.6 }
};
static const eqz_preset_t eqz_preset_soft_10b=
{
"soft", 10,
{ 4.8, 1.6, -1.11022e-15, -2.4, -1.11022e-15, 4, 8, 9.6, 11.2, 12 }
};
static const eqz_preset_t eqz_preset_softrock_10b=
{
"softrock", 10,
{ 4, 4, 2.4, -1.11022e-15, -4, -5.6, -3.2, -1.11022e-15, 2.4, 8.8 }
};
static const eqz_preset_t eqz_preset_techno_10b=
{
"techno", 10,
{ 8, 5.6, -1.11022e-15, -5.6, -4.8, -1.11022e-15, 8, 9.6, 9.6, 8.8 }
};
static const eqz_preset_t *eqz_preset_10b[] =
{
&eqz_preset_flat_10b,
&eqz_preset_classical_10b,
&eqz_preset_club_10b,
&eqz_preset_dance_10b,
&eqz_preset_fullbass_10b,
&eqz_preset_fullbasstreeble_10b,
&eqz_preset_fulltreeble_10b,
&eqz_preset_headphones_10b,
&eqz_preset_largehall_10b,
&eqz_preset_live_10b,
&eqz_preset_party_10b,
&eqz_preset_pop_10b,
&eqz_preset_reggae_10b,
&eqz_preset_rock_10b,
&eqz_preset_ska_10b,
&eqz_preset_soft_10b,
&eqz_preset_softrock_10b,
&eqz_preset_techno_10b,
NULL
};
static inline float EqzConvertdB( float db )
{
/* Map it to gain,
* (we do as if the input of iir is /EQZ_IN_FACTOR, but in fact it's the non iir data that is *EQZ_IN_FACTOR)
* db = 20*log( out / in ) with out = in + amp*iir(i/EQZ_IN_FACTOR)
* or iir(i) == i for the center freq so
* db = 20*log( 1 + amp/EQZ_IN_FACTOR )
* -> amp = EQZ_IN_FACTOR*(10^(db/20) - 1)
**/
if( db < -20.0 )
db = -20.0;
else if( db > 20.0 )
db = 20.0;
return EQZ_IN_FACTOR * ( pow( 10, db / 20.0 ) - 1.0 );
}
static int EqzInit( aout_filter_t *p_filter, int i_rate )
{
aout_filter_sys_t *p_sys = p_filter->p_sys;
const eqz_config_t *p_cfg;
char *psz;
int i, ch;
float f_float;
/* Select the config */
if( i_rate == 48000 )
{
p_cfg = &eqz_config_48000_10b;
}
else if( i_rate == 44100 )
{
p_cfg = &eqz_config_44100_10b;
}
else
{
/* TODO compute the coeffs on the fly */
msg_Err( p_filter, "unsupported rate" );
return VLC_EGENERIC;
}
/* Create the static filter config */
p_sys->i_band = p_cfg->i_band;
p_sys->f_alpha = malloc( p_sys->i_band * sizeof(float) );
p_sys->f_beta = malloc( p_sys->i_band * sizeof(float) );
p_sys->f_gamma = malloc( p_sys->i_band * sizeof(float) );
for( i = 0; i < p_sys->i_band; i++ )
{
p_sys->f_alpha[i] = p_cfg->band[i].f_alpha;
p_sys->f_beta[i] = p_cfg->band[i].f_beta;
p_sys->f_gamma[i] = p_cfg->band[i].f_gamma;
}
/* Filter dyn config */
p_sys->b_2eqz = VLC_FALSE;
p_sys->f_gamp = 1.0;
p_sys->f_amp = malloc( p_sys->i_band * sizeof(float) );
for( i = 0; i < p_sys->i_band; i++ )
{
p_sys->f_amp[i] = 0.0;
}
/* Filter state */
for( ch = 0; ch < 32; ch++ )
{
p_sys->x[ch][0] =
p_sys->x[ch][1] =
p_sys->x2[ch][0] =
p_sys->x2[ch][1] = 0.0;
for( i = 0; i < p_sys->i_band; i++ )
{
p_sys->y[ch][i][0] =
p_sys->y[ch][i][1] =
p_sys->y2[ch][i][0] =
p_sys->y2[ch][i][1] = 0.0;
}
}
/* Now parse config */
p_sys->b_2eqz = var_CreateGetBool( p_filter, "equalizer-2pass" );
f_float = var_CreateGetFloat( p_filter, "equalizer-preamp" );
if( f_float < -20.0 )
f_float = -20.0;
else if( f_float > 20.0 )
f_float = 20.0;
p_sys->f_gamp = pow( 10, f_float /20.0);
psz = var_CreateGetString( p_filter, "equalizer-preset" );
if( *psz && p_sys->i_band == 10 )
{
int i;
/* */
for( i = 0; eqz_preset_10b[i] != NULL; i++ )
{
if( !strcasecmp( eqz_preset_10b[i]->psz_name, psz ) )
{
int j;
for( j = 0; j < p_sys->i_band; j++ )
p_sys->f_amp[j] = EqzConvertdB( eqz_preset_10b[i]->f_amp[j] );
break;
}
}
if( eqz_preset_10b[i] == NULL )
{
msg_Err( p_filter, "equalizer preset '%s' not found", psz );
msg_Dbg( p_filter, "full list:" );
for( i = 0; eqz_preset_10b[i] != NULL; i++ )
msg_Dbg( p_filter, " - '%s'", eqz_preset_10b[i]->psz_name );
}
}
free( psz );
psz = var_CreateGetString( p_filter, "equalizer-bands" );
if( *psz )
{
char *p = psz;
int i;
for( i = 0; i < p_sys->i_band; i++ )
{
float f;
/* Read dB -20/20*/
f = strtof( p, &p );
p_sys->f_amp[i] = EqzConvertdB( f );
if( p == NULL )
break;
p++;
if( *p == '\0' )
break;
}
}
free( psz );
msg_Dbg( p_filter, "equalizer loaded for %d Hz with %d bands %d pass",
i_rate, p_sys->i_band, p_sys->b_2eqz ? 2 : 1 );
for( i = 0; i < p_sys->i_band; i++ )
{
msg_Dbg( p_filter, " %d Hz -> factor:%f alpha:%f beta:%f gamma:%f",
(int)p_cfg->band[i].f_frequency, p_sys->f_amp[i],
p_sys->f_alpha[i], p_sys->f_beta[i], p_sys->f_gamma[i]);
}
return VLC_SUCCESS;
}
static void EqzFilter( aout_filter_t *p_filter, float *out, float *in, int i_samples, int i_channels )
{
aout_filter_sys_t *p_sys = p_filter->p_sys;
int i, ch, j;
for( i = 0; i < i_samples; i++ )
{
for( ch = 0; ch < i_channels; ch++ )
{
const float x = in[ch];
float o = 0.0;
for( j = 0; j < p_sys->i_band; j++ )
{
float y = p_sys->f_alpha[j] * ( x - p_sys->x[ch][1] ) +
p_sys->f_gamma[j] * p_sys->y[ch][j][0] -
p_sys->f_beta[j] * p_sys->y[ch][j][1];
p_sys->y[ch][j][1] = p_sys->y[ch][j][0];
p_sys->y[ch][j][0] = y;
o += y * p_sys->f_amp[j];
}
p_sys->x[ch][1] = p_sys->x[ch][0];
p_sys->x[ch][0] = x;
/* Second filter */
if( p_sys->b_2eqz )
{
const float x2 = EQZ_IN_FACTOR * x + o;
o = 0.0;
for( j = 0; j < p_sys->i_band; j++ )
{
float y = p_sys->f_alpha[j] * ( x2 - p_sys->x2[ch][1] ) +
p_sys->f_gamma[j] * p_sys->y2[ch][j][0] -
p_sys->f_beta[j] * p_sys->y2[ch][j][1];
p_sys->y2[ch][j][1] = p_sys->y2[ch][j][0];
p_sys->y2[ch][j][0] = y;
o += y * p_sys->f_amp[j];
}
p_sys->x2[ch][1] = p_sys->x2[ch][0];
p_sys->x2[ch][0] = x2;
/* We add source PCM + filtered PCM */
out[ch] = p_sys->f_gamp *( EQZ_IN_FACTOR * x2 + o );
}
else
{
/* We add source PCM + filtered PCM */
out[ch] = p_sys->f_gamp *( EQZ_IN_FACTOR * x + o );
}
}
in += i_channels;
out += i_channels;
}
}
static void EqzClean( aout_filter_t *p_filter )
{
aout_filter_sys_t *p_sys = p_filter->p_sys;
free( p_sys->f_alpha );
free( p_sys->f_beta );
free( p_sys->f_gamma );
free( p_sys->f_amp );
}
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment