From: Laurent Aimar Date: Tue, 6 Jul 2004 21:41:15 +0000 (+0000) Subject: * equalizer: simple (and slow) 10 bands equalizer using a IIR filter. X-Git-Tag: 0.8.0~979 X-Git-Url: https://git.sesse.net/?a=commitdiff_plain;h=1f78d643f390731d4b5bf4f23800fef60f313655;p=vlc * equalizer: simple (and slow) 10 bands equalizer using a IIR filter. --- diff --git a/modules/audio_filter/equalizer.c b/modules/audio_filter/equalizer.c new file mode 100644 index 0000000000..622e294da5 --- /dev/null +++ b/modules/audio_filter/equalizer.c @@ -0,0 +1,571 @@ +/***************************************************************************** + * equalizer.c: + ***************************************************************************** + * Copyright (C) 2004 VideoLAN + * $Id: goom.c 8019 2004-06-22 19:35:01Z fenrir $ + * + * Authors: Laurent Aimar + * + * 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 /* malloc(), free() */ +#include + +#include + +#include +#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 ); +} +