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vlc/modules/audio_filter/param_eq.c

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/*****************************************************************************
* param_eq.c:
*****************************************************************************
* Copyright © 2006 VLC authors and VideoLAN
*
* Authors: Antti Huovilainen
* Sigmund A. Helberg <dnumgis@videolan.org>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <math.h>
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_aout.h>
#include <vlc_filter.h>
/*****************************************************************************
* Module descriptor
*****************************************************************************/
static int Open ( vlc_object_t * );
static void Close( filter_t * );
static void CalcPeakEQCoeffs( float, float, float, float, float * );
static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
static void ProcessEQ( const float *, float *, float *, unsigned, unsigned,
const float *, unsigned );
static block_t *DoWork( filter_t *, block_t * );
vlc_module_begin ()
set_description( N_("Parametric Equalizer") )
set_shortname( N_("Parametric Equalizer" ) )
set_capability( "audio filter", 0 )
set_subcategory( SUBCAT_AUDIO_AFILTER )
add_float( "param-eq-lowf", 100, N_("Low freq (Hz)"),NULL )
add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0,
N_("Low freq gain (dB)"), NULL )
add_float( "param-eq-highf", 10000, N_("High freq (Hz)"),NULL )
add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0,
N_("High freq gain (dB)"),NULL )
add_float( "param-eq-f1", 300, N_("Freq 1 (Hz)"),NULL )
add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0,
N_("Freq 1 gain (dB)"), NULL )
add_float_with_range( "param-eq-q1", 3, 0.1, 100.0,
N_("Freq 1 Q"), NULL )
add_float( "param-eq-f2", 1000, N_("Freq 2 (Hz)"),NULL )
add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0,
N_("Freq 2 gain (dB)"),NULL )
add_float_with_range( "param-eq-q2", 3, 0.1, 100.0,
N_("Freq 2 Q"),NULL )
add_float( "param-eq-f3", 3000, N_("Freq 3 (Hz)"),NULL )
add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0,
N_("Freq 3 gain (dB)"),NULL )
add_float_with_range( "param-eq-q3", 3, 0.1, 100.0,
N_("Freq 3 Q"),NULL )
set_callback( Open )
vlc_module_end ()
/*****************************************************************************
* Local prototypes
*****************************************************************************/
typedef struct
{
/* Filter static config */
float f_lowf, f_lowgain;
float f_f1, f_Q1, f_gain1;
float f_f2, f_Q2, f_gain2;
float f_f3, f_Q3, f_gain3;
float f_highf, f_highgain;
/* Filter computed coeffs */
float coeffs[5*5];
/* State */
float *p_state;
} filter_sys_t;
/*****************************************************************************
* Open:
*****************************************************************************/
static int Open( vlc_object_t *p_this )
{
filter_t *p_filter = (filter_t *)p_this;
unsigned i_samplerate;
/* Allocate structure */
filter_sys_t *p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
if( !p_sys )
return VLC_EGENERIC;
p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
p_filter->fmt_out.audio = p_filter->fmt_in.audio;
static const struct vlc_filter_operations filter_ops =
{
.filter_audio = DoWork, .close = Close,
};
p_filter->ops = &filter_ops;
p_sys->f_lowf = var_InheritFloat( p_this, "param-eq-lowf");
p_sys->f_lowgain = var_InheritFloat( p_this, "param-eq-lowgain");
p_sys->f_highf = var_InheritFloat( p_this, "param-eq-highf");
p_sys->f_highgain = var_InheritFloat( p_this, "param-eq-highgain");
p_sys->f_f1 = var_InheritFloat( p_this, "param-eq-f1");
p_sys->f_Q1 = var_InheritFloat( p_this, "param-eq-q1");
p_sys->f_gain1 = var_InheritFloat( p_this, "param-eq-gain1");
p_sys->f_f2 = var_InheritFloat( p_this, "param-eq-f2");
p_sys->f_Q2 = var_InheritFloat( p_this, "param-eq-q2");
p_sys->f_gain2 = var_InheritFloat( p_this, "param-eq-gain2");
p_sys->f_f3 = var_InheritFloat( p_this, "param-eq-f3");
p_sys->f_Q3 = var_InheritFloat( p_this, "param-eq-q3");
p_sys->f_gain3 = var_InheritFloat( p_this, "param-eq-gain3");
i_samplerate = p_filter->fmt_in.audio.i_rate;
CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,
i_samplerate, p_sys->coeffs+0*5);
CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,
i_samplerate, p_sys->coeffs+1*5);
CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,
i_samplerate, p_sys->coeffs+2*5);
CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,
i_samplerate, p_sys->coeffs+3*5);
CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,
i_samplerate, p_sys->coeffs+4*5);
p_sys->p_state = (float*)calloc( p_filter->fmt_in.audio.i_channels*5*4,
sizeof(float) );
return VLC_SUCCESS;
}
static void Close( filter_t *p_filter )
{
filter_sys_t *p_sys = p_filter->p_sys;
free( p_sys->p_state );
free( p_sys );
}
/*****************************************************************************
* DoWork: process samples buffer
*****************************************************************************
*
*****************************************************************************/
static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
{
filter_sys_t *p_sys = p_filter->p_sys;
ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_in_buf->p_buffer,
p_sys->p_state,
p_filter->fmt_in.audio.i_channels, p_in_buf->i_nb_samples,
p_sys->coeffs, 5 );
return p_in_buf;
}
/*
* Calculate direct form IIR coefficients for peaking EQ
* coeffs[0] = b0
* coeffs[1] = b1
* coeffs[2] = b2
* coeffs[3] = a1
* coeffs[4] = a2
*
* Equations taken from RBJ audio EQ cookbook
* (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
*/
static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,
float *coeffs )
{
float A;
float w0;
float alpha;
float b0, b1, b2;
float a0, a1, a2;
// Provide sane limits to avoid overflow
if (Q < 0.1f) Q = 0.1f;
if (Q > 100) Q = 100;
if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
if (gainDB < -40) gainDB = -40;
if (gainDB > 40) gainDB = 40;
A = powf(10, gainDB/40);
w0 = 2*((float)M_PI)*f0/Fs;
alpha = sinf(w0)/(2*Q);
b0 = 1 + alpha*A;
b1 = -2*cosf(w0);
b2 = 1 - alpha*A;
a0 = 1 + alpha/A;
a1 = -2*cosf(w0);
a2 = 1 - alpha/A;
// Store values to coeffs and normalize by 1/a0
coeffs[0] = b0/a0;
coeffs[1] = b1/a0;
coeffs[2] = b2/a0;
coeffs[3] = a1/a0;
coeffs[4] = a2/a0;
}
/*
* Calculate direct form IIR coefficients for low/high shelf EQ
* coeffs[0] = b0
* coeffs[1] = b1
* coeffs[2] = b2
* coeffs[3] = a1
* coeffs[4] = a2
*
* Equations taken from RBJ audio EQ cookbook
* (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
*/
static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,
float Fs, float *coeffs )
{
float A;
float w0;
float alpha;
float b0, b1, b2;
float a0, a1, a2;
// Provide sane limits to avoid overflow
if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
if (gainDB < -40) gainDB = -40;
if (gainDB > 40) gainDB = 40;
A = powf(10, gainDB/40);
w0 = 2*3.141593f*f0/Fs;
alpha = sinf(w0)/2 * sqrtf( (A + 1/A)*(1/slope - 1) + 2 );
if (high)
{
b0 = A*( (A+1) + (A-1)*cosf(w0) + 2*sqrtf(A)*alpha );
b1 = -2*A*( (A-1) + (A+1)*cosf(w0) );
b2 = A*( (A+1) + (A-1)*cosf(w0) - 2*sqrtf(A)*alpha );
a0 = (A+1) - (A-1)*cosf(w0) + 2*sqrtf(A)*alpha;
a1 = 2*( (A-1) - (A+1)*cosf(w0) );
a2 = (A+1) - (A-1)*cosf(w0) - 2*sqrtf(A)*alpha;
}
else
{
b0 = A*( (A+1) - (A-1)*cosf(w0) + 2*sqrtf(A)*alpha );
b1 = 2*A*( (A-1) - (A+1)*cosf(w0));
b2 = A*( (A+1) - (A-1)*cosf(w0) - 2*sqrtf(A)*alpha );
a0 = (A+1) + (A-1)*cosf(w0) + 2*sqrtf(A)*alpha;
a1 = -2*( (A-1) + (A+1)*cosf(w0));
a2 = (A+1) + (A-1)*cosf(w0) - 2*sqrtf(A)*alpha;
}
// Store values to coeffs and normalize by 1/a0
coeffs[0] = b0/a0;
coeffs[1] = b1/a0;
coeffs[2] = b2/a0;
coeffs[3] = a1/a0;
coeffs[4] = a2/a0;
}
/*
src is assumed to be interleaved
dest is assumed to be interleaved
size of state is 4*channels*eqCount
samples is not premultiplied by channels
size of coeffs is 5*eqCount
*/
void ProcessEQ( const float *src, float *dest, float *state,
unsigned channels, unsigned samples, const float *coeffs,
unsigned eqCount )
{
unsigned i, chn, eq;
float b0, b1, b2, a1, a2;
float x, y = 0;
const float *src1 = src;
float *dest1 = dest;
for (i = 0; i < samples; i++)
{
float *state1 = state;
for (chn = 0; chn < channels; chn++)
{
const float *coeffs1 = coeffs;
x = *src1++;
/* Direct form 1 IIRs */
for (eq = 0; eq < eqCount; eq++)
{
b0 = coeffs1[0];
b1 = coeffs1[1];
b2 = coeffs1[2];
a1 = coeffs1[3];
a2 = coeffs1[4];
coeffs1 += 5;
y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;
state1[1] = state1[0];
state1[0] = x;
state1[3] = state1[2];
state1[2] = y;
x = y;
state1 += 4;
}
*dest1++ = y;
}
}
}
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