<commit_before>/*****************************************************************************
 * param_eq.c:
 *****************************************************************************
 * Copyright © 2006 the VideoLAN team
 * $Id$
 *
 * Authors: <NAME>
 *          <NAME> <<EMAIL>>
 *
 * 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.
 *****************************************************************************/

/*****************************************************************************
 * Preamble
 *****************************************************************************/
#include <stdlib.h>                                      /* malloc(), free() */
#include <string.h>
#include <math.h>

#include <vlc/vlc.h>
#include <vlc_aout.h>

/*****************************************************************************
 * Module descriptor
 *****************************************************************************/
static int  Open ( vlc_object_t * );
static void Close( vlc_object_t * );
static void CalcPeakEQCoeffs( float, float, float, float, float * );
static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
static void ProcessEQ( float *, float *, float *, int, int, float *, int );
static void DoWork( aout_instance_t *, aout_filter_t *,
                    aout_buffer_t *, aout_buffer_t * );

vlc_module_begin();
    set_description( _("Parametric Equalizer") );
    set_shortname( _("Parametric Equalizer" ) );
    set_capability( "audio filter", 0 );
    set_category( CAT_AUDIO );
    set_subcategory( SUBCAT_AUDIO_AFILTER );

    add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
                          N_("Low freq gain (dB)"), "",VLC_FALSE );
    add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
                          N_("High freq gain (dB)"),"",VLC_FALSE );
    add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
                          N_("Freq 1 gain (dB)"), "",VLC_FALSE );
    add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
                          N_("Freq 1 Q"), "",VLC_FALSE );
    add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
                          N_("Freq 2 gain (dB)"),"",VLC_FALSE );
    add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
                          N_("Freq 2 Q"),"",VLC_FALSE );
    add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
                          N_("Freq 3 gain (dB)"),"",VLC_FALSE );
    add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
                          N_("Freq 3 Q"),"",VLC_FALSE );

    set_callbacks( Open, Close );
vlc_module_end();

/*****************************************************************************
 * Local prototypes
 *****************************************************************************/
typedef struct aout_filter_sys_t
{
    /* 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;
       
} aout_filter_sys_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;
    vlc_bool_t         b_fit = VLC_TRUE;
    int                i_samplerate;

    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;

    /* Allocate structure */
    p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );

    p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");
    p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");
    p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");
    p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");
    
    p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");
    p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");
    p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");
    
    p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");
    p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");
    p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");

    p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");
    p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");
    p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");
  

    i_samplerate = p_filter->input.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->input.i_channels*5*4,
                                     sizeof(float) );

    return VLC_SUCCESS;
}

static void Close( vlc_object_t *p_this )
{
    aout_filter_t *p_filter = (aout_filter_t *)p_this;
    free( p_filter->p_sys->p_state );
    free( p_filter->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;

    ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer,
               p_filter->p_sys->p_state, 
               p_filter->input.i_channels, p_in_buf->i_nb_samples,
               p_filter->p_sys->coeffs, 5 );
}

/*
 * 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 = pow(10, gainDB/40);
    w0 = 2*3.141593f*f0/Fs;
    alpha = sin(w0)/(2*Q);
    
    b0 = 1 + alpha*A;
    b1 = -2*cos(w0);
    b2 = 1 - alpha*A;
    a0 = 1 + alpha/A;
    a1 = -2*cos(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 = pow(10, gainDB/40);
    w0 = 2*3.141593f*f0/Fs;
    alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );

    if (high)
    {
        b0 =    A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
        b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
        b2 =    A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
        a0 =        (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
        a1 =    2*( (A-1) - (A+1)*cos(w0) );
        a2 =        (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
    }
    else
    {
        b0 =    A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
        b1 =  2*A*( (A-1) - (A+1)*cos(w0));
        b2 =    A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
        a0 =        (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
        a1 =   -2*( (A-1) + (A+1)*cos(w0));
        a2 =        (A+1) + (A-1)*cos(w0) - 2*sqrt(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( float *src, float *dest, float *state, 
                int channels, int samples, float *coeffs, 
                int eqCount )
{
    int i, chn, eq;
    float   b0, b1, b2, a1, a2;
    float   x, y = 0;
    float   *src1, *dest1;
    float   *coeffs1, *state1;
    src1 = src;
    dest1 = dest;
    for (i = 0; i < samples; i++)
    {
        state1 = state;
        for (chn = 0; chn < channels; chn++)
        {
            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;
        }
    }
}

<commit_msg>Use unsigned<commit_after>/*****************************************************************************
 * param_eq.c:
 *****************************************************************************
 * Copyright © 2006 the VideoLAN team
 * $Id$
 *
 * Authors: <NAME>
 *          <NAME> <<EMAIL>>
 *
 * 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.
 *****************************************************************************/

/*****************************************************************************
 * Preamble
 *****************************************************************************/
#include <stdlib.h>                                      /* malloc(), free() */
#include <string.h>
#include <math.h>

#include <vlc/vlc.h>
#include <vlc_aout.h>

/*****************************************************************************
 * Module descriptor
 *****************************************************************************/
static int  Open ( vlc_object_t * );
static void Close( vlc_object_t * );
static void CalcPeakEQCoeffs( float, float, float, float, float * );
static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
static void ProcessEQ( float *, float *, float *, unsigned, unsigned, float *, unsigned );
static void DoWork( aout_instance_t *, aout_filter_t *,
                    aout_buffer_t *, aout_buffer_t * );

vlc_module_begin();
    set_description( _("Parametric Equalizer") );
    set_shortname( _("Parametric Equalizer" ) );
    set_capability( "audio filter", 0 );
    set_category( CAT_AUDIO );
    set_subcategory( SUBCAT_AUDIO_AFILTER );

    add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
                          N_("Low freq gain (dB)"), "",VLC_FALSE );
    add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
                          N_("High freq gain (dB)"),"",VLC_FALSE );
    add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
                          N_("Freq 1 gain (dB)"), "",VLC_FALSE );
    add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
                          N_("Freq 1 Q"), "",VLC_FALSE );
    add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
                          N_("Freq 2 gain (dB)"),"",VLC_FALSE );
    add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
                          N_("Freq 2 Q"),"",VLC_FALSE );
    add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", VLC_FALSE );
    add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
                          N_("Freq 3 gain (dB)"),"",VLC_FALSE );
    add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
                          N_("Freq 3 Q"),"",VLC_FALSE );

    set_callbacks( Open, Close );
vlc_module_end();

/*****************************************************************************
 * Local prototypes
 *****************************************************************************/
typedef struct aout_filter_sys_t
{
    /* 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;
       
} aout_filter_sys_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;
    vlc_bool_t         b_fit = VLC_TRUE;
    int                i_samplerate;

    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;

    /* Allocate structure */
    p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );

    p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");
    p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");
    p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");
    p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");
    
    p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");
    p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");
    p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");
    
    p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");
    p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");
    p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");

    p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");
    p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");
    p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");
  

    i_samplerate = p_filter->input.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->input.i_channels*5*4,
                                     sizeof(float) );

    return VLC_SUCCESS;
}

static void Close( vlc_object_t *p_this )
{
    aout_filter_t *p_filter = (aout_filter_t *)p_this;
    free( p_filter->p_sys->p_state );
    free( p_filter->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;

    ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer,
               p_filter->p_sys->p_state, 
               p_filter->input.i_channels, p_in_buf->i_nb_samples,
               p_filter->p_sys->coeffs, 5 );
}

/*
 * 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 = pow(10, gainDB/40);
    w0 = 2*3.141593f*f0/Fs;
    alpha = sin(w0)/(2*Q);
    
    b0 = 1 + alpha*A;
    b1 = -2*cos(w0);
    b2 = 1 - alpha*A;
    a0 = 1 + alpha/A;
    a1 = -2*cos(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 = pow(10, gainDB/40);
    w0 = 2*3.141593f*f0/Fs;
    alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );

    if (high)
    {
        b0 =    A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
        b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
        b2 =    A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
        a0 =        (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
        a1 =    2*( (A-1) - (A+1)*cos(w0) );
        a2 =        (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
    }
    else
    {
        b0 =    A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
        b1 =  2*A*( (A-1) - (A+1)*cos(w0));
        b2 =    A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
        a0 =        (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
        a1 =   -2*( (A-1) + (A+1)*cos(w0));
        a2 =        (A+1) + (A-1)*cos(w0) - 2*sqrt(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( float *src, float *dest, float *state, 
                unsigned channels, unsigned samples, float *coeffs, 
                unsigned eqCount )
{
    unsigned i, chn, eq;
    float   b0, b1, b2, a1, a2;
    float   x, y = 0;
    float   *src1, *dest1;
    float   *coeffs1, *state1;
    src1 = src;
    dest1 = dest;
    for (i = 0; i < samples; i++)
    {
        state1 = state;
        for (chn = 0; chn < channels; chn++)
        {
            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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