modules/audio_filter/param_eq.c

   1 /*****************************************************************************
   2  * param_eq.c:
   3  *****************************************************************************
   4  * Copyright © 2006 VLC authors and VideoLAN
   5  * $Id$
   6  *
   7  * Authors: Antti Huovilainen
   8  *          Sigmund A. Helberg <dnumgis@videolan.org>
   9  *
  10  * This program is free software; you can redistribute it and/or modify it
  11  * under the terms of the GNU Lesser General Public License as published by
  12  * the Free Software Foundation; either version 2.1 of the License, or
  13  * (at your option) any later version.
  14  *
  15  * This program is distributed in the hope that it will be useful,
  16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
  18  * GNU Lesser General Public License for more details.
  19  *
  20  * You should have received a copy of the GNU Lesser General Public License
  21  * along with this program; if not, write to the Free Software Foundation,
  22  * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
  23  *****************************************************************************/
  24
  25 /*****************************************************************************
  26  * Preamble
  27  *****************************************************************************/
  28
  29 #ifdef HAVE_CONFIG_H
  30 # include "config.h"
  31 #endif
  32
  33 #include <math.h>
  34
  35 #include <vlc_common.h>
  36 #include <vlc_plugin.h>
  37 #include <vlc_aout.h>
  38 #include <vlc_filter.h>
  39
  40 /*****************************************************************************
  41  * Module descriptor
  42  *****************************************************************************/
  43 static int  Open ( vlc_object_t * );
  44 static void Close( vlc_object_t * );
  45 static void CalcPeakEQCoeffs( float, float, float, float, float * );
  46 static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
  47 static void ProcessEQ( const float *, float *, float *, unsigned, unsigned,
  48                        const float *, unsigned );
  49 static block_t *DoWork( filter_t *, block_t * );
  50
  51 vlc_module_begin ()
  52     set_description( N_("Parametric Equalizer") )
  53     set_shortname( N_("Parametric Equalizer" ) )
  54     set_capability( "audio filter", 0 )
  55     set_category( CAT_AUDIO )
  56     set_subcategory( SUBCAT_AUDIO_AFILTER )
  57
  58     add_float( "param-eq-lowf", 100, N_("Low freq (Hz)"),NULL, false )
  59     add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0,
  60                           N_("Low freq gain (dB)"), NULL,false )
  61     add_float( "param-eq-highf", 10000, N_("High freq (Hz)"),NULL, false )
  62     add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0,
  63                           N_("High freq gain (dB)"),NULL,false )
  64     add_float( "param-eq-f1", 300, N_("Freq 1 (Hz)"),NULL, false )
  65     add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0,
  66                           N_("Freq 1 gain (dB)"), NULL,false )
  67     add_float_with_range( "param-eq-q1", 3, 0.1, 100.0,
  68                           N_("Freq 1 Q"), NULL,false )
  69     add_float( "param-eq-f2", 1000, N_("Freq 2 (Hz)"),NULL, false )
  70     add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0,
  71                           N_("Freq 2 gain (dB)"),NULL,false )
  72     add_float_with_range( "param-eq-q2", 3, 0.1, 100.0,
  73                           N_("Freq 2 Q"),NULL,false )
  74     add_float( "param-eq-f3", 3000, N_("Freq 3 (Hz)"),NULL, false )
  75     add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0,
  76                           N_("Freq 3 gain (dB)"),NULL,false )
  77     add_float_with_range( "param-eq-q3", 3, 0.1, 100.0,
  78                           N_("Freq 3 Q"),NULL,false )
  79
  80     set_callbacks( Open, Close )
  81 vlc_module_end ()
  82
  83 /*****************************************************************************
  84  * Local prototypes
  85  *****************************************************************************/
  86 struct filter_sys_t
  87 {
  88     /* Filter static config */
  89     float   f_lowf, f_lowgain;
  90     float   f_f1, f_Q1, f_gain1;
  91     float   f_f2, f_Q2, f_gain2;
  92     float   f_f3, f_Q3, f_gain3;
  93     float   f_highf, f_highgain;
  94     /* Filter computed coeffs */
  95     float   coeffs[5*5];
  96     /* State */
  97     float  *p_state;
  98 };
  99
 100
 101
 102
 103 /*****************************************************************************
 104  * Open:
 105  *****************************************************************************/
 106 static int Open( vlc_object_t *p_this )
 107 {
 108     filter_t     *p_filter = (filter_t *)p_this;
 109     unsigned     i_samplerate;
 110
 111     /* Allocate structure */
 112     filter_sys_t *p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
 113     if( !p_sys )
 114         return VLC_EGENERIC;
 115
 116     p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
 117     p_filter->fmt_out.audio = p_filter->fmt_in.audio;
 118     p_filter->pf_audio_filter = DoWork;
 119
 120     p_sys->f_lowf = var_InheritFloat( p_this, "param-eq-lowf");
 121     p_sys->f_lowgain = var_InheritFloat( p_this, "param-eq-lowgain");
 122     p_sys->f_highf = var_InheritFloat( p_this, "param-eq-highf");
 123     p_sys->f_highgain = var_InheritFloat( p_this, "param-eq-highgain");
 124
 125     p_sys->f_f1 = var_InheritFloat( p_this, "param-eq-f1");
 126     p_sys->f_Q1 = var_InheritFloat( p_this, "param-eq-q1");
 127     p_sys->f_gain1 = var_InheritFloat( p_this, "param-eq-gain1");
 128
 129     p_sys->f_f2 = var_InheritFloat( p_this, "param-eq-f2");
 130     p_sys->f_Q2 = var_InheritFloat( p_this, "param-eq-q2");
 131     p_sys->f_gain2 = var_InheritFloat( p_this, "param-eq-gain2");
 132
 133     p_sys->f_f3 = var_InheritFloat( p_this, "param-eq-f3");
 134     p_sys->f_Q3 = var_InheritFloat( p_this, "param-eq-q3");
 135     p_sys->f_gain3 = var_InheritFloat( p_this, "param-eq-gain3");
 136
 137
 138     i_samplerate = p_filter->fmt_in.audio.i_rate;
 139     CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,
 140                      i_samplerate, p_sys->coeffs+0*5);
 141     CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,
 142                      i_samplerate, p_sys->coeffs+1*5);
 143     CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,
 144                      i_samplerate, p_sys->coeffs+2*5);
 145     CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,
 146                       i_samplerate, p_sys->coeffs+3*5);
 147     CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,
 148                       i_samplerate, p_sys->coeffs+4*5);
 149     p_sys->p_state = (float*)calloc( p_filter->fmt_in.audio.i_channels*5*4,
 150                                      sizeof(float) );
 151
 152     return VLC_SUCCESS;
 153 }
 154
 155 static void Close( vlc_object_t *p_this )
 156 {
 157     filter_t *p_filter = (filter_t *)p_this;
 158     free( p_filter->p_sys->p_state );
 159     free( p_filter->p_sys );
 160 }
 161
 162 /*****************************************************************************
 163  * DoWork: process samples buffer
 164  *****************************************************************************
 165  *
 166  *****************************************************************************/
 167 static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
 168 {
 169     ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_in_buf->p_buffer,
 170                p_filter->p_sys->p_state,
 171                p_filter->fmt_in.audio.i_channels, p_in_buf->i_nb_samples,
 172                p_filter->p_sys->coeffs, 5 );
 173     return p_in_buf;
 174 }
 175
 176 /*
 177  * Calculate direct form IIR coefficients for peaking EQ
 178  * coeffs[0] = b0
 179  * coeffs[1] = b1
 180  * coeffs[2] = b2
 181  * coeffs[3] = a1
 182  * coeffs[4] = a2
 183  *
 184  * Equations taken from RBJ audio EQ cookbook
 185  * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
 186  */
 187 static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,
 188                               float *coeffs )
 189 {
 190     float A;
 191     float w0;
 192     float alpha;
 193     float b0, b1, b2;
 194     float a0, a1, a2;
 195
 196     // Provide sane limits to avoid overflow
 197     if (Q < 0.1f) Q = 0.1f;
 198     if (Q > 100) Q = 100;
 199     if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
 200     if (gainDB < -40) gainDB = -40;
 201     if (gainDB > 40) gainDB = 40;
 202
 203     A = powf(10, gainDB/40);
 204     w0 = 2*((float)M_PI)*f0/Fs;
 205     alpha = sinf(w0)/(2*Q);
 206
 207     b0 = 1 + alpha*A;
 208     b1 = -2*cosf(w0);
 209     b2 = 1 - alpha*A;
 210     a0 = 1 + alpha/A;
 211     a1 = -2*cosf(w0);
 212     a2 = 1 - alpha/A;
 213
 214     // Store values to coeffs and normalize by 1/a0
 215     coeffs[0] = b0/a0;
 216     coeffs[1] = b1/a0;
 217     coeffs[2] = b2/a0;
 218     coeffs[3] = a1/a0;
 219     coeffs[4] = a2/a0;
 220 }
 221
 222 /*
 223  * Calculate direct form IIR coefficients for low/high shelf EQ
 224  * coeffs[0] = b0
 225  * coeffs[1] = b1
 226  * coeffs[2] = b2
 227  * coeffs[3] = a1
 228  * coeffs[4] = a2
 229  *
 230  * Equations taken from RBJ audio EQ cookbook
 231  * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
 232  */
 233 static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,
 234                                float Fs, float *coeffs )
 235 {
 236     float A;
 237     float w0;
 238     float alpha;
 239     float b0, b1, b2;
 240     float a0, a1, a2;
 241
 242     // Provide sane limits to avoid overflow
 243     if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
 244     if (gainDB < -40) gainDB = -40;
 245     if (gainDB > 40) gainDB = 40;
 246
 247     A = powf(10, gainDB/40);
 248     w0 = 2*3.141593f*f0/Fs;
 249     alpha = sinf(w0)/2 * sqrtf( (A + 1/A)*(1/slope - 1) + 2 );
 250
 251     if (high)
 252     {
 253         b0 =    A*( (A+1) + (A-1)*cosf(w0) + 2*sqrtf(A)*alpha );
 254         b1 = -2*A*( (A-1) + (A+1)*cosf(w0) );
 255         b2 =    A*( (A+1) + (A-1)*cosf(w0) - 2*sqrtf(A)*alpha );
 256         a0 =        (A+1) - (A-1)*cosf(w0) + 2*sqrtf(A)*alpha;
 257         a1 =    2*( (A-1) - (A+1)*cosf(w0) );
 258         a2 =        (A+1) - (A-1)*cosf(w0) - 2*sqrtf(A)*alpha;
 259     }
 260     else
 261     {
 262         b0 =    A*( (A+1) - (A-1)*cosf(w0) + 2*sqrtf(A)*alpha );
 263         b1 =  2*A*( (A-1) - (A+1)*cosf(w0));
 264         b2 =    A*( (A+1) - (A-1)*cosf(w0) - 2*sqrtf(A)*alpha );
 265         a0 =        (A+1) + (A-1)*cosf(w0) + 2*sqrtf(A)*alpha;
 266         a1 =   -2*( (A-1) + (A+1)*cosf(w0));
 267         a2 =        (A+1) + (A-1)*cosf(w0) - 2*sqrtf(A)*alpha;
 268     }
 269     // Store values to coeffs and normalize by 1/a0
 270     coeffs[0] = b0/a0;
 271     coeffs[1] = b1/a0;
 272     coeffs[2] = b2/a0;
 273     coeffs[3] = a1/a0;
 274     coeffs[4] = a2/a0;
 275 }
 276
 277 /*
 278   src is assumed to be interleaved
 279   dest is assumed to be interleaved
 280   size of state is 4*channels*eqCount
 281   samples is not premultiplied by channels
 282   size of coeffs is 5*eqCount
 283 */
 284 void ProcessEQ( const float *src, float *dest, float *state,
 285                 unsigned channels, unsigned samples, const float *coeffs,
 286                 unsigned eqCount )
 287 {
 288     unsigned i, chn, eq;
 289     float   b0, b1, b2, a1, a2;
 290     float   x, y = 0;
 291     const float *src1 = src;
 292     float *dest1 = dest;
 293
 294     for (i = 0; i < samples; i++)
 295     {
 296         float *state1 = state;
 297         for (chn = 0; chn < channels; chn++)
 298         {
 299             const float *coeffs1 = coeffs;
 300             x = *src1++;
 301             /* Direct form 1 IIRs */
 302             for (eq = 0; eq < eqCount; eq++)
 303             {
 304                 b0 = coeffs1[0];
 305                 b1 = coeffs1[1];
 306                 b2 = coeffs1[2];
 307                 a1 = coeffs1[3];
 308                 a2 = coeffs1[4];
 309                 coeffs1 += 5;
 310                 y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;
 311                 state1[1] = state1[0];
 312                 state1[0] = x;
 313                 state1[3] = state1[2];
 314                 state1[2] = y;
 315                 x = y;
 316                 state1 += 4;
 317             }
 318             *dest1++ = y;
 319         }
 320     }
 321 }
 322