all: parametric equalizer by Antti Huovilainen (adapted for vlc by me)

configure.ac

@@ -523,7 +523,7 @@ AC_CHECK_LIB(m,cos,[
   VLC_ADD_LDFLAGS([adjust distort a52tofloat32 dtstofloat32 x264 goom],[-lm])
 ])
 AC_CHECK_LIB(m,pow,[
-  VLC_ADD_LDFLAGS([ffmpeg ffmpegaltivec stream_out_transrate i420_rgb faad twolame equalizer vlc freetype mpc],[-lm])
+  VLC_ADD_LDFLAGS([ffmpeg ffmpegaltivec stream_out_transrate i420_rgb faad twolame equalizer param_eq vlc freetype mpc],[-lm])
 ])
 AC_CHECK_LIB(m,sqrt,[
   VLC_ADD_LDFLAGS([headphone_channel_mixer normvol speex],[-lm])
@@ -1069,7 +1069,7 @@ dnl  VLC_ADD_PLUGINS([externrun])
   VLC_ADD_PLUGINS([i420_yuy2 i422_yuy2 i420_ymga])
   VLC_ADD_PLUGINS([aout_file linear_resampler bandlimited_resampler])
   VLC_ADD_PLUGINS([float32_mixer spdif_mixer simple_channel_mixer])
-  VLC_ADD_PLUGINS([dolby_surround_decoder headphone_channel_mixer normvol equalizer])
+  VLC_ADD_PLUGINS([dolby_surround_decoder headphone_channel_mixer normvol equalizer param_eq])
   VLC_ADD_PLUGINS([fixed32tofloat32 float32tos16 float32tos8 float32tou16 float32tou8 a52tospdif dtstospdif s16tofloat32 s16tofloat32swab s8tofloat32 u8tofloat32 audio_format])
 fi
 

modules/audio_filter/Modules.am

 SOURCES_equalizer = equalizer.c equalizer_presets.h
 SOURCES_normvol = normvol.c
 SOURCES_audio_format = format.c
+SOURCES_param_eq = param_eq.c

modules/audio_filter/param_eq.c

+/*****************************************************************************
+ * param_eq.c:
+ *****************************************************************************
+ * Copyright (C) 2006 the VideoLAN team
+ * $Id: equalizer.c 13905 2006-01-12 23:10:04Z dionoea $
+ *
+ * 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 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>
+#include "aout_internal.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( N_("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)"),NULL, VLC_FALSE );
+    add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
+                          N_("Low freq gain (Db)"), NULL,VLC_FALSE );
+    add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),NULL, VLC_FALSE );
+    add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
+                          N_("High freq gain (Db)"), NULL,VLC_FALSE );
+    add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),NULL, VLC_FALSE );
+    add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
+                          N_("Freq 1 gain (Db)"), NULL,VLC_FALSE );
+    add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
+                          N_("Freq 1 Q"), NULL,VLC_FALSE );
+    add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),NULL, VLC_FALSE );
+    add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
+                          N_("Freq 2 gain (Db)"), NULL,VLC_FALSE );
+    add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
+                          N_("Freq 2 Q"), NULL,VLC_FALSE );
+    add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),NULL, VLC_FALSE );
+    add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
+                          N_("Freq 3 gain (Db)"), NULL,VLC_FALSE );
+    add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
+                          N_("Freq 3 Q"), NULL,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;
+        }
+    }
+}
+