-/*****************************************************************************\r
- * param_eq.c:\r
- *****************************************************************************\r
- * Copyright (C) 2006 the VideoLAN team\r
- * $Id: equalizer.c 13905 2006-01-12 23:10:04Z dionoea $\r
- *\r
- * Authors: Antti Huovilainen\r
- * Sigmund A. Helberg <dnumgis@videolan.org>\r
- *\r
- * This program is free software; you can redistribute it and/or modify\r
- * it under the terms of the GNU General Public License as published by\r
- * the Free Software Foundation; either version 2 of the License, or\r
- * (at your option) any later version.\r
- *\r
- * This program is distributed in the hope that it will be useful,\r
- * but WITHOUT ANY WARRANTY; without even the implied warranty of\r
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\r
- * GNU General Public License for more details.\r
- *\r
- * You should have received a copy of the GNU General Public License\r
- * along with this program; if not, write to the Free Software\r
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.\r
- *****************************************************************************/\r
-\r
-/*****************************************************************************\r
- * Preamble\r
- *****************************************************************************/\r
-#include <stdlib.h> /* malloc(), free() */\r
-#include <string.h>\r
-#include <math.h>\r
-\r
-#include <vlc/vlc.h>\r
-\r
-#include <vlc/aout.h>\r
-#include "aout_internal.h"\r
-\r
-/*****************************************************************************\r
- * Module descriptor\r
- *****************************************************************************/\r
-static int Open ( vlc_object_t * );\r
-static void Close( vlc_object_t * );\r
-static void CalcPeakEQCoeffs( float, float, float, float, float * );\r
-static void CalcShelfEQCoeffs( float, float, float, int, float, float * );\r
-static void ProcessEQ( float *, float *, float *, int, int, float *, int );\r
-static void DoWork( aout_instance_t *, aout_filter_t *,\r
- aout_buffer_t *, aout_buffer_t * );\r
-\r
-vlc_module_begin();\r
- set_description( _("Parametric Equalizer") );\r
- set_shortname( N_("Parametric Equalizer" ) );\r
- set_capability( "audio filter", 0 );\r
- set_category( CAT_AUDIO );\r
- set_subcategory( SUBCAT_AUDIO_AFILTER );\r
-\r
- add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),NULL, VLC_FALSE );\r
- add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,\r
- N_("Low freq gain (Db)"), NULL,VLC_FALSE );\r
- add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),NULL, VLC_FALSE );\r
- add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,\r
- N_("High freq gain (Db)"), NULL,VLC_FALSE );\r
- add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),NULL, VLC_FALSE );\r
- add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,\r
- N_("Freq 1 gain (Db)"), NULL,VLC_FALSE );\r
- add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,\r
- N_("Freq 1 Q"), NULL,VLC_FALSE );\r
- add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),NULL, VLC_FALSE );\r
- add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,\r
- N_("Freq 2 gain (Db)"), NULL,VLC_FALSE );\r
- add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,\r
- N_("Freq 2 Q"), NULL,VLC_FALSE );\r
- add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),NULL, VLC_FALSE );\r
- add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,\r
- N_("Freq 3 gain (Db)"), NULL,VLC_FALSE );\r
- add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,\r
- N_("Freq 3 Q"), NULL,VLC_FALSE );\r
-\r
- set_callbacks( Open, Close );\r
-vlc_module_end();\r
-\r
-/*****************************************************************************\r
- * Local prototypes\r
- *****************************************************************************/\r
-typedef struct aout_filter_sys_t\r
-{\r
- /* Filter static config */\r
- float f_lowf, f_lowgain;\r
- float f_f1, f_Q1, f_gain1;\r
- float f_f2, f_Q2, f_gain2;\r
- float f_f3, f_Q3, f_gain3;\r
- float f_highf, f_highgain;\r
- /* Filter computed coeffs */\r
- float coeffs[5*5];\r
- /* State */\r
- float *p_state;\r
- \r
-} aout_filter_sys_t;\r
-\r
-\r
-\r
-\r
-/*****************************************************************************\r
- * Open:\r
- *****************************************************************************/\r
-static int Open( vlc_object_t *p_this )\r
-{\r
- aout_filter_t *p_filter = (aout_filter_t *)p_this;\r
- aout_filter_sys_t *p_sys;\r
- vlc_bool_t b_fit = VLC_TRUE;\r
- int i_samplerate;\r
-\r
- if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||\r
- p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )\r
- {\r
- b_fit = VLC_FALSE;\r
- p_filter->input.i_format = VLC_FOURCC('f','l','3','2');\r
- p_filter->output.i_format = VLC_FOURCC('f','l','3','2');\r
- msg_Warn( p_filter, "Bad input or output format" );\r
- }\r
- if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )\r
- {\r
- b_fit = VLC_FALSE;\r
- memcpy( &p_filter->output, &p_filter->input,\r
- sizeof(audio_sample_format_t) );\r
- msg_Warn( p_filter, "input and output formats are not similar" );\r
- }\r
-\r
- if ( ! b_fit )\r
- {\r
- return VLC_EGENERIC;\r
- }\r
-\r
- p_filter->pf_do_work = DoWork;\r
- p_filter->b_in_place = VLC_TRUE;\r
-\r
- /* Allocate structure */\r
- p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );\r
-\r
- p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");\r
- p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");\r
- p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");\r
- p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");\r
- \r
- p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");\r
- p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");\r
- p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");\r
- \r
- p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");\r
- p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");\r
- p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");\r
-\r
- p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");\r
- p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");\r
- p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");\r
- \r
-\r
- i_samplerate = p_filter->input.i_rate;\r
- CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,\r
- i_samplerate, p_sys->coeffs+0*5);\r
- CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,\r
- i_samplerate, p_sys->coeffs+1*5);\r
- CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,\r
- i_samplerate, p_sys->coeffs+2*5);\r
- CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,\r
- i_samplerate, p_sys->coeffs+3*5);\r
- CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,\r
- i_samplerate, p_sys->coeffs+4*5);\r
- p_sys->p_state = (float*)calloc( p_filter->input.i_channels*5*4,\r
- sizeof(float) );\r
-\r
- return VLC_SUCCESS;\r
-}\r
-\r
-static void Close( vlc_object_t *p_this )\r
-{\r
- aout_filter_t *p_filter = (aout_filter_t *)p_this;\r
- free( p_filter->p_sys->p_state );\r
- free( p_filter->p_sys );\r
-}\r
-\r
-/*****************************************************************************\r
- * DoWork: process samples buffer\r
- *****************************************************************************\r
- *\r
- *****************************************************************************/\r
-static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,\r
- aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )\r
-{\r
- p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;\r
- p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;\r
-\r
- ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer,\r
- p_filter->p_sys->p_state, \r
- p_filter->input.i_channels, p_in_buf->i_nb_samples,\r
- p_filter->p_sys->coeffs, 5 );\r
-}\r
-\r
-/*\r
- * Calculate direct form IIR coefficients for peaking EQ\r
- * coeffs[0] = b0\r
- * coeffs[1] = b1\r
- * coeffs[2] = b2\r
- * coeffs[3] = a1\r
- * coeffs[4] = a2\r
- *\r
- * Equations taken from RBJ audio EQ cookbook \r
- * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)\r
- */\r
-static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,\r
- float *coeffs )\r
-{\r
- float A;\r
- float w0;\r
- float alpha;\r
- float b0, b1, b2;\r
- float a0, a1, a2;\r
-\r
- // Provide sane limits to avoid overflow\r
- if (Q < 0.1f) Q = 0.1f; \r
- if (Q > 100) Q = 100;\r
- if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;\r
- if (gainDB < -40) gainDB = -40;\r
- if (gainDB > 40) gainDB = 40;\r
- \r
- A = pow(10, gainDB/40);\r
- w0 = 2*3.141593f*f0/Fs;\r
- alpha = sin(w0)/(2*Q);\r
- \r
- b0 = 1 + alpha*A;\r
- b1 = -2*cos(w0);\r
- b2 = 1 - alpha*A;\r
- a0 = 1 + alpha/A;\r
- a1 = -2*cos(w0);\r
- a2 = 1 - alpha/A;\r
- \r
- // Store values to coeffs and normalize by 1/a0\r
- coeffs[0] = b0/a0;\r
- coeffs[1] = b1/a0;\r
- coeffs[2] = b2/a0;\r
- coeffs[3] = a1/a0;\r
- coeffs[4] = a2/a0;\r
-}\r
-\r
-/*\r
- * Calculate direct form IIR coefficients for low/high shelf EQ\r
- * coeffs[0] = b0\r
- * coeffs[1] = b1\r
- * coeffs[2] = b2\r
- * coeffs[3] = a1\r
- * coeffs[4] = a2\r
- *\r
- * Equations taken from RBJ audio EQ cookbook \r
- * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)\r
- */\r
-static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,\r
- float Fs, float *coeffs )\r
-{\r
- float A;\r
- float w0;\r
- float alpha;\r
- float b0, b1, b2;\r
- float a0, a1, a2;\r
-\r
- // Provide sane limits to avoid overflow\r
- if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;\r
- if (gainDB < -40) gainDB = -40;\r
- if (gainDB > 40) gainDB = 40;\r
-\r
- A = pow(10, gainDB/40);\r
- w0 = 2*3.141593f*f0/Fs;\r
- alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );\r
-\r
- if (high)\r
- {\r
- b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );\r
- b1 = -2*A*( (A-1) + (A+1)*cos(w0) );\r
- b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );\r
- a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;\r
- a1 = 2*( (A-1) - (A+1)*cos(w0) );\r
- a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;\r
- }\r
- else\r
- {\r
- b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );\r
- b1 = 2*A*( (A-1) - (A+1)*cos(w0));\r
- b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );\r
- a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;\r
- a1 = -2*( (A-1) + (A+1)*cos(w0));\r
- a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;\r
- }\r
- // Store values to coeffs and normalize by 1/a0\r
- coeffs[0] = b0/a0;\r
- coeffs[1] = b1/a0;\r
- coeffs[2] = b2/a0;\r
- coeffs[3] = a1/a0;\r
- coeffs[4] = a2/a0;\r
-}\r
-\r
-/*\r
- src is assumed to be interleaved\r
- dest is assumed to be interleaved\r
- size of state is 4*channels*eqCount\r
- samples is not premultiplied by channels\r
- size of coeffs is 5*eqCount\r
-*/\r
-void ProcessEQ( float *src, float *dest, float *state, \r
- int channels, int samples, float *coeffs, \r
- int eqCount )\r
-{\r
- int i, chn, eq;\r
- float b0, b1, b2, a1, a2;\r
- float x, y = 0;\r
- float *src1, *dest1;\r
- float *coeffs1, *state1;\r
- src1 = src;\r
- dest1 = dest;\r
- for (i = 0; i < samples; i++)\r
- {\r
- state1 = state;\r
- for (chn = 0; chn < channels; chn++)\r
- {\r
- coeffs1 = coeffs;\r
- x = *src1++;\r
- /* Direct form 1 IIRs */\r
- for (eq = 0; eq < eqCount; eq++)\r
- {\r
- b0 = coeffs1[0];\r
- b1 = coeffs1[1];\r
- b2 = coeffs1[2];\r
- a1 = coeffs1[3];\r
- a2 = coeffs1[4];\r
- coeffs1 += 5;\r
- y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;\r
- state1[1] = state1[0];\r
- state1[0] = x;\r
- state1[3] = state1[2];\r
- state1[2] = y;\r
- x = y;\r
- state1 += 4;\r
- }\r
- *dest1++ = y;\r
- }\r
- }\r
-}\r
-\r
+/*****************************************************************************
+ * param_eq.c:
+ *****************************************************************************
+ * Copyright (C) 2006 the VideoLAN team
+ * $Id$
+ *
+ * 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;
+ }
+ }
+}
+
projects / vlc / commitdiff