1 /*****************************************************************************
3 *****************************************************************************
4 * Copyright (C) 2006 the VideoLAN team
7 * Authors: Antti Huovilainen
8 * Sigmund A. Helberg <dnumgis@videolan.org>
10 * This program is free software; you can redistribute it and/or modify
11 * it under the terms of the GNU General Public License as published by
12 * the Free Software Foundation; either version 2 of the License, or
13 * (at your option) any later version.
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 General Public License for more details.
20 * You should have received a copy of the GNU General Public License
21 * along with this program; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
23 *****************************************************************************/
25 /*****************************************************************************
27 *****************************************************************************/
28 #include <stdlib.h> /* malloc(), free() */
35 #include "aout_internal.h"
37 /*****************************************************************************
39 *****************************************************************************/
40 static int Open ( vlc_object_t * );
41 static void Close( vlc_object_t * );
42 static void CalcPeakEQCoeffs( float, float, float, float, float * );
43 static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
44 static void ProcessEQ( float *, float *, float *, int, int, float *, int );
45 static void DoWork( aout_instance_t *, aout_filter_t *,
46 aout_buffer_t *, aout_buffer_t * );
49 set_description( _("Parametric Equalizer") );
50 set_shortname( N_("Parametric Equalizer" ) );
51 set_capability( "audio filter", 0 );
52 set_category( CAT_AUDIO );
53 set_subcategory( SUBCAT_AUDIO_AFILTER );
55 add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", VLC_FALSE );
57 add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
58 N_("Low freq gain (Db)"), "",VLC_FALSE );
59 add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", VLC_FALSE );
61 add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
62 N_("High freq gain (Db)"),"",VLC_FALSE );
63 add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", VLC_FALSE );
65 add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
66 N_("Freq 1 gain (Db)"), "",VLC_FALSE );
67 add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
68 N_("Freq 1 Q"), "",VLC_FALSE );
69 add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", VLC_FALSE );
71 add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
72 N_("Freq 2 gain (Db)"),"",VLC_FALSE );
73 add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
74 N_("Freq 2 Q"),"",VLC_FALSE );
75 add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", VLC_FALSE );
77 add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
78 N_("Freq 3 gain (Db)"),"",VLC_FALSE );
79 add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
80 N_("Freq 3 Q"),"",VLC_FALSE );
82 set_callbacks( Open, Close );
85 /*****************************************************************************
87 *****************************************************************************/
88 typedef struct aout_filter_sys_t
90 /* Filter static config */
91 float f_lowf, f_lowgain;
92 float f_f1, f_Q1, f_gain1;
93 float f_f2, f_Q2, f_gain2;
94 float f_f3, f_Q3, f_gain3;
95 float f_highf, f_highgain;
96 /* Filter computed coeffs */
106 /*****************************************************************************
108 *****************************************************************************/
109 static int Open( vlc_object_t *p_this )
111 aout_filter_t *p_filter = (aout_filter_t *)p_this;
112 aout_filter_sys_t *p_sys;
113 vlc_bool_t b_fit = VLC_TRUE;
116 if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||
117 p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )
120 p_filter->input.i_format = VLC_FOURCC('f','l','3','2');
121 p_filter->output.i_format = VLC_FOURCC('f','l','3','2');
122 msg_Warn( p_filter, "bad input or output format" );
124 if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
127 memcpy( &p_filter->output, &p_filter->input,
128 sizeof(audio_sample_format_t) );
129 msg_Warn( p_filter, "input and output formats are not similar" );
137 p_filter->pf_do_work = DoWork;
138 p_filter->b_in_place = VLC_TRUE;
140 /* Allocate structure */
141 p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
143 p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");
144 p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");
145 p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");
146 p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");
148 p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");
149 p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");
150 p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");
152 p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");
153 p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");
154 p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");
156 p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");
157 p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");
158 p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");
161 i_samplerate = p_filter->input.i_rate;
162 CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,
163 i_samplerate, p_sys->coeffs+0*5);
164 CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,
165 i_samplerate, p_sys->coeffs+1*5);
166 CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,
167 i_samplerate, p_sys->coeffs+2*5);
168 CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,
169 i_samplerate, p_sys->coeffs+3*5);
170 CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,
171 i_samplerate, p_sys->coeffs+4*5);
172 p_sys->p_state = (float*)calloc( p_filter->input.i_channels*5*4,
178 static void Close( vlc_object_t *p_this )
180 aout_filter_t *p_filter = (aout_filter_t *)p_this;
181 free( p_filter->p_sys->p_state );
182 free( p_filter->p_sys );
185 /*****************************************************************************
186 * DoWork: process samples buffer
187 *****************************************************************************
189 *****************************************************************************/
190 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
191 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
193 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
194 p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
196 ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer,
197 p_filter->p_sys->p_state,
198 p_filter->input.i_channels, p_in_buf->i_nb_samples,
199 p_filter->p_sys->coeffs, 5 );
203 * Calculate direct form IIR coefficients for peaking EQ
210 * Equations taken from RBJ audio EQ cookbook
211 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
213 static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,
222 // Provide sane limits to avoid overflow
223 if (Q < 0.1f) Q = 0.1f;
224 if (Q > 100) Q = 100;
225 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
226 if (gainDB < -40) gainDB = -40;
227 if (gainDB > 40) gainDB = 40;
229 A = pow(10, gainDB/40);
230 w0 = 2*3.141593f*f0/Fs;
231 alpha = sin(w0)/(2*Q);
240 // Store values to coeffs and normalize by 1/a0
249 * Calculate direct form IIR coefficients for low/high shelf EQ
256 * Equations taken from RBJ audio EQ cookbook
257 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
259 static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,
260 float Fs, float *coeffs )
268 // Provide sane limits to avoid overflow
269 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
270 if (gainDB < -40) gainDB = -40;
271 if (gainDB > 40) gainDB = 40;
273 A = pow(10, gainDB/40);
274 w0 = 2*3.141593f*f0/Fs;
275 alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );
279 b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
280 b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
281 b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
282 a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
283 a1 = 2*( (A-1) - (A+1)*cos(w0) );
284 a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
288 b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
289 b1 = 2*A*( (A-1) - (A+1)*cos(w0));
290 b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
291 a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
292 a1 = -2*( (A-1) + (A+1)*cos(w0));
293 a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
295 // Store values to coeffs and normalize by 1/a0
304 src is assumed to be interleaved
305 dest is assumed to be interleaved
306 size of state is 4*channels*eqCount
307 samples is not premultiplied by channels
308 size of coeffs is 5*eqCount
310 void ProcessEQ( float *src, float *dest, float *state,
311 int channels, int samples, float *coeffs,
315 float b0, b1, b2, a1, a2;
318 float *coeffs1, *state1;
321 for (i = 0; i < samples; i++)
324 for (chn = 0; chn < channels; chn++)
328 /* Direct form 1 IIRs */
329 for (eq = 0; eq < eqCount; eq++)
337 y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;
338 state1[1] = state1[0];
340 state1[3] = state1[2];