1 /*****************************************************************************
3 *****************************************************************************
4 * Copyright © 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 *****************************************************************************/
35 #include <vlc_common.h>
36 #include <vlc_plugin.h>
39 /*****************************************************************************
41 *****************************************************************************/
42 static int Open ( vlc_object_t * );
43 static void Close( vlc_object_t * );
44 static void CalcPeakEQCoeffs( float, float, float, float, float * );
45 static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
46 static void ProcessEQ( float *, float *, float *, unsigned, unsigned, float *, unsigned );
47 static void DoWork( aout_instance_t *, aout_filter_t *,
48 aout_buffer_t *, aout_buffer_t * );
51 set_description( N_("Parametric Equalizer") )
52 set_shortname( N_("Parametric Equalizer" ) )
53 set_capability( "audio filter", 0 )
54 set_category( CAT_AUDIO )
55 set_subcategory( SUBCAT_AUDIO_AFILTER )
57 add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", false )
58 add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
59 N_("Low freq gain (dB)"), "",false )
60 add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", false )
61 add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
62 N_("High freq gain (dB)"),"",false )
63 add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", false )
64 add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
65 N_("Freq 1 gain (dB)"), "",false )
66 add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
67 N_("Freq 1 Q"), "",false )
68 add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", false )
69 add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
70 N_("Freq 2 gain (dB)"),"",false )
71 add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
72 N_("Freq 2 Q"),"",false )
73 add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", false )
74 add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
75 N_("Freq 3 gain (dB)"),"",false )
76 add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
77 N_("Freq 3 Q"),"",false )
79 set_callbacks( Open, Close )
82 /*****************************************************************************
84 *****************************************************************************/
85 struct aout_filter_sys_t
87 /* Filter static config */
88 float f_lowf, f_lowgain;
89 float f_f1, f_Q1, f_gain1;
90 float f_f2, f_Q2, f_gain2;
91 float f_f3, f_Q3, f_gain3;
92 float f_highf, f_highgain;
93 /* Filter computed coeffs */
103 /*****************************************************************************
105 *****************************************************************************/
106 static int Open( vlc_object_t *p_this )
108 aout_filter_t *p_filter = (aout_filter_t *)p_this;
109 aout_filter_sys_t *p_sys;
113 if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||
114 p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )
117 p_filter->input.i_format = VLC_FOURCC('f','l','3','2');
118 p_filter->output.i_format = VLC_FOURCC('f','l','3','2');
119 msg_Warn( p_filter, "bad input or output format" );
121 if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
124 memcpy( &p_filter->output, &p_filter->input,
125 sizeof(audio_sample_format_t) );
126 msg_Warn( p_filter, "input and output formats are not similar" );
134 p_filter->pf_do_work = DoWork;
135 p_filter->b_in_place = true;
137 /* Allocate structure */
138 p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
140 p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");
141 p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");
142 p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");
143 p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");
145 p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");
146 p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");
147 p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");
149 p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");
150 p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");
151 p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");
153 p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");
154 p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");
155 p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");
158 i_samplerate = p_filter->input.i_rate;
159 CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,
160 i_samplerate, p_sys->coeffs+0*5);
161 CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,
162 i_samplerate, p_sys->coeffs+1*5);
163 CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,
164 i_samplerate, p_sys->coeffs+2*5);
165 CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,
166 i_samplerate, p_sys->coeffs+3*5);
167 CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,
168 i_samplerate, p_sys->coeffs+4*5);
169 p_sys->p_state = (float*)calloc( p_filter->input.i_channels*5*4,
175 static void Close( vlc_object_t *p_this )
177 aout_filter_t *p_filter = (aout_filter_t *)p_this;
178 free( p_filter->p_sys->p_state );
179 free( p_filter->p_sys );
182 /*****************************************************************************
183 * DoWork: process samples buffer
184 *****************************************************************************
186 *****************************************************************************/
187 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
188 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
191 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
192 p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
194 ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer,
195 p_filter->p_sys->p_state,
196 p_filter->input.i_channels, p_in_buf->i_nb_samples,
197 p_filter->p_sys->coeffs, 5 );
201 * Calculate direct form IIR coefficients for peaking EQ
208 * Equations taken from RBJ audio EQ cookbook
209 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
211 static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,
220 // Provide sane limits to avoid overflow
221 if (Q < 0.1f) Q = 0.1f;
222 if (Q > 100) Q = 100;
223 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
224 if (gainDB < -40) gainDB = -40;
225 if (gainDB > 40) gainDB = 40;
227 A = pow(10, gainDB/40);
228 w0 = 2*3.141593f*f0/Fs;
229 alpha = sin(w0)/(2*Q);
238 // Store values to coeffs and normalize by 1/a0
247 * Calculate direct form IIR coefficients for low/high shelf EQ
254 * Equations taken from RBJ audio EQ cookbook
255 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
257 static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,
258 float Fs, float *coeffs )
266 // Provide sane limits to avoid overflow
267 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
268 if (gainDB < -40) gainDB = -40;
269 if (gainDB > 40) gainDB = 40;
271 A = pow(10, gainDB/40);
272 w0 = 2*3.141593f*f0/Fs;
273 alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );
277 b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
278 b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
279 b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
280 a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
281 a1 = 2*( (A-1) - (A+1)*cos(w0) );
282 a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
286 b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
287 b1 = 2*A*( (A-1) - (A+1)*cos(w0));
288 b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
289 a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
290 a1 = -2*( (A-1) + (A+1)*cos(w0));
291 a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
293 // Store values to coeffs and normalize by 1/a0
302 src is assumed to be interleaved
303 dest is assumed to be interleaved
304 size of state is 4*channels*eqCount
305 samples is not premultiplied by channels
306 size of coeffs is 5*eqCount
308 void ProcessEQ( float *src, float *dest, float *state,
309 unsigned channels, unsigned samples, float *coeffs,
313 float b0, b1, b2, a1, a2;
316 float *coeffs1, *state1;
319 for (i = 0; i < samples; i++)
322 for (chn = 0; chn < channels; chn++)
326 /* Direct form 1 IIRs */
327 for (eq = 0; eq < eqCount; eq++)
335 y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;
336 state1[1] = state1[0];
338 state1[3] = state1[2];