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 *****************************************************************************/
33 /*****************************************************************************
35 *****************************************************************************/
36 static int Open ( vlc_object_t * );
37 static void Close( vlc_object_t * );
38 static void CalcPeakEQCoeffs( float, float, float, float, float * );
39 static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
40 static void ProcessEQ( float *, float *, float *, unsigned, unsigned, float *, unsigned );
41 static void DoWork( aout_instance_t *, aout_filter_t *,
42 aout_buffer_t *, aout_buffer_t * );
45 set_description( _("Parametric Equalizer") );
46 set_shortname( _("Parametric Equalizer" ) );
47 set_capability( "audio filter", 0 );
48 set_category( CAT_AUDIO );
49 set_subcategory( SUBCAT_AUDIO_AFILTER );
51 add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", VLC_FALSE );
52 add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
53 N_("Low freq gain (dB)"), "",VLC_FALSE );
54 add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", VLC_FALSE );
55 add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
56 N_("High freq gain (dB)"),"",VLC_FALSE );
57 add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", VLC_FALSE );
58 add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
59 N_("Freq 1 gain (dB)"), "",VLC_FALSE );
60 add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
61 N_("Freq 1 Q"), "",VLC_FALSE );
62 add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", VLC_FALSE );
63 add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
64 N_("Freq 2 gain (dB)"),"",VLC_FALSE );
65 add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
66 N_("Freq 2 Q"),"",VLC_FALSE );
67 add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", VLC_FALSE );
68 add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
69 N_("Freq 3 gain (dB)"),"",VLC_FALSE );
70 add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
71 N_("Freq 3 Q"),"",VLC_FALSE );
73 set_callbacks( Open, Close );
76 /*****************************************************************************
78 *****************************************************************************/
79 typedef struct aout_filter_sys_t
81 /* Filter static config */
82 float f_lowf, f_lowgain;
83 float f_f1, f_Q1, f_gain1;
84 float f_f2, f_Q2, f_gain2;
85 float f_f3, f_Q3, f_gain3;
86 float f_highf, f_highgain;
87 /* Filter computed coeffs */
97 /*****************************************************************************
99 *****************************************************************************/
100 static int Open( vlc_object_t *p_this )
102 aout_filter_t *p_filter = (aout_filter_t *)p_this;
103 aout_filter_sys_t *p_sys;
104 vlc_bool_t b_fit = VLC_TRUE;
107 if( p_filter->input.i_format != VLC_FOURCC('f','l','3','2' ) ||
108 p_filter->output.i_format != VLC_FOURCC('f','l','3','2') )
111 p_filter->input.i_format = VLC_FOURCC('f','l','3','2');
112 p_filter->output.i_format = VLC_FOURCC('f','l','3','2');
113 msg_Warn( p_filter, "bad input or output format" );
115 if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
118 memcpy( &p_filter->output, &p_filter->input,
119 sizeof(audio_sample_format_t) );
120 msg_Warn( p_filter, "input and output formats are not similar" );
128 p_filter->pf_do_work = DoWork;
129 p_filter->b_in_place = VLC_TRUE;
131 /* Allocate structure */
132 p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
134 p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");
135 p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");
136 p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");
137 p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");
139 p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");
140 p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");
141 p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");
143 p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");
144 p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");
145 p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");
147 p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");
148 p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");
149 p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");
152 i_samplerate = p_filter->input.i_rate;
153 CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,
154 i_samplerate, p_sys->coeffs+0*5);
155 CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,
156 i_samplerate, p_sys->coeffs+1*5);
157 CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,
158 i_samplerate, p_sys->coeffs+2*5);
159 CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,
160 i_samplerate, p_sys->coeffs+3*5);
161 CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,
162 i_samplerate, p_sys->coeffs+4*5);
163 p_sys->p_state = (float*)calloc( p_filter->input.i_channels*5*4,
169 static void Close( vlc_object_t *p_this )
171 aout_filter_t *p_filter = (aout_filter_t *)p_this;
172 free( p_filter->p_sys->p_state );
173 free( p_filter->p_sys );
176 /*****************************************************************************
177 * DoWork: process samples buffer
178 *****************************************************************************
180 *****************************************************************************/
181 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
182 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
184 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
185 p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
187 ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_out_buf->p_buffer,
188 p_filter->p_sys->p_state,
189 p_filter->input.i_channels, p_in_buf->i_nb_samples,
190 p_filter->p_sys->coeffs, 5 );
194 * Calculate direct form IIR coefficients for peaking EQ
201 * Equations taken from RBJ audio EQ cookbook
202 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
204 static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,
213 // Provide sane limits to avoid overflow
214 if (Q < 0.1f) Q = 0.1f;
215 if (Q > 100) Q = 100;
216 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
217 if (gainDB < -40) gainDB = -40;
218 if (gainDB > 40) gainDB = 40;
220 A = pow(10, gainDB/40);
221 w0 = 2*3.141593f*f0/Fs;
222 alpha = sin(w0)/(2*Q);
231 // Store values to coeffs and normalize by 1/a0
240 * Calculate direct form IIR coefficients for low/high shelf EQ
247 * Equations taken from RBJ audio EQ cookbook
248 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
250 static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,
251 float Fs, float *coeffs )
259 // Provide sane limits to avoid overflow
260 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
261 if (gainDB < -40) gainDB = -40;
262 if (gainDB > 40) gainDB = 40;
264 A = pow(10, gainDB/40);
265 w0 = 2*3.141593f*f0/Fs;
266 alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );
270 b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
271 b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
272 b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
273 a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
274 a1 = 2*( (A-1) - (A+1)*cos(w0) );
275 a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
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;
286 // Store values to coeffs and normalize by 1/a0
295 src is assumed to be interleaved
296 dest is assumed to be interleaved
297 size of state is 4*channels*eqCount
298 samples is not premultiplied by channels
299 size of coeffs is 5*eqCount
301 void ProcessEQ( float *src, float *dest, float *state,
302 unsigned channels, unsigned samples, float *coeffs,
306 float b0, b1, b2, a1, a2;
309 float *coeffs1, *state1;
312 for (i = 0; i < samples; i++)
315 for (chn = 0; chn < channels; chn++)
319 /* Direct form 1 IIRs */
320 for (eq = 0; eq < eqCount; eq++)
328 y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;
329 state1[1] = state1[0];
331 state1[3] = state1[2];