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>
38 #include <vlc_filter.h>
40 /*****************************************************************************
42 *****************************************************************************/
43 static int Open ( vlc_object_t * );
44 static void Close( vlc_object_t * );
45 static void CalcPeakEQCoeffs( float, float, float, float, float * );
46 static void CalcShelfEQCoeffs( float, float, float, int, float, float * );
47 static void ProcessEQ( const float *, float *, float *, unsigned, unsigned,
48 const float *, unsigned );
49 static block_t *DoWork( filter_t *, block_t * );
52 set_description( N_("Parametric Equalizer") )
53 set_shortname( N_("Parametric Equalizer" ) )
54 set_capability( "audio filter", 0 )
55 set_category( CAT_AUDIO )
56 set_subcategory( SUBCAT_AUDIO_AFILTER )
58 add_float( "param-eq-lowf", 100, NULL, N_("Low freq (Hz)"),"", false )
59 add_float_with_range( "param-eq-lowgain", 0, -20.0, 20.0, NULL,
60 N_("Low freq gain (dB)"), "",false )
61 add_float( "param-eq-highf", 10000, NULL, N_("High freq (Hz)"),"", false )
62 add_float_with_range( "param-eq-highgain", 0, -20.0, 20.0, NULL,
63 N_("High freq gain (dB)"),"",false )
64 add_float( "param-eq-f1", 300, NULL, N_("Freq 1 (Hz)"),"", false )
65 add_float_with_range( "param-eq-gain1", 0, -20.0, 20.0, NULL,
66 N_("Freq 1 gain (dB)"), "",false )
67 add_float_with_range( "param-eq-q1", 3, 0.1, 100.0, NULL,
68 N_("Freq 1 Q"), "",false )
69 add_float( "param-eq-f2", 1000, NULL, N_("Freq 2 (Hz)"),"", false )
70 add_float_with_range( "param-eq-gain2", 0, -20.0, 20.0, NULL,
71 N_("Freq 2 gain (dB)"),"",false )
72 add_float_with_range( "param-eq-q2", 3, 0.1, 100.0, NULL,
73 N_("Freq 2 Q"),"",false )
74 add_float( "param-eq-f3", 3000, NULL, N_("Freq 3 (Hz)"),"", false )
75 add_float_with_range( "param-eq-gain3", 0, -20.0, 20.0, NULL,
76 N_("Freq 3 gain (dB)"),"",false )
77 add_float_with_range( "param-eq-q3", 3, 0.1, 100.0, NULL,
78 N_("Freq 3 Q"),"",false )
80 set_callbacks( Open, Close )
83 /*****************************************************************************
85 *****************************************************************************/
88 /* Filter static config */
89 float f_lowf, f_lowgain;
90 float f_f1, f_Q1, f_gain1;
91 float f_f2, f_Q2, f_gain2;
92 float f_f3, f_Q3, f_gain3;
93 float f_highf, f_highgain;
94 /* Filter computed coeffs */
103 /*****************************************************************************
105 *****************************************************************************/
106 static int Open( vlc_object_t *p_this )
108 filter_t *p_filter = (filter_t *)p_this;
111 unsigned i_samplerate;
113 if( p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ||
114 p_filter->fmt_out.audio.i_format != VLC_CODEC_FL32 )
117 p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
118 p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
119 msg_Warn( p_filter, "bad input or output format" );
121 if ( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio, &p_filter->fmt_out.audio ) )
124 memcpy( &p_filter->fmt_out.audio, &p_filter->fmt_in.audio,
125 sizeof(audio_sample_format_t) );
126 msg_Warn( p_filter, "input and output formats are not similar" );
134 /* Allocate structure */
135 p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
139 p_filter->pf_audio_filter = DoWork;
141 p_sys->f_lowf = config_GetFloat( p_this, "param-eq-lowf");
142 p_sys->f_lowgain = config_GetFloat( p_this, "param-eq-lowgain");
143 p_sys->f_highf = config_GetFloat( p_this, "param-eq-highf");
144 p_sys->f_highgain = config_GetFloat( p_this, "param-eq-highgain");
146 p_sys->f_f1 = config_GetFloat( p_this, "param-eq-f1");
147 p_sys->f_Q1 = config_GetFloat( p_this, "param-eq-q1");
148 p_sys->f_gain1 = config_GetFloat( p_this, "param-eq-gain1");
150 p_sys->f_f2 = config_GetFloat( p_this, "param-eq-f2");
151 p_sys->f_Q2 = config_GetFloat( p_this, "param-eq-q2");
152 p_sys->f_gain2 = config_GetFloat( p_this, "param-eq-gain2");
154 p_sys->f_f3 = config_GetFloat( p_this, "param-eq-f3");
155 p_sys->f_Q3 = config_GetFloat( p_this, "param-eq-q3");
156 p_sys->f_gain3 = config_GetFloat( p_this, "param-eq-gain3");
159 i_samplerate = p_filter->fmt_in.audio.i_rate;
160 CalcPeakEQCoeffs(p_sys->f_f1, p_sys->f_Q1, p_sys->f_gain1,
161 i_samplerate, p_sys->coeffs+0*5);
162 CalcPeakEQCoeffs(p_sys->f_f2, p_sys->f_Q2, p_sys->f_gain2,
163 i_samplerate, p_sys->coeffs+1*5);
164 CalcPeakEQCoeffs(p_sys->f_f3, p_sys->f_Q3, p_sys->f_gain3,
165 i_samplerate, p_sys->coeffs+2*5);
166 CalcShelfEQCoeffs(p_sys->f_lowf, 1, p_sys->f_lowgain, 0,
167 i_samplerate, p_sys->coeffs+3*5);
168 CalcShelfEQCoeffs(p_sys->f_highf, 1, p_sys->f_highgain, 0,
169 i_samplerate, p_sys->coeffs+4*5);
170 p_sys->p_state = (float*)calloc( p_filter->fmt_in.audio.i_channels*5*4,
176 static void Close( vlc_object_t *p_this )
178 filter_t *p_filter = (filter_t *)p_this;
179 free( p_filter->p_sys->p_state );
180 free( p_filter->p_sys );
183 /*****************************************************************************
184 * DoWork: process samples buffer
185 *****************************************************************************
187 *****************************************************************************/
188 static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
190 ProcessEQ( (float*)p_in_buf->p_buffer, (float*)p_in_buf->p_buffer,
191 p_filter->p_sys->p_state,
192 p_filter->fmt_in.audio.i_channels, p_in_buf->i_nb_samples,
193 p_filter->p_sys->coeffs, 5 );
198 * Calculate direct form IIR coefficients for peaking EQ
205 * Equations taken from RBJ audio EQ cookbook
206 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
208 static void CalcPeakEQCoeffs( float f0, float Q, float gainDB, float Fs,
217 // Provide sane limits to avoid overflow
218 if (Q < 0.1f) Q = 0.1f;
219 if (Q > 100) Q = 100;
220 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
221 if (gainDB < -40) gainDB = -40;
222 if (gainDB > 40) gainDB = 40;
224 A = pow(10, gainDB/40);
225 w0 = 2*3.141593f*f0/Fs;
226 alpha = sin(w0)/(2*Q);
235 // Store values to coeffs and normalize by 1/a0
244 * Calculate direct form IIR coefficients for low/high shelf EQ
251 * Equations taken from RBJ audio EQ cookbook
252 * (http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
254 static void CalcShelfEQCoeffs( float f0, float slope, float gainDB, int high,
255 float Fs, float *coeffs )
263 // Provide sane limits to avoid overflow
264 if (f0 > Fs/2*0.95f) f0 = Fs/2*0.95f;
265 if (gainDB < -40) gainDB = -40;
266 if (gainDB > 40) gainDB = 40;
268 A = pow(10, gainDB/40);
269 w0 = 2*3.141593f*f0/Fs;
270 alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/slope - 1) + 2 );
274 b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
275 b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
276 b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
277 a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
278 a1 = 2*( (A-1) - (A+1)*cos(w0) );
279 a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
283 b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
284 b1 = 2*A*( (A-1) - (A+1)*cos(w0));
285 b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
286 a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
287 a1 = -2*( (A-1) + (A+1)*cos(w0));
288 a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
290 // Store values to coeffs and normalize by 1/a0
299 src is assumed to be interleaved
300 dest is assumed to be interleaved
301 size of state is 4*channels*eqCount
302 samples is not premultiplied by channels
303 size of coeffs is 5*eqCount
305 void ProcessEQ( const float *src, float *dest, float *state,
306 unsigned channels, unsigned samples, const float *coeffs,
310 float b0, b1, b2, a1, a2;
312 const float *src1 = src;
315 for (i = 0; i < samples; i++)
317 float *state1 = state;
318 for (chn = 0; chn < channels; chn++)
320 const float *coeffs1 = coeffs;
322 /* Direct form 1 IIRs */
323 for (eq = 0; eq < eqCount; eq++)
331 y = x*b0 + state1[0]*b1 + state1[1]*b2 - state1[2]*a1 - state1[3]*a2;
332 state1[1] = state1[0];
334 state1[3] = state1[2];