Mercurial > mplayer.hg
annotate libaf/filter.c @ 33263:5f527a9a9521
Add an exit function.
This function will allow performing clean-up operations.
(MPlayer calls guiDone() before exiting, but only if the GUI has been
initialized, i.e. if guiInit() has been called successfully. Any
exit_player()/exit_player_with_rc() after GUI's cfg_read() until
guiInit(), or any exit_player() during guiInit() itself will end the GUI
without calling guiDone(). This exit function will at least handle
abortions during guiInit() itself. It will be called twice in case of an
guiExit() after GUI initialization - first directly, next by guiDone()
via MPlayer's exit_player_with_rc().)
author | ib |
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date | Tue, 03 May 2011 12:19:22 +0000 |
parents | 32725ca88fed |
children |
rev | line source |
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1 /* |
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2 * design and implementation of different types of digital filters |
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3 * |
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4 * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au |
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5 * |
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6 * This file is part of MPlayer. |
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7 * |
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8 * MPlayer is free software; you can redistribute it and/or modify |
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9 * it under the terms of the GNU General Public License as published by |
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10 * the Free Software Foundation; either version 2 of the License, or |
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11 * (at your option) any later version. |
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12 * |
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13 * MPlayer is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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16 * GNU General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU General Public License along |
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19 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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21 */ |
7568 | 22 |
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23 #include <string.h> |
7568 | 24 #include <math.h> |
25 #include "dsp.h" | |
26 | |
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27 /****************************************************************************** |
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28 * FIR filter implementations |
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29 ******************************************************************************/ |
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30 |
7568 | 31 /* C implementation of FIR filter y=w*x |
32 | |
33 n number of filter taps, where mod(n,4)==0 | |
34 w filter taps | |
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35 x input signal must be a circular buffer which is indexed backwards |
7568 | 36 */ |
27258 | 37 inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w, |
38 const FLOAT_TYPE* x) | |
7568 | 39 { |
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40 register FLOAT_TYPE y; // Output |
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41 y = 0.0; |
7568 | 42 do{ |
43 n--; | |
44 y+=w[n]*x[n]; | |
45 }while(n != 0); | |
46 return y; | |
47 } | |
48 | |
49 /* C implementation of parallel FIR filter y(k)=w(k) * x(k) (where * denotes convolution) | |
50 | |
51 n number of filter taps, where mod(n,4)==0 | |
52 d number of filters | |
53 xi current index in xq | |
54 w filter taps k by n big | |
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55 x input signal must be a circular buffers which are indexed backwards |
7568 | 56 y output buffer |
57 s output buffer stride | |
58 */ | |
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59 FLOAT_TYPE* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, |
27258 | 60 const FLOAT_TYPE** w, const FLOAT_TYPE** x, FLOAT_TYPE* y, |
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61 unsigned int s) |
7568 | 62 { |
27258 | 63 register const FLOAT_TYPE* xt = *x + xi; |
64 register const FLOAT_TYPE* wt = *w; | |
7568 | 65 register int nt = 2*n; |
66 while(d-- > 0){ | |
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67 *y = af_filter_fir(n,wt,xt); |
7568 | 68 wt+=n; |
69 xt+=nt; | |
70 y+=s; | |
71 } | |
72 return y; | |
73 } | |
74 | |
75 /* Add new data to circular queue designed to be used with a parallel | |
76 FIR filter, with d filters. xq is the circular queue, in pointing | |
77 at the new samples, xi current index in xq and n the length of the | |
78 filter. xq must be n*2 by k big, s is the index for in. | |
79 */ | |
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80 int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, |
27258 | 81 FLOAT_TYPE** xq, const FLOAT_TYPE* in, unsigned int s) |
7568 | 82 { |
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83 register FLOAT_TYPE* txq = *xq + xi; |
7568 | 84 register int nt = n*2; |
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85 |
7568 | 86 while(d-- >0){ |
87 *txq= *(txq+n) = *in; | |
88 txq+=nt; | |
89 in+=s; | |
90 } | |
91 return (++xi)&(n-1); | |
92 } | |
93 | |
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94 /****************************************************************************** |
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95 * FIR filter design |
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96 ******************************************************************************/ |
7568 | 97 |
98 /* Design FIR filter using the Window method | |
99 | |
100 n filter length must be odd for HP and BS filters | |
101 w buffer for the filter taps (must be n long) | |
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102 fc cutoff frequencies (1 for LP and HP, 2 for BP and BS) |
7568 | 103 0 < fc < 1 where 1 <=> Fs/2 |
104 flags window and filter type as defined in filter.h | |
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105 variables are ored together: i.e. LP|HAMMING will give a |
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106 low pass filter designed using a hamming window |
7568 | 107 opt beta constant used only when designing using kaiser windows |
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108 |
7568 | 109 returns 0 if OK, -1 if fail |
110 */ | |
27258 | 111 int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc, |
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112 unsigned int flags, FLOAT_TYPE opt) |
7568 | 113 { |
114 unsigned int o = n & 1; // Indicator for odd filter length | |
115 unsigned int end = ((n + 1) >> 1) - o; // Loop end | |
116 unsigned int i; // Loop index | |
117 | |
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118 FLOAT_TYPE k1 = 2 * M_PI; // 2*pi*fc1 |
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119 FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length |
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120 FLOAT_TYPE k3; // 2*pi*fc2 Constant used in BP and BS design |
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121 FLOAT_TYPE g = 0.0; // Gain |
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122 FLOAT_TYPE t1,t2,t3; // Temporary variables |
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123 FLOAT_TYPE fc1,fc2; // Cutoff frequencies |
7568 | 124 |
125 // Sanity check | |
126 if(!w || (n == 0)) return -1; | |
127 | |
128 // Get window coefficients | |
129 switch(flags & WINDOW_MASK){ | |
130 case(BOXCAR): | |
14274 | 131 af_window_boxcar(n,w); break; |
7568 | 132 case(TRIANG): |
14274 | 133 af_window_triang(n,w); break; |
7568 | 134 case(HAMMING): |
14274 | 135 af_window_hamming(n,w); break; |
7568 | 136 case(HANNING): |
14274 | 137 af_window_hanning(n,w); break; |
7568 | 138 case(BLACKMAN): |
14274 | 139 af_window_blackman(n,w); break; |
7568 | 140 case(FLATTOP): |
14274 | 141 af_window_flattop(n,w); break; |
7568 | 142 case(KAISER): |
14274 | 143 af_window_kaiser(n,w,opt); break; |
7568 | 144 default: |
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145 return -1; |
7568 | 146 } |
147 | |
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148 if(flags & (LP | HP)){ |
7568 | 149 fc1=*fc; |
150 // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2 | |
151 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25; | |
152 k1 *= fc1; | |
153 | |
154 if(flags & LP){ // Low pass filter | |
155 | |
156 // If the filter length is odd, there is one point which is exactly | |
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157 // in the middle. The value at this point is 2*fCutoff*sin(x)/x, |
7568 | 158 // where x is zero. To make sure nothing strange happens, we set this |
159 // value separately. | |
160 if (o){ | |
161 w[end] = fc1 * w[end] * 2.0; | |
162 g=w[end]; | |
163 } | |
164 | |
165 // Create filter | |
166 for (i=0 ; i<end ; i++){ | |
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167 t1 = (FLOAT_TYPE)(i+1) - k2; |
7568 | 168 w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc |
169 g += 2*w[end-i-1]; // Total gain in filter | |
170 } | |
171 } | |
172 else{ // High pass filter | |
173 if (!o) // High pass filters must have odd length | |
174 return -1; | |
175 w[end] = 1.0 - (fc1 * w[end] * 2.0); | |
176 g= w[end]; | |
177 | |
178 // Create filter | |
179 for (i=0 ; i<end ; i++){ | |
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180 t1 = (FLOAT_TYPE)(i+1); |
7568 | 181 w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc |
182 g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter | |
183 } | |
184 } | |
185 } | |
186 | |
187 if(flags & (BP | BS)){ | |
188 fc1=fc[0]; | |
189 fc2=fc[1]; | |
190 // Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2 | |
191 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25; | |
192 fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25; | |
193 k3 = k1 * fc2; // 2*pi*fc2 | |
194 k1 *= fc1; // 2*pi*fc1 | |
195 | |
196 if(flags & BP){ // Band pass | |
197 // Calculate center tap | |
198 if (o){ | |
199 g=w[end]*(fc1+fc2); | |
200 w[end] = (fc2 - fc1) * w[end] * 2.0; | |
201 } | |
202 | |
203 // Create filter | |
204 for (i=0 ; i<end ; i++){ | |
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205 t1 = (FLOAT_TYPE)(i+1) - k2; |
7568 | 206 t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2 |
207 t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1 | |
208 g += w[end-i-1] * (t3 + t2); // Total gain in filter | |
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209 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); |
7568 | 210 } |
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211 } |
7568 | 212 else{ // Band stop |
213 if (!o) // Band stop filters must have odd length | |
214 return -1; | |
215 w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0; | |
216 g= w[end]; | |
217 | |
218 // Create filter | |
219 for (i=0 ; i<end ; i++){ | |
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220 t1 = (FLOAT_TYPE)(i+1); |
7568 | 221 t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1 |
222 t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2 | |
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223 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); |
7568 | 224 g += 2*w[end-i-1]; // Total gain in filter |
225 } | |
226 } | |
227 } | |
228 | |
229 // Normalize gain | |
230 g=1/g; | |
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231 for (i=0; i<n; i++) |
7568 | 232 w[i] *= g; |
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233 |
7568 | 234 return 0; |
235 } | |
236 | |
237 /* Design polyphase FIR filter from prototype filter | |
238 | |
239 n length of prototype filter | |
240 k number of polyphase components | |
241 w prototype filter taps | |
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242 pw Parallel FIR filter |
7568 | 243 g Filter gain |
244 flags FWD forward indexing | |
245 REW reverse indexing | |
246 ODD multiply every 2nd filter tap by -1 => HP filter | |
247 | |
248 returns 0 if OK, -1 if fail | |
249 */ | |
27258 | 250 int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w, |
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251 FLOAT_TYPE** pw, FLOAT_TYPE g, unsigned int flags) |
7568 | 252 { |
253 int l = (int)n/k; // Length of individual FIR filters | |
254 int i; // Counters | |
255 int j; | |
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256 FLOAT_TYPE t; // g * w[i] |
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257 |
7568 | 258 // Sanity check |
259 if(l<1 || k<1 || !w || !pw) | |
260 return -1; | |
261 | |
262 // Do the stuff | |
263 if(flags&REW){ | |
264 for(j=l-1;j>-1;j--){//Columns | |
265 for(i=0;i<(int)k;i++){//Rows | |
266 t=g * *w++; | |
267 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? -1 : 1) : 1); | |
268 } | |
269 } | |
270 } | |
271 else{ | |
272 for(j=0;j<l;j++){//Columns | |
273 for(i=0;i<(int)k;i++){//Rows | |
274 t=g * *w++; | |
275 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? 1 : -1) : 1); | |
276 } | |
277 } | |
278 } | |
279 return -1; | |
280 } | |
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281 |
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282 /****************************************************************************** |
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283 * IIR filter design |
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284 ******************************************************************************/ |
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285 |
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286 /* Helper functions for the bilinear transform */ |
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287 |
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288 /* Pre-warp the coefficients of a numerator or denominator. |
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289 Note that a0 is assumed to be 1, so there is no wrapping |
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290 of it. |
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291 */ |
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292 static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs) |
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293 { |
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294 FLOAT_TYPE wp; |
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295 wp = 2.0 * fs * tan(M_PI * fc / fs); |
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296 a[2] = a[2]/(wp * wp); |
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297 a[1] = a[1]/wp; |
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298 } |
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299 |
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300 /* Transform the numerator and denominator coefficients of s-domain |
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301 biquad section into corresponding z-domain coefficients. |
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302 |
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303 The transfer function for z-domain is: |
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304 |
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305 1 + alpha1 * z^(-1) + alpha2 * z^(-2) |
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306 H(z) = ------------------------------------- |
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307 1 + beta1 * z^(-1) + beta2 * z^(-2) |
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308 |
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309 Store the 4 IIR coefficients in array pointed by coef in following |
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310 order: |
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311 beta1, beta2 (denominator) |
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312 alpha1, alpha2 (numerator) |
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313 |
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314 Arguments: |
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315 a - s-domain numerator coefficients |
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316 b - s-domain denominator coefficients |
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317 k - filter gain factor. Initially set to 1 and modified by each |
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318 biquad section in such a way, as to make it the |
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319 coefficient by which to multiply the overall filter gain |
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320 in order to achieve a desired overall filter gain, |
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321 specified in initial value of k. |
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322 fs - sampling rate (Hz) |
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323 coef - array of z-domain coefficients to be filled in. |
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324 |
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325 Return: On return, set coef z-domain coefficients and k to the gain |
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326 required to maintain overall gain = 1.0; |
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327 */ |
27258 | 328 static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE* k, |
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329 FLOAT_TYPE fs, FLOAT_TYPE *coef) |
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330 { |
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331 FLOAT_TYPE ad, bd; |
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332 |
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333 /* alpha (Numerator in s-domain) */ |
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334 ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0]; |
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335 /* beta (Denominator in s-domain) */ |
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336 bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0]; |
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337 |
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338 /* Update gain constant for this section */ |
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339 *k *= ad/bd; |
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340 |
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341 /* Denominator */ |
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342 *coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */ |
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343 *coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */ |
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344 |
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345 /* Numerator */ |
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346 *coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */ |
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347 *coef = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad; /* alpha2 */ |
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348 } |
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349 |
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350 |
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351 |
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352 /* IIR filter design using bilinear transform and prewarp. Transforms |
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353 2nd order s domain analog filter into a digital IIR biquad link. To |
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354 create a filter fill in a, b, Q and fs and make space for coef and k. |
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355 |
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356 |
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357 Example Butterworth design: |
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358 |
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359 Below are Butterworth polynomials, arranged as a series of 2nd |
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360 order sections: |
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361 |
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362 Note: n is filter order. |
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363 |
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364 n Polynomials |
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365 ------------------------------------------------------------------- |
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366 2 s^2 + 1.4142s + 1 |
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367 4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1) |
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368 6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1) |
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369 |
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370 For n=4 we have following equation for the filter transfer function: |
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371 1 1 |
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372 T(s) = --------------------------- * ---------------------------- |
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373 s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1 |
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374 |
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375 The filter consists of two 2nd order sections since highest s power |
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376 is 2. Now we can take the coefficients, or the numbers by which s |
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377 is multiplied and plug them into a standard formula to be used by |
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378 bilinear transform. |
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379 |
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380 Our standard form for each 2nd order section is: |
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381 |
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382 a2 * s^2 + a1 * s + a0 |
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383 H(s) = ---------------------- |
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384 b2 * s^2 + b1 * s + b0 |
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385 |
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386 Note that Butterworth numerator is 1 for all filter sections, which |
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387 means s^2 = 0 and s^1 = 0 |
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388 |
11000 | 389 Let's convert standard Butterworth polynomials into this form: |
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390 |
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391 0 + 0 + 1 0 + 0 + 1 |
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392 --------------------------- * -------------------------- |
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393 1 + ((1/Q) * 0.765367) + 1 1 + ((1/Q) * 1.847759) + 1 |
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394 |
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395 Section 1: |
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396 a2 = 0; a1 = 0; a0 = 1; |
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397 b2 = 1; b1 = 0.765367; b0 = 1; |
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398 |
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399 Section 2: |
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400 a2 = 0; a1 = 0; a0 = 1; |
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401 b2 = 1; b1 = 1.847759; b0 = 1; |
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402 |
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403 Q is filter quality factor or resonance, in the range of 1 to |
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404 1000. The overall filter Q is a product of all 2nd order stages. |
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405 For example, the 6th order filter (3 stages, or biquads) with |
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406 individual Q of 2 will have filter Q = 2 * 2 * 2 = 8. |
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407 |
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408 |
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409 Arguments: |
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410 a - s-domain numerator coefficients, a[1] is always assumed to be 1.0 |
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411 b - s-domain denominator coefficients |
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412 Q - Q value for the filter |
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413 k - filter gain factor. Initially set to 1 and modified by each |
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414 biquad section in such a way, as to make it the |
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415 coefficient by which to multiply the overall filter gain |
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416 in order to achieve a desired overall filter gain, |
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417 specified in initial value of k. |
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418 fs - sampling rate (Hz) |
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419 coef - array of z-domain coefficients to be filled in. |
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420 |
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421 Note: Upon return from each call, the k argument will be set to a |
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422 value, by which to multiply our actual signal in order for the gain |
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423 to be one. On second call to szxform() we provide k that was |
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424 changed by the previous section. During actual audio filtering |
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425 k can be used for gain compensation. |
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426 |
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427 return -1 if fail 0 if success. |
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428 */ |
27258 | 429 int af_filter_szxform(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_TYPE Q, FLOAT_TYPE fc, |
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430 FLOAT_TYPE fs, FLOAT_TYPE *k, FLOAT_TYPE *coef) |
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431 { |
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432 FLOAT_TYPE at[3]; |
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433 FLOAT_TYPE bt[3]; |
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434 |
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435 if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0)) |
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436 return -1; |
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437 |
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438 memcpy(at,a,3*sizeof(FLOAT_TYPE)); |
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439 memcpy(bt,b,3*sizeof(FLOAT_TYPE)); |
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440 |
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441 bt[1]/=Q; |
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442 |
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443 /* Calculate a and b and overwrite the original values */ |
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444 af_filter_prewarp(at, fc, fs); |
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445 af_filter_prewarp(bt, fc, fs); |
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446 /* Execute bilinear transform */ |
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447 af_filter_bilinear(at, bt, k, fs, coef); |
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448 |
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449 return 0; |
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450 } |