Mercurial > mplayer.hg
annotate libaf/filter.c @ 26877:4ab1cfcee430
Add a slave command to stop stream playback.
Mostly useful when used with -idle mode.
Patch by Mathieu Schroeter ( mathieu dot schroeter at gamesover dot ch )
author | ben |
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date | Tue, 27 May 2008 18:39:57 +0000 |
parents | 07abe94a9cc4 |
children | 82fd0e4c93c9 |
rev | line source |
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7568 | 1 /*============================================================================= |
2 // | |
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The full name of the GPL is GNU General Public License.
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3 // This software has been released under the terms of the GNU General Public |
7568 | 4 // license. See http://www.gnu.org/copyleft/gpl.html for details. |
5 // | |
6 // Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au | |
7 // | |
8 //============================================================================= | |
9 */ | |
10 | |
11 /* Design and implementation of different types of digital filters | |
12 | |
13 */ | |
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14 #include <string.h> |
7568 | 15 #include <math.h> |
16 #include "dsp.h" | |
17 | |
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18 /****************************************************************************** |
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19 * FIR filter implementations |
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20 ******************************************************************************/ |
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21 |
7568 | 22 /* C implementation of FIR filter y=w*x |
23 | |
24 n number of filter taps, where mod(n,4)==0 | |
25 w filter taps | |
26 x input signal must be a circular buffer which is indexed backwards | |
27 */ | |
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28 inline FLOAT_TYPE af_filter_fir(register unsigned int n, FLOAT_TYPE* w, |
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29 FLOAT_TYPE* x) |
7568 | 30 { |
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31 register FLOAT_TYPE y; // Output |
7568 | 32 y = 0.0; |
33 do{ | |
34 n--; | |
35 y+=w[n]*x[n]; | |
36 }while(n != 0); | |
37 return y; | |
38 } | |
39 | |
40 /* C implementation of parallel FIR filter y(k)=w(k) * x(k) (where * denotes convolution) | |
41 | |
42 n number of filter taps, where mod(n,4)==0 | |
43 d number of filters | |
44 xi current index in xq | |
45 w filter taps k by n big | |
46 x input signal must be a circular buffers which are indexed backwards | |
47 y output buffer | |
48 s output buffer stride | |
49 */ | |
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50 FLOAT_TYPE* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, |
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51 FLOAT_TYPE** w, FLOAT_TYPE** x, FLOAT_TYPE* y, |
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52 unsigned int s) |
7568 | 53 { |
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54 register FLOAT_TYPE* xt = *x + xi; |
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55 register FLOAT_TYPE* wt = *w; |
7568 | 56 register int nt = 2*n; |
57 while(d-- > 0){ | |
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58 *y = af_filter_fir(n,wt,xt); |
7568 | 59 wt+=n; |
60 xt+=nt; | |
61 y+=s; | |
62 } | |
63 return y; | |
64 } | |
65 | |
66 /* Add new data to circular queue designed to be used with a parallel | |
67 FIR filter, with d filters. xq is the circular queue, in pointing | |
68 at the new samples, xi current index in xq and n the length of the | |
69 filter. xq must be n*2 by k big, s is the index for in. | |
70 */ | |
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71 int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, |
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72 FLOAT_TYPE** xq, FLOAT_TYPE* in, unsigned int s) |
7568 | 73 { |
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74 register FLOAT_TYPE* txq = *xq + xi; |
7568 | 75 register int nt = n*2; |
76 | |
77 while(d-- >0){ | |
78 *txq= *(txq+n) = *in; | |
79 txq+=nt; | |
80 in+=s; | |
81 } | |
82 return (++xi)&(n-1); | |
83 } | |
84 | |
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85 /****************************************************************************** |
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86 * FIR filter design |
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87 ******************************************************************************/ |
7568 | 88 |
89 /* Design FIR filter using the Window method | |
90 | |
91 n filter length must be odd for HP and BS filters | |
92 w buffer for the filter taps (must be n long) | |
93 fc cutoff frequencies (1 for LP and HP, 2 for BP and BS) | |
94 0 < fc < 1 where 1 <=> Fs/2 | |
95 flags window and filter type as defined in filter.h | |
96 variables are ored together: i.e. LP|HAMMING will give a | |
97 low pass filter designed using a hamming window | |
98 opt beta constant used only when designing using kaiser windows | |
99 | |
100 returns 0 if OK, -1 if fail | |
101 */ | |
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102 int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, FLOAT_TYPE* fc, |
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103 unsigned int flags, FLOAT_TYPE opt) |
7568 | 104 { |
105 unsigned int o = n & 1; // Indicator for odd filter length | |
106 unsigned int end = ((n + 1) >> 1) - o; // Loop end | |
107 unsigned int i; // Loop index | |
108 | |
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109 FLOAT_TYPE k1 = 2 * M_PI; // 2*pi*fc1 |
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110 FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length |
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111 FLOAT_TYPE k3; // 2*pi*fc2 Constant used in BP and BS design |
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112 FLOAT_TYPE g = 0.0; // Gain |
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113 FLOAT_TYPE t1,t2,t3; // Temporary variables |
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114 FLOAT_TYPE fc1,fc2; // Cutoff frequencies |
7568 | 115 |
116 // Sanity check | |
117 if(!w || (n == 0)) return -1; | |
118 | |
119 // Get window coefficients | |
120 switch(flags & WINDOW_MASK){ | |
121 case(BOXCAR): | |
14274 | 122 af_window_boxcar(n,w); break; |
7568 | 123 case(TRIANG): |
14274 | 124 af_window_triang(n,w); break; |
7568 | 125 case(HAMMING): |
14274 | 126 af_window_hamming(n,w); break; |
7568 | 127 case(HANNING): |
14274 | 128 af_window_hanning(n,w); break; |
7568 | 129 case(BLACKMAN): |
14274 | 130 af_window_blackman(n,w); break; |
7568 | 131 case(FLATTOP): |
14274 | 132 af_window_flattop(n,w); break; |
7568 | 133 case(KAISER): |
14274 | 134 af_window_kaiser(n,w,opt); break; |
7568 | 135 default: |
136 return -1; | |
137 } | |
138 | |
139 if(flags & (LP | HP)){ | |
140 fc1=*fc; | |
141 // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2 | |
142 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25; | |
143 k1 *= fc1; | |
144 | |
145 if(flags & LP){ // Low pass filter | |
146 | |
147 // If the filter length is odd, there is one point which is exactly | |
148 // in the middle. The value at this point is 2*fCutoff*sin(x)/x, | |
149 // where x is zero. To make sure nothing strange happens, we set this | |
150 // value separately. | |
151 if (o){ | |
152 w[end] = fc1 * w[end] * 2.0; | |
153 g=w[end]; | |
154 } | |
155 | |
156 // Create filter | |
157 for (i=0 ; i<end ; i++){ | |
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158 t1 = (FLOAT_TYPE)(i+1) - k2; |
7568 | 159 w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc |
160 g += 2*w[end-i-1]; // Total gain in filter | |
161 } | |
162 } | |
163 else{ // High pass filter | |
164 if (!o) // High pass filters must have odd length | |
165 return -1; | |
166 w[end] = 1.0 - (fc1 * w[end] * 2.0); | |
167 g= w[end]; | |
168 | |
169 // Create filter | |
170 for (i=0 ; i<end ; i++){ | |
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171 t1 = (FLOAT_TYPE)(i+1); |
7568 | 172 w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc |
173 g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter | |
174 } | |
175 } | |
176 } | |
177 | |
178 if(flags & (BP | BS)){ | |
179 fc1=fc[0]; | |
180 fc2=fc[1]; | |
181 // Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2 | |
182 fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25; | |
183 fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25; | |
184 k3 = k1 * fc2; // 2*pi*fc2 | |
185 k1 *= fc1; // 2*pi*fc1 | |
186 | |
187 if(flags & BP){ // Band pass | |
188 // Calculate center tap | |
189 if (o){ | |
190 g=w[end]*(fc1+fc2); | |
191 w[end] = (fc2 - fc1) * w[end] * 2.0; | |
192 } | |
193 | |
194 // Create filter | |
195 for (i=0 ; i<end ; i++){ | |
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196 t1 = (FLOAT_TYPE)(i+1) - k2; |
7568 | 197 t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2 |
198 t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1 | |
199 g += w[end-i-1] * (t3 + t2); // Total gain in filter | |
200 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); | |
201 } | |
202 } | |
203 else{ // Band stop | |
204 if (!o) // Band stop filters must have odd length | |
205 return -1; | |
206 w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0; | |
207 g= w[end]; | |
208 | |
209 // Create filter | |
210 for (i=0 ; i<end ; i++){ | |
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211 t1 = (FLOAT_TYPE)(i+1); |
7568 | 212 t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1 |
213 t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2 | |
214 w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3); | |
215 g += 2*w[end-i-1]; // Total gain in filter | |
216 } | |
217 } | |
218 } | |
219 | |
220 // Normalize gain | |
221 g=1/g; | |
222 for (i=0; i<n; i++) | |
223 w[i] *= g; | |
224 | |
225 return 0; | |
226 } | |
227 | |
228 /* Design polyphase FIR filter from prototype filter | |
229 | |
230 n length of prototype filter | |
231 k number of polyphase components | |
232 w prototype filter taps | |
233 pw Parallel FIR filter | |
234 g Filter gain | |
235 flags FWD forward indexing | |
236 REW reverse indexing | |
237 ODD multiply every 2nd filter tap by -1 => HP filter | |
238 | |
239 returns 0 if OK, -1 if fail | |
240 */ | |
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241 int af_filter_design_pfir(unsigned int n, unsigned int k, FLOAT_TYPE* w, |
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242 FLOAT_TYPE** pw, FLOAT_TYPE g, unsigned int flags) |
7568 | 243 { |
244 int l = (int)n/k; // Length of individual FIR filters | |
245 int i; // Counters | |
246 int j; | |
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247 FLOAT_TYPE t; // g * w[i] |
7568 | 248 |
249 // Sanity check | |
250 if(l<1 || k<1 || !w || !pw) | |
251 return -1; | |
252 | |
253 // Do the stuff | |
254 if(flags&REW){ | |
255 for(j=l-1;j>-1;j--){//Columns | |
256 for(i=0;i<(int)k;i++){//Rows | |
257 t=g * *w++; | |
258 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? -1 : 1) : 1); | |
259 } | |
260 } | |
261 } | |
262 else{ | |
263 for(j=0;j<l;j++){//Columns | |
264 for(i=0;i<(int)k;i++){//Rows | |
265 t=g * *w++; | |
266 pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? 1 : -1) : 1); | |
267 } | |
268 } | |
269 } | |
270 return -1; | |
271 } | |
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272 |
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273 /****************************************************************************** |
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274 * IIR filter design |
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275 ******************************************************************************/ |
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276 |
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277 /* Helper functions for the bilinear transform */ |
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278 |
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279 /* Pre-warp the coefficients of a numerator or denominator. |
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280 Note that a0 is assumed to be 1, so there is no wrapping |
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281 of it. |
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282 */ |
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283 static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs) |
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284 { |
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285 FLOAT_TYPE wp; |
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286 wp = 2.0 * fs * tan(M_PI * fc / fs); |
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287 a[2] = a[2]/(wp * wp); |
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288 a[1] = a[1]/wp; |
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289 } |
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290 |
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291 /* Transform the numerator and denominator coefficients of s-domain |
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292 biquad section into corresponding z-domain coefficients. |
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293 |
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294 The transfer function for z-domain is: |
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295 |
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296 1 + alpha1 * z^(-1) + alpha2 * z^(-2) |
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297 H(z) = ------------------------------------- |
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298 1 + beta1 * z^(-1) + beta2 * z^(-2) |
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299 |
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300 Store the 4 IIR coefficients in array pointed by coef in following |
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301 order: |
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302 beta1, beta2 (denominator) |
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303 alpha1, alpha2 (numerator) |
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304 |
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305 Arguments: |
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306 a - s-domain numerator coefficients |
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307 b - s-domain denominator coefficients |
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308 k - filter gain factor. Initially set to 1 and modified by each |
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309 biquad section in such a way, as to make it the |
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310 coefficient by which to multiply the overall filter gain |
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311 in order to achieve a desired overall filter gain, |
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312 specified in initial value of k. |
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313 fs - sampling rate (Hz) |
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314 coef - array of z-domain coefficients to be filled in. |
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315 |
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316 Return: On return, set coef z-domain coefficients and k to the gain |
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317 required to maintain overall gain = 1.0; |
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318 */ |
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319 static void af_filter_bilinear(FLOAT_TYPE* a, FLOAT_TYPE* b, FLOAT_TYPE* k, |
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320 FLOAT_TYPE fs, FLOAT_TYPE *coef) |
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321 { |
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322 FLOAT_TYPE ad, bd; |
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323 |
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324 /* alpha (Numerator in s-domain) */ |
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325 ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0]; |
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326 /* beta (Denominator in s-domain) */ |
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327 bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0]; |
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328 |
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329 /* Update gain constant for this section */ |
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330 *k *= ad/bd; |
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331 |
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332 /* Denominator */ |
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333 *coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */ |
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334 *coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */ |
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335 |
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336 /* Numerator */ |
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337 *coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */ |
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338 *coef = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad; /* alpha2 */ |
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339 } |
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340 |
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341 |
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342 |
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343 /* IIR filter design using bilinear transform and prewarp. Transforms |
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344 2nd order s domain analog filter into a digital IIR biquad link. To |
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345 create a filter fill in a, b, Q and fs and make space for coef and k. |
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346 |
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347 |
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348 Example Butterworth design: |
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349 |
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350 Below are Butterworth polynomials, arranged as a series of 2nd |
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351 order sections: |
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352 |
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353 Note: n is filter order. |
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354 |
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355 n Polynomials |
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356 ------------------------------------------------------------------- |
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357 2 s^2 + 1.4142s + 1 |
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358 4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1) |
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359 6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1) |
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360 |
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361 For n=4 we have following equation for the filter transfer function: |
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362 1 1 |
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363 T(s) = --------------------------- * ---------------------------- |
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364 s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1 |
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365 |
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366 The filter consists of two 2nd order sections since highest s power |
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367 is 2. Now we can take the coefficients, or the numbers by which s |
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368 is multiplied and plug them into a standard formula to be used by |
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369 bilinear transform. |
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370 |
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371 Our standard form for each 2nd order section is: |
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372 |
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373 a2 * s^2 + a1 * s + a0 |
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374 H(s) = ---------------------- |
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375 b2 * s^2 + b1 * s + b0 |
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376 |
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377 Note that Butterworth numerator is 1 for all filter sections, which |
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378 means s^2 = 0 and s^1 = 0 |
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379 |
11000 | 380 Let's convert standard Butterworth polynomials into this form: |
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381 |
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382 0 + 0 + 1 0 + 0 + 1 |
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383 --------------------------- * -------------------------- |
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384 1 + ((1/Q) * 0.765367) + 1 1 + ((1/Q) * 1.847759) + 1 |
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385 |
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386 Section 1: |
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387 a2 = 0; a1 = 0; a0 = 1; |
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388 b2 = 1; b1 = 0.765367; b0 = 1; |
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389 |
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390 Section 2: |
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391 a2 = 0; a1 = 0; a0 = 1; |
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392 b2 = 1; b1 = 1.847759; b0 = 1; |
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393 |
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394 Q is filter quality factor or resonance, in the range of 1 to |
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395 1000. The overall filter Q is a product of all 2nd order stages. |
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396 For example, the 6th order filter (3 stages, or biquads) with |
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397 individual Q of 2 will have filter Q = 2 * 2 * 2 = 8. |
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398 |
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399 |
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400 Arguments: |
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401 a - s-domain numerator coefficients, a[1] is always assumed to be 1.0 |
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402 b - s-domain denominator coefficients |
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403 Q - Q value for the filter |
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404 k - filter gain factor. Initially set to 1 and modified by each |
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405 biquad section in such a way, as to make it the |
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406 coefficient by which to multiply the overall filter gain |
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407 in order to achieve a desired overall filter gain, |
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408 specified in initial value of k. |
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409 fs - sampling rate (Hz) |
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410 coef - array of z-domain coefficients to be filled in. |
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411 |
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412 Note: Upon return from each call, the k argument will be set to a |
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413 value, by which to multiply our actual signal in order for the gain |
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414 to be one. On second call to szxform() we provide k that was |
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415 changed by the previous section. During actual audio filtering |
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416 k can be used for gain compensation. |
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417 |
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418 return -1 if fail 0 if success. |
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419 */ |
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420 int af_filter_szxform(FLOAT_TYPE* a, FLOAT_TYPE* b, FLOAT_TYPE Q, FLOAT_TYPE fc, |
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421 FLOAT_TYPE fs, FLOAT_TYPE *k, FLOAT_TYPE *coef) |
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422 { |
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423 FLOAT_TYPE at[3]; |
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424 FLOAT_TYPE bt[3]; |
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425 |
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426 if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0)) |
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427 return -1; |
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428 |
26350
07abe94a9cc4
Fix illegal identifier: Rename _ftype_t macro to FLOAT_TYPE.
diego
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24890
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429 memcpy(at,a,3*sizeof(FLOAT_TYPE)); |
07abe94a9cc4
Fix illegal identifier: Rename _ftype_t macro to FLOAT_TYPE.
diego
parents:
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430 memcpy(bt,b,3*sizeof(FLOAT_TYPE)); |
8832
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
parents:
7568
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changeset
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431 |
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
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432 bt[1]/=Q; |
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
parents:
7568
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changeset
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433 |
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
parents:
7568
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changeset
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434 /* Calculate a and b and overwrite the original values */ |
14275
de13fd557440
less namespace pollution #2 (prefixed globals in filter.c with af_filter_)
alex
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14274
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435 af_filter_prewarp(at, fc, fs); |
de13fd557440
less namespace pollution #2 (prefixed globals in filter.c with af_filter_)
alex
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14274
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436 af_filter_prewarp(bt, fc, fs); |
8832
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
parents:
7568
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changeset
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437 /* Execute bilinear transform */ |
14275
de13fd557440
less namespace pollution #2 (prefixed globals in filter.c with af_filter_)
alex
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438 af_filter_bilinear(at, bt, k, fs, coef); |
8832
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
parents:
7568
diff
changeset
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439 |
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
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440 return 0; |
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
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441 } |
a1578b329cc0
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
anders
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7568
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changeset
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442 |