audio: add af_lavrresample, remove old resampling filters
[mplayer2.git] / libaf / af_hrtf.c
blobd8ec33e11b24184ad5c770408d8058d64ffdb70e
1 /*
2 * Experimental audio filter that mixes 5.1 and 5.1 with matrix
3 * encoded rear channels into headphone signal using FIR filtering
4 * with HRTF.
6 * This file is part of MPlayer.
8 * MPlayer is free software; you can redistribute it and/or modify
9 * it under the terms of the GNU General Public License as published by
10 * the Free Software Foundation; either version 2 of the License, or
11 * (at your option) any later version.
13 * MPlayer is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 * GNU General Public License for more details.
18 * You should have received a copy of the GNU General Public License along
19 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
23 //#include <stdio.h>
24 #include <stdlib.h>
25 #include <string.h>
26 #include <inttypes.h>
28 #include <math.h>
30 #include "af.h"
31 #include "dsp.h"
33 /* HRTF filter coefficients and adjustable parameters */
34 #include "af_hrtf.h"
36 typedef struct af_hrtf_s {
37 /* Lengths */
38 int dlbuflen, hrflen, basslen;
39 /* L, C, R, Ls, Rs channels */
40 float *lf, *rf, *lr, *rr, *cf, *cr;
41 const float *cf_ir, *af_ir, *of_ir, *ar_ir, *or_ir, *cr_ir;
42 int cf_o, af_o, of_o, ar_o, or_o, cr_o;
43 /* Bass */
44 float *ba_l, *ba_r;
45 float *ba_ir;
46 /* Whether to matrix decode the rear center channel */
47 int matrix_mode;
48 /* How to decode the input:
49 0 = 5/5+1 channels
50 1 = 2 channels
51 2 = matrix encoded 2 channels */
52 int decode_mode;
53 /* Full wave rectified (FWR) amplitudes and gain used to steer the
54 active matrix decoding of front channels (variable names
55 lpr/lmr means Lt + Rt, Lt - Rt) */
56 float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
57 float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
58 /* Matrix input decoding require special FWR buffer, since the
59 decoding is done in place. */
60 float *fwrbuf_l, *fwrbuf_r, *fwrbuf_lr, *fwrbuf_rr;
61 /* Rear channel delay buffer for matrix decoding */
62 float *rear_dlbuf;
63 /* Full wave rectified amplitude and gain used to steer the active
64 matrix decoding of center rear channel */
65 float lr_fwr, rr_fwr, lrprr_fwr, lrmrr_fwr;
66 float adapt_lr_gain, adapt_rr_gain;
67 float adapt_lrprr_gain, adapt_lrmrr_gain;
68 /* Cyclic position on the ring buffer */
69 int cyc_pos;
70 int print_flag;
71 } af_hrtf_t;
73 /* Convolution on a ring buffer
74 * nx: length of the ring buffer
75 * nk: length of the convolution kernel
76 * sx: ring buffer
77 * sk: convolution kernel
78 * offset: offset on the ring buffer, can be
80 static float conv(const int nx, const int nk, const float *sx, const float *sk,
81 const int offset)
83 /* k = reminder of offset / nx */
84 int k = offset >= 0 ? offset % nx : nx + (offset % nx);
86 if(nk + k <= nx)
87 return af_filter_fir(nk, sx + k, sk);
88 else
89 return af_filter_fir(nk + k - nx, sx, sk + nx - k) +
90 af_filter_fir(nx - k, sx + k, sk);
93 /* Detect when the impulse response starts (significantly) */
94 static int pulse_detect(const float *sx)
96 /* nmax must be the reference impulse response length (128) minus
97 s->hrflen */
98 const int nmax = 128 - HRTFFILTLEN;
99 const float thresh = IRTHRESH;
100 int i;
102 for(i = 0; i < nmax; i++)
103 if(fabs(sx[i]) > thresh)
104 return i;
105 return 0;
108 /* Fuzzy matrix coefficient transfer function to "lock" the matrix on
109 a effectively passive mode if the gain is approximately 1 */
110 static inline float passive_lock(float x)
112 const float x1 = x - 1;
113 const float ax1s = fabs(x - 1) * (1.0 / MATAGCLOCK);
115 return x1 - x1 / (1 + ax1s * ax1s) + 1;
118 /* Unified active matrix decoder for 2 channel matrix encoded surround
119 sources */
120 static inline void matrix_decode(short *in, const int k, const int il,
121 const int ir, const int decode_rear,
122 const int dlbuflen,
123 float l_fwr, float r_fwr,
124 float lpr_fwr, float lmr_fwr,
125 float *adapt_l_gain, float *adapt_r_gain,
126 float *adapt_lpr_gain, float *adapt_lmr_gain,
127 float *lf, float *rf, float *lr,
128 float *rr, float *cf)
130 const int kr = (k + MATREARDELAY) % dlbuflen;
131 float l_gain = (l_fwr + r_fwr) /
132 (1 + l_fwr + l_fwr);
133 float r_gain = (l_fwr + r_fwr) /
134 (1 + r_fwr + r_fwr);
135 /* The 2nd axis has strong gain fluctuations, and therefore require
136 limits. The factor corresponds to the 1 / amplification of (Lt
137 - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
138 dialogues). It should be bigger than -12 dB to prevent
139 distortion. */
140 float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ?
141 lmr_fwr : M9_03DB * lpr_fwr;
142 float lpr_gain = (lpr_fwr + lmr_lim_fwr) /
143 (1 + lpr_fwr + lpr_fwr);
144 float lmr_gain = (lpr_fwr + lmr_lim_fwr) /
145 (1 + lmr_lim_fwr + lmr_lim_fwr);
146 float lmr_unlim_gain = (lpr_fwr + lmr_fwr) /
147 (1 + lmr_fwr + lmr_fwr);
148 float lpr, lmr;
149 float l_agc, r_agc, lpr_agc, lmr_agc;
150 float f, d_gain, c_gain, c_agc_cfk;
152 #if 0
153 static int counter = 0;
154 static FILE *fp_out;
156 if(counter == 0)
157 fp_out = fopen("af_hrtf.log", "w");
158 if(counter % 240 == 0)
159 fprintf(fp_out, "%g %g %g %g %g ", counter * (1.0 / 48000),
160 l_gain, r_gain, lpr_gain, lmr_gain);
161 #endif
163 /*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
164 /* AGC adaption */
165 d_gain = (fabs(l_gain - *adapt_l_gain) +
166 fabs(r_gain - *adapt_r_gain)) * 0.5;
167 f = d_gain * (1.0 / MATAGCTRIG);
168 f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
169 *adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
170 *adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
171 /* Matrix */
172 l_agc = in[il] * passive_lock(*adapt_l_gain);
173 r_agc = in[ir] * passive_lock(*adapt_r_gain);
174 cf[k] = (l_agc + r_agc) * M_SQRT1_2;
175 if(decode_rear) {
176 lr[kr] = rr[kr] = (l_agc - r_agc) * M_SQRT1_2;
177 /* Stereo rear channel is steered with the same AGC steering as
178 the decoding matrix. Note this requires a fast updating AGC
179 at the order of 20 ms (which is the case here). */
180 lr[kr] *= (l_fwr + l_fwr) /
181 (1 + l_fwr + r_fwr);
182 rr[kr] *= (r_fwr + r_fwr) /
183 (1 + l_fwr + r_fwr);
186 /*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
187 lpr = (in[il] + in[ir]) * M_SQRT1_2;
188 lmr = (in[il] - in[ir]) * M_SQRT1_2;
189 /* AGC adaption */
190 d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
191 f = d_gain * (1.0 / MATAGCTRIG);
192 f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
193 *adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
194 *adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
195 /* Matrix */
196 lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
197 lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
198 lf[k] = (lpr_agc + lmr_agc) * M_SQRT1_2;
199 rf[k] = (lpr_agc - lmr_agc) * M_SQRT1_2;
201 /*** CENTER FRONT CANCELLATION ***/
202 /* A heuristic approach exploits that Lt + Rt gain contains the
203 information about Lt, Rt correlation. This effectively reshapes
204 the front and rear "cones" to concentrate Lt + Rt to C and
205 introduce Lt - Rt in L, R. */
206 /* 0.67677 is the emprical lower bound for lpr_gain. */
207 c_gain = 8 * (*adapt_lpr_gain - 0.67677);
208 c_gain = c_gain > 0 ? c_gain : 0;
209 /* c_gain should not be too high, not even reaching full
210 cancellation (~ 0.50 - 0.55 at current AGC implementation), or
211 the center will s0und too narrow. */
212 c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
213 c_agc_cfk = c_gain * cf[k];
214 lf[k] -= c_agc_cfk;
215 rf[k] -= c_agc_cfk;
216 cf[k] += c_agc_cfk + c_agc_cfk;
217 #if 0
218 if(counter % 240 == 0)
219 fprintf(fp_out, "%g %g %g %g %g\n",
220 *adapt_l_gain, *adapt_r_gain,
221 *adapt_lpr_gain, *adapt_lmr_gain,
222 c_gain);
223 counter++;
224 #endif
227 static inline void update_ch(af_hrtf_t *s, short *in, const int k)
229 const int fwr_pos = (k + FWRDURATION) % s->dlbuflen;
230 /* Update the full wave rectified total amplitude */
231 /* Input matrix decoder */
232 if(s->decode_mode == HRTF_MIX_MATRIX2CH) {
233 s->l_fwr += abs(in[0]) - fabs(s->fwrbuf_l[fwr_pos]);
234 s->r_fwr += abs(in[1]) - fabs(s->fwrbuf_r[fwr_pos]);
235 s->lpr_fwr += abs(in[0] + in[1]) -
236 fabs(s->fwrbuf_l[fwr_pos] + s->fwrbuf_r[fwr_pos]);
237 s->lmr_fwr += abs(in[0] - in[1]) -
238 fabs(s->fwrbuf_l[fwr_pos] - s->fwrbuf_r[fwr_pos]);
240 /* Rear matrix decoder */
241 if(s->matrix_mode) {
242 s->lr_fwr += abs(in[2]) - fabs(s->fwrbuf_lr[fwr_pos]);
243 s->rr_fwr += abs(in[3]) - fabs(s->fwrbuf_rr[fwr_pos]);
244 s->lrprr_fwr += abs(in[2] + in[3]) -
245 fabs(s->fwrbuf_lr[fwr_pos] + s->fwrbuf_rr[fwr_pos]);
246 s->lrmrr_fwr += abs(in[2] - in[3]) -
247 fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
250 switch (s->decode_mode) {
251 case HRTF_MIX_51:
252 /* 5/5+1 channel sources */
253 s->lf[k] = in[0];
254 s->cf[k] = in[4];
255 s->rf[k] = in[1];
256 s->fwrbuf_lr[k] = s->lr[k] = in[2];
257 s->fwrbuf_rr[k] = s->rr[k] = in[3];
258 break;
259 case HRTF_MIX_MATRIX2CH:
260 /* Matrix encoded 2 channel sources */
261 s->fwrbuf_l[k] = in[0];
262 s->fwrbuf_r[k] = in[1];
263 matrix_decode(in, k, 0, 1, 1, s->dlbuflen,
264 s->l_fwr, s->r_fwr,
265 s->lpr_fwr, s->lmr_fwr,
266 &(s->adapt_l_gain), &(s->adapt_r_gain),
267 &(s->adapt_lpr_gain), &(s->adapt_lmr_gain),
268 s->lf, s->rf, s->lr, s->rr, s->cf);
269 break;
270 case HRTF_MIX_STEREO:
271 /* Stereo sources */
272 s->lf[k] = in[0];
273 s->rf[k] = in[1];
274 s->cf[k] = s->lr[k] = s->rr[k] = 0;
275 break;
278 /* We need to update the bass compensation delay line, too. */
279 s->ba_l[k] = in[0] + in[4] + in[2];
280 s->ba_r[k] = in[4] + in[1] + in[3];
283 /* Initialization and runtime control */
284 static int control(struct af_instance_s *af, int cmd, void* arg)
286 af_hrtf_t *s = af->setup;
287 int test_output_res;
288 char mode;
290 switch(cmd) {
291 case AF_CONTROL_REINIT:
292 af->data->rate = ((af_data_t*)arg)->rate;
293 if(af->data->rate != 48000) {
294 // automatic samplerate adjustment in the filter chain
295 // is not yet supported.
296 mp_msg(MSGT_AFILTER, MSGL_ERR,
297 "[hrtf] ERROR: Sampling rate is not 48000 Hz (%d)!\n",
298 af->data->rate);
299 return AF_ERROR;
301 af->data->nch = ((af_data_t*)arg)->nch;
302 if(af->data->nch == 2) {
303 /* 2 channel input */
304 if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
305 /* Default behavior is stereo mixing. */
306 s->decode_mode = HRTF_MIX_STEREO;
309 else if (af->data->nch < 5)
310 af->data->nch = 5;
311 af->data->format = AF_FORMAT_S16_NE;
312 af->data->bps = 2;
313 test_output_res = af_test_output(af, (af_data_t*)arg);
314 af->mul = 2.0 / af->data->nch;
315 // after testing input set the real output format
316 af->data->nch = 2;
317 s->print_flag = 1;
318 return test_output_res;
319 case AF_CONTROL_COMMAND_LINE:
320 sscanf((char*)arg, "%c", &mode);
321 switch(mode) {
322 case 'm':
323 /* Use matrix rear decoding. */
324 s->matrix_mode = 1;
325 break;
326 case 's':
327 /* Input needs matrix decoding. */
328 s->decode_mode = HRTF_MIX_MATRIX2CH;
329 break;
330 case '0':
331 s->matrix_mode = 0;
332 break;
333 default:
334 mp_msg(MSGT_AFILTER, MSGL_ERR,
335 "[hrtf] Mode is neither 'm', 's', nor '0' (%c).\n",
336 mode);
337 return AF_ERROR;
339 s->print_flag = 1;
340 return AF_OK;
343 return AF_UNKNOWN;
346 /* Deallocate memory */
347 static void uninit(struct af_instance_s *af)
349 if(af->setup) {
350 af_hrtf_t *s = af->setup;
352 free(s->lf);
353 free(s->rf);
354 free(s->lr);
355 free(s->rr);
356 free(s->cf);
357 free(s->cr);
358 free(s->ba_l);
359 free(s->ba_r);
360 free(s->ba_ir);
361 free(s->fwrbuf_l);
362 free(s->fwrbuf_r);
363 free(s->fwrbuf_lr);
364 free(s->fwrbuf_rr);
365 free(af->setup);
367 if(af->data)
368 free(af->data->audio);
369 free(af->data);
372 /* Filter data through filter
374 Two "tricks" are used to compensate the "color" of the KEMAR data:
376 1. The KEMAR data is refiltered to ensure that the front L, R channels
377 on the same side of the ear are equalized (especially in the high
378 frequencies).
380 2. A bass compensation is introduced to ensure that 0-200 Hz are not
381 damped (without any real 3D acoustical image, however).
383 static af_data_t* play(struct af_instance_s *af, af_data_t *data)
385 af_hrtf_t *s = af->setup;
386 short *in = data->audio; // Input audio data
387 short *out = NULL; // Output audio data
388 short *end = in + data->len / sizeof(short); // Loop end
389 float common, left, right, diff, left_b, right_b;
390 const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
392 if(AF_OK != RESIZE_LOCAL_BUFFER(af, data))
393 return NULL;
395 if(s->print_flag) {
396 s->print_flag = 0;
397 switch (s->decode_mode) {
398 case HRTF_MIX_51:
399 mp_msg(MSGT_AFILTER, MSGL_INFO,
400 "[hrtf] Using HRTF to mix %s discrete surround into "
401 "L, R channels\n", s->matrix_mode ? "5+1" : "5");
402 break;
403 case HRTF_MIX_STEREO:
404 mp_msg(MSGT_AFILTER, MSGL_INFO,
405 "[hrtf] Using HRTF to mix stereo into "
406 "L, R channels\n");
407 break;
408 case HRTF_MIX_MATRIX2CH:
409 mp_msg(MSGT_AFILTER, MSGL_INFO,
410 "[hrtf] Using active matrix to decode 2 channel "
411 "input, HRTF to mix %s matrix surround into "
412 "L, R channels\n", "3/2");
413 break;
414 default:
415 mp_msg(MSGT_AFILTER, MSGL_WARN,
416 "[hrtf] bogus decode_mode: %d\n", s->decode_mode);
417 break;
420 if(s->matrix_mode)
421 mp_msg(MSGT_AFILTER, MSGL_INFO,
422 "[hrtf] Using active matrix to decode rear center "
423 "channel\n");
426 out = af->data->audio;
428 /* MPlayer's 5 channel layout (notation for the variable):
430 * 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix
431 * encoded: Cs (CR)
433 * or: L = left, C = center, R = right, F = front, R = rear
435 * Filter notation:
437 * CF
438 * OF AF
439 * Ear->
440 * OR AR
441 * CR
443 * or: C = center, A = same side, O = opposite, F = front, R = rear
446 while(in < end) {
447 const int k = s->cyc_pos;
449 update_ch(s, in, k);
451 /* Simulate a 7.5 ms -20 dB echo of the center channel in the
452 front channels (like reflection from a room wall) - a kind of
453 psycho-acoustically "cheating" to focus the center front
454 channel, which is normally hard to be perceived as front */
455 s->lf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
456 s->rf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
458 switch (s->decode_mode) {
459 case HRTF_MIX_51:
460 case HRTF_MIX_MATRIX2CH:
461 /* Mixer filter matrix */
462 common = conv(dblen, hlen, s->cf, s->cf_ir, k + s->cf_o);
463 if(s->matrix_mode) {
464 /* In matrix decoding mode, the rear channel gain must be
465 renormalized, as there is an additional channel. */
466 matrix_decode(in, k, 2, 3, 0, s->dlbuflen,
467 s->lr_fwr, s->rr_fwr,
468 s->lrprr_fwr, s->lrmrr_fwr,
469 &(s->adapt_lr_gain), &(s->adapt_rr_gain),
470 &(s->adapt_lrprr_gain), &(s->adapt_lrmrr_gain),
471 s->lr, s->rr, NULL, NULL, s->cr);
472 common +=
473 conv(dblen, hlen, s->cr, s->cr_ir, k + s->cr_o) *
474 M1_76DB;
475 left =
476 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
477 conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
478 (conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
479 conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o)) *
480 M1_76DB + common);
481 right =
482 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
483 conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
484 (conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
485 conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o)) *
486 M1_76DB + common);
487 } else {
488 left =
489 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
490 conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
491 conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
492 conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o) +
493 common);
494 right =
495 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
496 conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
497 conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
498 conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o) +
499 common);
501 break;
502 case HRTF_MIX_STEREO:
503 left =
504 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
505 conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o));
506 right =
507 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
508 conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o));
509 break;
510 default:
511 /* make gcc happy */
512 left = 0.0;
513 right = 0.0;
514 break;
517 /* Bass compensation for the lower frequency cut of the HRTF. A
518 cross talk of the left and right channel is introduced to
519 match the directional characteristics of higher frequencies.
520 The bass will not have any real 3D perception, but that is
521 OK (note at 180 Hz, the wavelength is about 2 m, and any
522 spatial perception is impossible). */
523 left_b = conv(dblen, blen, s->ba_l, s->ba_ir, k);
524 right_b = conv(dblen, blen, s->ba_r, s->ba_ir, k);
525 left += (1 - BASSCROSS) * left_b + BASSCROSS * right_b;
526 right += (1 - BASSCROSS) * right_b + BASSCROSS * left_b;
527 /* Also mix the LFE channel (if available) */
528 if(data->nch >= 6) {
529 left += in[5] * M3_01DB;
530 right += in[5] * M3_01DB;
533 /* Amplitude renormalization. */
534 left *= AMPLNORM;
535 right *= AMPLNORM;
537 switch (s->decode_mode) {
538 case HRTF_MIX_51:
539 case HRTF_MIX_STEREO:
540 /* "Cheating": linear stereo expansion to amplify the 3D
541 perception. Note: Too much will destroy the acoustic space
542 and may even result in headaches. */
543 diff = STEXPAND2 * (left - right);
544 out[0] = (int16_t)(left + diff);
545 out[1] = (int16_t)(right - diff);
546 break;
547 case HRTF_MIX_MATRIX2CH:
548 /* Do attempt any stereo expansion with matrix encoded
549 sources. The L, R channels are already stereo expanded
550 by the steering, any further stereo expansion will sound
551 very unnatural. */
552 out[0] = (int16_t)left;
553 out[1] = (int16_t)right;
554 break;
557 /* Next sample... */
558 in = &in[data->nch];
559 out = &out[af->data->nch];
560 (s->cyc_pos)--;
561 if(s->cyc_pos < 0)
562 s->cyc_pos += dblen;
565 /* Set output data */
566 data->audio = af->data->audio;
567 data->len = data->len / data->nch * 2;
568 data->nch = 2;
570 return data;
573 static int allocate(af_hrtf_t *s)
575 if ((s->lf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
576 if ((s->rf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
577 if ((s->lr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
578 if ((s->rr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
579 if ((s->cf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
580 if ((s->cr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
581 if ((s->ba_l = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
582 if ((s->ba_r = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
583 if ((s->fwrbuf_l =
584 malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
585 if ((s->fwrbuf_r =
586 malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
587 if ((s->fwrbuf_lr =
588 malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
589 if ((s->fwrbuf_rr =
590 malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
591 return 0;
594 /* Allocate memory and set function pointers */
595 static int af_open(af_instance_t* af)
597 int i;
598 af_hrtf_t *s;
599 float fc;
601 af->control = control;
602 af->uninit = uninit;
603 af->play = play;
604 af->mul = 1;
605 af->data = calloc(1, sizeof(af_data_t));
606 af->setup = calloc(1, sizeof(af_hrtf_t));
607 if((af->data == NULL) || (af->setup == NULL))
608 return AF_ERROR;
610 s = af->setup;
612 s->dlbuflen = DELAYBUFLEN;
613 s->hrflen = HRTFFILTLEN;
614 s->basslen = BASSFILTLEN;
616 s->cyc_pos = s->dlbuflen - 1;
617 /* With a full (two axis) steering matrix decoder, s->matrix_mode
618 should not be enabled lightly (it will also steer the Ls, Rs
619 channels). */
620 s->matrix_mode = 0;
621 s->decode_mode = HRTF_MIX_51;
623 s->print_flag = 1;
625 if (allocate(s) != 0) {
626 mp_msg(MSGT_AFILTER, MSGL_ERR, "[hrtf] Memory allocation error.\n");
627 return AF_ERROR;
630 for(i = 0; i < s->dlbuflen; i++)
631 s->lf[i] = s->rf[i] = s->lr[i] = s->rr[i] = s->cf[i] =
632 s->cr[i] = 0;
634 s->lr_fwr =
635 s->rr_fwr = 0;
637 s->cf_ir = cf_filt + (s->cf_o = pulse_detect(cf_filt));
638 s->af_ir = af_filt + (s->af_o = pulse_detect(af_filt));
639 s->of_ir = of_filt + (s->of_o = pulse_detect(of_filt));
640 s->ar_ir = ar_filt + (s->ar_o = pulse_detect(ar_filt));
641 s->or_ir = or_filt + (s->or_o = pulse_detect(or_filt));
642 s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt));
644 if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) {
645 mp_msg(MSGT_AFILTER, MSGL_ERR, "[hrtf] Memory allocation error.\n");
646 return AF_ERROR;
648 fc = 2.0 * BASSFILTFREQ / (float)af->data->rate;
649 if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) ==
650 -1) {
651 mp_msg(MSGT_AFILTER, MSGL_ERR, "[hrtf] Unable to design low-pass "
652 "filter.\n");
653 return AF_ERROR;
655 for(i = 0; i < s->basslen; i++)
656 s->ba_ir[i] *= BASSGAIN;
658 return AF_OK;
661 /* Description of this filter */
662 af_info_t af_info_hrtf = {
663 "HRTF Headphone",
664 "hrtf",
665 "ylai",
667 AF_FLAGS_REENTRANT,
668 af_open