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FreeBSD/Linux Kernel Cross Reference
sys/dev/audio.c
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1 /* $NetBSD: audio.c,v 1.192.4.7 2007/08/26 19:08:02 bouyer Exp $ */ 2 3 /* 4 * Copyright (c) 1991-1993 Regents of the University of California. 5 * All rights reserved. 6 * 7 * Redistribution and use in source and binary forms, with or without 8 * modification, are permitted provided that the following conditions 9 * are met: 10 * 1. Redistributions of source code must retain the above copyright 11 * notice, this list of conditions and the following disclaimer. 12 * 2. Redistributions in binary form must reproduce the above copyright 13 * notice, this list of conditions and the following disclaimer in the 14 * documentation and/or other materials provided with the distribution. 15 * 3. All advertising materials mentioning features or use of this software 16 * must display the following acknowledgement: 17 * This product includes software developed by the Computer Systems 18 * Engineering Group at Lawrence Berkeley Laboratory. 19 * 4. Neither the name of the University nor of the Laboratory may be used 20 * to endorse or promote products derived from this software without 21 * specific prior written permission. 22 * 23 * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND 24 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE 25 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE 26 * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE 27 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL 28 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS 29 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) 30 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT 31 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY 32 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF 33 * SUCH DAMAGE. 34 */ 35 36 /* 37 * This is a (partially) SunOS-compatible /dev/audio driver for NetBSD. 38 * 39 * This code tries to do something half-way sensible with 40 * half-duplex hardware, such as with the SoundBlaster hardware. With 41 * half-duplex hardware allowing O_RDWR access doesn't really make 42 * sense. However, closing and opening the device to "turn around the 43 * line" is relatively expensive and costs a card reset (which can 44 * take some time, at least for the SoundBlaster hardware). Instead 45 * we allow O_RDWR access, and provide an ioctl to set the "mode", 46 * i.e. playing or recording. 47 * 48 * If you write to a half-duplex device in record mode, the data is 49 * tossed. If you read from the device in play mode, you get silence 50 * filled buffers at the rate at which samples are naturally 51 * generated. 52 * 53 * If you try to set both play and record mode on a half-duplex 54 * device, playing takes precedence. 55 */ 56 57 /* 58 * Todo: 59 * - Add softaudio() isr processing for wakeup, poll, signals, 60 * and silence fill. 61 */ 62 63 #include <sys/cdefs.h> 64 __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.192.4.7 2007/08/26 19:08:02 bouyer Exp $"); 65 66 #include "audio.h" 67 #if NAUDIO > 0 68 69 #include <sys/param.h> 70 #include <sys/ioctl.h> 71 #include <sys/fcntl.h> 72 #include <sys/vnode.h> 73 #include <sys/select.h> 74 #include <sys/poll.h> 75 #include <sys/malloc.h> 76 #include <sys/proc.h> 77 #include <sys/systm.h> 78 #include <sys/syslog.h> 79 #include <sys/kernel.h> 80 #include <sys/signalvar.h> 81 #include <sys/conf.h> 82 #include <sys/audioio.h> 83 #include <sys/device.h> 84 85 #include <dev/audio_if.h> 86 #include <dev/audiovar.h> 87 88 #include <machine/endian.h> 89 90 /* #define AUDIO_DEBUG 1 */ 91 #ifdef AUDIO_DEBUG 92 #define DPRINTF(x) if (audiodebug) printf x 93 #define DPRINTFN(n,x) if (audiodebug>(n)) printf x 94 int audiodebug = AUDIO_DEBUG; 95 #else 96 #define DPRINTF(x) 97 #define DPRINTFN(n,x) 98 #endif 99 100 #define ROUNDSIZE(x) x &= -16 /* round to nice boundary */ 101 #define SPECIFIED(x) (x != ~0) 102 #define SPECIFIED_CH(x) (x != (u_char)~0) 103 104 int audio_blk_ms = AUDIO_BLK_MS; 105 106 int audiosetinfo(struct audio_softc *, struct audio_info *); 107 int audiogetinfo(struct audio_softc *, struct audio_info *); 108 109 int audio_open(dev_t, struct audio_softc *, int, int, struct proc *); 110 int audio_close(struct audio_softc *, int, int, struct proc *); 111 int audio_read(struct audio_softc *, struct uio *, int); 112 int audio_write(struct audio_softc *, struct uio *, int); 113 int audio_ioctl(struct audio_softc *, u_long, caddr_t, int, struct proc *); 114 int audio_poll(struct audio_softc *, int, struct proc *); 115 int audio_kqfilter(struct audio_softc *, struct knote *); 116 paddr_t audio_mmap(struct audio_softc *, off_t, int); 117 118 int mixer_open(dev_t, struct audio_softc *, int, int, struct proc *); 119 int mixer_close(struct audio_softc *, int, int, struct proc *); 120 int mixer_ioctl(struct audio_softc *, u_long, caddr_t, int, struct proc *); 121 static void mixer_remove(struct audio_softc *, struct proc *); 122 static void mixer_signal(struct audio_softc *); 123 124 void audio_init_record(struct audio_softc *); 125 void audio_init_play(struct audio_softc *); 126 int audiostartr(struct audio_softc *); 127 int audiostartp(struct audio_softc *); 128 void audio_rint(void *); 129 void audio_pint(void *); 130 int audio_check_params(struct audio_params *); 131 132 void audio_calc_blksize(struct audio_softc *, int); 133 void audio_fill_silence(struct audio_params *, uint8_t *, int); 134 int audio_silence_copyout(struct audio_softc *, int, struct uio *); 135 136 void audio_init_ringbuffer(struct audio_softc *, 137 struct audio_ringbuffer *, int); 138 int audio_initbufs(struct audio_softc *); 139 void audio_calcwater(struct audio_softc *); 140 static __inline int audio_sleep_timo(int *, const char *, int); 141 static __inline int audio_sleep(int *, const char *); 142 static __inline void audio_wakeup(int *); 143 int audio_drain(struct audio_softc *); 144 void audio_clear(struct audio_softc *); 145 static __inline void audio_pint_silence 146 (struct audio_softc *, struct audio_ringbuffer *, uint8_t *, int); 147 148 int audio_alloc_ring 149 (struct audio_softc *, struct audio_ringbuffer *, int, size_t); 150 void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *); 151 static int audio_setup_pfilters(struct audio_softc *, const audio_params_t *, 152 stream_filter_list_t *); 153 static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *, 154 stream_filter_list_t *); 155 static void audio_destruct_pfilters(struct audio_softc *); 156 static void audio_destruct_rfilters(struct audio_softc *); 157 static void audio_stream_dtor(audio_stream_t *); 158 static int audio_stream_ctor(audio_stream_t *, const audio_params_t *, int); 159 static void stream_filter_list_append 160 (stream_filter_list_t *, stream_filter_factory_t, 161 const audio_params_t *); 162 static void stream_filter_list_prepend 163 (stream_filter_list_t *, stream_filter_factory_t, 164 const audio_params_t *); 165 static void stream_filter_list_set 166 (stream_filter_list_t *, int, stream_filter_factory_t, 167 const audio_params_t *); 168 int audio_set_defaults(struct audio_softc *, u_int); 169 170 int audioprobe(struct device *, struct cfdata *, void *); 171 void audioattach(struct device *, struct device *, void *); 172 int audiodetach(struct device *, int); 173 int audioactivate(struct device *, enum devact); 174 175 struct portname { 176 const char *name; 177 int mask; 178 }; 179 static const struct portname itable[] = { 180 { AudioNmicrophone, AUDIO_MICROPHONE }, 181 { AudioNline, AUDIO_LINE_IN }, 182 { AudioNcd, AUDIO_CD }, 183 { 0 } 184 }; 185 static const struct portname otable[] = { 186 { AudioNspeaker, AUDIO_SPEAKER }, 187 { AudioNheadphone, AUDIO_HEADPHONE }, 188 { AudioNline, AUDIO_LINE_OUT }, 189 { 0 } 190 }; 191 void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, 192 mixer_devinfo_t *, const struct portname *); 193 int au_set_gain(struct audio_softc *, struct au_mixer_ports *, 194 int, int); 195 void au_get_gain(struct audio_softc *, struct au_mixer_ports *, 196 u_int *, u_char *); 197 int au_set_port(struct audio_softc *, struct au_mixer_ports *, 198 u_int); 199 int au_get_port(struct audio_softc *, struct au_mixer_ports *); 200 int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, 201 int *, int *); 202 int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, 203 int, int); 204 int au_portof(struct audio_softc *, char *, int); 205 206 typedef struct uio_fetcher { 207 stream_fetcher_t base; 208 struct uio *uio; 209 int usedhigh; 210 int last_used; 211 } uio_fetcher_t; 212 213 static void uio_fetcher_ctor(uio_fetcher_t *, struct uio *, int); 214 static int uio_fetcher_fetch_to(stream_fetcher_t *, 215 audio_stream_t *, int); 216 static int null_fetcher_fetch_to(stream_fetcher_t *, 217 audio_stream_t *, int); 218 219 dev_type_open(audioopen); 220 dev_type_close(audioclose); 221 dev_type_read(audioread); 222 dev_type_write(audiowrite); 223 dev_type_ioctl(audioioctl); 224 dev_type_poll(audiopoll); 225 dev_type_mmap(audiommap); 226 dev_type_kqfilter(audiokqfilter); 227 228 const struct cdevsw audio_cdevsw = { 229 audioopen, audioclose, audioread, audiowrite, audioioctl, 230 nostop, notty, audiopoll, audiommap, audiokqfilter, 231 }; 232 233 /* The default audio mode: 8 kHz mono mu-law */ 234 const struct audio_params audio_default = { 235 .sample_rate = 8000, 236 .encoding = AUDIO_ENCODING_ULAW, 237 .precision = 8, 238 .validbits = 8, 239 .channels = 1, 240 }; 241 242 CFATTACH_DECL(audio, sizeof(struct audio_softc), 243 audioprobe, audioattach, audiodetach, audioactivate); 244 245 extern struct cfdriver audio_cd; 246 247 int 248 audioprobe(struct device *parent, struct cfdata *match, void *aux) 249 { 250 struct audio_attach_args *sa; 251 252 sa = aux; 253 DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n", 254 sa->type, sa, sa->hwif)); 255 return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; 256 } 257 258 void 259 audioattach(struct device *parent, struct device *self, void *aux) 260 { 261 struct audio_softc *sc; 262 struct audio_attach_args *sa; 263 const struct audio_hw_if *hwp; 264 void *hdlp; 265 int error; 266 mixer_devinfo_t mi; 267 int iclass, mclass, oclass, rclass, props; 268 int record_master_found, record_source_found; 269 270 sc = (void *)self; 271 sa = aux; 272 hwp = sa->hwif; 273 hdlp = sa->hdl; 274 #ifdef DIAGNOSTIC 275 if (hwp == 0 || 276 hwp->query_encoding == 0 || 277 hwp->set_params == 0 || 278 (hwp->start_output == 0 && hwp->trigger_output == 0) || 279 (hwp->start_input == 0 && hwp->trigger_input == 0) || 280 hwp->halt_output == 0 || 281 hwp->halt_input == 0 || 282 hwp->getdev == 0 || 283 hwp->set_port == 0 || 284 hwp->get_port == 0 || 285 hwp->query_devinfo == 0 || 286 hwp->get_props == 0) { 287 printf(": missing method\n"); 288 sc->hw_if = 0; 289 return; 290 } 291 #endif 292 293 props = hwp->get_props(hdlp); 294 295 if (props & AUDIO_PROP_FULLDUPLEX) 296 printf(": full duplex"); 297 else 298 printf(": half duplex"); 299 300 if (props & AUDIO_PROP_MMAP) 301 printf(", mmap"); 302 if (props & AUDIO_PROP_INDEPENDENT) 303 printf(", independent"); 304 305 printf("\n"); 306 307 sc->hw_if = hwp; 308 sc->hw_hdl = hdlp; 309 sc->sc_dev = parent; 310 sc->sc_writing = sc->sc_waitcomp = 0; 311 312 error = audio_alloc_ring(sc, &sc->sc_pr, AUMODE_PLAY, AU_RING_SIZE); 313 if (error) { 314 sc->hw_if = NULL; 315 printf("audio: could not allocate play buffer\n"); 316 return; 317 } 318 error = audio_alloc_ring(sc, &sc->sc_rr, AUMODE_RECORD, AU_RING_SIZE); 319 if (error) { 320 audio_free_ring(sc, &sc->sc_pr); 321 sc->hw_if = NULL; 322 printf("audio: could not allocate record buffer\n"); 323 return; 324 } 325 326 if ((error = audio_set_defaults(sc, 0))) { 327 printf("audioattach: audio_set_defaults() failed\n"); 328 sc->hw_if = NULL; 329 return; 330 } 331 332 iclass = mclass = oclass = rclass = -1; 333 sc->sc_inports.index = -1; 334 sc->sc_inports.master = -1; 335 sc->sc_inports.nports = 0; 336 sc->sc_inports.isenum = FALSE; 337 sc->sc_inports.allports = 0; 338 sc->sc_inports.isdual = FALSE; 339 sc->sc_inports.mixerout = -1; 340 sc->sc_inports.cur_port = -1; 341 sc->sc_outports.index = -1; 342 sc->sc_outports.master = -1; 343 sc->sc_outports.nports = 0; 344 sc->sc_outports.isenum = FALSE; 345 sc->sc_outports.allports = 0; 346 sc->sc_outports.isdual = FALSE; 347 sc->sc_outports.mixerout = -1; 348 sc->sc_outports.cur_port = -1; 349 sc->sc_monitor_port = -1; 350 /* 351 * Read through the underlying driver's list, picking out the class 352 * names from the mixer descriptions. We'll need them to decode the 353 * mixer descriptions on the next pass through the loop. 354 */ 355 for(mi.index = 0; ; mi.index++) { 356 if (hwp->query_devinfo(hdlp, &mi) != 0) 357 break; 358 /* 359 * The type of AUDIO_MIXER_CLASS merely introduces a class. 360 * All the other types describe an actual mixer. 361 */ 362 if (mi.type == AUDIO_MIXER_CLASS) { 363 if (strcmp(mi.label.name, AudioCinputs) == 0) 364 iclass = mi.mixer_class; 365 if (strcmp(mi.label.name, AudioCmonitor) == 0) 366 mclass = mi.mixer_class; 367 if (strcmp(mi.label.name, AudioCoutputs) == 0) 368 oclass = mi.mixer_class; 369 if (strcmp(mi.label.name, AudioCrecord) == 0) 370 rclass = mi.mixer_class; 371 } 372 } 373 /* 374 * This is where we assign each control in the "audio" model, to the 375 * underlying "mixer" control. We walk through the whole list once, 376 * assigning likely candidates as we come across them. 377 */ 378 record_master_found = 0; 379 record_source_found = 0; 380 for(mi.index = 0; ; mi.index++) { 381 if (hwp->query_devinfo(hdlp, &mi) != 0) 382 break; 383 if (mi.type == AUDIO_MIXER_CLASS) 384 continue; 385 if (mi.mixer_class == iclass) { 386 /* 387 * AudioCinputs is only a fallback, when we don't 388 * find what we're looking for in AudioCrecord, so 389 * check the flags before accepting one of these. 390 */ 391 if (strcmp(mi.label.name, AudioNmaster) == 0 392 && record_master_found == 0) 393 sc->sc_inports.master = mi.index; 394 if (strcmp(mi.label.name, AudioNsource) == 0 395 && record_source_found == 0) { 396 if (mi.type == AUDIO_MIXER_ENUM) { 397 int i; 398 for(i = 0; i < mi.un.e.num_mem; i++) 399 if (strcmp(mi.un.e.member[i].label.name, 400 AudioNmixerout) == 0) 401 sc->sc_inports.mixerout = 402 mi.un.e.member[i].ord; 403 } 404 au_setup_ports(sc, &sc->sc_inports, &mi, 405 itable); 406 } 407 } else if (mi.mixer_class == mclass) { 408 if (strcmp(mi.label.name, AudioNmonitor) == 0) 409 sc->sc_monitor_port = mi.index; 410 } else if (mi.mixer_class == oclass) { 411 if (strcmp(mi.label.name, AudioNmaster) == 0) 412 sc->sc_outports.master = mi.index; 413 if (strcmp(mi.label.name, AudioNselect) == 0) 414 au_setup_ports(sc, &sc->sc_outports, &mi, 415 otable); 416 } else if (mi.mixer_class == rclass) { 417 /* 418 * These are the preferred mixers for the audio record 419 * controls, so set the flags here, but don't check. 420 */ 421 if (strcmp(mi.label.name, AudioNmaster) == 0) { 422 sc->sc_inports.master = mi.index; 423 record_master_found = 1; 424 } 425 #if 1 /* Deprecated. Use AudioNmaster. */ 426 if (strcmp(mi.label.name, AudioNrecord) == 0) { 427 sc->sc_inports.master = mi.index; 428 record_master_found = 1; 429 } 430 if (strcmp(mi.label.name, AudioNvolume) == 0) { 431 sc->sc_inports.master = mi.index; 432 record_master_found = 1; 433 } 434 #endif 435 if (strcmp(mi.label.name, AudioNsource) == 0) { 436 if (mi.type == AUDIO_MIXER_ENUM) { 437 int i; 438 for(i = 0; i < mi.un.e.num_mem; i++) 439 if (strcmp(mi.un.e.member[i].label.name, 440 AudioNmixerout) == 0) 441 sc->sc_inports.mixerout = 442 mi.un.e.member[i].ord; 443 } 444 au_setup_ports(sc, &sc->sc_inports, &mi, 445 itable); 446 record_source_found = 1; 447 } 448 } 449 } 450 DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, " 451 "output ports=0x%x, output master=%d\n", 452 sc->sc_inports.allports, sc->sc_inports.master, 453 sc->sc_outports.allports, sc->sc_outports.master)); 454 } 455 456 int 457 audioactivate(struct device *self, enum devact act) 458 { 459 struct audio_softc *sc; 460 461 sc = (struct audio_softc *)self; 462 switch (act) { 463 case DVACT_ACTIVATE: 464 return EOPNOTSUPP; 465 466 case DVACT_DEACTIVATE: 467 sc->sc_dying = TRUE; 468 break; 469 } 470 return 0; 471 } 472 473 int 474 audiodetach(struct device *self, int flags) 475 { 476 struct audio_softc *sc; 477 int maj, mn; 478 int s; 479 480 sc = (struct audio_softc *)self; 481 DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags)); 482 483 sc->sc_dying = TRUE; 484 485 wakeup(&sc->sc_wchan); 486 wakeup(&sc->sc_rchan); 487 s = splaudio(); 488 if (--sc->sc_refcnt >= 0) { 489 if (tsleep(&sc->sc_refcnt, PZERO, "auddet", hz * 120)) 490 printf("audiodetach: %s didn't detach\n", 491 sc->dev.dv_xname); 492 } 493 splx(s); 494 495 /* free resources */ 496 audio_free_ring(sc, &sc->sc_pr); 497 audio_free_ring(sc, &sc->sc_rr); 498 audio_destruct_pfilters(sc); 499 audio_destruct_rfilters(sc); 500 501 /* locate the major number */ 502 maj = cdevsw_lookup_major(&audio_cdevsw); 503 504 /* Nuke the vnodes for any open instances (calls close). */ 505 mn = self->dv_unit; 506 vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); 507 vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); 508 vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); 509 vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); 510 511 return 0; 512 } 513 514 int 515 au_portof(struct audio_softc *sc, char *name, int class) 516 { 517 mixer_devinfo_t mi; 518 519 for(mi.index = 0; 520 sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0; 521 mi.index++) 522 if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) 523 return mi.index; 524 return -1; 525 } 526 527 void 528 au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, 529 mixer_devinfo_t *mi, const struct portname *tbl) 530 { 531 int i, j; 532 533 ports->index = mi->index; 534 if (mi->type == AUDIO_MIXER_ENUM) { 535 ports->isenum = TRUE; 536 for(i = 0; tbl[i].name; i++) 537 for(j = 0; j < mi->un.e.num_mem; j++) 538 if (strcmp(mi->un.e.member[j].label.name, 539 tbl[i].name) == 0) { 540 ports->allports |= tbl[i].mask; 541 ports->aumask[ports->nports] = tbl[i].mask; 542 ports->misel[ports->nports] = 543 mi->un.e.member[j].ord; 544 ports->miport[ports->nports] = 545 au_portof(sc, mi->un.e.member[j].label.name, 546 mi->mixer_class); 547 if (ports->mixerout != -1 && 548 ports->miport[ports->nports] != -1) 549 ports->isdual = TRUE; 550 ++ports->nports; 551 } 552 } else if (mi->type == AUDIO_MIXER_SET) { 553 for(i = 0; tbl[i].name; i++) 554 for(j = 0; j < mi->un.s.num_mem; j++) 555 if (strcmp(mi->un.s.member[j].label.name, 556 tbl[i].name) == 0) { 557 ports->allports |= tbl[i].mask; 558 ports->aumask[ports->nports] = tbl[i].mask; 559 ports->misel[ports->nports] = 560 mi->un.s.member[j].mask; 561 ports->miport[ports->nports] = 562 au_portof(sc, mi->un.s.member[j].label.name, 563 mi->mixer_class); 564 ++ports->nports; 565 } 566 } 567 } 568 569 /* 570 * Called from hardware driver. This is where the MI audio driver gets 571 * probed/attached to the hardware driver. 572 */ 573 struct device * 574 audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, struct device *dev) 575 { 576 struct audio_attach_args arg; 577 578 #ifdef DIAGNOSTIC 579 if (ahwp == NULL) { 580 aprint_error("audio_attach_mi: NULL\n"); 581 return 0; 582 } 583 #endif 584 arg.type = AUDIODEV_TYPE_AUDIO; 585 arg.hwif = ahwp; 586 arg.hdl = hdlp; 587 return config_found(dev, &arg, audioprint); 588 } 589 590 #ifdef AUDIO_DEBUG 591 void audio_printsc(struct audio_softc *); 592 void audio_print_params(const char *, struct audio_params *); 593 594 void 595 audio_printsc(struct audio_softc *sc) 596 { 597 printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if); 598 printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode); 599 printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan); 600 printf("rring used 0x%x pring used=%d\n", 601 audio_stream_get_used(&sc->sc_rr.s), 602 audio_stream_get_used(&sc->sc_pr.s)); 603 printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus); 604 printf("blksize %d", sc->sc_pr.blksize); 605 printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow); 606 } 607 608 void 609 audio_print_params(const char *s, struct audio_params *p) 610 { 611 printf("%s enc=%u %uch %u/%ubit %uHz\n", s, p->encoding, p->channels, 612 p->validbits, p->precision, p->sample_rate); 613 } 614 #endif 615 616 int 617 audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r, 618 int direction, size_t bufsize) 619 { 620 const struct audio_hw_if *hw; 621 void *hdl; 622 623 hw = sc->hw_if; 624 hdl = sc->hw_hdl; 625 /* 626 * Alloc DMA play and record buffers 627 */ 628 if (bufsize < AUMINBUF) 629 bufsize = AUMINBUF; 630 ROUNDSIZE(bufsize); 631 if (hw->round_buffersize) 632 bufsize = hw->round_buffersize(hdl, direction, bufsize); 633 if (hw->allocm) 634 r->s.start = hw->allocm(hdl, direction, bufsize, 635 M_DEVBUF, M_WAITOK); 636 else 637 r->s.start = malloc(bufsize, M_DEVBUF, M_WAITOK); 638 if (r->s.start == 0) 639 return ENOMEM; 640 r->s.bufsize = bufsize; 641 return 0; 642 } 643 644 void 645 audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r) 646 { 647 648 if (sc->hw_if->freem) 649 sc->hw_if->freem(sc->hw_hdl, r->s.start, M_DEVBUF); 650 else 651 free(r->s.start, M_DEVBUF); 652 r->s.start = 0; 653 } 654 655 static int 656 audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp, 657 stream_filter_list_t *pfilters) 658 { 659 stream_filter_t *pf[AUDIO_MAX_FILTERS]; 660 audio_stream_t ps[AUDIO_MAX_FILTERS]; 661 const audio_params_t *from_param; 662 audio_params_t *to_param; 663 int i, n; 664 665 while (sc->sc_writing) { 666 sc->sc_waitcomp = 1; 667 (void)tsleep(sc, 0, "audioch", 10*hz); 668 } 669 670 memset(pf, 0, sizeof(pf)); 671 memset(ps, 0, sizeof(ps)); 672 from_param = pp; 673 for (i = 0; i < pfilters->req_size; i++) { 674 n = pfilters->req_size - i - 1; 675 to_param = &pfilters->filters[n].param; 676 audio_check_params(to_param); 677 pf[i] = pfilters->filters[n].factory(sc, from_param, to_param); 678 if (pf[i] == NULL) 679 break; 680 if (audio_stream_ctor(&ps[i], from_param, AU_RING_SIZE)) 681 break; 682 if (i > 0) 683 pf[i]->set_fetcher(pf[i], &pf[i - 1]->base); 684 from_param = to_param; 685 } 686 if (i < pfilters->req_size) { /* failure */ 687 DPRINTF(("%s: pfilters failure\n", __func__)); 688 for (; i >= 0; i--) { 689 if (pf[i] != NULL) 690 pf[i]->dtor(pf[i]); 691 audio_stream_dtor(&ps[i]); 692 } 693 sc->sc_waitcomp = 0; 694 return EINVAL; 695 } 696 697 audio_destruct_pfilters(sc); 698 memcpy(sc->sc_pfilters, pf, sizeof(pf)); 699 memcpy(sc->sc_pstreams, ps, sizeof(ps)); 700 sc->sc_npfilters = pfilters->req_size; 701 for (i = 0; i < pfilters->req_size; i++) { 702 pf[i]->set_inputbuffer(pf[i], &sc->sc_pstreams[i]); 703 } 704 705 /* hardware format and the buffer near to userland */ 706 if (pfilters->req_size <= 0) { 707 sc->sc_pr.s.param = *pp; 708 sc->sc_pustream = &sc->sc_pr.s; 709 } else { 710 sc->sc_pr.s.param = pfilters->filters[0].param; 711 sc->sc_pustream = &sc->sc_pstreams[0]; 712 } 713 #ifdef AUDIO_DEBUG 714 printf("%s: HW-buffer=%p pustream=%p\n", 715 __func__, &sc->sc_pr.s, sc->sc_pustream); 716 for (i = 0; i < pfilters->req_size; i++) { 717 char num[100]; 718 snprintf(num, 100, "[%d]", i); 719 audio_print_params(num, &sc->sc_pstreams[i].param); 720 } 721 audio_print_params("[HW]", &sc->sc_pr.s.param); 722 #endif /* AUDIO_DEBUG */ 723 724 sc->sc_waitcomp = 0; 725 return 0; 726 } 727 728 static int 729 audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp, 730 stream_filter_list_t *rfilters) 731 { 732 stream_filter_t *rf[AUDIO_MAX_FILTERS]; 733 audio_stream_t rs[AUDIO_MAX_FILTERS]; 734 const audio_params_t *to_param; 735 audio_params_t *from_param; 736 int i; 737 738 memset(rf, 0, sizeof(rf)); 739 memset(rs, 0, sizeof(rs)); 740 for (i = 0; i < rfilters->req_size; i++) { 741 from_param = &rfilters->filters[i].param; 742 audio_check_params(from_param); 743 to_param = i + 1 < rfilters->req_size 744 ? &rfilters->filters[i + 1].param : rp; 745 rf[i] = rfilters->filters[i].factory(sc, from_param, to_param); 746 if (rf[i] == NULL) 747 break; 748 if (audio_stream_ctor(&rs[i], to_param, AU_RING_SIZE)) 749 break; 750 if (i > 0) { 751 rf[i]->set_fetcher(rf[i], &rf[i - 1]->base); 752 } else { 753 /* rf[0] has no previous fetcher because 754 * the audio hardware fills data to the 755 * input buffer. */ 756 rf[0]->set_inputbuffer(rf[0], &sc->sc_rr.s); 757 } 758 } 759 if (i < rfilters->req_size) { /* failure */ 760 DPRINTF(("%s: rfilters failure\n", __func__)); 761 for (; i >= 0; i--) { 762 if (rf[i] != NULL) 763 rf[i]->dtor(rf[i]); 764 audio_stream_dtor(&rs[i]); 765 } 766 return EINVAL; 767 } 768 769 audio_destruct_rfilters(sc); 770 memcpy(sc->sc_rfilters, rf, sizeof(rf)); 771 memcpy(sc->sc_rstreams, rs, sizeof(rs)); 772 sc->sc_nrfilters = rfilters->req_size; 773 for (i = 1; i < rfilters->req_size; i++) { 774 rf[i]->set_inputbuffer(rf[i], &sc->sc_rstreams[i - 1]); 775 } 776 777 /* hardware format and the buffer near to userland */ 778 if (rfilters->req_size <= 0) { 779 sc->sc_rr.s.param = *rp; 780 sc->sc_rustream = &sc->sc_rr.s; 781 } else { 782 sc->sc_rr.s.param = rfilters->filters[0].param; 783 sc->sc_rustream = &sc->sc_rstreams[rfilters->req_size - 1]; 784 } 785 #ifdef AUDIO_DEBUG 786 printf("%s: HW-buffer=%p pustream=%p\n", 787 __func__, &sc->sc_rr.s, sc->sc_rustream); 788 audio_print_params("[HW]", &sc->sc_rr.s.param); 789 for (i = 0; i < rfilters->req_size; i++) { 790 char num[100]; 791 snprintf(num, 100, "[%d]", i); 792 audio_print_params(num, &sc->sc_rstreams[i].param); 793 } 794 #endif /* AUDIO_DEBUG */ 795 return 0; 796 } 797 798 static void 799 audio_destruct_pfilters(struct audio_softc *sc) 800 { 801 int i; 802 803 for (i = 0; i < sc->sc_npfilters; i++) { 804 sc->sc_pfilters[i]->dtor(sc->sc_pfilters[i]); 805 sc->sc_pfilters[i] = NULL; 806 audio_stream_dtor(&sc->sc_pstreams[i]); 807 } 808 sc->sc_npfilters = 0; 809 } 810 811 static void 812 audio_destruct_rfilters(struct audio_softc *sc) 813 { 814 int i; 815 816 for (i = 0; i < sc->sc_nrfilters; i++) { 817 sc->sc_rfilters[i]->dtor(sc->sc_rfilters[i]); 818 sc->sc_rfilters[i] = NULL; 819 audio_stream_dtor(&sc->sc_rstreams[i]); 820 } 821 sc->sc_nrfilters = 0; 822 } 823 824 static void 825 audio_stream_dtor(audio_stream_t *stream) 826 { 827 828 if (stream->start != NULL) 829 free(stream->start, M_DEVBUF); 830 memset(stream, 0, sizeof(audio_stream_t)); 831 } 832 833 static int 834 audio_stream_ctor(audio_stream_t *stream, const audio_params_t *param, int size) 835 { 836 int frame_size; 837 838 size = min(size, AU_RING_SIZE); 839 stream->bufsize = size; 840 stream->start = malloc(size, M_DEVBUF, M_WAITOK); 841 if (stream->start == NULL) 842 return ENOMEM; 843 frame_size = (param->precision + 7) / 8 * param->channels; 844 size = (size / frame_size) * frame_size; 845 stream->end = stream->start + size; 846 stream->inp = stream->start; 847 stream->outp = stream->start; 848 stream->used = 0; 849 stream->param = *param; 850 stream->loop = FALSE; 851 return 0; 852 } 853 854 static void 855 stream_filter_list_append(stream_filter_list_t *list, 856 stream_filter_factory_t factory, 857 const audio_params_t *param) 858 { 859 860 if (list->req_size >= AUDIO_MAX_FILTERS) { 861 printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n", 862 __func__); 863 return; 864 } 865 list->filters[list->req_size].factory = factory; 866 list->filters[list->req_size].param = *param; 867 list->req_size++; 868 } 869 870 static void 871 stream_filter_list_set(stream_filter_list_t *list, int i, 872 stream_filter_factory_t factory, 873 const audio_params_t *param) 874 { 875 876 if (i < 0 || i >= AUDIO_MAX_FILTERS) { 877 printf("%s: invalid index: %d\n", __func__, i); 878 return; 879 } 880 881 list->filters[i].factory = factory; 882 list->filters[i].param = *param; 883 if (list->req_size <= i) 884 list->req_size = i + 1; 885 } 886 887 static void 888 stream_filter_list_prepend(stream_filter_list_t *list, 889 stream_filter_factory_t factory, 890 const audio_params_t *param) 891 { 892 893 if (list->req_size >= AUDIO_MAX_FILTERS) { 894 printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n", 895 __func__); 896 return; 897 } 898 memmove(&list->filters[1], &list->filters[0], 899 sizeof(struct stream_filter_req) * list->req_size); 900 list->filters[0].factory = factory; 901 list->filters[0].param = *param; 902 list->req_size++; 903 } 904 905 int 906 audioopen(dev_t dev, int flags, int ifmt, struct proc *p) 907 { 908 struct audio_softc *sc; 909 int error; 910 911 sc = device_lookup(&audio_cd, AUDIOUNIT(dev)); 912 if (sc == NULL) 913 return ENXIO; 914 915 if (sc->sc_dying) 916 return EIO; 917 918 sc->sc_refcnt++; 919 switch (AUDIODEV(dev)) { 920 case SOUND_DEVICE: 921 case AUDIO_DEVICE: 922 error = audio_open(dev, sc, flags, ifmt, p); 923 break; 924 case AUDIOCTL_DEVICE: 925 error = 0; 926 break; 927 case MIXER_DEVICE: 928 error = mixer_open(dev, sc, flags, ifmt, p); 929 break; 930 default: 931 error = ENXIO; 932 break; 933 } 934 if (--sc->sc_refcnt < 0) 935 wakeup(&sc->sc_refcnt); 936 return error; 937 } 938 939 int 940 audioclose(dev_t dev, int flags, int ifmt, struct proc *p) 941 { 942 struct audio_softc *sc; 943 int unit; 944 int error; 945 946 unit = AUDIOUNIT(dev); 947 sc = audio_cd.cd_devs[unit]; 948 switch (AUDIODEV(dev)) { 949 case SOUND_DEVICE: 950 case AUDIO_DEVICE: 951 error = audio_close(sc, flags, ifmt, p); 952 break; 953 case MIXER_DEVICE: 954 error = mixer_close(sc, flags, ifmt, p); 955 break; 956 case AUDIOCTL_DEVICE: 957 error = 0; 958 break; 959 default: 960 error = ENXIO; 961 break; 962 } 963 return error; 964 } 965 966 int 967 audioread(dev_t dev, struct uio *uio, int ioflag) 968 { 969 struct audio_softc *sc; 970 int error; 971 972 sc = device_lookup(&audio_cd, AUDIOUNIT(dev)); 973 if (sc == NULL) 974 return ENXIO; 975 976 if (sc->sc_dying) 977 return EIO; 978 979 sc->sc_refcnt++; 980 switch (AUDIODEV(dev)) { 981 case SOUND_DEVICE: 982 case AUDIO_DEVICE: 983 error = audio_read(sc, uio, ioflag); 984 break; 985 case AUDIOCTL_DEVICE: 986 case MIXER_DEVICE: 987 error = ENODEV; 988 break; 989 default: 990 error = ENXIO; 991 break; 992 } 993 if (--sc->sc_refcnt < 0) 994 wakeup(&sc->sc_refcnt); 995 return error; 996 } 997 998 int 999 audiowrite(dev_t dev, struct uio *uio, int ioflag) 1000 { 1001 struct audio_softc *sc; 1002 int error; 1003 1004 sc = device_lookup(&audio_cd, AUDIOUNIT(dev)); 1005 if (sc == NULL) 1006 return ENXIO; 1007 1008 if (sc->sc_dying) 1009 return EIO; 1010 1011 sc->sc_refcnt++; 1012 switch (AUDIODEV(dev)) { 1013 case SOUND_DEVICE: 1014 case AUDIO_DEVICE: 1015 error = audio_write(sc, uio, ioflag); 1016 break; 1017 case AUDIOCTL_DEVICE: 1018 case MIXER_DEVICE: 1019 error = ENODEV; 1020 break; 1021 default: 1022 error = ENXIO; 1023 break; 1024 } 1025 if (--sc->sc_refcnt < 0) 1026 wakeup(&sc->sc_refcnt); 1027 return error; 1028 } 1029 1030 int 1031 audioioctl(dev_t dev, u_long cmd, caddr_t addr, int flag, struct proc *p) 1032 { 1033 struct audio_softc *sc; 1034 int error; 1035 1036 sc = audio_cd.cd_devs[AUDIOUNIT(dev)]; 1037 if (sc->sc_dying) 1038 return EIO; 1039 1040 sc->sc_refcnt++; 1041 switch (AUDIODEV(dev)) { 1042 case SOUND_DEVICE: 1043 case AUDIO_DEVICE: 1044 case AUDIOCTL_DEVICE: 1045 error = audio_ioctl(sc, cmd, addr, flag, p); 1046 break; 1047 case MIXER_DEVICE: 1048 error = mixer_ioctl(sc, cmd, addr, flag, p); 1049 break; 1050 default: 1051 error = ENXIO; 1052 break; 1053 } 1054 if (--sc->sc_refcnt < 0) 1055 wakeup(&sc->sc_refcnt); 1056 return error; 1057 } 1058 1059 int 1060 audiopoll(dev_t dev, int events, struct proc *p) 1061 { 1062 struct audio_softc *sc; 1063 int error; 1064 1065 sc = audio_cd.cd_devs[AUDIOUNIT(dev)]; 1066 if (sc->sc_dying) 1067 return EIO; 1068 1069 sc->sc_refcnt++; 1070 switch (AUDIODEV(dev)) { 1071 case SOUND_DEVICE: 1072 case AUDIO_DEVICE: 1073 error = audio_poll(sc, events, p); 1074 break; 1075 case AUDIOCTL_DEVICE: 1076 case MIXER_DEVICE: 1077 error = ENODEV; 1078 break; 1079 default: 1080 error = ENXIO; 1081 break; 1082 } 1083 if (--sc->sc_refcnt < 0) 1084 wakeup(&sc->sc_refcnt); 1085 return error; 1086 } 1087 1088 int 1089 audiokqfilter(dev_t dev, struct knote *kn) 1090 { 1091 struct audio_softc *sc; 1092 int rv; 1093 1094 sc = audio_cd.cd_devs[AUDIOUNIT(dev)]; 1095 if (sc->sc_dying) 1096 return 1; 1097 1098 sc->sc_refcnt++; 1099 switch (AUDIODEV(dev)) { 1100 case SOUND_DEVICE: 1101 case AUDIO_DEVICE: 1102 rv = audio_kqfilter(sc, kn); 1103 break; 1104 case AUDIOCTL_DEVICE: 1105 case MIXER_DEVICE: 1106 rv = 1; 1107 break; 1108 default: 1109 rv = 1; 1110 } 1111 if (--sc->sc_refcnt < 0) 1112 wakeup(&sc->sc_refcnt); 1113 return rv; 1114 } 1115 1116 paddr_t 1117 audiommap(dev_t dev, off_t off, int prot) 1118 { 1119 struct audio_softc *sc; 1120 paddr_t error; 1121 1122 sc = audio_cd.cd_devs[AUDIOUNIT(dev)]; 1123 if (sc->sc_dying) 1124 return -1; 1125 1126 sc->sc_refcnt++; 1127 switch (AUDIODEV(dev)) { 1128 case SOUND_DEVICE: 1129 case AUDIO_DEVICE: 1130 error = audio_mmap(sc, off, prot); 1131 break; 1132 case AUDIOCTL_DEVICE: 1133 case MIXER_DEVICE: 1134 error = -1; 1135 break; 1136 default: 1137 error = -1; 1138 break; 1139 } 1140 if (--sc->sc_refcnt < 0) 1141 wakeup(&sc->sc_refcnt); 1142 return error; 1143 } 1144 1145 /* 1146 * Audio driver 1147 */ 1148 void 1149 audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp, 1150 int mode) 1151 { 1152 int nblks; 1153 int blksize; 1154 1155 blksize = rp->blksize; 1156 if (blksize < AUMINBLK) 1157 blksize = AUMINBLK; 1158 if (blksize > rp->s.bufsize / AUMINNOBLK) 1159 blksize = rp->s.bufsize / AUMINNOBLK; 1160 ROUNDSIZE(blksize); 1161 DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n", blksize)); 1162 if (sc->hw_if->round_blocksize) 1163 blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, 1164 mode, &rp->s.param); 1165 if (blksize <= 0) 1166 panic("audio_init_ringbuffer: blksize"); 1167 nblks = rp->s.bufsize / blksize; 1168 1169 DPRINTF(("audio_init_ringbuffer: final blksize=%d\n", blksize)); 1170 rp->blksize = blksize; 1171 rp->maxblks = nblks; 1172 rp->s.end = rp->s.start + nblks * blksize; 1173 rp->s.outp = rp->s.inp = rp->s.start; 1174 rp->s.used = 0; 1175 rp->stamp = 0; 1176 rp->stamp_last = 0; 1177 rp->fstamp = 0; 1178 rp->drops = 0; 1179 rp->pause = FALSE; 1180 rp->copying = FALSE; 1181 rp->needfill = FALSE; 1182 rp->mmapped = FALSE; 1183 } 1184 1185 int 1186 audio_initbufs(struct audio_softc *sc) 1187 { 1188 const struct audio_hw_if *hw; 1189 int error; 1190 1191 DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode)); 1192 hw = sc->hw_if; 1193 audio_init_ringbuffer(sc, &sc->sc_rr, AUMODE_RECORD); 1194 if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) { 1195 error = hw->init_input(sc->hw_hdl, sc->sc_rr.s.start, 1196 sc->sc_rr.s.end - sc->sc_rr.s.start); 1197 if (error) 1198 return error; 1199 } 1200 1201 audio_init_ringbuffer(sc, &sc->sc_pr, AUMODE_PLAY); 1202 sc->sc_sil_count = 0; 1203 if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) { 1204 error = hw->init_output(sc->hw_hdl, sc->sc_pr.s.start, 1205 sc->sc_pr.s.end - sc->sc_pr.s.start); 1206 if (error) 1207 return error; 1208 } 1209 1210 #ifdef AUDIO_INTR_TIME 1211 #define double u_long 1212 sc->sc_pnintr = 0; 1213 sc->sc_pblktime = (u_long)( 1214 (double)sc->sc_pr.blksize * 100000 / 1215 (double)(sc->sc_pparams.precision / NBBY * 1216 sc->sc_pparams.channels * 1217 sc->sc_pparams.sample_rate)) * 10; 1218 DPRINTF(("audio: play blktime = %lu for %d\n", 1219 sc->sc_pblktime, sc->sc_pr.blksize)); 1220 sc->sc_rnintr = 0; 1221 sc->sc_rblktime = (u_long)( 1222 (double)sc->sc_rr.blksize * 100000 / 1223 (double)(sc->sc_rparams.precision / NBBY * 1224 sc->sc_rparams.channels * 1225 sc->sc_rparams.sample_rate)) * 10; 1226 DPRINTF(("audio: record blktime = %lu for %d\n", 1227 sc->sc_rblktime, sc->sc_rr.blksize)); 1228 #undef double 1229 #endif 1230 1231 return 0; 1232 } 1233 1234 void 1235 audio_calcwater(struct audio_softc *sc) 1236 { 1237 1238 /* set high at 100% */ 1239 sc->sc_pr.usedhigh = sc->sc_pustream->end - sc->sc_pustream->start; 1240 /* set low at 75% of usedhigh */ 1241 sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4; 1242 if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh) 1243 sc->sc_pr.usedlow -= sc->sc_pr.blksize; 1244 1245 sc->sc_rr.usedhigh = sc->sc_rustream->end - sc->sc_rustream->start 1246 - sc->sc_rr.blksize; 1247 sc->sc_rr.usedlow = 0; 1248 DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n", __func__, 1249 sc->sc_pr.usedlow, sc->sc_pr.usedhigh, 1250 sc->sc_rr.usedlow, sc->sc_rr.usedhigh)); 1251 } 1252 1253 static __inline int 1254 audio_sleep_timo(int *chan, const char *label, int timo) 1255 { 1256 int st; 1257 1258 if (label == NULL) 1259 label = "audio"; 1260 1261 DPRINTFN(3, ("audio_sleep_timo: chan=%p, label=%s, timo=%d\n", 1262 chan, label, timo)); 1263 *chan = 1; 1264 st = tsleep(chan, PWAIT | PCATCH, label, timo); 1265 *chan = 0; 1266 #ifdef AUDIO_DEBUG 1267 if (st != 0 && st != EINTR) 1268 DPRINTF(("audio_sleep: woke up st=%d\n", st)); 1269 #endif 1270 return st; 1271 } 1272 1273 static __inline int 1274 audio_sleep(int *chan, const char *label) 1275 { 1276 1277 return audio_sleep_timo(chan, label, 0); 1278 } 1279 1280 /* call at splaudio() */ 1281 static __inline void 1282 audio_wakeup(int *chan) 1283 { 1284 1285 DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan)); 1286 if (*chan) { 1287 wakeup(chan); 1288 *chan = 0; 1289 } 1290 } 1291 1292 int 1293 audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 1294 struct proc *p) 1295 { 1296 int error; 1297 u_int mode; 1298 const struct audio_hw_if *hw; 1299 1300 hw = sc->hw_if; 1301 if (hw == NULL) 1302 return ENXIO; 1303 1304 DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n", 1305 flags, sc, sc->hw_hdl)); 1306 1307 if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0) 1308 return EBUSY; 1309 1310 if (hw->open != NULL) { 1311 error = hw->open(sc->hw_hdl, flags); 1312 if (error) 1313 return error; 1314 } 1315 1316 sc->sc_async_audio = 0; 1317 sc->sc_rchan = 0; 1318 sc->sc_wchan = 0; 1319 sc->sc_sil_count = 0; 1320 sc->sc_rbus = FALSE; 1321 sc->sc_pbus = FALSE; 1322 sc->sc_eof = 0; 1323 sc->sc_playdrop = 0; 1324 1325 sc->sc_full_duplex = 1326 (flags & (FWRITE|FREAD)) == (FWRITE|FREAD) && 1327 (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX); 1328 1329 mode = 0; 1330 if (flags & FREAD) { 1331 sc->sc_open |= AUOPEN_READ; 1332 mode |= AUMODE_RECORD; 1333 } 1334 if (flags & FWRITE) { 1335 sc->sc_open |= AUOPEN_WRITE; 1336 mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; 1337 } 1338 1339 /* 1340 * Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear) 1341 * The /dev/audio is always (re)set to 8-bit MU-Law mono 1342 * For the other devices, you get what they were last set to. 1343 */ 1344 if (ISDEVAUDIO(dev)) { 1345 error = audio_set_defaults(sc, mode); 1346 } else { 1347 struct audio_info ai; 1348 1349 AUDIO_INITINFO(&ai); 1350 ai.mode = mode; 1351 error = audiosetinfo(sc, &ai); 1352 } 1353 if (error) 1354 goto bad; 1355 1356 #ifdef DIAGNOSTIC 1357 /* 1358 * Sample rate and precision are supposed to be set to proper 1359 * default values by the hardware driver, so that it may give 1360 * us these values. 1361 */ 1362 if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) { 1363 printf("audio_open: 0 precision\n"); 1364 return EINVAL; 1365 } 1366 #endif 1367 1368 /* audio_close() decreases sc_pr.usedlow, recalculate here */ 1369 audio_calcwater(sc); 1370 1371 DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode)); 1372 1373 return 0; 1374 1375 bad: 1376 if (hw->close != NULL) 1377 hw->close(sc->hw_hdl); 1378 sc->sc_open = 0; 1379 sc->sc_mode = 0; 1380 sc->sc_full_duplex = 0; 1381 return error; 1382 } 1383 1384 /* 1385 * Must be called from task context. 1386 */ 1387 void 1388 audio_init_record(struct audio_softc *sc) 1389 { 1390 int s; 1391 1392 s = splaudio(); 1393 if (sc->hw_if->speaker_ctl && 1394 (!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0)) 1395 sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF); 1396 splx(s); 1397 } 1398 1399 /* 1400 * Must be called from task context. 1401 */ 1402 void 1403 audio_init_play(struct audio_softc *sc) 1404 { 1405 int s; 1406 1407 s = splaudio(); 1408 sc->sc_wstamp = sc->sc_pr.stamp; 1409 if (sc->hw_if->speaker_ctl) 1410 sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON); 1411 splx(s); 1412 } 1413 1414 int 1415 audio_drain(struct audio_softc *sc) 1416 { 1417 struct audio_ringbuffer *cb; 1418 int error, drops; 1419 int s; 1420 int i, used; 1421 1422 DPRINTF(("audio_drain: enter busy=%d\n", sc->sc_pbus)); 1423 cb = &sc->sc_pr; 1424 if (cb->mmapped) 1425 return 0; 1426 1427 used = audio_stream_get_used(&sc->sc_pr.s); 1428 s = splaudio(); 1429 for (i = 0; i < sc->sc_npfilters; i++) 1430 used += audio_stream_get_used(&sc->sc_pstreams[i]); 1431 splx(s); 1432 if (used <= 0) 1433 return 0; 1434 1435 if (!sc->sc_pbus) { 1436 /* We've never started playing, probably because the 1437 * block was too short. Pad it and start now. 1438 */ 1439 int cc; 1440 uint8_t *inp = cb->s.inp; 1441 1442 cc = cb->blksize - (inp - cb->s.start) % cb->blksize; 1443 audio_fill_silence(&cb->s.param, inp, cc); 1444 s = splaudio(); 1445 cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); 1446 error = audiostartp(sc); 1447 splx(s); 1448 if (error) 1449 return error; 1450 } 1451 /* 1452 * Play until a silence block has been played, then we 1453 * know all has been drained. 1454 * XXX This should be done some other way to avoid 1455 * playing silence. 1456 */ 1457 #ifdef DIAGNOSTIC 1458 if (cb->copying) { 1459 printf("audio_drain: copying in progress!?!\n"); 1460 cb->copying = FALSE; 1461 } 1462 #endif 1463 drops = cb->drops; 1464 error = 0; 1465 s = splaudio(); 1466 while (cb->drops == drops && !error) { 1467 DPRINTF(("audio_drain: used=%d, drops=%ld\n", 1468 audio_stream_get_used(&sc->sc_pr.s), cb->drops)); 1469 /* 1470 * When the process is exiting, it ignores all signals and 1471 * we can't interrupt this sleep, so we set a timeout 1472 * just in case. 1473 */ 1474 error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz); 1475 if (sc->sc_dying) 1476 error = EIO; 1477 } 1478 splx(s); 1479 return error; 1480 } 1481 1482 /* 1483 * Close an audio chip. 1484 */ 1485 /* ARGSUSED */ 1486 int 1487 audio_close(struct audio_softc *sc, int flags, int ifmt, struct proc *p) 1488 { 1489 const struct audio_hw_if *hw; 1490 int s; 1491 1492 DPRINTF(("audio_close: sc=%p\n", sc)); 1493 hw = sc->hw_if; 1494 s = splaudio(); 1495 /* Stop recording. */ 1496 if ((flags & FREAD) && sc->sc_rbus) { 1497 /* 1498 * XXX Some drivers (e.g. SB) use the same routine 1499 * to halt input and output so don't halt input if 1500 * in full duplex mode. These drivers should be fixed. 1501 */ 1502 if (!sc->sc_full_duplex || hw->halt_input != hw->halt_output) 1503 hw->halt_input(sc->hw_hdl); 1504 sc->sc_rbus = FALSE; 1505 } 1506 /* 1507 * Block until output drains, but allow ^C interrupt. 1508 */ 1509 sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */ 1510 /* 1511 * If there is pending output, let it drain (unless 1512 * the output is paused). 1513 */ 1514 if ((flags & FWRITE) && sc->sc_pbus) { 1515 if (!sc->sc_pr.pause && !audio_drain(sc) && hw->drain) 1516 (void)hw->drain(sc->hw_hdl); 1517 hw->halt_output(sc->hw_hdl); 1518 sc->sc_pbus = FALSE; 1519 } 1520 1521 if (hw->close != NULL) 1522 hw->close(sc->hw_hdl); 1523 1524 if (flags & FREAD) { 1525 sc->sc_open &= ~AUOPEN_READ; 1526 sc->sc_mode &= ~AUMODE_RECORD; 1527 } 1528 if (flags & FWRITE) { 1529 sc->sc_open &= ~AUOPEN_WRITE; 1530 sc->sc_mode &= ~(AUMODE_PLAY|AUMODE_PLAY_ALL); 1531 } 1532 1533 sc->sc_async_audio = 0; 1534 sc->sc_full_duplex = 0; 1535 splx(s); 1536 DPRINTF(("audio_close: done\n")); 1537 1538 return 0; 1539 } 1540 1541 int 1542 audio_read(struct audio_softc *sc, struct uio *uio, int ioflag) 1543 { 1544 struct audio_ringbuffer *cb; 1545 const uint8_t *outp; 1546 uint8_t *inp; 1547 int error, s, used, cc, n; 1548 1549 cb = &sc->sc_rr; 1550 if (cb->mmapped) 1551 return EINVAL; 1552 1553 DPRINTFN(1,("audio_read: cc=%zu mode=%d\n", 1554 uio->uio_resid, sc->sc_mode)); 1555 1556 error = 0; 1557 /* 1558 * If hardware is half-duplex and currently playing, return 1559 * silence blocks based on the number of blocks we have output. 1560 */ 1561 if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) { 1562 while (uio->uio_resid > 0 && !error) { 1563 s = splaudio(); 1564 for(;;) { 1565 cc = sc->sc_pr.stamp - sc->sc_wstamp; 1566 if (cc > 0) 1567 break; 1568 DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n", 1569 sc->sc_pr.stamp, sc->sc_wstamp)); 1570 if (ioflag & IO_NDELAY) { 1571 splx(s); 1572 return EWOULDBLOCK; 1573 } 1574 error = audio_sleep(&sc->sc_rchan, "aud_hr"); 1575 if (sc->sc_dying) 1576 error = EIO; 1577 if (error) { 1578 splx(s); 1579 return error; 1580 } 1581 } 1582 splx(s); 1583 1584 if (uio->uio_resid < cc) 1585 cc = uio->uio_resid; 1586 DPRINTFN(1,("audio_read: reading in write mode, " 1587 "cc=%d\n", cc)); 1588 error = audio_silence_copyout(sc, cc, uio); 1589 sc->sc_wstamp += cc; 1590 } 1591 return error; 1592 } 1593 1594 while (uio->uio_resid > 0 && !error) { 1595 s = splaudio(); 1596 while ((used = audio_stream_get_used(sc->sc_rustream)) <= 0) { 1597 if (!sc->sc_rbus) { 1598 error = audiostartr(sc); 1599 if (error) { 1600 splx(s); 1601 return error; 1602 } 1603 } 1604 if (ioflag & IO_NDELAY) { 1605 splx(s); 1606 return EWOULDBLOCK; 1607 } 1608 DPRINTFN(2, ("audio_read: sleep used=%d\n", used)); 1609 error = audio_sleep(&sc->sc_rchan, "aud_rd"); 1610 if (sc->sc_dying) 1611 error = EIO; 1612 if (error) { 1613 splx(s); 1614 return error; 1615 } 1616 } 1617 1618 outp = sc->sc_rustream->outp; 1619 inp = sc->sc_rustream->inp; 1620 cb->copying = TRUE; 1621 splx(s); 1622 1623 /* 1624 * cc is the amount of data in the sc_rustream excluding 1625 * wrapped data 1626 */ 1627 cc = outp <= inp ? inp - outp :sc->sc_rustream->end - outp; 1628 DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc)); 1629 1630 n = uio->uio_resid; 1631 error = uiomove(__UNCONST(outp), cc, uio); 1632 n -= uio->uio_resid; /* number of bytes actually moved */ 1633 1634 s = splaudio(); 1635 sc->sc_rustream->outp = audio_stream_add_outp 1636 (sc->sc_rustream, outp, n); 1637 cb->copying = FALSE; 1638 splx(s); 1639 } 1640 return error; 1641 } 1642 1643 void 1644 audio_clear(struct audio_softc *sc) 1645 { 1646 int s; 1647 1648 s = splaudio(); 1649 if (sc->sc_rbus) { 1650 audio_wakeup(&sc->sc_rchan); 1651 sc->hw_if->halt_input(sc->hw_hdl); 1652 sc->sc_rbus = FALSE; 1653 } 1654 if (sc->sc_pbus) { 1655 audio_wakeup(&sc->sc_wchan); 1656 sc->hw_if->halt_output(sc->hw_hdl); 1657 sc->sc_pbus = FALSE; 1658 } 1659 splx(s); 1660 } 1661 1662 void 1663 audio_calc_blksize(struct audio_softc *sc, int mode) 1664 { 1665 const audio_params_t *parm; 1666 struct audio_ringbuffer *rb; 1667 1668 if (sc->sc_blkset) 1669 return; 1670 1671 if (mode == AUMODE_PLAY) { 1672 rb = &sc->sc_pr; 1673 parm = &rb->s.param; 1674 } else { 1675 rb = &sc->sc_rr; 1676 parm = &rb->s.param; 1677 } 1678 1679 rb->blksize = parm->sample_rate * audio_blk_ms / 1000 * 1680 parm->channels * parm->precision / NBBY; 1681 1682 DPRINTF(("audio_calc_blksize: %s blksize=%d\n", 1683 mode == AUMODE_PLAY ? "play" : "record", rb->blksize)); 1684 } 1685 1686 void 1687 audio_fill_silence(struct audio_params *params, uint8_t *p, int n) 1688 { 1689 uint8_t auzero0, auzero1; 1690 int nfill; 1691 1692 auzero1 = 0; /* initialize to please gcc */ 1693 nfill = 1; 1694 switch (params->encoding) { 1695 case AUDIO_ENCODING_ULAW: 1696 auzero0 = 0x7f; 1697 break; 1698 case AUDIO_ENCODING_ALAW: 1699 auzero0 = 0x55; 1700 break; 1701 case AUDIO_ENCODING_MPEG_L1_STREAM: 1702 case AUDIO_ENCODING_MPEG_L1_PACKETS: 1703 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 1704 case AUDIO_ENCODING_MPEG_L2_STREAM: 1705 case AUDIO_ENCODING_MPEG_L2_PACKETS: 1706 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 1707 case AUDIO_ENCODING_ADPCM: /* is this right XXX */ 1708 case AUDIO_ENCODING_SLINEAR_LE: 1709 case AUDIO_ENCODING_SLINEAR_BE: 1710 auzero0 = 0;/* fortunately this works for any number of bits */ 1711 break; 1712 case AUDIO_ENCODING_ULINEAR_LE: 1713 case AUDIO_ENCODING_ULINEAR_BE: 1714 if (params->precision > 8) { 1715 nfill = (params->precision + NBBY - 1)/ NBBY; 1716 auzero0 = 0x80; 1717 auzero1 = 0; 1718 } else 1719 auzero0 = 0x80; 1720 break; 1721 default: 1722 DPRINTF(("audio: bad encoding %d\n", params->encoding)); 1723 auzero0 = 0; 1724 break; 1725 } 1726 if (nfill == 1) { 1727 while (--n >= 0) 1728 *p++ = auzero0; /* XXX memset */ 1729 } else /* nfill must no longer be 2 */ { 1730 if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) { 1731 int k = nfill; 1732 while (--k > 0) 1733 *p++ = auzero1; 1734 n -= nfill - 1; 1735 } 1736 while (n >= nfill) { 1737 int k = nfill; 1738 *p++ = auzero0; 1739 while (--k > 0) 1740 *p++ = auzero1; 1741 1742 n -= nfill; 1743 } 1744 if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */ 1745 *p++ = auzero0; 1746 } 1747 } 1748 1749 int 1750 audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio) 1751 { 1752 uint8_t zerobuf[128]; 1753 int error; 1754 int k; 1755 1756 audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf); 1757 1758 error = 0; 1759 while (n > 0 && uio->uio_resid > 0 && !error) { 1760 k = min(n, min(uio->uio_resid, sizeof zerobuf)); 1761 error = uiomove(zerobuf, k, uio); 1762 n -= k; 1763 } 1764 return error; 1765 } 1766 1767 static int 1768 uio_fetcher_fetch_to(stream_fetcher_t *self, audio_stream_t *p, 1769 int max_used) 1770 { 1771 uio_fetcher_t *this; 1772 int size; 1773 int stream_space; 1774 int error; 1775 1776 this = (uio_fetcher_t *)self; 1777 this->last_used = audio_stream_get_used(p); 1778 if (this->last_used >= this->usedhigh) 1779 return 0; 1780 /* 1781 * uio_fetcher ignores max_used and move the data as 1782 * much as possible in order to return the correct value 1783 * for audio_prinfo::seek and kfilters. 1784 */ 1785 stream_space = audio_stream_get_space(p); 1786 size = min(this->uio->uio_resid, stream_space); 1787 1788 /* the first fragment of the space */ 1789 stream_space = p->end - p->inp; 1790 if (stream_space >= size) { 1791 error = uiomove(p->inp, size, this->uio); 1792 if (error) 1793 return error; 1794 p->inp = audio_stream_add_inp(p, p->inp, size); 1795 } else { 1796 error = uiomove(p->inp, stream_space, this->uio); 1797 if (error) 1798 return error; 1799 p->inp = audio_stream_add_inp(p, p->inp, stream_space); 1800 error = uiomove(p->start, size - stream_space, this->uio); 1801 if (error) 1802 return error; 1803 p->inp = audio_stream_add_inp(p, p->inp, size - stream_space); 1804 } 1805 this->last_used = audio_stream_get_used(p); 1806 return 0; 1807 } 1808 1809 static int 1810 null_fetcher_fetch_to(stream_fetcher_t *self, audio_stream_t *p, 1811 int max_used) 1812 { 1813 1814 return 0; 1815 } 1816 1817 static void 1818 uio_fetcher_ctor(uio_fetcher_t *this, struct uio *u, int h) 1819 { 1820 1821 this->base.fetch_to = uio_fetcher_fetch_to; 1822 this->uio = u; 1823 this->usedhigh = h; 1824 } 1825 1826 int 1827 audio_write(struct audio_softc *sc, struct uio *uio, int ioflag) 1828 { 1829 uio_fetcher_t ufetcher; 1830 audio_stream_t stream; 1831 struct audio_ringbuffer *cb; 1832 stream_fetcher_t *fetcher; 1833 stream_filter_t *filter; 1834 uint8_t *inp, *einp; 1835 int saveerror, error, s, n, cc, used; 1836 1837 DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n", 1838 sc, uio->uio_resid, audio_stream_get_used(sc->sc_pustream), 1839 sc->sc_pr.usedhigh)); 1840 cb = &sc->sc_pr; 1841 if (cb->mmapped) 1842 return EINVAL; 1843 1844 if (uio->uio_resid == 0) { 1845 sc->sc_eof++; 1846 return 0; 1847 } 1848 1849 /* 1850 * If half-duplex and currently recording, throw away data. 1851 */ 1852 if (!sc->sc_full_duplex && 1853 (sc->sc_mode & AUMODE_RECORD)) { 1854 uio->uio_offset += uio->uio_resid; 1855 uio->uio_resid = 0; 1856 DPRINTF(("audio_write: half-dpx read busy\n")); 1857 return 0; 1858 } 1859 1860 if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) { 1861 n = min(sc->sc_playdrop, uio->uio_resid); 1862 DPRINTF(("audio_write: playdrop %d\n", n)); 1863 uio->uio_offset += n; 1864 uio->uio_resid -= n; 1865 sc->sc_playdrop -= n; 1866 if (uio->uio_resid == 0) 1867 return 0; 1868 } 1869 1870 /** 1871 * setup filter pipeline 1872 */ 1873 uio_fetcher_ctor(&ufetcher, uio, cb->usedhigh); 1874 if (sc->sc_npfilters > 0) { 1875 fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; 1876 } else { 1877 fetcher = &ufetcher.base; 1878 } 1879 1880 error = 0; 1881 while (uio->uio_resid > 0 && !error) { 1882 s = splaudio(); 1883 /* wait if the first buffer is occupied */ 1884 while ((used = audio_stream_get_used(sc->sc_pustream)) 1885 >= cb->usedhigh) { 1886 DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d " 1887 "hiwat=%d\n", used, 1888 cb->usedlow, cb->usedhigh)); 1889 if (ioflag & IO_NDELAY) { 1890 splx(s); 1891 return EWOULDBLOCK; 1892 } 1893 error = audio_sleep(&sc->sc_wchan, "aud_wr"); 1894 if (sc->sc_dying) 1895 error = EIO; 1896 if (error) { 1897 splx(s); 1898 return error; 1899 } 1900 } 1901 inp = cb->s.inp; 1902 cb->copying = TRUE; 1903 stream = cb->s; 1904 used = stream.used; 1905 splx(s); 1906 1907 /* 1908 * write to the cb as much as possible 1909 * 1910 * work with a temporary audio_stream_t to narrow 1911 * splaudio() enclosure 1912 */ 1913 1914 sc->sc_writing = 1; 1915 1916 if (sc->sc_npfilters > 0) { 1917 filter = sc->sc_pfilters[0]; 1918 filter->set_fetcher(filter, &ufetcher.base); 1919 } 1920 cc = stream.end - stream.start; 1921 if (sc->sc_npfilters > 0) { 1922 fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; 1923 } else { 1924 fetcher = &ufetcher.base; 1925 } 1926 error = fetcher->fetch_to(fetcher, &stream, cc); 1927 1928 sc->sc_writing = 0; 1929 if (sc->sc_waitcomp) 1930 wakeup(sc); 1931 1932 s = splaudio(); 1933 if (sc->sc_npfilters > 0) { 1934 cb->fstamp += ufetcher.last_used 1935 - audio_stream_get_used(sc->sc_pustream); 1936 } 1937 cb->s.used += stream.used - used; 1938 cb->s.inp = stream.inp; 1939 einp = cb->s.inp; 1940 1941 /* 1942 * This is a very suboptimal way of keeping track of 1943 * silence in the buffer, but it is simple. 1944 */ 1945 sc->sc_sil_count = 0; 1946 1947 /* 1948 * If the interrupt routine wants the last block filled AND 1949 * the copy did not fill the last block completely it needs to 1950 * be padded. 1951 */ 1952 if (cb->needfill && inp < einp && 1953 (inp - cb->s.start) / cb->blksize == 1954 (einp - cb->s.start) / cb->blksize) { 1955 /* Figure out how many bytes to a block boundary. */ 1956 cc = cb->blksize - (einp - cb->s.start) % cb->blksize; 1957 DPRINTF(("audio_write: partial fill %d\n", cc)); 1958 } else 1959 cc = 0; 1960 cb->needfill = FALSE; 1961 cb->copying = FALSE; 1962 if (!sc->sc_pbus && !cb->pause) { 1963 saveerror = error; 1964 error = audiostartp(sc); 1965 if (saveerror != 0) { 1966 /* Report the first error that occurred. */ 1967 error = saveerror; 1968 } 1969 } 1970 splx(s); 1971 if (cc != 0) { 1972 DPRINTFN(1, ("audio_write: fill %d\n", cc)); 1973 audio_fill_silence(&cb->s.param, einp, cc); 1974 } 1975 } 1976 1977 return error; 1978 } 1979 1980 int 1981 audio_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag, 1982 struct proc *p) 1983 { 1984 const struct audio_hw_if *hw; 1985 struct audio_offset *ao; 1986 u_long stamp; 1987 int error, s, offs, fd; 1988 boolean_t rbus, pbus; 1989 1990 DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n", 1991 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); 1992 hw = sc->hw_if; 1993 error = 0; 1994 switch (cmd) { 1995 case FIONBIO: 1996 /* All handled in the upper FS layer. */ 1997 break; 1998 1999 case FIONREAD: 2000 *(int *)addr = audio_stream_get_used(sc->sc_rustream); 2001 break; 2002 2003 case FIOASYNC: 2004 if (*(int *)addr) { 2005 if (sc->sc_async_audio) 2006 return EBUSY; 2007 sc->sc_async_audio = p; 2008 DPRINTF(("audio_ioctl: FIOASYNC %p\n", p)); 2009 } else 2010 sc->sc_async_audio = 0; 2011 break; 2012 2013 case AUDIO_FLUSH: 2014 DPRINTF(("AUDIO_FLUSH\n")); 2015 rbus = sc->sc_rbus; 2016 pbus = sc->sc_pbus; 2017 audio_clear(sc); 2018 s = splaudio(); 2019 error = audio_initbufs(sc); 2020 if (error) { 2021 splx(s); 2022 return error; 2023 } 2024 if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus) 2025 error = audiostartp(sc); 2026 if (!error && 2027 (sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus) 2028 error = audiostartr(sc); 2029 splx(s); 2030 break; 2031 2032 /* 2033 * Number of read (write) samples dropped. We don't know where or 2034 * when they were dropped. 2035 */ 2036 case AUDIO_RERROR: 2037 *(int *)addr = sc->sc_rr.drops; 2038 break; 2039 2040 case AUDIO_PERROR: 2041 *(int *)addr = sc->sc_pr.drops; 2042 break; 2043 2044 /* 2045 * Offsets into buffer. 2046 */ 2047 case AUDIO_GETIOFFS: 2048 ao = (struct audio_offset *)addr; 2049 s = splaudio(); 2050 /* figure out where next DMA will start */ 2051 stamp = sc->sc_rustream == &sc->sc_rr.s 2052 ? sc->sc_rr.stamp : sc->sc_rr.fstamp; 2053 offs = sc->sc_rustream->inp - sc->sc_rustream->start; 2054 splx(s); 2055 ao->samples = stamp; 2056 ao->deltablks = 2057 (stamp / sc->sc_rr.blksize) - 2058 (sc->sc_rr.stamp_last / sc->sc_rr.blksize); 2059 sc->sc_rr.stamp_last = stamp; 2060 ao->offset = offs; 2061 break; 2062 2063 case AUDIO_GETOOFFS: 2064 ao = (struct audio_offset *)addr; 2065 s = splaudio(); 2066 /* figure out where next DMA will start */ 2067 stamp = sc->sc_pustream == &sc->sc_pr.s 2068 ? sc->sc_pr.stamp : sc->sc_pr.fstamp; 2069 offs = sc->sc_pustream->outp - sc->sc_pustream->start 2070 + sc->sc_pr.blksize; 2071 splx(s); 2072 ao->samples = stamp; 2073 ao->deltablks = 2074 (stamp / sc->sc_pr.blksize) - 2075 (sc->sc_pr.stamp_last / sc->sc_pr.blksize); 2076 sc->sc_pr.stamp_last = stamp; 2077 if (sc->sc_pustream->start + offs >= sc->sc_pustream->end) 2078 offs = 0; 2079 ao->offset = offs; 2080 break; 2081 2082 /* 2083 * How many bytes will elapse until mike hears the first 2084 * sample of what we write next? 2085 */ 2086 case AUDIO_WSEEK: 2087 *(u_long *)addr = audio_stream_get_used(sc->sc_pustream); 2088 break; 2089 2090 case AUDIO_SETINFO: 2091 DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode)); 2092 error = audiosetinfo(sc, (struct audio_info *)addr); 2093 break; 2094 2095 case AUDIO_GETINFO: 2096 DPRINTF(("AUDIO_GETINFO\n")); 2097 error = audiogetinfo(sc, (struct audio_info *)addr); 2098 break; 2099 2100 case AUDIO_DRAIN: 2101 DPRINTF(("AUDIO_DRAIN\n")); 2102 error = audio_drain(sc); 2103 if (!error && hw->drain) 2104 error = hw->drain(sc->hw_hdl); 2105 break; 2106 2107 case AUDIO_GETDEV: 2108 DPRINTF(("AUDIO_GETDEV\n")); 2109 error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); 2110 break; 2111 2112 case AUDIO_GETENC: 2113 DPRINTF(("AUDIO_GETENC\n")); 2114 error = hw->query_encoding(sc->hw_hdl, 2115 (struct audio_encoding *)addr); 2116 break; 2117 2118 case AUDIO_GETFD: 2119 DPRINTF(("AUDIO_GETFD\n")); 2120 *(int *)addr = sc->sc_full_duplex; 2121 break; 2122 2123 case AUDIO_SETFD: 2124 DPRINTF(("AUDIO_SETFD\n")); 2125 fd = *(int *)addr; 2126 if (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) { 2127 if (hw->setfd) 2128 error = hw->setfd(sc->hw_hdl, fd); 2129 else 2130 error = 0; 2131 if (!error) 2132 sc->sc_full_duplex = fd; 2133 } else { 2134 if (fd) 2135 error = ENOTTY; 2136 else 2137 error = 0; 2138 } 2139 break; 2140 2141 case AUDIO_GETPROPS: 2142 DPRINTF(("AUDIO_GETPROPS\n")); 2143 *(int *)addr = hw->get_props(sc->hw_hdl); 2144 break; 2145 2146 default: 2147 if (hw->dev_ioctl) { 2148 error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, p); 2149 } else { 2150 DPRINTF(("audio_ioctl: unknown ioctl\n")); 2151 error = EINVAL; 2152 } 2153 break; 2154 } 2155 DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n", 2156 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); 2157 return error; 2158 } 2159 2160 int 2161 audio_poll(struct audio_softc *sc, int events, struct proc *p) 2162 { 2163 int revents; 2164 int s; 2165 int used; 2166 2167 DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode)); 2168 2169 revents = 0; 2170 s = splaudio(); 2171 if (events & (POLLIN | POLLRDNORM)) { 2172 used = audio_stream_get_used(sc->sc_rustream); 2173 /* 2174 * If half duplex and playing, audio_read() will generate 2175 * silence at the play rate; poll for silence being 2176 * available. Otherwise, poll for recorded sound. 2177 */ 2178 if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) ? 2179 sc->sc_pr.stamp > sc->sc_wstamp : 2180 used > sc->sc_rr.usedlow) 2181 revents |= events & (POLLIN | POLLRDNORM); 2182 } 2183 2184 if (events & (POLLOUT | POLLWRNORM)) { 2185 used = audio_stream_get_used(sc->sc_pustream); 2186 /* 2187 * If half duplex and recording, audio_write() will throw 2188 * away play data, which means we are always ready to write. 2189 * Otherwise, poll for play buffer being below its low water 2190 * mark. 2191 */ 2192 if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_RECORD)) || 2193 used <= sc->sc_pr.usedlow) 2194 revents |= events & (POLLOUT | POLLWRNORM); 2195 } 2196 2197 if (revents == 0) { 2198 if (events & (POLLIN | POLLRDNORM)) 2199 selrecord(p, &sc->sc_rsel); 2200 2201 if (events & (POLLOUT | POLLWRNORM)) 2202 selrecord(p, &sc->sc_wsel); 2203 } 2204 2205 splx(s); 2206 return revents; 2207 } 2208 2209 static void 2210 filt_audiordetach(struct knote *kn) 2211 { 2212 struct audio_softc *sc; 2213 int s; 2214 2215 sc = kn->kn_hook; 2216 s = splaudio(); 2217 SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); 2218 splx(s); 2219 } 2220 2221 static int 2222 filt_audioread(struct knote *kn, long hint) 2223 { 2224 struct audio_softc *sc; 2225 int s; 2226 2227 sc = kn->kn_hook; 2228 s = splaudio(); 2229 if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) 2230 kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp; 2231 else 2232 kn->kn_data = audio_stream_get_used(sc->sc_rustream) 2233 - sc->sc_rr.usedlow; 2234 splx(s); 2235 2236 return kn->kn_data > 0; 2237 } 2238 2239 static const struct filterops audioread_filtops = 2240 { 1, NULL, filt_audiordetach, filt_audioread }; 2241 2242 static void 2243 filt_audiowdetach(struct knote *kn) 2244 { 2245 struct audio_softc *sc; 2246 int s; 2247 2248 sc = kn->kn_hook; 2249 s = splaudio(); 2250 SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); 2251 splx(s); 2252 } 2253 2254 static int 2255 filt_audiowrite(struct knote *kn, long hint) 2256 { 2257 struct audio_softc *sc; 2258 audio_stream_t *stream; 2259 int s; 2260 2261 sc = kn->kn_hook; 2262 s = splaudio(); 2263 stream = sc->sc_pustream; 2264 kn->kn_data = (stream->end - stream->start) 2265 - audio_stream_get_used(stream); 2266 splx(s); 2267 2268 return kn->kn_data > 0; 2269 } 2270 2271 static const struct filterops audiowrite_filtops = 2272 { 1, NULL, filt_audiowdetach, filt_audiowrite }; 2273 2274 int 2275 audio_kqfilter(struct audio_softc *sc, struct knote *kn) 2276 { 2277 struct klist *klist; 2278 int s; 2279 2280 switch (kn->kn_filter) { 2281 case EVFILT_READ: 2282 klist = &sc->sc_rsel.sel_klist; 2283 kn->kn_fop = &audioread_filtops; 2284 break; 2285 2286 case EVFILT_WRITE: 2287 klist = &sc->sc_wsel.sel_klist; 2288 kn->kn_fop = &audiowrite_filtops; 2289 break; 2290 2291 default: 2292 return 1; 2293 } 2294 2295 kn->kn_hook = sc; 2296 2297 s = splaudio(); 2298 SLIST_INSERT_HEAD(klist, kn, kn_selnext); 2299 splx(s); 2300 2301 return 0; 2302 } 2303 2304 paddr_t 2305 audio_mmap(struct audio_softc *sc, off_t off, int prot) 2306 { 2307 const struct audio_hw_if *hw; 2308 struct audio_ringbuffer *cb; 2309 int s; 2310 2311 DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)off, prot)); 2312 hw = sc->hw_if; 2313 if (!(hw->get_props(sc->hw_hdl) & AUDIO_PROP_MMAP) || !hw->mappage) 2314 return -1; 2315 #if 0 2316 /* XXX 2317 * The idea here was to use the protection to determine if 2318 * we are mapping the read or write buffer, but it fails. 2319 * The VM system is broken in (at least) two ways. 2320 * 1) If you map memory VM_PROT_WRITE you SIGSEGV 2321 * when writing to it, so VM_PROT_READ|VM_PROT_WRITE 2322 * has to be used for mmapping the play buffer. 2323 * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE 2324 * audio_mmap will get called at some point with VM_PROT_READ 2325 * only. 2326 * So, alas, we always map the play buffer for now. 2327 */ 2328 if (prot == (VM_PROT_READ|VM_PROT_WRITE) || 2329 prot == VM_PROT_WRITE) 2330 cb = &sc->sc_pr; 2331 else if (prot == VM_PROT_READ) 2332 cb = &sc->sc_rr; 2333 else 2334 return -1; 2335 #else 2336 cb = &sc->sc_pr; 2337 #endif 2338 2339 if ((u_int)off >= cb->s.bufsize) 2340 return -1; 2341 if (!cb->mmapped) { 2342 cb->mmapped = TRUE; 2343 if (cb == &sc->sc_pr) { 2344 audio_fill_silence(&cb->s.param, cb->s.start, 2345 cb->s.bufsize); 2346 s = splaudio(); 2347 sc->sc_pustream = &cb->s; 2348 if (!sc->sc_pbus) 2349 (void)audiostartp(sc); 2350 splx(s); 2351 } else { 2352 s = splaudio(); 2353 sc->sc_rustream = &cb->s; 2354 if (!sc->sc_rbus) 2355 (void)audiostartr(sc); 2356 splx(s); 2357 } 2358 } 2359 2360 return hw->mappage(sc->hw_hdl, cb->s.start, off, prot); 2361 } 2362 2363 int 2364 audiostartr(struct audio_softc *sc) 2365 { 2366 int error; 2367 2368 DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n", 2369 sc->sc_rr.s.start, audio_stream_get_used(&sc->sc_rr.s), 2370 sc->sc_rr.usedhigh, sc->sc_rr.mmapped)); 2371 2372 if (sc->hw_if->trigger_input) 2373 error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.s.start, 2374 sc->sc_rr.s.end, sc->sc_rr.blksize, 2375 audio_rint, (void *)sc, &sc->sc_rr.s.param); 2376 else 2377 error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.s.start, 2378 sc->sc_rr.blksize, audio_rint, (void *)sc); 2379 if (error) { 2380 DPRINTF(("audiostartr failed: %d\n", error)); 2381 return error; 2382 } 2383 sc->sc_rbus = TRUE; 2384 return 0; 2385 } 2386 2387 int 2388 audiostartp(struct audio_softc *sc) 2389 { 2390 int error; 2391 int used; 2392 2393 used = audio_stream_get_used(&sc->sc_pr.s); 2394 DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n", 2395 sc->sc_pr.s.start, used, sc->sc_pr.usedhigh, 2396 sc->sc_pr.blksize, sc->sc_pr.mmapped)); 2397 2398 if (!sc->sc_pr.mmapped && used < sc->sc_pr.blksize) { 2399 wakeup(&sc->sc_wchan); 2400 DPRINTF(("%s: wakeup and return\n", __func__)); 2401 return 0; 2402 } 2403 2404 if (sc->hw_if->trigger_output) { 2405 DPRINTF(("%s: call trigger_output\n", __func__)); 2406 error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.s.start, 2407 sc->sc_pr.s.end, sc->sc_pr.blksize, 2408 audio_pint, (void *)sc, &sc->sc_pr.s.param); 2409 } else { 2410 DPRINTF(("%s: call start_output\n", __func__)); 2411 error = sc->hw_if->start_output(sc->hw_hdl, 2412 __UNCONST(sc->sc_pr.s.outp), sc->sc_pr.blksize, 2413 audio_pint, (void *)sc); 2414 } 2415 if (error) { 2416 DPRINTF(("audiostartp failed: %d\n", error)); 2417 return error; 2418 } 2419 sc->sc_pbus = TRUE; 2420 return 0; 2421 } 2422 2423 /* 2424 * When the play interrupt routine finds that the write isn't keeping 2425 * the buffer filled it will insert silence in the buffer to make up 2426 * for this. The part of the buffer that is filled with silence 2427 * is kept track of in a very approximate way: it starts at sc_sil_start 2428 * and extends sc_sil_count bytes. If there is already silence in 2429 * the requested area nothing is done; so when the whole buffer is 2430 * silent nothing happens. When the writer starts again sc_sil_count 2431 * is set to 0. 2432 */ 2433 /* XXX 2434 * Putting silence into the output buffer should not really be done 2435 * at splaudio, but there is no softaudio level to do it at yet. 2436 */ 2437 static __inline void 2438 audio_pint_silence(struct audio_softc *sc, struct audio_ringbuffer *cb, 2439 uint8_t *inp, int cc) 2440 { 2441 uint8_t *s, *e, *p, *q; 2442 2443 if (sc->sc_sil_count > 0) { 2444 s = sc->sc_sil_start; /* start of silence */ 2445 e = s + sc->sc_sil_count; /* end of sil., may be beyond end */ 2446 p = inp; /* adjusted pointer to area to fill */ 2447 if (p < s) 2448 p += cb->s.end - cb->s.start; 2449 q = p + cc; 2450 /* Check if there is already silence. */ 2451 if (!(s <= p && p < e && 2452 s <= q && q <= e)) { 2453 if (s <= p) 2454 sc->sc_sil_count = max(sc->sc_sil_count, q-s); 2455 DPRINTFN(5,("audio_pint_silence: fill cc=%d inp=%p, " 2456 "count=%d size=%d\n", 2457 cc, inp, sc->sc_sil_count, 2458 (int)(cb->s.end - cb->s.start))); 2459 audio_fill_silence(&cb->s.param, inp, cc); 2460 } else { 2461 DPRINTFN(5,("audio_pint_silence: already silent " 2462 "cc=%d inp=%p\n", cc, inp)); 2463 2464 } 2465 } else { 2466 sc->sc_sil_start = inp; 2467 sc->sc_sil_count = cc; 2468 DPRINTFN(5, ("audio_pint_silence: start fill %p %d\n", 2469 inp, cc)); 2470 audio_fill_silence(&cb->s.param, inp, cc); 2471 } 2472 } 2473 2474 /* 2475 * Called from HW driver module on completion of DMA output. 2476 * Start output of new block, wrap in ring buffer if needed. 2477 * If no more buffers to play, output zero instead. 2478 * Do a wakeup if necessary. 2479 */ 2480 void 2481 audio_pint(void *v) 2482 { 2483 stream_fetcher_t null_fetcher; 2484 struct audio_softc *sc; 2485 const struct audio_hw_if *hw; 2486 struct audio_ringbuffer *cb; 2487 stream_fetcher_t *fetcher; 2488 uint8_t *inp; 2489 int cc, used; 2490 int blksize; 2491 int error; 2492 2493 sc = v; 2494 if (!sc->sc_open) 2495 return; /* ignore interrupt if not open */ 2496 2497 hw = sc->hw_if; 2498 cb = &sc->sc_pr; 2499 blksize = cb->blksize; 2500 cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); 2501 cb->stamp += blksize; 2502 if (cb->mmapped) { 2503 DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n", 2504 cb->s.outp, blksize, cb->s.inp)); 2505 if (hw->trigger_output == NULL) 2506 (void)hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp), 2507 blksize, audio_pint, (void *)sc); 2508 return; 2509 } 2510 2511 #ifdef AUDIO_INTR_TIME 2512 { 2513 struct timeval tv; 2514 u_long t; 2515 microtime(&tv); 2516 t = tv.tv_usec + 1000000 * tv.tv_sec; 2517 if (sc->sc_pnintr) { 2518 long lastdelta, totdelta; 2519 lastdelta = t - sc->sc_plastintr - sc->sc_pblktime; 2520 if (lastdelta > sc->sc_pblktime / 3) { 2521 printf("audio: play interrupt(%d) off " 2522 "relative by %ld us (%lu)\n", 2523 sc->sc_pnintr, lastdelta, 2524 sc->sc_pblktime); 2525 } 2526 totdelta = t - sc->sc_pfirstintr - 2527 sc->sc_pblktime * sc->sc_pnintr; 2528 if (totdelta > sc->sc_pblktime) { 2529 printf("audio: play interrupt(%d) off " 2530 "absolute by %ld us (%lu) (LOST)\n", 2531 sc->sc_pnintr, totdelta, 2532 sc->sc_pblktime); 2533 sc->sc_pnintr++; /* avoid repeated messages */ 2534 } 2535 } else 2536 sc->sc_pfirstintr = t; 2537 sc->sc_plastintr = t; 2538 sc->sc_pnintr++; 2539 } 2540 #endif 2541 2542 used = audio_stream_get_used(&cb->s); 2543 /* 2544 * "used <= cb->usedlow" should be "used < blksize" ideally. 2545 * Some HW drivers such as uaudio(4) does not call audio_pint() 2546 * at accurate timing. If used < blksize, uaudio(4) already 2547 * request transfer of garbage data. 2548 */ 2549 if (used <= cb->usedlow && !cb->copying && sc->sc_npfilters > 0) { 2550 /* we might have data in filter pipeline */ 2551 null_fetcher.fetch_to = null_fetcher_fetch_to; 2552 fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base; 2553 sc->sc_pfilters[0]->set_fetcher(sc->sc_pfilters[0], 2554 &null_fetcher); 2555 used = audio_stream_get_used(sc->sc_pustream); 2556 cc = cb->s.end - cb->s.start; 2557 fetcher->fetch_to(fetcher, &cb->s, cc); 2558 cb->fstamp += used - audio_stream_get_used(sc->sc_pustream); 2559 used = audio_stream_get_used(&cb->s); 2560 } 2561 if (used < blksize) { 2562 /* we don't have a full block to use */ 2563 if (cb->copying) { 2564 /* writer is in progress, don't disturb */ 2565 cb->needfill = TRUE; 2566 DPRINTFN(1, ("audio_pint: copying in progress\n")); 2567 } else { 2568 inp = cb->s.inp; 2569 cc = blksize - (inp - cb->s.start) % blksize; 2570 if (cb->pause) 2571 cb->pdrops += cc; 2572 else { 2573 cb->drops += cc; 2574 sc->sc_playdrop += cc; 2575 } 2576 audio_pint_silence(sc, cb, inp, cc); 2577 cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); 2578 2579 /* Clear next block so we keep ahead of the DMA. */ 2580 used = audio_stream_get_used(&cb->s); 2581 if (used + cc < cb->s.end - cb->s.start) 2582 audio_pint_silence(sc, cb, inp, blksize); 2583 } 2584 } 2585 2586 DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n", cb->s.outp, blksize)); 2587 if (hw->trigger_output == NULL) { 2588 error = hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp), 2589 blksize, audio_pint, (void *)sc); 2590 if (error) { 2591 /* XXX does this really help? */ 2592 DPRINTF(("audio_pint restart failed: %d\n", error)); 2593 audio_clear(sc); 2594 } 2595 } 2596 2597 DPRINTFN(2, ("audio_pint: mode=%d pause=%d used=%d lowat=%d\n", 2598 sc->sc_mode, cb->pause, 2599 audio_stream_get_used(sc->sc_pustream), cb->usedlow)); 2600 if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) { 2601 if (audio_stream_get_used(sc->sc_pustream) <= cb->usedlow) { 2602 audio_wakeup(&sc->sc_wchan); 2603 selnotify(&sc->sc_wsel, 0); 2604 if (sc->sc_async_audio) { 2605 DPRINTFN(3, ("audio_pint: sending SIGIO %p\n", 2606 sc->sc_async_audio)); 2607 psignal(sc->sc_async_audio, SIGIO); 2608 } 2609 } 2610 } 2611 2612 /* Possible to return one or more "phantom blocks" now. */ 2613 if (!sc->sc_full_duplex && sc->sc_rchan) { 2614 audio_wakeup(&sc->sc_rchan); 2615 selnotify(&sc->sc_rsel, 0); 2616 if (sc->sc_async_audio) 2617 psignal(sc->sc_async_audio, SIGIO); 2618 } 2619 } 2620 2621 /* 2622 * Called from HW driver module on completion of DMA input. 2623 * Mark it as input in the ring buffer (fiddle pointers). 2624 * Do a wakeup if necessary. 2625 */ 2626 void 2627 audio_rint(void *v) 2628 { 2629 stream_fetcher_t null_fetcher; 2630 struct audio_softc *sc; 2631 const struct audio_hw_if *hw; 2632 struct audio_ringbuffer *cb; 2633 stream_fetcher_t *last_fetcher; 2634 int cc; 2635 int used; 2636 int blksize; 2637 int error; 2638 2639 sc = v; 2640 cb = &sc->sc_rr; 2641 if (!sc->sc_open) 2642 return; /* ignore interrupt if not open */ 2643 2644 hw = sc->hw_if; 2645 blksize = cb->blksize; 2646 cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, blksize); 2647 cb->stamp += blksize; 2648 if (cb->mmapped) { 2649 DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n", 2650 cb->s.inp, blksize)); 2651 if (hw->trigger_input == NULL) 2652 (void)hw->start_input(sc->hw_hdl, cb->s.inp, blksize, 2653 audio_rint, (void *)sc); 2654 return; 2655 } 2656 2657 #ifdef AUDIO_INTR_TIME 2658 { 2659 struct timeval tv; 2660 u_long t; 2661 microtime(&tv); 2662 t = tv.tv_usec + 1000000 * tv.tv_sec; 2663 if (sc->sc_rnintr) { 2664 long lastdelta, totdelta; 2665 lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime; 2666 if (lastdelta > sc->sc_rblktime / 5) { 2667 printf("audio: record interrupt(%d) off " 2668 "relative by %ld us (%lu)\n", 2669 sc->sc_rnintr, lastdelta, 2670 sc->sc_rblktime); 2671 } 2672 totdelta = t - sc->sc_rfirstintr - 2673 sc->sc_rblktime * sc->sc_rnintr; 2674 if (totdelta > sc->sc_rblktime / 2) { 2675 sc->sc_rnintr++; 2676 printf("audio: record interrupt(%d) off " 2677 "absolute by %ld us (%lu)\n", 2678 sc->sc_rnintr, totdelta, 2679 sc->sc_rblktime); 2680 sc->sc_rnintr++; /* avoid repeated messages */ 2681 } 2682 } else 2683 sc->sc_rfirstintr = t; 2684 sc->sc_rlastintr = t; 2685 sc->sc_rnintr++; 2686 } 2687 #endif 2688 2689 if (!cb->pause && sc->sc_nrfilters > 0) { 2690 null_fetcher.fetch_to = null_fetcher_fetch_to; 2691 last_fetcher = &sc->sc_rfilters[sc->sc_nrfilters - 1]->base; 2692 sc->sc_rfilters[0]->set_fetcher(sc->sc_rfilters[0], 2693 &null_fetcher); 2694 used = audio_stream_get_used(sc->sc_rustream); 2695 cc = sc->sc_rustream->end - sc->sc_rustream->start; 2696 error = last_fetcher->fetch_to 2697 (last_fetcher, sc->sc_rustream, cc); 2698 cb->fstamp += audio_stream_get_used(sc->sc_rustream) - used; 2699 /* XXX what should do for error? */ 2700 } 2701 used = audio_stream_get_used(&sc->sc_rr.s); 2702 if (cb->pause) { 2703 DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops)); 2704 cb->pdrops += blksize; 2705 cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); 2706 } else if (used + blksize > cb->s.end - cb->s.start && !cb->copying) { 2707 DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops)); 2708 cb->drops += blksize; 2709 cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); 2710 } 2711 2712 DPRINTFN(2, ("audio_rint: inp=%p cc=%d\n", cb->s.inp, blksize)); 2713 if (hw->trigger_input == NULL) { 2714 error = hw->start_input(sc->hw_hdl, cb->s.inp, blksize, 2715 audio_rint, (void *)sc); 2716 if (error) { 2717 /* XXX does this really help? */ 2718 DPRINTF(("audio_rint: restart failed: %d\n", error)); 2719 audio_clear(sc); 2720 } 2721 } 2722 2723 audio_wakeup(&sc->sc_rchan); 2724 selnotify(&sc->sc_rsel, 0); 2725 if (sc->sc_async_audio) 2726 psignal(sc->sc_async_audio, SIGIO); 2727 } 2728 2729 int 2730 audio_check_params(struct audio_params *p) 2731 { 2732 2733 if (p->encoding == AUDIO_ENCODING_PCM16) { 2734 if (p->precision == 8) 2735 p->encoding = AUDIO_ENCODING_ULINEAR; 2736 else 2737 p->encoding = AUDIO_ENCODING_SLINEAR; 2738 } else if (p->encoding == AUDIO_ENCODING_PCM8) { 2739 if (p->precision == 8) 2740 p->encoding = AUDIO_ENCODING_ULINEAR; 2741 else 2742 return EINVAL; 2743 } 2744 2745 if (p->encoding == AUDIO_ENCODING_SLINEAR) 2746 #if BYTE_ORDER == LITTLE_ENDIAN 2747 p->encoding = AUDIO_ENCODING_SLINEAR_LE; 2748 #else 2749 p->encoding = AUDIO_ENCODING_SLINEAR_BE; 2750 #endif 2751 if (p->encoding == AUDIO_ENCODING_ULINEAR) 2752 #if BYTE_ORDER == LITTLE_ENDIAN 2753 p->encoding = AUDIO_ENCODING_ULINEAR_LE; 2754 #else 2755 p->encoding = AUDIO_ENCODING_ULINEAR_BE; 2756 #endif 2757 2758 switch (p->encoding) { 2759 case AUDIO_ENCODING_ULAW: 2760 case AUDIO_ENCODING_ALAW: 2761 if (p->precision != 8) 2762 return EINVAL; 2763 break; 2764 case AUDIO_ENCODING_ADPCM: 2765 if (p->precision != 4 && p->precision != 8) 2766 return EINVAL; 2767 break; 2768 case AUDIO_ENCODING_SLINEAR_LE: 2769 case AUDIO_ENCODING_SLINEAR_BE: 2770 case AUDIO_ENCODING_ULINEAR_LE: 2771 case AUDIO_ENCODING_ULINEAR_BE: 2772 /* XXX is: our zero-fill can handle any multiple of 8 */ 2773 if (p->precision != 8 && p->precision != 16 && 2774 p->precision != 24 && p->precision != 32) 2775 return EINVAL; 2776 if (p->precision == 8 && p->encoding == AUDIO_ENCODING_SLINEAR_BE) 2777 p->encoding = AUDIO_ENCODING_SLINEAR_LE; 2778 if (p->precision == 8 && p->encoding == AUDIO_ENCODING_ULINEAR_BE) 2779 p->encoding = AUDIO_ENCODING_ULINEAR_LE; 2780 if (p->validbits > p->precision) 2781 return EINVAL; 2782 break; 2783 case AUDIO_ENCODING_MPEG_L1_STREAM: 2784 case AUDIO_ENCODING_MPEG_L1_PACKETS: 2785 case AUDIO_ENCODING_MPEG_L1_SYSTEM: 2786 case AUDIO_ENCODING_MPEG_L2_STREAM: 2787 case AUDIO_ENCODING_MPEG_L2_PACKETS: 2788 case AUDIO_ENCODING_MPEG_L2_SYSTEM: 2789 break; 2790 default: 2791 return EINVAL; 2792 } 2793 2794 /* sanity check # of channels*/ 2795 if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) 2796 return EINVAL; 2797 2798 return 0; 2799 } 2800 2801 int 2802 audio_set_defaults(struct audio_softc *sc, u_int mode) 2803 { 2804 struct audio_info ai; 2805 2806 /* default parameters */ 2807 sc->sc_rparams = audio_default; 2808 sc->sc_pparams = audio_default; 2809 sc->sc_blkset = FALSE; 2810 2811 AUDIO_INITINFO(&ai); 2812 ai.record.sample_rate = sc->sc_rparams.sample_rate; 2813 ai.record.encoding = sc->sc_rparams.encoding; 2814 ai.record.channels = sc->sc_rparams.channels; 2815 ai.record.precision = sc->sc_rparams.precision; 2816 ai.play.sample_rate = sc->sc_pparams.sample_rate; 2817 ai.play.encoding = sc->sc_pparams.encoding; 2818 ai.play.channels = sc->sc_pparams.channels; 2819 ai.play.precision = sc->sc_pparams.precision; 2820 ai.mode = mode; 2821 2822 return audiosetinfo(sc, &ai); 2823 } 2824 2825 int 2826 au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) 2827 { 2828 2829 ct->type = AUDIO_MIXER_VALUE; 2830 ct->un.value.num_channels = 2; 2831 ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; 2832 ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; 2833 if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0) 2834 return 0; 2835 ct->un.value.num_channels = 1; 2836 ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; 2837 return sc->hw_if->set_port(sc->hw_hdl, ct); 2838 } 2839 2840 int 2841 au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 2842 int gain, int balance) 2843 { 2844 mixer_ctrl_t ct; 2845 int i, error; 2846 int l, r; 2847 u_int mask; 2848 int nset; 2849 2850 if (balance == AUDIO_MID_BALANCE) { 2851 l = r = gain; 2852 } else if (balance < AUDIO_MID_BALANCE) { 2853 l = gain; 2854 r = (balance * gain) / AUDIO_MID_BALANCE; 2855 } else { 2856 r = gain; 2857 l = ((AUDIO_RIGHT_BALANCE - balance) * gain) 2858 / AUDIO_MID_BALANCE; 2859 } 2860 DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n", 2861 gain, balance, l, r)); 2862 2863 if (ports->index == -1) { 2864 usemaster: 2865 if (ports->master == -1) 2866 return 0; /* just ignore it silently */ 2867 ct.dev = ports->master; 2868 error = au_set_lr_value(sc, &ct, l, r); 2869 } else { 2870 ct.dev = ports->index; 2871 if (ports->isenum) { 2872 ct.type = AUDIO_MIXER_ENUM; 2873 error = sc->hw_if->get_port(sc->hw_hdl, &ct); 2874 if (error) 2875 return error; 2876 if (ports->isdual) { 2877 if (ports->cur_port == -1) 2878 ct.dev = ports->master; 2879 else 2880 ct.dev = ports->miport[ports->cur_port]; 2881 error = au_set_lr_value(sc, &ct, l, r); 2882 } else { 2883 for(i = 0; i < ports->nports; i++) 2884 if (ports->misel[i] == ct.un.ord) { 2885 ct.dev = ports->miport[i]; 2886 if (ct.dev == -1 || 2887 au_set_lr_value(sc, &ct, l, r)) 2888 goto usemaster; 2889 else 2890 break; 2891 } 2892 } 2893 } else { 2894 ct.type = AUDIO_MIXER_SET; 2895 error = sc->hw_if->get_port(sc->hw_hdl, &ct); 2896 if (error) 2897 return error; 2898 mask = ct.un.mask; 2899 nset = 0; 2900 for(i = 0; i < ports->nports; i++) { 2901 if (ports->misel[i] & mask) { 2902 ct.dev = ports->miport[i]; 2903 if (ct.dev != -1 && 2904 au_set_lr_value(sc, &ct, l, r) == 0) 2905 nset++; 2906 } 2907 } 2908 if (nset == 0) 2909 goto usemaster; 2910 } 2911 } 2912 if (!error) 2913 mixer_signal(sc); 2914 return error; 2915 } 2916 2917 int 2918 au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) 2919 { 2920 int error; 2921 2922 ct->un.value.num_channels = 2; 2923 if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) { 2924 *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; 2925 *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; 2926 } else { 2927 ct->un.value.num_channels = 1; 2928 error = sc->hw_if->get_port(sc->hw_hdl, ct); 2929 if (error) 2930 return error; 2931 *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; 2932 } 2933 return 0; 2934 } 2935 2936 void 2937 au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, 2938 u_int *pgain, u_char *pbalance) 2939 { 2940 mixer_ctrl_t ct; 2941 int i, l, r, n; 2942 int lgain, rgain; 2943 2944 lgain = AUDIO_MAX_GAIN / 2; 2945 rgain = AUDIO_MAX_GAIN / 2; 2946 if (ports->index == -1) { 2947 usemaster: 2948 if (ports->master == -1) 2949 goto bad; 2950 ct.dev = ports->master; 2951 ct.type = AUDIO_MIXER_VALUE; 2952 if (au_get_lr_value(sc, &ct, &lgain, &rgain)) 2953 goto bad; 2954 } else { 2955 ct.dev = ports->index; 2956 if (ports->isenum) { 2957 ct.type = AUDIO_MIXER_ENUM; 2958 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 2959 goto bad; 2960 ct.type = AUDIO_MIXER_VALUE; 2961 if (ports->isdual) { 2962 if (ports->cur_port == -1) 2963 ct.dev = ports->master; 2964 else 2965 ct.dev = ports->miport[ports->cur_port]; 2966 au_get_lr_value(sc, &ct, &lgain, &rgain); 2967 } else { 2968 for(i = 0; i < ports->nports; i++) 2969 if (ports->misel[i] == ct.un.ord) { 2970 ct.dev = ports->miport[i]; 2971 if (ct.dev == -1 || 2972 au_get_lr_value(sc, &ct, 2973 &lgain, &rgain)) 2974 goto usemaster; 2975 else 2976 break; 2977 } 2978 } 2979 } else { 2980 ct.type = AUDIO_MIXER_SET; 2981 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 2982 goto bad; 2983 ct.type = AUDIO_MIXER_VALUE; 2984 lgain = rgain = n = 0; 2985 for(i = 0; i < ports->nports; i++) { 2986 if (ports->misel[i] & ct.un.mask) { 2987 ct.dev = ports->miport[i]; 2988 if (ct.dev == -1 || 2989 au_get_lr_value(sc, &ct, &l, &r)) 2990 goto usemaster; 2991 else { 2992 lgain += l; 2993 rgain += r; 2994 n++; 2995 } 2996 } 2997 } 2998 if (n != 0) { 2999 lgain /= n; 3000 rgain /= n; 3001 } 3002 } 3003 } 3004 bad: 3005 if (lgain == rgain) { /* handles lgain==rgain==0 */ 3006 *pgain = lgain; 3007 *pbalance = AUDIO_MID_BALANCE; 3008 } else if (lgain < rgain) { 3009 *pgain = rgain; 3010 /* balance should be > AUDIO_MID_BALANCE */ 3011 *pbalance = AUDIO_RIGHT_BALANCE - 3012 (AUDIO_MID_BALANCE * lgain) / rgain; 3013 } else /* lgain > rgain */ { 3014 *pgain = lgain; 3015 /* balance should be < AUDIO_MID_BALANCE */ 3016 *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; 3017 } 3018 } 3019 3020 int 3021 au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) 3022 { 3023 mixer_ctrl_t ct; 3024 int i, error, use_mixerout; 3025 3026 use_mixerout = 1; 3027 if (port == 0) { 3028 if (ports->allports == 0) 3029 return 0; /* Allow this special case. */ 3030 else if (ports->isdual) { 3031 if (ports->cur_port == -1) { 3032 return 0; 3033 } else { 3034 port = ports->aumask[ports->cur_port]; 3035 ports->cur_port = -1; 3036 use_mixerout = 0; 3037 } 3038 } 3039 } 3040 if (ports->index == -1) 3041 return EINVAL; 3042 ct.dev = ports->index; 3043 if (ports->isenum) { 3044 if (port & (port-1)) 3045 return EINVAL; /* Only one port allowed */ 3046 ct.type = AUDIO_MIXER_ENUM; 3047 error = EINVAL; 3048 for(i = 0; i < ports->nports; i++) 3049 if (ports->aumask[i] == port) { 3050 if (ports->isdual && use_mixerout) { 3051 ct.un.ord = ports->mixerout; 3052 ports->cur_port = i; 3053 } else { 3054 ct.un.ord = ports->misel[i]; 3055 } 3056 error = sc->hw_if->set_port(sc->hw_hdl, &ct); 3057 break; 3058 } 3059 } else { 3060 ct.type = AUDIO_MIXER_SET; 3061 ct.un.mask = 0; 3062 for(i = 0; i < ports->nports; i++) 3063 if (ports->aumask[i] & port) 3064 ct.un.mask |= ports->misel[i]; 3065 if (port != 0 && ct.un.mask == 0) 3066 error = EINVAL; 3067 else 3068 error = sc->hw_if->set_port(sc->hw_hdl, &ct); 3069 } 3070 if (!error) 3071 mixer_signal(sc); 3072 return error; 3073 } 3074 3075 int 3076 au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) 3077 { 3078 mixer_ctrl_t ct; 3079 int i, aumask; 3080 3081 if (ports->index == -1) 3082 return 0; 3083 ct.dev = ports->index; 3084 ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; 3085 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 3086 return 0; 3087 aumask = 0; 3088 if (ports->isenum) { 3089 if (ports->isdual && ports->cur_port != -1) { 3090 if (ports->mixerout == ct.un.ord) 3091 aumask = ports->aumask[ports->cur_port]; 3092 else 3093 ports->cur_port = -1; 3094 } 3095 if (aumask == 0) 3096 for(i = 0; i < ports->nports; i++) 3097 if (ports->misel[i] == ct.un.ord) 3098 aumask = ports->aumask[i]; 3099 } else { 3100 for(i = 0; i < ports->nports; i++) 3101 if (ct.un.mask & ports->misel[i]) 3102 aumask |= ports->aumask[i]; 3103 } 3104 return aumask; 3105 } 3106 3107 int 3108 audiosetinfo(struct audio_softc *sc, struct audio_info *ai) 3109 { 3110 stream_filter_list_t pfilters, rfilters; 3111 audio_params_t pp, rp; 3112 struct audio_prinfo *r, *p; 3113 const struct audio_hw_if *hw; 3114 audio_stream_t *oldpus, *oldrus; 3115 int s, setmode; 3116 int error; 3117 int np, nr; 3118 unsigned int blks; 3119 int oldpblksize, oldrblksize; 3120 u_int gain; 3121 boolean_t rbus, pbus; 3122 boolean_t cleared, modechange, pausechange; 3123 u_char balance; 3124 3125 hw = sc->hw_if; 3126 if (hw == NULL) /* HW has not attached */ 3127 return ENXIO; 3128 3129 DPRINTF(("%s sc=%p ai=%p\n", __func__, sc, ai)); 3130 r = &ai->record; 3131 p = &ai->play; 3132 rbus = sc->sc_rbus; 3133 pbus = sc->sc_pbus; 3134 error = 0; 3135 cleared = FALSE; 3136 modechange = FALSE; 3137 pausechange = FALSE; 3138 3139 pp = sc->sc_pparams; /* Temporary encoding storage in */ 3140 rp = sc->sc_rparams; /* case setting the modes fails. */ 3141 nr = np = 0; 3142 3143 if (SPECIFIED(p->sample_rate)) { 3144 pp.sample_rate = p->sample_rate; 3145 np++; 3146 } 3147 if (SPECIFIED(r->sample_rate)) { 3148 rp.sample_rate = r->sample_rate; 3149 nr++; 3150 } 3151 if (SPECIFIED(p->encoding)) { 3152 pp.encoding = p->encoding; 3153 np++; 3154 } 3155 if (SPECIFIED(r->encoding)) { 3156 rp.encoding = r->encoding; 3157 nr++; 3158 } 3159 if (SPECIFIED(p->precision)) { 3160 pp.precision = p->precision; 3161 /* we don't have API to specify validbits */ 3162 pp.validbits = p->precision; 3163 np++; 3164 } 3165 if (SPECIFIED(r->precision)) { 3166 rp.precision = r->precision; 3167 /* we don't have API to specify validbits */ 3168 rp.validbits = r->precision; 3169 nr++; 3170 } 3171 if (SPECIFIED(p->channels)) { 3172 pp.channels = p->channels; 3173 np++; 3174 } 3175 if (SPECIFIED(r->channels)) { 3176 rp.channels = r->channels; 3177 nr++; 3178 } 3179 #ifdef AUDIO_DEBUG 3180 if (audiodebug && nr > 0) 3181 audio_print_params("audiosetinfo() Setting record params:", &rp); 3182 if (audiodebug && np > 0) 3183 audio_print_params("audiosetinfo() Setting play params:", &pp); 3184 #endif 3185 if (nr > 0 && (error = audio_check_params(&rp))) 3186 return error; 3187 if (np > 0 && (error = audio_check_params(&pp))) 3188 return error; 3189 setmode = 0; 3190 if (nr > 0) { 3191 if (!cleared) { 3192 audio_clear(sc); 3193 cleared = TRUE; 3194 } 3195 modechange = TRUE; 3196 setmode |= AUMODE_RECORD; 3197 } 3198 if (np > 0) { 3199 if (!cleared) { 3200 audio_clear(sc); 3201 cleared = TRUE; 3202 } 3203 modechange = TRUE; 3204 setmode |= AUMODE_PLAY; 3205 } 3206 3207 if (SPECIFIED(ai->mode)) { 3208 if (!cleared) { 3209 audio_clear(sc); 3210 cleared = TRUE; 3211 } 3212 modechange = TRUE; 3213 sc->sc_mode = ai->mode; 3214 if (sc->sc_mode & AUMODE_PLAY_ALL) 3215 sc->sc_mode |= AUMODE_PLAY; 3216 if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex) 3217 /* Play takes precedence */ 3218 sc->sc_mode &= ~AUMODE_RECORD; 3219 } 3220 3221 oldpus = sc->sc_pustream; 3222 oldrus = sc->sc_rustream; 3223 if (modechange) { 3224 int indep; 3225 3226 indep = hw->get_props(sc->hw_hdl) & AUDIO_PROP_INDEPENDENT; 3227 if (!indep) { 3228 if (setmode == AUMODE_RECORD) 3229 pp = rp; 3230 else if (setmode == AUMODE_PLAY) 3231 rp = pp; 3232 } 3233 memset(&pfilters, 0, sizeof(pfilters)); 3234 memset(&rfilters, 0, sizeof(rfilters)); 3235 pfilters.append = stream_filter_list_append; 3236 pfilters.prepend = stream_filter_list_prepend; 3237 pfilters.set = stream_filter_list_set; 3238 rfilters.append = stream_filter_list_append; 3239 rfilters.prepend = stream_filter_list_prepend; 3240 rfilters.set = stream_filter_list_set; 3241 /* Some device drivers change channels/sample_rate and change 3242 * no channels/sample_rate. */ 3243 error = hw->set_params(sc->hw_hdl, setmode, 3244 sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp, 3245 &pfilters, &rfilters); 3246 if (error) { 3247 DPRINTF(("%s: hw->set_params() failed with %d\n", 3248 __func__, error)); 3249 return error; 3250 } 3251 3252 audio_check_params(&pp); 3253 audio_check_params(&rp); 3254 if (!indep) { 3255 /* XXX for !indep device, we have to use the same 3256 * parameters for the hardware, not userland */ 3257 if (setmode == AUMODE_RECORD) { 3258 pp = rp; 3259 } else if (setmode == AUMODE_PLAY) { 3260 rp = pp; 3261 } 3262 } 3263 3264 if (sc->sc_pr.mmapped && pfilters.req_size > 0) { 3265 DPRINTF(("%s: mmapped, and filters are requested.\n", 3266 __func__)); 3267 return EINVAL; 3268 } 3269 3270 /* construct new filter chain */ 3271 if (setmode & AUMODE_PLAY) { 3272 error = audio_setup_pfilters(sc, &pp, &pfilters); 3273 if (error) 3274 return error; 3275 } 3276 if (setmode & AUMODE_RECORD) { 3277 error = audio_setup_rfilters(sc, &rp, &rfilters); 3278 if (error) 3279 return error; 3280 } 3281 DPRINTF(("%s: filter setup is completed.\n", __func__)); 3282 3283 /* userland formats */ 3284 sc->sc_pparams = pp; 3285 sc->sc_rparams = rp; 3286 } 3287 3288 oldpblksize = sc->sc_pr.blksize; 3289 oldrblksize = sc->sc_rr.blksize; 3290 /* Play params can affect the record params, so recalculate blksize. */ 3291 if (nr > 0 || np > 0) { 3292 audio_calc_blksize(sc, AUMODE_RECORD); 3293 audio_calc_blksize(sc, AUMODE_PLAY); 3294 } 3295 #ifdef AUDIO_DEBUG 3296 if (audiodebug > 1 && nr > 0) 3297 audio_print_params("audiosetinfo() After setting record params:", &sc->sc_rparams); 3298 if (audiodebug > 1 && np > 0) 3299 audio_print_params("audiosetinfo() After setting play params:", &sc->sc_pparams); 3300 #endif 3301 3302 if (SPECIFIED(p->port)) { 3303 if (!cleared) { 3304 audio_clear(sc); 3305 cleared = TRUE; 3306 } 3307 error = au_set_port(sc, &sc->sc_outports, p->port); 3308 if (error) 3309 return error; 3310 } 3311 if (SPECIFIED(r->port)) { 3312 if (!cleared) { 3313 audio_clear(sc); 3314 cleared = TRUE; 3315 } 3316 error = au_set_port(sc, &sc->sc_inports, r->port); 3317 if (error) 3318 return error; 3319 } 3320 if (SPECIFIED(p->gain)) { 3321 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 3322 error = au_set_gain(sc, &sc->sc_outports, p->gain, balance); 3323 if (error) 3324 return error; 3325 } 3326 if (SPECIFIED(r->gain)) { 3327 au_get_gain(sc, &sc->sc_inports, &gain, &balance); 3328 error = au_set_gain(sc, &sc->sc_inports, r->gain, balance); 3329 if (error) 3330 return error; 3331 } 3332 3333 if (SPECIFIED_CH(p->balance)) { 3334 au_get_gain(sc, &sc->sc_outports, &gain, &balance); 3335 error = au_set_gain(sc, &sc->sc_outports, gain, p->balance); 3336 if (error) 3337 return error; 3338 } 3339 if (SPECIFIED_CH(r->balance)) { 3340 au_get_gain(sc, &sc->sc_inports, &gain, &balance); 3341 error = au_set_gain(sc, &sc->sc_inports, gain, r->balance); 3342 if (error) 3343 return error; 3344 } 3345 3346 if (SPECIFIED(ai->monitor_gain) && sc->sc_monitor_port != -1) { 3347 mixer_ctrl_t ct; 3348 3349 ct.dev = sc->sc_monitor_port; 3350 ct.type = AUDIO_MIXER_VALUE; 3351 ct.un.value.num_channels = 1; 3352 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain; 3353 error = sc->hw_if->set_port(sc->hw_hdl, &ct); 3354 if (error) 3355 return error; 3356 } 3357 3358 if (SPECIFIED_CH(p->pause)) { 3359 sc->sc_pr.pause = p->pause; 3360 pbus = !p->pause; 3361 pausechange = TRUE; 3362 } 3363 if (SPECIFIED_CH(r->pause)) { 3364 sc->sc_rr.pause = r->pause; 3365 rbus = !r->pause; 3366 pausechange = TRUE; 3367 } 3368 3369 if (SPECIFIED(ai->blocksize)) { 3370 int pblksize, rblksize; 3371 3372 /* Block size specified explicitly. */ 3373 if (ai->blocksize == 0) { 3374 if (!cleared) { 3375 audio_clear(sc); 3376 cleared = TRUE; 3377 } 3378 sc->sc_blkset = FALSE; 3379 audio_calc_blksize(sc, AUMODE_RECORD); 3380 audio_calc_blksize(sc, AUMODE_PLAY); 3381 } else { 3382 sc->sc_blkset = TRUE; 3383 /* check whether new blocksize changes actually */ 3384 if (hw->round_blocksize == NULL) { 3385 if (!cleared) { 3386 audio_clear(sc); 3387 cleared = TRUE; 3388 } 3389 sc->sc_pr.blksize = ai->blocksize; 3390 sc->sc_rr.blksize = ai->blocksize; 3391 } else { 3392 pblksize = hw->round_blocksize(sc->hw_hdl, 3393 ai->blocksize, AUMODE_PLAY, &sc->sc_pr.s.param); 3394 rblksize = hw->round_blocksize(sc->hw_hdl, 3395 ai->blocksize, AUMODE_RECORD, &sc->sc_rr.s.param); 3396 if (pblksize != sc->sc_pr.blksize || 3397 rblksize != sc->sc_rr.blksize) { 3398 if (!cleared) { 3399 audio_clear(sc); 3400 cleared = TRUE; 3401 } 3402 sc->sc_pr.blksize = ai->blocksize; 3403 sc->sc_rr.blksize = ai->blocksize; 3404 } 3405 } 3406 } 3407 } 3408 3409 if (SPECIFIED(ai->mode)) { 3410 if (sc->sc_mode & AUMODE_PLAY) 3411 audio_init_play(sc); 3412 if (sc->sc_mode & AUMODE_RECORD) 3413 audio_init_record(sc); 3414 } 3415 3416 if (hw->commit_settings) { 3417 error = hw->commit_settings(sc->hw_hdl); 3418 if (error) 3419 return error; 3420 } 3421 3422 if (cleared || pausechange) { 3423 s = splaudio(); 3424 error = audio_initbufs(sc); 3425 if (error) goto err; 3426 if (sc->sc_pr.blksize != oldpblksize || 3427 sc->sc_rr.blksize != oldrblksize || 3428 sc->sc_pustream != oldpus || 3429 sc->sc_rustream != oldrus) 3430 audio_calcwater(sc); 3431 if ((sc->sc_mode & AUMODE_PLAY) && 3432 pbus && !sc->sc_pbus) 3433 error = audiostartp(sc); 3434 if (!error && 3435 (sc->sc_mode & AUMODE_RECORD) && 3436 rbus && !sc->sc_rbus) 3437 error = audiostartr(sc); 3438 err: 3439 splx(s); 3440 if (error) 3441 return error; 3442 } 3443 3444 /* Change water marks after initializing the buffers. */ 3445 if (SPECIFIED(ai->hiwat)) { 3446 blks = ai->hiwat; 3447 if (blks > sc->sc_pr.maxblks) 3448 blks = sc->sc_pr.maxblks; 3449 if (blks < 2) 3450 blks = 2; 3451 sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize; 3452 } 3453 if (SPECIFIED(ai->lowat)) { 3454 blks = ai->lowat; 3455 if (blks > sc->sc_pr.maxblks - 1) 3456 blks = sc->sc_pr.maxblks - 1; 3457 sc->sc_pr.usedlow = blks * sc->sc_pr.blksize; 3458 } 3459 if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { 3460 if (sc->sc_pr.usedlow > sc->sc_pr.usedhigh - sc->sc_pr.blksize) 3461 sc->sc_pr.usedlow = 3462 sc->sc_pr.usedhigh - sc->sc_pr.blksize; 3463 } 3464 3465 return 0; 3466 } 3467 3468 int 3469 audiogetinfo(struct audio_softc *sc, struct audio_info *ai) 3470 { 3471 struct audio_prinfo *r, *p; 3472 const struct audio_hw_if *hw; 3473 3474 r = &ai->record; 3475 p = &ai->play; 3476 hw = sc->hw_if; 3477 if (hw == NULL) /* HW has not attached */ 3478 return ENXIO; 3479 3480 p->sample_rate = sc->sc_pparams.sample_rate; 3481 r->sample_rate = sc->sc_rparams.sample_rate; 3482 p->channels = sc->sc_pparams.channels; 3483 r->channels = sc->sc_rparams.channels; 3484 p->precision = sc->sc_pparams.precision; 3485 r->precision = sc->sc_rparams.precision; 3486 p->encoding = sc->sc_pparams.encoding; 3487 r->encoding = sc->sc_rparams.encoding; 3488 3489 r->port = au_get_port(sc, &sc->sc_inports); 3490 p->port = au_get_port(sc, &sc->sc_outports); 3491 3492 r->avail_ports = sc->sc_inports.allports; 3493 p->avail_ports = sc->sc_outports.allports; 3494 3495 au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance); 3496 au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance); 3497 3498 if (sc->sc_monitor_port != -1) { 3499 mixer_ctrl_t ct; 3500 3501 ct.dev = sc->sc_monitor_port; 3502 ct.type = AUDIO_MIXER_VALUE; 3503 ct.un.value.num_channels = 1; 3504 if (sc->hw_if->get_port(sc->hw_hdl, &ct)) 3505 ai->monitor_gain = 0; 3506 else 3507 ai->monitor_gain = 3508 ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; 3509 } else 3510 ai->monitor_gain = 0; 3511 3512 p->seek = audio_stream_get_used(sc->sc_pustream); 3513 r->seek = audio_stream_get_used(sc->sc_rustream); 3514 3515 /* 3516 * XXX samples should be a value for userland data. 3517 * But drops is a value for HW data. 3518 */ 3519 p->samples = (sc->sc_pustream == &sc->sc_pr.s 3520 ? sc->sc_pr.stamp : sc->sc_pr.fstamp) - sc->sc_pr.drops; 3521 r->samples = (sc->sc_rustream == &sc->sc_rr.s 3522 ? sc->sc_rr.stamp : sc->sc_rr.fstamp) - sc->sc_rr.drops; 3523 3524 p->eof = sc->sc_eof; 3525 r->eof = 0; 3526 3527 p->pause = sc->sc_pr.pause; 3528 r->pause = sc->sc_rr.pause; 3529 3530 p->error = sc->sc_pr.drops != 0; 3531 r->error = sc->sc_rr.drops != 0; 3532 3533 p->waiting = r->waiting = 0; /* open never hangs */ 3534 3535 p->open = (sc->sc_open & AUOPEN_WRITE) != 0; 3536 r->open = (sc->sc_open & AUOPEN_READ) != 0; 3537 3538 p->active = sc->sc_pbus; 3539 r->active = sc->sc_rbus; 3540 3541 p->buffer_size = sc->sc_pustream->bufsize; 3542 r->buffer_size = sc->sc_rustream->bufsize; 3543 3544 ai->blocksize = sc->sc_pr.blksize; 3545 ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize; 3546 ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize; 3547 ai->mode = sc->sc_mode; 3548 3549 return 0; 3550 } 3551 3552 /* 3553 * Mixer driver 3554 */ 3555 int 3556 mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, 3557 struct proc *p) 3558 { 3559 if (sc->hw_if == NULL) 3560 return ENXIO; 3561 3562 DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc)); 3563 3564 return 0; 3565 } 3566 3567 /* 3568 * Remove a process from those to be signalled on mixer activity. 3569 */ 3570 static void 3571 mixer_remove(struct audio_softc *sc, struct proc *p) 3572 { 3573 struct mixer_asyncs **pm, *m; 3574 3575 for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) { 3576 if ((*pm)->proc == p) { 3577 m = *pm; 3578 *pm = m->next; 3579 free(m, M_DEVBUF); 3580 return; 3581 } 3582 } 3583 } 3584 3585 /* 3586 * Signal all processes waiting for the mixer. 3587 */ 3588 static void 3589 mixer_signal(struct audio_softc *sc) 3590 { 3591 struct mixer_asyncs *m; 3592 3593 for (m = sc->sc_async_mixer; m; m = m->next) 3594 psignal(m->proc, SIGIO); 3595 } 3596 3597 /* 3598 * Close a mixer device 3599 */ 3600 /* ARGSUSED */ 3601 int 3602 mixer_close(struct audio_softc *sc, int flags, int ifmt, struct proc *p) 3603 { 3604 3605 DPRINTF(("mixer_close: sc %p\n", sc)); 3606 mixer_remove(sc, p); 3607 return 0; 3608 } 3609 3610 int 3611 mixer_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag, 3612 struct proc *p) 3613 { 3614 const struct audio_hw_if *hw; 3615 int error; 3616 3617 DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n", 3618 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); 3619 hw = sc->hw_if; 3620 error = EINVAL; 3621 switch (cmd) { 3622 case FIOASYNC: 3623 mixer_remove(sc, p); /* remove old entry */ 3624 if (*(int *)addr) { 3625 struct mixer_asyncs *ma; 3626 ma = malloc(sizeof (struct mixer_asyncs), 3627 M_DEVBUF, M_WAITOK); 3628 ma->next = sc->sc_async_mixer; 3629 ma->proc = p; 3630 sc->sc_async_mixer = ma; 3631 } 3632 error = 0; 3633 break; 3634 3635 case AUDIO_GETDEV: 3636 DPRINTF(("AUDIO_GETDEV\n")); 3637 error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); 3638 break; 3639 3640 case AUDIO_MIXER_DEVINFO: 3641 DPRINTF(("AUDIO_MIXER_DEVINFO\n")); 3642 ((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */ 3643 error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr); 3644 break; 3645 3646 case AUDIO_MIXER_READ: 3647 DPRINTF(("AUDIO_MIXER_READ\n")); 3648 error = hw->get_port(sc->hw_hdl, (mixer_ctrl_t *)addr); 3649 break; 3650 3651 case AUDIO_MIXER_WRITE: 3652 DPRINTF(("AUDIO_MIXER_WRITE\n")); 3653 error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr); 3654 if (!error && hw->commit_settings) 3655 error = hw->commit_settings(sc->hw_hdl); 3656 if (!error) 3657 mixer_signal(sc); 3658 break; 3659 3660 default: 3661 if (hw->dev_ioctl) 3662 error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, p); 3663 else 3664 error = EINVAL; 3665 break; 3666 } 3667 DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n", 3668 IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); 3669 return error; 3670 } 3671 #endif /* NAUDIO > 0 */ 3672 3673 #include "midi.h" 3674 3675 #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) 3676 #include <sys/param.h> 3677 #include <sys/systm.h> 3678 #include <sys/device.h> 3679 #include <sys/audioio.h> 3680 #include <dev/audio_if.h> 3681 #endif 3682 3683 #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) 3684 int 3685 audioprint(void *aux, const char *pnp) 3686 { 3687 struct audio_attach_args *arg; 3688 const char *type; 3689 3690 if (pnp != NULL) { 3691 arg = aux; 3692 switch (arg->type) { 3693 case AUDIODEV_TYPE_AUDIO: 3694 type = "audio"; 3695 break; 3696 case AUDIODEV_TYPE_MIDI: 3697 type = "midi"; 3698 break; 3699 case AUDIODEV_TYPE_OPL: 3700 type = "opl"; 3701 break; 3702 case AUDIODEV_TYPE_MPU: 3703 type = "mpu"; 3704 break; 3705 default: 3706 panic("audioprint: unknown type %d", arg->type); 3707 } 3708 aprint_normal("%s at %s", type, pnp); 3709 } 3710 return UNCONF; 3711 } 3712 3713 #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */
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